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Diffstat (limited to 'ffmpeg1/doc')
91 files changed, 31679 insertions, 0 deletions
diff --git a/ffmpeg1/doc/APIchanges b/ffmpeg1/doc/APIchanges new file mode 100644 index 0000000..255f914 --- /dev/null +++ b/ffmpeg1/doc/APIchanges @@ -0,0 +1,1586 @@ +Never assume the API of libav* to be stable unless at least 1 month has passed +since the last major version increase or the API was added. + +The last version increases were: +libavcodec: 2013-03-xx +libavdevice: 2013-03-xx +libavfilter: 2012-06-22 +libavformat: 2013-03-xx +libavresample: 2012-10-05 +libpostproc: 2011-04-18 +libswresample: 2011-09-19 +libswscale: 2011-06-20 +libavutil: 2012-10-22 + + +API changes, most recent first: + +2013-03-20 - xxxxxxx - lavu 52.22.100 - opt.h + Add AV_OPT_TYPE_DURATION value to AVOptionType enum. + +2013-03-17 - xxxxxx - lavu 52.20.100 - opt.h + Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum. + +2013-03-07 - xxxxxx - lavu 52.18.100 - avstring.h,bprint.h + Add av_escape() and av_bprint_escape() API. + +2013-02-24 - xxxxxx - lavfi 3.41.100 - buffersink.h + Add sample_rates field to AVABufferSinkParams. + +2013-01-17 - a1a707f - lavf 54.61.100 + Add av_codec_get_tag2(). + +2013-01-01 - 2eb2e17 - lavfi 3.34.100 + Add avfilter_get_audio_buffer_ref_from_arrays_channels. + +2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h + Add AVFilterLink.channels, avfilter_link_get_channels() + and avfilter_ref_get_channels(). + +2012-12-15 - 2ada584d - lavc 54.80.100 - avcodec.h + Add pkt_size field to AVFrame. + +2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h + Add the following convenience functions to opt.h: + av_opt_get_image_size + av_opt_get_pixel_fmt + av_opt_get_sample_fmt + av_opt_set_image_size + av_opt_set_pixel_fmt + av_opt_set_sample_fmt + +2012-11-17 - 4cd74c81 - lavu 52.8.100 - bprint.h + Add av_bprint_strftime(). + +2012-11-15 - 92648107 - lavu 52.7.100 - opt.h + Add av_opt_get_key_value(). + +2012-11-13 - 79456652 - lavfi 3.23.100 - avfilter.h + Add channels field to AVFilterBufferRefAudioProps. + +2012-11-03 - 481fdeee - lavu 52.3.100 - opt.h + Add AV_OPT_TYPE_SAMPLE_FMT value to AVOptionType enum. + +2012-10-21 - 6fb2fd8 - lavc 54.68.100 - avcodec.h + lavfi 3.20.100 - avfilter.h + Add AV_PKT_DATA_STRINGS_METADATA side data type, used to transmit key/value + strings between AVPacket and AVFrame, and add metadata field to + AVCodecContext (which shall not be accessed by users; see AVFrame metadata + instead). + +2012-09-27 - a70b493 - lavd 54.3.100 - version.h + Add LIBAVDEVICE_IDENT symbol. + +2012-09-27 - a70b493 - lavfi 3.18.100 - version.h + Add LIBAVFILTER_IDENT symbol. + +2012-09-27 - a70b493 - libswr 0.16.100 - version.h + Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD + and LIBSWRESAMPLE_IDENT symbols. + +2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h + Add av_small_strptime() time parsing function. + + Can be used as a stripped-down replacement for strptime(), on + systems which do not support it. + +2012-08-25 - 2626cc4 - lavf 54.28.100 + Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP instead + of AV_CODEC_ID_TEXT. + +2012-08-13 - 5c0d8bc - lavfi 3.8.100 - avfilter.h + Add avfilter_get_class() function, and priv_class field to AVFilter + struct. + +2012-08-12 - a25346e - lavu 51.69.100 - opt.h + Add AV_OPT_FLAG_FILTERING_PARAM symbol in opt.h. + +2012-07-31 - 23fc4dd - lavc 54.46.100 + Add channels field to AVFrame. + +2012-07-30 - f893904 - lavu 51.66.100 + Add av_get_channel_description() + and av_get_standard_channel_layout() functions. + +2012-07-21 - 016a472 - lavc 54.43.100 + Add decode_error_flags field to AVFrame. + +2012-07-20 - b062936 - lavf 54.18.100 + Add avformat_match_stream_specifier() function. + +2012-07-14 - f49ec1b - lavc 54.38.100 - avcodec.h + Add metadata to AVFrame, and the accessor functions + av_frame_get_metadata() and av_frame_set_metadata(). + +2012-07-10 - 0e003d8 - lavc 54.33.100 + Add av_fast_padded_mallocz(). + +2012-07-10 - 21d5609 - lavfi 3.2.0 - avfilter.h + Add init_opaque() callback to AVFilter struct. + +2012-06-26 - e6674e4 - lavu 51.63.100 - imgutils.h + Add functions to libavutil/imgutils.h: + av_image_get_buffer_size() + av_image_fill_arrays() + av_image_copy_to_buffer() + +2012-06-24 - c41899a - lavu 51.62.100 - version.h + version moved from avutil.h to version.h + +2012-04-11 - 359abb1 - lavu 51.58.100 - error.h + Add av_make_error_string() and av_err2str() utilities to + libavutil/error.h. + +2012-06-05 - 62b39d4 - lavc 54.24.100 + Add pkt_duration field to AVFrame. + +2012-05-24 - f2ee065 - lavu 51.54.100 + Move AVPALETTE_SIZE and AVPALETTE_COUNT macros from + libavcodec/avcodec.h to libavutil/pixfmt.h. + +2012-05-14 - 94a9ac1 - lavf 54.5.100 + Add av_guess_sample_aspect_ratio() function. + +2012-04-20 - 65fa7bc - lavfi 2.70.100 + Add avfilter_unref_bufferp() to avfilter.h. + +2012-04-13 - 162e400 - lavfi 2.68.100 + Install libavfilter/asrc_abuffer.h public header. + +2012-03-26 - a67d9cf - lavfi 2.66.100 + Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions. + +2013-xx-xx - lavu 52.9.0 - pixdesc.h + Add av_pix_fmt_count_planes() function for counting planes in a pixel format. + +2013-xx-xx - lavfi 3.6.0 + Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count. + +2013-03-xx - Reference counted buffers - lavu 52.8.0, lavc 55.0.0, lavf 55.0.0, +lavd 54.0.0, lavfi 3.5.0 + xxxxxxx, xxxxxxx - add a new API for reference counted buffers and buffer + pools (new header libavutil/buffer.h). + xxxxxxx - add AVPacket.buf to allow reference counting for the AVPacket data. + Add av_packet_from_data() function for constructing packets from + av_malloc()ed data. + xxxxxxx - move AVFrame from lavc to lavu (new header libavutil/frame.h), add + AVFrame.buf/extended_buf to allow reference counting for the AVFrame + data. Add new API for working with reference-counted AVFrames. + xxxxxxx - add the refcounted_frames field to AVCodecContext to make audio and + video decoders return reference-counted frames. Add get_buffer2() + callback to AVCodecContext which allocates reference-counted frames. + Add avcodec_default_get_buffer2() as the default get_buffer2() + implementation. + Deprecate AVCodecContext.get_buffer() / release_buffer() / + reget_buffer(), avcodec_default_get_buffer(), + avcodec_default_reget_buffer(), avcodec_default_release_buffer(). + Remove avcodec_default_free_buffers(), which should not have ever + been called from outside of lavc. + Deprecate the following AVFrame fields: + * base -- is now stored in AVBufferRef + * reference, type, buffer_hints -- are unnecessary in the new API + * hwaccel_picture_private, owner, thread_opaque -- should not + have been acessed from outside of lavc + * qscale_table, qstride, qscale_type, mbskip_table, motion_val, + mb_type, dct_coeff, ref_index -- mpegvideo-specific tables, + which are not exported anymore. + xxxxxxx - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add + av_buffersrc_add_frame(), deprecate av_buffersrc_buffer(). + Add av_buffersink_get_frame() and av_buffersink_get_samples(), + deprecate av_buffersink_read() and av_buffersink_read_samples(). + Deprecate AVFilterBufferRef and all functions for working with it. + +2013-xx-xx - xxxxxxx - lavu 52.8.0 - avstring.h + Add av_isdigit, av_isgraph, av_isspace, av_isxdigit. + +2013-xx-xx - xxxxxxx - lavfi 3.4.0 - avfiltergraph.h + Add resample_lavr_opts to AVFilterGraph for setting libavresample options + for auto-inserted resample filters. + +2013-xx-xx - xxxxxxx - lavu 52.7.0 - dict.h + Add av_dict_parse_string() to set multiple key/value pairs at once from a + string. + +2013-01-xx - xxxxxxx - lavu 52.6.0 - avstring.h + Add av_strnstr() + +2013-01-xx - xxxxxxx - lavu 52.5.0 - hmac.h + Add AVHMAC. + +2013-01-13 - xxxxxxx - lavc 54.87.100 / 54.36.0 - vdpau.h + Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding. + +2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h + Add AV_PIX_FMT_VDPAU flag. + +2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0 + Add avresample_set_channel_mapping() for input channel reordering, + duplication, and silencing. + +2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h + Add av_basename() and av_dirname(). + +2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h + Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated. + +2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h + Add av_ctz() for trailing zero bit count + +2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h + Add AVERROR_EXPERIMENTAL + +2012-10-12 - a33ed6b / d2fcb35 - lavu 51.76.100 / 51.44.0 - pixdesc.h + Add functions for accessing pixel format descriptors. + Accessing the av_pix_fmt_descriptors array directly is now + deprecated. + +2012-10-11 - f391e40 / 9a92aea - lavu 51.75.100 / 51.43.0 - aes.h, md5.h, sha.h, tree.h + Add functions for allocating the opaque contexts for the algorithms, + +2012-10-10 - de31814 / b522000 - lavf 54.32.100 / 54.18.0 - avio.h + Add avio_closep to complement avio_close. + +2012-10-08 - ae77266 / 78071a1 - lavu 51.74.100 / 51.42.0 - pixfmt.h + Rename PixelFormat to AVPixelFormat and all PIX_FMT_* to AV_PIX_FMT_*. + To provide backwards compatibility, PixelFormat is now #defined as + AVPixelFormat. + Note that this can break user code that includes pixfmt.h and uses the + 'PixelFormat' identifier. Such code should either #undef PixelFormat + or stop using the PixelFormat name. + +2012-10-05 - 55c49af / e7ba5b1 - lavr 1.0.0 - avresample.h + Data planes parameters to avresample_convert() and + avresample_read() are now uint8_t** instead of void**. + Libavresample is now stable. + +2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h + Add avcodec_free_frame(). This function must now + be used for freeing an AVFrame. + +2012-09-12 - e3e09f2 / 8919fee - lavu 51.73.100 / 51.41.0 - audioconvert.h + Added AV_CH_LOW_FREQUENCY_2 channel mask value. + +2012-09-04 - b21b5b0 / 686a329 - lavu 51.71.100 / 51.40.0 - opt.h + Reordered the fields in default_val in AVOption, changed which + default_val field is used for which AVOptionType. + +2012-08-30 - 98298eb / a231832 - lavc 54.54.101 / 54.26.1 - avcodec.h + Add codec descriptor properties AV_CODEC_PROP_LOSSY and + AV_CODEC_PROP_LOSSLESS. + +2012-08-18 - lavc 54.26 - avcodec.h + Add codec descriptors for accessing codec properties without having + to refer to a specific decoder or encoder. + + f5f3684 / c223d79 - Add an AVCodecDescriptor struct and functions + avcodec_descriptor_get() and avcodec_descriptor_next(). + f5f3684 / 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY. + 6c180b3 / 91e59fe - Add avcodec_descriptor_get_by_name(). + +2012-08-08 - f5f3684 / 987170c - lavu 51.68.100 / 51.38.0 - dict.h + Add av_dict_count(). + +2012-08-07 - 7a72695 / 104e10f - lavc 54.51.100 / 54.25.0 - avcodec.h + Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*. + To provide backwards compatibility, CodecID is now #defined as AVCodecID. + Note that this can break user code that includes avcodec.h and uses the + 'CodecID' identifier. Such code should either #undef CodecID or stop using the + CodecID name. + +2012-08-03 - e776ee8 / 239fdf1 - lavu 51.66.101 / 51.37.1 - cpu.h + lsws 2.1.1 - swscale.h + Rename AV_CPU_FLAG_MMX2 ---> AV_CPU_FLAG_MMXEXT. + Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT. + +2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h + Add AVFMT_FLAG_NOBUFFER for low latency use cases. + +2012-07-10 - 5fade8a - lavu 51.37.0 + Add av_malloc_array() and av_mallocz_array() + +2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0 + Add av_usleep() + +2012-06-20 - 4da42eb / ae0a301 - lavu 51.60.100 / 51.33.0 + Move av_gettime() to libavutil, add libavutil/time.h + +2012-06-09 - 82edf67 / 3971be0 - lavr 0.0.3 + Add a parameter to avresample_build_matrix() for Dolby/DPLII downmixing. + +2012-06-12 - c7b9eab / 9baeff9 - lavfi 2.79.100 / 2.23.0 - avfilter.h + Add AVFilterContext.nb_inputs/outputs. Deprecate + AVFilterContext.input/output_count. + +2012-06-12 - c7b9eab / 84b9fbe - lavfi 2.79.100 / 2.22.0 - avfilter.h + Add avfilter_pad_get_type() and avfilter_pad_get_name(). Those + should now be used instead of accessing AVFilterPad members + directly. + +2012-06-12 - 3630a07 / b0f0dfc - lavu 51.57.100 / 51.32.0 - audioconvert.h + Add av_get_channel_layout_channel_index(), av_get_channel_name() + and av_channel_layout_extract_channel(). + +2012-05-25 - 53ce990 / 154486f - lavu 51.55.100 / 51.31.0 - opt.h + Add av_opt_set_bin() + +2012-05-15 - lavfi 2.74.100 / 2.17.0 + Add support for audio filters + 61930bd / ac71230, 1cbf7fb / a2cd9be - add video/audio buffer sink in a new installed + header buffersink.h + 1cbf7fb / 720c6b7 - add av_buffersrc_write_frame(), deprecate + av_vsrc_buffer_add_frame() + 61930bd / ab16504 - add avfilter_copy_buf_props() + 61930bd / 9453c9e - add extended_data to AVFilterBuffer + 61930bd / 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays() + +2012-05-09 - lavu 51.53.100 / 51.30.0 - samplefmt.h + 61930bd / 142e740 - add av_samples_copy() + 61930bd / 6d7f617 - add av_samples_set_silence() + +2012-05-09 - 61930bd / a5117a2 - lavc 54.21.101 / 54.13.1 + For audio formats with fixed frame size, the last frame + no longer needs to be padded with silence, libavcodec + will handle this internally (effectively all encoders + behave as if they had CODEC_CAP_SMALL_LAST_FRAME set). + +2012-05-07 - 653d117 / 828bd08 - lavc 54.20.100 / 54.13.0 - avcodec.h + Add sample_rate and channel_layout fields to AVFrame. + +2012-05-01 - 2330eb1 / 4010d72 - lavr 0.0.1 + Change AV_MIX_COEFF_TYPE_Q6 to AV_MIX_COEFF_TYPE_Q8. + +2012-04-25 - e890b68 / 3527a73 - lavu 51.48.100 / 51.29.0 - cpu.h + Add av_parse_cpu_flags() + +2012-04-24 - 3ead79e / c8af852 - lavr 0.0.0 + Add libavresample audio conversion library + +2012-04-20 - 3194ab7 / 0c0d1bc - lavu 51.47.100 / 51.28.0 - audio_fifo.h + Add audio FIFO functions: + av_audio_fifo_free() + av_audio_fifo_alloc() + av_audio_fifo_realloc() + av_audio_fifo_write() + av_audio_fifo_read() + av_audio_fifo_drain() + av_audio_fifo_reset() + av_audio_fifo_size() + av_audio_fifo_space() + +2012-04-14 - lavfi 2.70.100 / 2.16.0 - avfiltergraph.h + 7432bcf / d7bcc71 Add avfilter_graph_parse2(). + +2012-04-08 - 6bfb304 / 4d693b0 - lavu 51.46.100 / 51.27.0 - samplefmt.h + Add av_get_packed_sample_fmt() and av_get_planar_sample_fmt() + +2012-03-21 - b75c67d - lavu 51.43.100 + Add bprint.h for bprint API. + +2012-02-21 - 9cbf17e - lavc 54.4.100 + Add av_get_pcm_codec() function. + +2012-02-16 - 560b224 - libswr 0.7.100 + Add swr_set_matrix() function. + +2012-02-09 - c28e7af - lavu 51.39.100 + Add a new installed header libavutil/timestamp.h with timestamp + utilities. + +2012-02-06 - 70ffda3 - lavu 51.38.100 + Add av_parse_ratio() function to parseutils.h. + +2012-02-06 - 70ffda3 - lavu 51.38.100 + Add AV_LOG_MAX_OFFSET macro to log.h. + +2012-02-02 - 0eaa123 - lavu 51.37.100 + Add public timecode helpers. + +2012-01-24 - 0c3577b - lavfi 2.60.100 + Add avfilter_graph_dump. + +2012-03-20 - 0ebd836 / 3c90cc2 - lavfo 54.2.0 + Deprecate av_read_packet(), use av_read_frame() with + AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN in AVFormatContext.flags + +2012-03-05 - lavc 54.10.100 / 54.8.0 + f095391 / 6699d07 Add av_get_exact_bits_per_sample() + f095391 / 9524cf7 Add av_get_audio_frame_duration() + +2012-03-04 - 2af8f2c / 44fe77b - lavc 54.8.100 / 54.7.0 - avcodec.h + Add av_codec_is_encoder/decoder(). + +2012-03-01 - 1eb7f39 / 442c132 - lavc 54.5.100 / 54.3.0 - avcodec.h + Add av_packet_shrink_side_data. + +2012-02-29 - 79ae084 / dd2a4bc - lavf 54.2.100 / 54.2.0 - avformat.h + Add AVStream.attached_pic and AV_DISPOSITION_ATTACHED_PIC, + used for dealing with attached pictures/cover art. + +2012-02-25 - 305e4b3 / c9bca80 - lavu 51.41.100 / 51.24.0 - error.h + Add AVERROR_UNKNOWN + NOTE: this was backported to 0.8 + +2012-02-20 - eadd426 / e9cda85 - lavc 54.2.100 / 54.2.0 + Add duration field to AVCodecParserContext + +2012-02-20 - eadd426 / 0b42a93 - lavu 51.40.100 / 51.23.1 - mathematics.h + Add av_rescale_q_rnd() + +2012-02-08 - f2b20b7 / 38d5533 - lavu 51.38.101 / 51.22.1 - pixdesc.h + Add PIX_FMT_PSEUDOPAL flag. + +2012-02-08 - f2b20b7 / 52f82a1 - lavc 54.2.100 / 54.1.0 + Add avcodec_encode_video2() and deprecate avcodec_encode_video(). + +2012-02-01 - 4c677df / 316fc74 - lavc 54.1.0 + Add av_fast_padded_malloc() as alternative for av_realloc() when aligned + memory is required. The buffer will always have FF_INPUT_BUFFER_PADDING_SIZE + zero-padded bytes at the end. + +2012-01-31 - a369a6b / dd6d3b0 - lavf 54.1.0 + Add avformat_get_riff_video_tags() and avformat_get_riff_audio_tags(). + NOTE: this was backported to 0.8 + +2012-01-31 - a369a6b / af08d9a - lavc 54.1.0 + Add avcodec_is_open() function. + NOTE: this was backported to 0.8 + +2012-01-30 - 151ecc2 / 8b93312 - lavu 51.36.100 / 51.22.0 - intfloat.h + Add a new installed header libavutil/intfloat.h with int/float punning + functions. + NOTE: this was backported to 0.8 + +2012-01-25 - lavf 53.31.100 / 53.22.0 + 3c5fe5b / f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible + buffered data within a muxer. Added AVFMT_ALLOW_FLUSH for + muxers supporting it (av_write_frame makes sure it is called + only for muxers with this flag). + +2012-01-15 - lavc 53.56.105 / 53.34.0 + New audio encoding API: + 67f5650 / b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio + encoders. + 67f5650 / 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function. + 67f5650 / b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio(). + Add AVCodec.encode2(). + +2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0 + Add a new installed header -- libavfilter/version.h -- with version macros. + +2011-12-08 - a502939 - lavfi 2.52.0 + Add av_buffersink_poll_frame() to buffersink.h. + +2011-12-08 - 26c6fec - lavu 51.31.0 + Add av_log_format_line. + +2011-12-03 - 976b095 - lavu 51.30.0 + Add AVERROR_BUG. + +2011-11-24 - 573ffbb - lavu 51.28.1 + Add av_get_alt_sample_fmt() to samplefmt.h. + +2011-11-03 - 96949da - lavu 51.23.0 + Add av_strcasecmp() and av_strncasecmp() to avstring.h. + +2011-10-20 - b35e9e1 - lavu 51.22.0 + Add av_strtok() to avstring.h. + +2012-01-03 - ad1c8dd / b73ec05 - lavu 51.34.100 / 51.21.0 + Add av_popcount64 + +2011-12-18 - 7c29313 / 8400b12 - lavc 53.46.1 / 53.28.1 + Deprecate AVFrame.age. The field is unused. + +2011-12-12 - 8bc7fe4 / 5266045 - lavf 53.25.0 / 53.17.0 + Add avformat_close_input(). + Deprecate av_close_input_file() and av_close_input_stream(). + +2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0 + Add nb_samples and extended_data fields to AVFrame. + Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE. + Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4(). + avcodec_decode_audio4() writes output samples to an AVFrame, which allows + audio decoders to use get_buffer(). + +2011-12-04 - e4de716 / 560f773 - lavc 53.40.0 / 53.24.0 + Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump. + Change AVPicture.data[4]/linesize[4] to [8] at next major bump. + Change AVCodecContext.error[4] to [8] at next major bump. + Add AV_NUM_DATA_POINTERS to simplify the bump transition. + +2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0 + Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and + av_samples_alloc(), to samplefmt.h. + +2011-11-23 - 8e576d5 / 8889cc4 - lavu 51.27.0 / 51.17.0 + Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h. + +2011-11-19 - dbb38bc / f3a29b7 - lavc 53.36.0 / 53.21.0 + Move some AVCodecContext fields to a new private struct, AVCodecInternal, + which is accessed from a new field, AVCodecContext.internal. + - fields moved: + AVCodecContext.internal_buffer --> AVCodecInternal.buffer + AVCodecContext.internal_buffer_count --> AVCodecInternal.buffer_count + AVCodecContext.is_copy --> AVCodecInternal.is_copy + +2011-11-16 - 8709ba9 / 6270671 - lavu 51.26.0 / 51.16.0 + Add av_timegm() + +2011-11-13 - lavf 53.21.0 / 53.15.0 + New interrupt callback API, allowing per-AVFormatContext/AVIOContext + interrupt callbacks. + 5f268ca / 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to + AVFormatContext. + 5f268ca / 1dee0ac Add avio_open2() with additional parameters. Those are + an interrupt callback and an options AVDictionary. + This will allow passing AVOptions to protocols after lavf + 54.0. + +2011-11-06 - 13b7781 / ba04ecf - lavu 51.24.0 / 51.14.0 + Add av_strcasecmp() and av_strncasecmp() to avstring.h. + +2011-11-06 - 13b7781 / 07b172f - lavu 51.24.0 / 51.13.0 + Add av_toupper()/av_tolower() + +2011-11-05 - d8cab5c / b6d08f4 - lavf 53.19.0 / 53.13.0 + Add avformat_network_init()/avformat_network_deinit() + +2011-10-27 - 6faf0a2 / 512557b - lavc 53.24.0 / 53.15.0 + Remove avcodec_parse_frame. + Deprecate AVCodecContext.parse_only and CODEC_CAP_PARSE_ONLY. + +2011-10-19 - d049257 / 569129a - lavf 53.17.0 / 53.10.0 + Add avformat_new_stream(). Deprecate av_new_stream(). + +2011-10-13 - 91eb1b1 / b631fba - lavf 53.16.0 / 53.9.0 + Add AVFMT_NO_BYTE_SEEK AVInputFormat flag. + +2011-10-12 - lavu 51.21.0 / 51.12.0 + AVOptions API rewrite. + + - f884ef0 / 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_* + - new setting/getting functions with slightly different semantics: + f884ef0 / dac66da av_set_string3 -> av_opt_set + av_set_double -> av_opt_set_double + av_set_q -> av_opt_set_q + av_set_int -> av_opt_set_int + + f884ef0 / 41d9d51 av_get_string -> av_opt_get + av_get_double -> av_opt_get_double + av_get_q -> av_opt_get_q + av_get_int -> av_opt_get_int + + - f884ef0 / 8c5dcaa trivial rename av_next_option -> av_opt_next + - f884ef0 / 641c7af new functions - av_opt_child_next, av_opt_child_class_next + and av_opt_find2() + +2011-09-22 - a70e787 - lavu 51.17.0 + Add av_x_if_null(). + +2011-09-18 - 645cebb - lavc 53.16.0 + Add showall flag2 + +2011-09-16 - ea8de10 - lavfi 2.42.0 + Add avfilter_all_channel_layouts. + +2011-09-16 - 9899037 - lavfi 2.41.0 + Rename avfilter_all_* function names to avfilter_make_all_*. + + In particular, apply the renames: + avfilter_all_formats -> avfilter_make_all_formats + avfilter_all_channel_layouts -> avfilter_make_all_channel_layouts + avfilter_all_packing_formats -> avfilter_make_all_packing_formats + +2011-09-12 - 4381bdd - lavfi 2.40.0 + Change AVFilterBufferRefAudioProps.sample_rate type from uint32_t to int. + +2011-09-12 - 2c03174 - lavfi 2.40.0 + Simplify signature for avfilter_get_audio_buffer(), make it + consistent with avfilter_get_video_buffer(). + +2011-09-06 - 4f7dfe1 - lavfi 2.39.0 + Rename libavfilter/vsink_buffer.h to libavfilter/buffersink.h. + +2011-09-06 - c4415f6 - lavfi 2.38.0 + Unify video and audio sink API. + + In particular, add av_buffersink_get_buffer_ref(), deprecate + av_vsink_buffer_get_video_buffer_ref() and change the value for the + opaque field passed to the abuffersink init function. + +2011-09-04 - 61e2e29 - lavu 51.16.0 + Add av_asprintf(). + +2011-08-22 - dacd827 - lavf 53.10.0 + Add av_find_program_from_stream(). + +2011-08-20 - 69e2c1a - lavu 51.13.0 + Add av_get_media_type_string(). + +2011-09-03 - 1889c67 / fb4ca26 - lavc 53.13.0 + lavf 53.11.0 + lsws 2.1.0 + Add {avcodec,avformat,sws}_get_class(). + +2011-08-03 - 1889c67 / c11fb82 - lavu 51.15.0 + Add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find() function. + +2011-08-14 - 323b930 - lavu 51.12.0 + Add av_fifo_peek2(), deprecate av_fifo_peek(). + +2011-08-26 - lavu 51.14.0 / 51.9.0 + - 976a8b2 / add41de..976a8b2 / abc78a5 Do not include intfloat_readwrite.h, + mathematics.h, rational.h, pixfmt.h, or log.h from avutil.h. + +2011-08-16 - 27fbe31 / 48f9e45 - lavf 53.11.0 / 53.8.0 + Add avformat_query_codec(). + +2011-08-16 - 27fbe31 / bca06e7 - lavc 53.11.0 + Add avcodec_get_type(). + +2011-08-06 - 0cb233c / 2f63440 - lavf 53.7.0 + Add error_recognition to AVFormatContext. + +2011-08-02 - 1d186e9 / 9d39cbf - lavc 53.9.1 + Add AV_PKT_FLAG_CORRUPT AVPacket flag. + +2011-07-16 - b57df29 - lavfi 2.27.0 + Add audio packing negotiation fields and helper functions. + + In particular, add AVFilterPacking enum, planar, in_packings and + out_packings fields to AVFilterLink, and the functions: + avfilter_set_common_packing_formats() + avfilter_all_packing_formats() + +2011-07-10 - 3602ad7 / a67c061 - lavf 53.6.0 + Add avformat_find_stream_info(), deprecate av_find_stream_info(). + NOTE: this was backported to 0.7 + +2011-07-10 - 3602ad7 / 0b950fe - lavc 53.8.0 + Add avcodec_open2(), deprecate avcodec_open(). + NOTE: this was backported to 0.7 + + Add avcodec_alloc_context3. Deprecate avcodec_alloc_context() and + avcodec_alloc_context2(). + +2011-07-01 - b442ca6 - lavf 53.5.0 - avformat.h + Add function av_get_output_timestamp(). + +2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h + Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum. + +2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h + Add layout negotiation fields and helper functions. + + In particular, add in_chlayouts and out_chlayouts to AVFilterLink, + and the functions: + avfilter_set_common_sample_formats() + avfilter_set_common_channel_layouts() + avfilter_all_channel_layouts() + +2011-06-19 - 527ca39 - lavfi 2.22.0 - AVFilterFormats + Change type of AVFilterFormats.formats from int * to int64_t *, + and update formats handling API accordingly. + + avfilter_make_format_list() still takes a int32_t array and converts + it to int64_t. A new function, avfilter_make_format64_list(), that + takes int64_t arrays has been added. + +2011-06-19 - 44f669e - lavfi 2.21.0 - vsink_buffer.h + Add video sink buffer and vsink_buffer.h public header. + +2011-06-12 - 9fdf772 - lavfi 2.18.0 - avcodec.h + Add avfilter_get_video_buffer_ref_from_frame() function in + libavfilter/avcodec.h. + +2011-06-12 - c535494 - lavfi 2.17.0 - avfiltergraph.h + Add avfilter_inout_alloc() and avfilter_inout_free() functions. + +2011-06-12 - 6119b23 - lavfi 2.16.0 - avfilter_graph_parse() + Change avfilter_graph_parse() signature. + +2011-06-23 - 686959e / 67e9ae1 - lavu 51.10.0 / 51.8.0 - attributes.h + Add av_printf_format(). + +2011-06-16 - 2905e3f / 05e84c9, 2905e3f / 25de595 - lavf 53.4.0 / 53.2.0 - avformat.h + Add avformat_open_input and avformat_write_header(). + Deprecate av_open_input_stream, av_open_input_file, + AVFormatParameters and av_write_header. + +2011-06-16 - 2905e3f / 7e83e1c, 2905e3f / dc59ec5 - lavu 51.9.0 / 51.7.0 - opt.h + Add av_opt_set_dict() and av_opt_find(). + Deprecate av_find_opt(). + Add AV_DICT_APPEND flag. + +2011-06-10 - 45fb647 / cb7c11c - lavu 51.6.0 - opt.h + Add av_opt_flag_is_set(). + +2011-06-10 - c381960 - lavfi 2.15.0 - avfilter_get_audio_buffer_ref_from_arrays + Add avfilter_get_audio_buffer_ref_from_arrays() to avfilter.h. + +2011-06-09 - f9ecb84 / d9f80ea - lavu 51.8.0 - AVMetadata + Move AVMetadata from lavf to lavu and rename it to + AVDictionary -- new installed header dict.h. + All av_metadata_* functions renamed to av_dict_*. + +2011-06-07 - d552f61 / a6703fa - lavu 51.8.0 - av_get_bytes_per_sample() + Add av_get_bytes_per_sample() in libavutil/samplefmt.h. + Deprecate av_get_bits_per_sample_fmt(). + +2011-06-05 - f956924 / b39b062 - lavu 51.8.0 - opt.h + Add av_opt_free convenience function. + +2011-06-06 - 95a0242 - lavfi 2.14.0 - AVFilterBufferRefAudioProps + Remove AVFilterBufferRefAudioProps.size, and use nb_samples in + avfilter_get_audio_buffer() and avfilter_default_get_audio_buffer() in + place of size. + +2011-06-06 - 0bc2cca - lavu 51.6.0 - av_samples_alloc() + Switch nb_channels and nb_samples parameters order in + av_samples_alloc(). + +2011-06-06 - e1c7414 - lavu 51.5.0 - av_samples_* + Change the data layout created by av_samples_fill_arrays() and + av_samples_alloc(). + +2011-06-06 - 27bcf55 - lavfi 2.13.0 - vsrc_buffer.h + Make av_vsrc_buffer_add_video_buffer_ref() accepts an additional + flags parameter in input. + +2011-06-03 - e977ca2 - lavfi 2.12.0 - avfilter_link_free() + Add avfilter_link_free() function. + +2011-06-02 - 5ad38d9 - lavu 51.4.0 - av_force_cpu_flags() + Add av_cpu_flags() in libavutil/cpu.h. + +2011-05-28 - e71f260 - lavu 51.3.0 - pixdesc.h + Add av_get_pix_fmt_name() in libavutil/pixdesc.h, and deprecate + avcodec_get_pix_fmt_name() in libavcodec/avcodec.h in its favor. + +2011-05-25 - 39e4206 / 30315a8 - lavf 53.3.0 - avformat.h + Add fps_probe_size to AVFormatContext. + +2011-05-22 - 5ecdfd0 - lavf 53.2.0 - avformat.h + Introduce avformat_alloc_output_context2() and deprecate + avformat_alloc_output_context(). + +2011-05-22 - 83db719 - lavfi 2.10.0 - vsrc_buffer.h + Make libavfilter/vsrc_buffer.h public. + +2011-05-19 - c000a9f - lavfi 2.8.0 - avcodec.h + Add av_vsrc_buffer_add_frame() to libavfilter/avcodec.h. + +2011-05-14 - 9fdf772 - lavfi 2.6.0 - avcodec.h + Add avfilter_get_video_buffer_ref_from_frame() to libavfilter/avcodec.h. + +2011-05-18 - 75a37b5 / 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt + Add request_sample_fmt field to AVCodecContext. + +2011-05-10 - 59eb12f / 188dea1 - lavc 53.6.0 - avcodec.h + Deprecate AVLPCType and the following fields in + AVCodecContext: lpc_coeff_precision, prediction_order_method, + min_partition_order, max_partition_order, lpc_type, lpc_passes. + Corresponding FLAC encoder options should be used instead. + +2011-05-07 - 9fdf772 - lavfi 2.5.0 - avcodec.h + Add libavfilter/avcodec.h header and avfilter_copy_frame_props() + function. + +2011-05-07 - 18ded93 - lavc 53.5.0 - AVFrame + Add format field to AVFrame. + +2011-05-07 - 22333a6 - lavc 53.4.0 - AVFrame + Add width and height fields to AVFrame. + +2011-05-01 - 35fe66a - lavfi 2.4.0 - avfilter.h + Rename AVFilterBufferRefVideoProps.pixel_aspect to + sample_aspect_ratio. + +2011-05-01 - 77e9dee - lavc 53.3.0 - AVFrame + Add a sample_aspect_ratio field to AVFrame. + +2011-05-01 - 1ba5727 - lavc 53.2.0 - AVFrame + Add a pkt_pos field to AVFrame. + +2011-04-29 - 35ceaa7 - lavu 51.2.0 - mem.h + Add av_dynarray_add function for adding + an element to a dynamic array. + +2011-04-26 - d7e5aeb / bebe72f - lavu 51.1.0 - avutil.h + Add AVPictureType enum and av_get_picture_type_char(), deprecate + FF_*_TYPE defines and av_get_pict_type_char() defined in + libavcodec/avcodec.h. + +2011-04-26 - d7e5aeb / 10d3940 - lavfi 2.3.0 - avfilter.h + Add pict_type and key_frame fields to AVFilterBufferRefVideo. + +2011-04-26 - d7e5aeb / 7a11c82 - lavfi 2.2.0 - vsrc_buffer + Add sample_aspect_ratio fields to vsrc_buffer arguments + +2011-04-21 - 8772156 / 94f7451 - lavc 53.1.0 - avcodec.h + Add CODEC_CAP_SLICE_THREADS for codecs supporting sliced threading. + +2011-04-15 - lavc 52.120.0 - avcodec.h + AVPacket structure got additional members for passing side information: + c407984 / 4de339e introduce side information for AVPacket + c407984 / 2d8591c make containers pass palette change in AVPacket + +2011-04-12 - lavf 52.107.0 - avio.h + Avio cleanup, part II - deprecate the entire URLContext API: + c55780d / 175389c add avio_check as a replacement for url_exist + 9891004 / ff1ec0c add avio_pause and avio_seek_time as replacements + for _av_url_read_fseek/fpause + d4d0932 / cdc6a87 deprecate av_protocol_next(), avio_enum_protocols + should be used instead. + c88caa5 / 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb. + c88caa5 / f87b1b3 rename open flags: URL_* -> AVIO_* + d4d0932 / f8270bb add avio_enum_protocols. + d4d0932 / 5593f03 deprecate URLProtocol. + d4d0932 / c486dad deprecate URLContext. + d4d0932 / 026e175 deprecate the typedef for URLInterruptCB + c88caa5 / 8e76a19 deprecate av_register_protocol2. + 11d7841 / b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME + 11d7841 / 1305d93 deprecate av_url_read_seek + 11d7841 / fa104e1 deprecate av_url_read_pause + 434f248 / 727c7aa deprecate url_get_filename(). + 434f248 / 5958df3 deprecate url_max_packet_size(). + 434f248 / 1869ea0 deprecate url_get_file_handle(). + 434f248 / 32a97d4 deprecate url_filesize(). + 434f248 / e52a914 deprecate url_close(). + 434f248 / 58a48c6 deprecate url_seek(). + 434f248 / 925e908 deprecate url_write(). + 434f248 / dce3756 deprecate url_read_complete(). + 434f248 / bc371ac deprecate url_read(). + 434f248 / 0589da0 deprecate url_open(). + 434f248 / 62eaaea deprecate url_connect. + 434f248 / 5652bb9 deprecate url_alloc. + 434f248 / 333e894 deprecate url_open_protocol + 434f248 / e230705 deprecate url_poll and URLPollEntry + +2011-04-08 - lavf 52.106.0 - avformat.h + Minor avformat.h cleanup: + d4d0932 / a9bf9d8 deprecate av_guess_image2_codec + d4d0932 / c3675df rename avf_sdp_create->av_sdp_create + +2011-04-03 - lavf 52.105.0 - avio.h + Large-scale renaming/deprecating of AVIOContext-related functions: + 2cae980 / 724f6a0 deprecate url_fdopen + 2cae980 / 403ee83 deprecate url_open_dyn_packet_buf + 2cae980 / 6dc7d80 rename url_close_dyn_buf -> avio_close_dyn_buf + 2cae980 / b92c545 rename url_open_dyn_buf -> avio_open_dyn_buf + 2cae980 / 8978fed introduce an AVIOContext.seekable field as a replacement for + AVIOContext.is_streamed and url_is_streamed() + 1caa412 / b64030f deprecate get_checksum() + 1caa412 / 4c4427a deprecate init_checksum() + 2fd41c9 / 4ec153b deprecate udp_set_remote_url/get_local_port + 4fa0e24 / 933e90a deprecate av_url_read_fseek/fpause + 4fa0e24 / 8d9769a deprecate url_fileno + 0fecf26 / b7f2fdd rename put_flush_packet -> avio_flush + 0fecf26 / 35f1023 deprecate url_close_buf + 0fecf26 / 83fddae deprecate url_open_buf + 0fecf26 / d9d86e0 rename url_fprintf -> avio_printf + 0fecf26 / 59f65d9 deprecate url_setbufsize + 6947b0c / 3e68b3b deprecate url_ferror + e8bb2e2 deprecate url_fget_max_packet_size + 76aa876 rename url_fsize -> avio_size + e519753 deprecate url_fgetc + 655e45e deprecate url_fgets + a2704c9 rename url_ftell -> avio_tell + e16ead0 deprecate get_strz() in favor of avio_get_str + 0300db8,2af07d3 rename url_fskip -> avio_skip + 6b4aa5d rename url_fseek -> avio_seek + 61840b4 deprecate put_tag + 22a3212 rename url_fopen/fclose -> avio_open/close. + 0ac8e2b deprecate put_nbyte + 77eb550 rename put_byte -> avio_w8 + put_[b/l]e<type> -> avio_w[b/l]<type> + put_buffer -> avio_write + b7effd4 rename get_byte -> avio_r8, + get_[b/l]e<type> -> avio_r[b/l]<type> + get_buffer -> avio_read + b3db9ce deprecate get_partial_buffer + 8d9ac96 rename av_alloc_put_byte -> avio_alloc_context + +2011-03-25 - 27ef7b1 / 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type + Add audio_service_type field to AVCodecContext. + +2011-03-17 - e309fdc - lavu 50.40.0 - pixfmt.h + Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats + +2011-03-02 - 863c471 - lavf 52.103.0 - av_pkt_dump2, av_pkt_dump_log2 + Add new functions av_pkt_dump2, av_pkt_dump_log2 that uses the + source stream timebase for outputting timestamps. Deprecate + av_pkt_dump and av_pkt_dump_log. + +2011-02-20 - e731b8d - lavf 52.102.0 - avio.h + * e731b8d - rename init_put_byte() to ffio_init_context(), deprecating the + original, and move it to a private header so it is no longer + part of our public API. Instead, use av_alloc_put_byte(). + * ae628ec - rename ByteIOContext to AVIOContext. + +2011-02-16 - 09d171b - lavf 52.101.0 - avformat.h + lavu 52.39.0 - parseutils.h + * 610219a - Add av_ prefix to dump_format(). + * f6c7375 - Replace parse_date() in lavf with av_parse_time() in lavu. + * ab0287f - Move find_info_tag from lavf to lavu and add av_prefix to it. + +2011-02-15 - lavu 52.38.0 - merge libavcore + libavcore is merged back completely into libavutil + +2011-02-10 - 55bad0c - lavc 52.113.0 - vbv_delay + Add vbv_delay field to AVCodecContext + +2011-02-14 - 24a83bd - lavf 52.100.0 - AV_DISPOSITION_CLEAN_EFFECTS + Add AV_DISPOSITION_CLEAN_EFFECTS disposition flag. + +2011-02-14 - 910b5b8 - lavfi 1.76.0 - AVFilterLink sample_aspect_ratio + Add sample_aspect_ratio field to AVFilterLink. + +2011-02-10 - 12c14cd - lavf 52.99.0 - AVStream.disposition + Add AV_DISPOSITION_HEARING_IMPAIRED and AV_DISPOSITION_VISUAL_IMPAIRED. + +2011-02-09 - c0b102c - lavc 52.112.0 - avcodec_thread_init() + Deprecate avcodec_thread_init()/avcodec_thread_free() use; instead + set thread_count before calling avcodec_open. + +2011-02-09 - 37b00b4 - lavc 52.111.0 - threading API + Add CODEC_CAP_FRAME_THREADS with new restrictions on get_buffer()/ + release_buffer()/draw_horiz_band() callbacks for appropriate codecs. + Add thread_type and active_thread_type fields to AVCodecContext. + +2011-02-08 - 3940caa - lavf 52.98.0 - av_probe_input_buffer + Add av_probe_input_buffer() to avformat.h for probing format from a + ByteIOContext. + +2011-02-06 - fe174fc - lavf 52.97.0 - avio.h + Add flag for non-blocking protocols: URL_FLAG_NONBLOCK + +2011-02-04 - f124b08 - lavf 52.96.0 - avformat_free_context() + Add avformat_free_context() in avformat.h. + +2011-02-03 - f5b82f4 - lavc 52.109.0 - add CODEC_ID_PRORES + Add CODEC_ID_PRORES to avcodec.h. + +2011-02-03 - fe9a3fb - lavc 52.109.0 - H.264 profile defines + Add defines for H.264 * Constrained Baseline and Intra profiles + +2011-02-02 - lavf 52.95.0 + * 50196a9 - add a new installed header version.h. + * 4efd5cf, dccbd97, 93b78d1 - add several variants of public + avio_{put,get}_str* functions. Deprecate corresponding semi-public + {put,get}_str*. + +2011-02-02 - dfd2a00 - lavu 50.37.0 - log.h + Make av_dlog public. + +2011-01-31 - 7b3ea55 - lavfi 1.76.0 - vsrc_buffer + Add sample_aspect_ratio fields to vsrc_buffer arguments + +2011-01-31 - 910b5b8 - lavfi 1.75.0 - AVFilterLink sample_aspect_ratio + Add sample_aspect_ratio field to AVFilterLink. + +2011-01-15 - a242ac3 - lavfi 1.74.0 - AVFilterBufferRefAudioProps + Rename AVFilterBufferRefAudioProps.samples_nb to nb_samples. + +2011-01-14 - 7f88a5b - lavf 52.93.0 - av_metadata_copy() + Add av_metadata_copy() in avformat.h. + +2011-01-07 - 81c623f - lavc 52.107.0 - deprecate reordered_opaque + Deprecate reordered_opaque in favor of pkt_pts/dts. + +2011-01-07 - 1919fea - lavc 52.106.0 - pkt_dts + Add pkt_dts to AVFrame, this will in the future allow multithreading decoders + to not mess up dts. + +2011-01-07 - 393cbb9 - lavc 52.105.0 - pkt_pts + Add pkt_pts to AVFrame. + +2011-01-07 - 060ec0a - lavc 52.104.0 - av_get_profile_name() + Add av_get_profile_name to libavcodec/avcodec.h. + +2010-12-27 - 0ccabee - lavfi 1.71.0 - AV_PERM_NEG_LINESIZES + Add AV_PERM_NEG_LINESIZES in avfilter.h. + +2010-12-27 - 9128ae0 - lavf 52.91.0 - av_find_best_stream() + Add av_find_best_stream to libavformat/avformat.h. + +2010-12-27 - 107a7e3 - lavf 52.90.0 + Add AVFMT_NOSTREAMS flag for formats with no streams, + like e.g. text metadata. + +2010-12-22 - 0328b9e - lavu 50.36.0 - file.h + Add functions av_file_map() and av_file_unmap() in file.h. + +2010-12-19 - 0bc55f5 - lavu 50.35.0 - error.h + Add "not found" error codes: + AVERROR_DEMUXER_NOT_FOUND + AVERROR_MUXER_NOT_FOUND + AVERROR_DECODER_NOT_FOUND + AVERROR_ENCODER_NOT_FOUND + AVERROR_PROTOCOL_NOT_FOUND + AVERROR_FILTER_NOT_FOUND + AVERROR_BSF_NOT_FOUND + AVERROR_STREAM_NOT_FOUND + +2010-12-09 - c61cdd0 - lavcore 0.16.0 - avcore.h + Move AV_NOPTS_VALUE, AV_TIME_BASE, AV_TIME_BASE_Q symbols from + avcodec.h to avcore.h. + +2010-12-04 - 16cfc96 - lavc 52.98.0 - CODEC_CAP_NEG_LINESIZES + Add CODEC_CAP_NEG_LINESIZES codec capability flag in avcodec.h. + +2010-12-04 - bb4afa1 - lavu 50.34.0 - av_get_pix_fmt_string() + Deprecate avcodec_pix_fmt_string() in favor of + pixdesc.h/av_get_pix_fmt_string(). + +2010-12-04 - 4da12e3 - lavcore 0.15.0 - av_image_alloc() + Add av_image_alloc() to libavcore/imgutils.h. + +2010-12-02 - 037be76 - lavfi 1.67.0 - avfilter_graph_create_filter() + Add function avfilter_graph_create_filter() in avfiltergraph.h. + +2010-11-25 - 4723bc2 - lavfi 1.65.0 - avfilter_get_video_buffer_ref_from_arrays() + Add function avfilter_get_video_buffer_ref_from_arrays() in + avfilter.h. + +2010-11-21 - 176a615 - lavcore 0.14.0 - audioconvert.h + Add a public audio channel API in audioconvert.h, and deprecate the + corresponding functions in libavcodec: + avcodec_get_channel_name() + avcodec_get_channel_layout() + avcodec_get_channel_layout_string() + avcodec_channel_layout_num_channels() + and the CH_* macros defined in libavcodec/avcodec.h. + +2010-11-21 - 6bfc268 - lavf 52.85.0 - avformat.h + Add av_append_packet(). + +2010-11-21 - a08d918 - lavc 52.97.0 - avcodec.h + Add av_grow_packet(). + +2010-11-17 - 0985e1a - lavcore 0.13.0 - parseutils.h + Add av_parse_color() declared in libavcore/parseutils.h. + +2010-11-13 - cb2c971 - lavc 52.95.0 - AVCodecContext + Add AVCodecContext.subtitle_header and AVCodecContext.subtitle_header_size + fields. + +2010-11-13 - 5aaea02 - lavfi 1.62.0 - avfiltergraph.h + Make avfiltergraph.h public. + +2010-11-13 - 4fcbb2a - lavfi 1.61.0 - avfiltergraph.h + Remove declarations from avfiltergraph.h for the functions: + avfilter_graph_check_validity() + avfilter_graph_config_links() + avfilter_graph_config_formats() + which are now internal. + Use avfilter_graph_config() instead. + +2010-11-08 - d2af720 - lavu 50.33.0 - eval.h + Deprecate functions: + av_parse_and_eval_expr(), + av_parse_expr(), + av_eval_expr(), + av_free_expr(), + in favor of the functions: + av_expr_parse_and_eval(), + av_expr_parse(), + av_expr_eval(), + av_expr_free(). + +2010-11-08 - 24de0ed - lavfi 1.59.0 - avfilter_free() + Rename avfilter_destroy() to avfilter_free(). + This change breaks libavfilter API/ABI. + +2010-11-07 - 1e80a0e - lavfi 1.58.0 - avfiltergraph.h + Remove graphparser.h header, move AVFilterInOut and + avfilter_graph_parse() declarations to libavfilter/avfiltergraph.h. + +2010-11-07 - 7313132 - lavfi 1.57.0 - AVFilterInOut + Rename field AVFilterInOut.filter to AVFilterInOut.filter_ctx. + This change breaks libavfilter API. + +2010-11-04 - 97dd1e4 - lavfi 1.56.0 - avfilter_graph_free() + Rename avfilter_graph_destroy() to avfilter_graph_free(). + This change breaks libavfilter API/ABI. + +2010-11-04 - e15aeea - lavfi 1.55.0 - avfilter_graph_alloc() + Add avfilter_graph_alloc() to libavfilter/avfiltergraph.h. + +2010-11-02 - 6f84cd1 - lavcore 0.12.0 - av_get_bits_per_sample_fmt() + Add av_get_bits_per_sample_fmt() to libavcore/samplefmt.h and + deprecate av_get_bits_per_sample_format(). + +2010-11-02 - d63e456 - lavcore 0.11.0 - samplefmt.h + Add sample format functions in libavcore/samplefmt.h: + av_get_sample_fmt_name(), + av_get_sample_fmt(), + av_get_sample_fmt_string(), + and deprecate the corresponding libavcodec/audioconvert.h functions: + avcodec_get_sample_fmt_name(), + avcodec_get_sample_fmt(), + avcodec_sample_fmt_string(). + +2010-11-02 - 262d1c5 - lavcore 0.10.0 - samplefmt.h + Define enum AVSampleFormat in libavcore/samplefmt.h, deprecate enum + SampleFormat. + +2010-10-16 - 2a24df9 - lavfi 1.52.0 - avfilter_graph_config() + Add the function avfilter_graph_config() in avfiltergraph.h. + +2010-10-15 - 03700d3 - lavf 52.83.0 - metadata API + Change demuxers to export metadata in generic format and + muxers to accept generic format. Deprecate the public + conversion API. + +2010-10-10 - 867ae7a - lavfi 1.49.0 - AVFilterLink.time_base + Add time_base field to AVFilterLink. + +2010-09-27 - c85eef4 - lavu 50.31.0 - av_set_options_string() + Move av_set_options_string() from libavfilter/parseutils.h to + libavutil/opt.h. + +2010-09-27 - acc0490 - lavfi 1.47.0 - AVFilterLink + Make the AVFilterLink fields srcpad and dstpad store the pointers to + the source and destination pads, rather than their indexes. + +2010-09-27 - 372e288 - lavu 50.30.0 - av_get_token() + Move av_get_token() from libavfilter/parseutils.h to + libavutil/avstring.h. + +2010-09-26 - 635d4ae - lsws 0.12.0 - swscale.h + Add the functions sws_alloc_context() and sws_init_context(). + +2010-09-26 - 6ed0404 - lavu 50.29.0 - opt.h + Move libavcodec/opt.h to libavutil/opt.h. + +2010-09-24 - 1c1c80f - lavu 50.28.0 - av_log_set_flags() + Default of av_log() changed due to many problems to the old no repeat + detection. Read the docs of AV_LOG_SKIP_REPEATED in log.h before + enabling it for your app!. + +2010-09-24 - f66eb58 - lavc 52.90.0 - av_opt_show2() + Deprecate av_opt_show() in favor or av_opt_show2(). + +2010-09-14 - bc6f0af - lavu 50.27.0 - av_popcount() + Add av_popcount() to libavutil/common.h. + +2010-09-08 - c6c98d0 - lavu 50.26.0 - av_get_cpu_flags() + Add av_get_cpu_flags(). + +2010-09-07 - 34017fd - lavcore 0.9.0 - av_image_copy() + Add av_image_copy(). + +2010-09-07 - 9686abb - lavcore 0.8.0 - av_image_copy_plane() + Add av_image_copy_plane(). + +2010-09-07 - 9b7269e - lavcore 0.7.0 - imgutils.h + Adopt hierarchical scheme for the imgutils.h function names, + deprecate the old names. + +2010-09-04 - 7160bb7 - lavu 50.25.0 - AV_CPU_FLAG_* + Deprecate the FF_MM_* flags defined in libavcodec/avcodec.h in favor + of the AV_CPU_FLAG_* flags defined in libavutil/cpu.h. + +2010-08-26 - 5da19b5 - lavc 52.87.0 - avcodec_get_channel_layout() + Add avcodec_get_channel_layout() in audioconvert.h. + +2010-08-20 - e344336 - lavcore 0.6.0 - av_fill_image_max_pixsteps() + Rename av_fill_image_max_pixstep() to av_fill_image_max_pixsteps(). + +2010-08-18 - a6ddf8b - lavcore 0.5.0 - av_fill_image_max_pixstep() + Add av_fill_image_max_pixstep() in imgutils.h. + +2010-08-17 - 4f2d2e4 - lavu 50.24.0 - AV_NE() + Add the AV_NE macro. + +2010-08-17 - ad2c950 - lavfi 1.36.0 - audio framework + Implement AVFilterBufferRefAudioProps struct for audio properties, + get_audio_buffer(), filter_samples() functions and related changes. + +2010-08-12 - 81c1eca - lavcore 0.4.0 - av_get_image_linesize() + Add av_get_image_linesize() in imgutils.h. + +2010-08-11 - c1db7bf - lavfi 1.34.0 - AVFilterBufferRef + Resize data and linesize arrays in AVFilterBufferRef to 8. + + This change breaks libavfilter API/ABI. + +2010-08-11 - 9f08d80 - lavc 52.85.0 - av_picture_data_copy() + Add av_picture_data_copy in avcodec.h. + +2010-08-11 - 84c0386 - lavfi 1.33.0 - avfilter_open() + Change avfilter_open() signature: + AVFilterContext *avfilter_open(AVFilter *filter, const char *inst_name) -> + int avfilter_open(AVFilterContext **filter_ctx, AVFilter *filter, const char *inst_name); + + This change breaks libavfilter API/ABI. + +2010-08-11 - cc80caf - lavfi 1.32.0 - AVFilterBufferRef + Add a type field to AVFilterBufferRef, and move video specific + properties to AVFilterBufferRefVideoProps. + + This change breaks libavfilter API/ABI. + +2010-08-07 - 5d4890d - lavfi 1.31.0 - AVFilterLink + Rename AVFilterLink fields: + AVFilterLink.srcpic -> AVFilterLink.src_buf + AVFilterLink.cur_pic -> AVFilterLink.cur_buf + AVFilterLink.outpic -> AVFilterLink.out_buf + +2010-08-07 - 7fce481 - lavfi 1.30.0 + Rename functions and fields: + avfilter_(un)ref_pic -> avfilter_(un)ref_buffer + avfilter_copy_picref_props -> avfilter_copy_buffer_ref_props + AVFilterBufferRef.pic -> AVFilterBufferRef.buffer + +2010-08-07 - ecc8dad - lavfi 1.29.0 - AVFilterBufferRef + Rename AVFilterPicRef to AVFilterBufferRef. + +2010-08-07 - d54e094 - lavfi 1.28.0 - AVFilterBuffer + Move format field from AVFilterBuffer to AVFilterPicRef. + +2010-08-06 - bf176f5 - lavcore 0.3.0 - av_check_image_size() + Deprecate avcodec_check_dimensions() in favor of the function + av_check_image_size() defined in libavcore/imgutils.h. + +2010-07-30 - 56b5e9d - lavfi 1.27.0 - AVFilterBuffer + Increase size of the arrays AVFilterBuffer.data and + AVFilterBuffer.linesize from 4 to 8. + + This change breaks libavfilter ABI. + +2010-07-29 - e7bd48a - lavcore 0.2.0 - imgutils.h + Add functions av_fill_image_linesizes() and + av_fill_image_pointers(), declared in libavcore/imgutils.h. + +2010-07-27 - 126b638 - lavcore 0.1.0 - parseutils.h + Deprecate av_parse_video_frame_size() and av_parse_video_frame_rate() + defined in libavcodec in favor of the newly added functions + av_parse_video_size() and av_parse_video_rate() declared in + libavcore/parseutils.h. + +2010-07-23 - 4485247 - lavu 50.23.0 - mathematics.h + Add the M_PHI constant definition. + +2010-07-22 - bdab614 - lavfi 1.26.0 - media format generalization + Add a type field to AVFilterLink. + + Change the field types: + enum PixelFormat format -> int format in AVFilterBuffer + enum PixelFormat *formats -> int *formats in AVFilterFormats + enum PixelFormat *format -> int format in AVFilterLink + + Change the function signatures: + AVFilterFormats *avfilter_make_format_list(const enum PixelFormat *pix_fmts); -> + AVFilterFormats *avfilter_make_format_list(const int *fmts); + + int avfilter_add_colorspace(AVFilterFormats **avff, enum PixelFormat pix_fmt); -> + int avfilter_add_format (AVFilterFormats **avff, int fmt); + + AVFilterFormats *avfilter_all_colorspaces(void); -> + AVFilterFormats *avfilter_all_formats (enum AVMediaType type); + + This change breaks libavfilter API/ABI. + +2010-07-21 - aac6ca6 - lavcore 0.0.0 + Add libavcore. + +2010-07-17 - b5c582f - lavfi 1.25.0 - AVFilterBuffer + Remove w and h fields from AVFilterBuffer. + +2010-07-17 - f0d77b2 - lavfi 1.24.0 - AVFilterBuffer + Rename AVFilterPic to AVFilterBuffer. + +2010-07-17 - 57fe80f - lavf 52.74.0 - url_fskip() + Make url_fskip() return an int error code instead of void. + +2010-07-11 - 23940f1 - lavc 52.83.0 + Add AVCodecContext.lpc_type and AVCodecContext.lpc_passes fields. + Add AVLPCType enum. + Deprecate AVCodecContext.use_lpc. + +2010-07-11 - e1d7c88 - lavc 52.82.0 - avsubtitle_free() + Add a function for free the contents of a AVSubtitle generated by + avcodec_decode_subtitle. + +2010-07-11 - b91d08f - lavu 50.22.0 - bswap.h and intreadwrite.h + Make the bswap.h and intreadwrite.h API public. + +2010-07-08 - ce1cd1c - lavu 50.21.0 - pixdesc.h + Rename read/write_line() to av_read/write_image_line(). + +2010-07-07 - 4d508e4 - lavfi 1.21.0 - avfilter_copy_picref_props() + Add avfilter_copy_picref_props(). + +2010-07-03 - 2d525ef - lavc 52.79.0 + Add FF_COMPLIANCE_UNOFFICIAL and change all instances of + FF_COMPLIANCE_INOFFICIAL to use FF_COMPLIANCE_UNOFFICIAL. + +2010-07-02 - 89eec74 - lavu 50.20.0 - lfg.h + Export av_lfg_init(), av_lfg_get(), av_mlfg_get(), and av_bmg_get() through + lfg.h. + +2010-06-28 - a52e2c3 - lavfi 1.20.1 - av_parse_color() + Extend av_parse_color() syntax, make it accept an alpha value specifier and + set the alpha value to 255 by default. + +2010-06-22 - 735cf6b - lavf 52.71.0 - URLProtocol.priv_data_size, priv_data_class + Add priv_data_size and priv_data_class to URLProtocol. + +2010-06-22 - ffbb289 - lavf 52.70.0 - url_alloc(), url_connect() + Add url_alloc() and url_connect(). + +2010-06-22 - 9b07a2d - lavf 52.69.0 - av_register_protocol2() + Add av_register_protocol2(), deprecating av_register_protocol(). + +2010-06-09 - 65db058 - lavu 50.19.0 - av_compare_mod() + Add av_compare_mod() to libavutil/mathematics.h. + +2010-06-05 - 0b99215 - lavu 50.18.0 - eval API + Make the eval API public. + +2010-06-04 - 31878fc - lavu 50.17.0 - AV_BASE64_SIZE + Add AV_BASE64_SIZE() macro. + +2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string() + Add av_get_codec_tag_string(). + +2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API + Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24(). + +2010-05-26 - 93ebfee - lavc 52.72.0 - CODEC_CAP_EXPERIMENTAL + Add CODEC_CAP_EXPERIMENTAL flag. + NOTE: this was backported to 0.6 + +2010-05-23 - 9977863 - lavu 50.16.0 - av_get_random_seed() + Add av_get_random_seed(). + +2010-05-18 - 796ac23 - lavf 52.63.0 - AVFMT_FLAG_RTP_HINT + Add AVFMT_FLAG_RTP_HINT as possible value for AVFormatContext.flags. + NOTE: this was backported to 0.6 + +2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef + Add interlaced and top_field_first fields to AVFilterPicRef. + +------------------------------8<------------------------------------- + 0.6 branch was cut here +----------------------------->8-------------------------------------- + +2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function + Add av_probe_input_format2 to API, it allows ignoring probe + results below given score and returns the actual probe score. + +2010-04-01 - 3dd6180 - lavf 52.61.0 - metadata API + Add a flag for av_metadata_set2() to disable overwriting of + existing tags. + +2010-04-01 - 0fb49b5 - lavc 52.66.0 + Add avcodec_get_edge_width(). + +2010-03-31 - d103218 - lavc 52.65.0 + Add avcodec_copy_context(). + +2010-03-31 - 1a70d12 - lavf 52.60.0 - av_match_ext() + Make av_match_ext() public. + +2010-03-31 - 1149150 - lavu 50.14.0 - AVMediaType + Move AVMediaType enum from libavcodec to libavutil. + +2010-03-31 - 72415b2 - lavc 52.64.0 - AVMediaType + Define AVMediaType enum, and use it instead of enum CodecType, which + is deprecated and will be dropped at the next major bump. + +2010-03-25 - 8795823 - lavu 50.13.0 - av_strerror() + Implement av_strerror(). + +2010-03-23 - e1484eb - lavc 52.60.0 - av_dct_init() + Support DCT-I and DST-I. + +2010-03-15 - b8819c8 - lavf 52.56.0 - AVFormatContext.start_time_realtime + Add AVFormatContext.start_time_realtime field. + +2010-03-13 - 5bb5c1d - lavfi 1.18.0 - AVFilterPicRef.pos + Add AVFilterPicRef.pos field. + +2010-03-13 - 60c144f - lavu 50.12.0 - error.h + Move error code definitions from libavcodec/avcodec.h to + the new public header libavutil/error.h. + +2010-03-07 - c709483 - lavc 52.56.0 - avfft.h + Add public FFT interface. + +2010-03-06 - ac6ef86 - lavu 50.11.0 - av_stristr() + Add av_stristr(). + +2010-03-03 - 4b83fc0 - lavu 50.10.0 - av_tree_enumerate() + Add av_tree_enumerate(). + +2010-02-07 - b687c1a - lavu 50.9.0 - av_compare_ts() + Add av_compare_ts(). + +2010-02-05 - 3f3dc76 - lsws 0.10.0 - sws_getCoefficients() + Add sws_getCoefficients(). + +2010-02-01 - ca76a11 - lavf 52.50.0 - metadata API + Add a list of generic tag names, change 'author' -> 'artist', + 'year' -> 'date'. + +2010-01-30 - 80a07f6 - lavu 50.8.0 - av_get_pix_fmt() + Add av_get_pix_fmt(). + +2010-01-21 - 01cc47d - lsws 0.9.0 - sws_scale() + Change constness attributes of sws_scale() parameters. + +2010-01-10 - 3fb8e77 - lavfi 1.15.0 - avfilter_graph_config_links() + Add a log_ctx parameter to avfilter_graph_config_links(). + +2010-01-07 - 8e9767f - lsws 0.8.0 - sws_isSupported{In,Out}put() + Add sws_isSupportedInput() and sws_isSupportedOutput() functions. + +2010-01-06 - c1d662f - lavfi 1.14.0 - avfilter_add_colorspace() + Change the avfilter_add_colorspace() signature, make it accept an + (AVFilterFormats **) rather than an (AVFilterFormats *) as before. + +2010-01-03 - 4fd1f18 - lavfi 1.13.0 - avfilter_add_colorspace() + Add avfilter_add_colorspace(). + +2010-01-02 - 8eb631f - lavf 52.46.0 - av_match_ext() + Add av_match_ext(), it should be used in place of match_ext(). + +2010-01-01 - a1f547b - lavf 52.45.0 - av_guess_format() + Add av_guess_format(), it should be used in place of guess_format(). + +2009-12-13 - a181981 - lavf 52.43.0 - metadata API + Add av_metadata_set2(), AV_METADATA_DONT_STRDUP_KEY and + AV_METADATA_DONT_STRDUP_VAL. + +2009-12-13 - 277c733 - lavu 50.7.0 - avstring.h API + Add av_d2str(). + +2009-12-13 - 02b398e - lavc 52.42.0 - AVStream + Add avg_frame_rate. + +2009-12-12 - 3ba69a1 - lavu 50.6.0 - av_bmg_next() + Introduce the av_bmg_next() function. + +2009-12-05 - a13a543 - lavfi 1.12.0 - avfilter_draw_slice() + Add a slice_dir parameter to avfilter_draw_slice(). + +2009-11-26 - 4cc3f6a - lavfi 1.11.0 - AVFilter + Remove the next field from AVFilter, this is not anymore required. + +2009-11-25 - 1433c4a - lavfi 1.10.0 - avfilter_next() + Introduce the avfilter_next() function. + +2009-11-25 - 86a60fa - lavfi 1.9.0 - avfilter_register() + Change the signature of avfilter_register() to make it return an + int. This is required since now the registration operation may fail. + +2009-11-25 - 74a0059 - lavu 50.5.0 - pixdesc.h API + Make the pixdesc.h API public. + +2009-10-27 - 243110f - lavfi 1.5.0 - AVFilter.next + Add a next field to AVFilter, this is used for simplifying the + registration and management of the registered filters. + +2009-10-23 - cccd292 - lavfi 1.4.1 - AVFilter.description + Add a description field to AVFilter. + +2009-10-19 - 6b5dc05 - lavfi 1.3.0 - avfilter_make_format_list() + Change the interface of avfilter_make_format_list() from + avfilter_make_format_list(int n, ...) to + avfilter_make_format_list(enum PixelFormat *pix_fmts). + +2009-10-18 - 0eb4ff9 - lavfi 1.0.0 - avfilter_get_video_buffer() + Make avfilter_get_video_buffer() recursive and add the w and h + parameters to it. + +2009-10-07 - 46c40e4 - lavfi 0.5.1 - AVFilterPic + Add w and h fields to AVFilterPic. + +2009-06-22 - 92400be - lavf 52.34.1 - AVFormatContext.packet_size + This is now an unsigned int instead of a signed int. + +2009-06-19 - a4276ba - lavc 52.32.0 - AVSubtitle.pts + Add a pts field to AVSubtitle which gives the subtitle packet pts + in AV_TIME_BASE. Some subtitle de-/encoders (e.g. XSUB) will + not work right without this. + +2009-06-03 - 8f3f2e0 - lavc 52.30.2 - AV_PKT_FLAG_KEY + PKT_FLAG_KEY has been deprecated and will be dropped at the next + major version. Use AV_PKT_FLAG_KEY instead. + +2009-06-01 - f988ce6 - lavc 52.30.0 - av_lockmgr_register() + av_lockmgr_register() can be used to register a callback function + that lavc (and in the future, libraries that depend on lavc) can use + to implement mutexes. The application should provide a callback function + that implements the AV_LOCK_* operations described in avcodec.h. + When the lock manager is registered, FFmpeg is guaranteed to behave + correctly in a multi-threaded application. + +2009-04-30 - ce1d9c8 - lavc 52.28.0 - av_free_packet() + av_free_packet() is no longer an inline function. It is now exported. + +2009-04-11 - 80d403f - lavc 52.25.0 - deprecate av_destruct_packet_nofree() + Please use NULL instead. This has been supported since r16506 + (lavf > 52.23.1, lavc > 52.10.0). + +2009-04-07 - 7a00bba - lavc 52.23.0 - avcodec_decode_video/audio/subtitle + The old decoding functions are deprecated, all new code should use the + new functions avcodec_decode_video2(), avcodec_decode_audio3() and + avcodec_decode_subtitle2(). These new functions take an AVPacket *pkt + argument instead of a const uint8_t *buf / int buf_size pair. + +2009-04-03 - 7b09db3 - lavu 50.3.0 - av_fifo_space() + Introduce the av_fifo_space() function. + +2009-04-02 - fabd246 - lavc 52.23.0 - AVPacket + Move AVPacket declaration from libavformat/avformat.h to + libavcodec/avcodec.h. + +2009-03-22 - 6e08ca9 - lavu 50.2.0 - RGB32 pixel formats + Convert the pixel formats PIX_FMT_ARGB, PIX_FMT_RGBA, PIX_FMT_ABGR, + PIX_FMT_BGRA, which were defined as macros, into enum PixelFormat values. + Conversely PIX_FMT_RGB32, PIX_FMT_RGB32_1, PIX_FMT_BGR32 and + PIX_FMT_BGR32_1 are now macros. + avcodec_get_pix_fmt() now recognizes the "rgb32" and "bgr32" aliases. + Re-sort the enum PixelFormat list accordingly. + This change breaks API/ABI backward compatibility. + +2009-03-22 - f82674e - lavu 50.1.0 - PIX_FMT_RGB5X5 endian variants + Add the enum PixelFormat values: + PIX_FMT_RGB565BE, PIX_FMT_RGB565LE, PIX_FMT_RGB555BE, PIX_FMT_RGB555LE, + PIX_FMT_BGR565BE, PIX_FMT_BGR565LE, PIX_FMT_BGR555BE, PIX_FMT_BGR555LE. + +2009-03-21 - ee6624e - lavu 50.0.0 - av_random* + The Mersenne Twister PRNG implemented through the av_random* functions + was removed. Use the lagged Fibonacci PRNG through the av_lfg* functions + instead. + +2009-03-08 - 41dd680 - lavu 50.0.0 - AVFifoBuffer + av_fifo_init, av_fifo_read, av_fifo_write and av_fifo_realloc were dropped + and replaced by av_fifo_alloc, av_fifo_generic_read, av_fifo_generic_write + and av_fifo_realloc2. + In addition, the order of the function arguments of av_fifo_generic_read + was changed to match av_fifo_generic_write. + The AVFifoBuffer/struct AVFifoBuffer may only be used in an opaque way by + applications, they may not use sizeof() or directly access members. + +2009-03-01 - ec26457 - lavf 52.31.0 - Generic metadata API + Introduce a new metadata API (see av_metadata_get() and friends). + The old API is now deprecated and should not be used anymore. This especially + includes the following structure fields: + - AVFormatContext.title + - AVFormatContext.author + - AVFormatContext.copyright + - AVFormatContext.comment + - AVFormatContext.album + - AVFormatContext.year + - AVFormatContext.track + - AVFormatContext.genre + - AVStream.language + - AVStream.filename + - AVProgram.provider_name + - AVProgram.name + - AVChapter.title diff --git a/ffmpeg1/doc/Doxyfile b/ffmpeg1/doc/Doxyfile new file mode 100644 index 0000000..7e6d0f5 --- /dev/null +++ b/ffmpeg1/doc/Doxyfile @@ -0,0 +1,1624 @@ +# Doxyfile 1.7.1 + +# This file describes the settings to be used by the documentation system +# doxygen (www.doxygen.org) for a project +# +# All text after a hash (#) is considered a comment and will be ignored +# The format is: +# TAG = value [value, ...] +# For lists items can also be appended using: +# TAG += value [value, ...] +# Values that contain spaces should be placed between quotes (" ") + +#--------------------------------------------------------------------------- +# Project related configuration options +#--------------------------------------------------------------------------- + +# This tag specifies the encoding used for all characters in the config file +# that follow. The default is UTF-8 which is also the encoding used for all +# text before the first occurrence of this tag. Doxygen uses libiconv (or the +# iconv built into libc) for the transcoding. See +# http://www.gnu.org/software/libiconv for the list of possible encodings. + +DOXYFILE_ENCODING = UTF-8 + +# The PROJECT_NAME tag is a single word (or a sequence of words surrounded +# by quotes) that should identify the project. + +PROJECT_NAME = FFmpeg + +# The PROJECT_NUMBER tag can be used to enter a project or revision number. +# This could be handy for archiving the generated documentation or +# if some version control system is used. + +PROJECT_NUMBER = + +# With the PROJECT_LOGO tag one can specify an logo or icon that is included +# in the documentation. The maximum height of the logo should not exceed 55 +# pixels and the maximum width should not exceed 200 pixels. Doxygen will +# copy the logo to the output directory. +PROJECT_LOGO = + +# The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute) +# base path where the generated documentation will be put. +# If a relative path is entered, it will be relative to the location +# where doxygen was started. If left blank the current directory will be used. + +OUTPUT_DIRECTORY = doc/doxy + +# If the CREATE_SUBDIRS tag is set to YES, then doxygen will create +# 4096 sub-directories (in 2 levels) under the output directory of each output +# format and will distribute the generated files over these directories. +# Enabling this option can be useful when feeding doxygen a huge amount of +# source files, where putting all generated files in the same directory would +# otherwise cause performance problems for the file system. + +CREATE_SUBDIRS = NO + +# The OUTPUT_LANGUAGE tag is used to specify the language in which all +# documentation generated by doxygen is written. Doxygen will use this +# information to generate all constant output in the proper language. +# The default language is English, other supported languages are: +# Afrikaans, Arabic, Brazilian, Catalan, Chinese, Chinese-Traditional, +# Croatian, Czech, Danish, Dutch, Esperanto, Farsi, Finnish, French, German, +# Greek, Hungarian, Italian, Japanese, Japanese-en (Japanese with English +# messages), Korean, Korean-en, Lithuanian, Norwegian, Macedonian, Persian, +# Polish, Portuguese, Romanian, Russian, Serbian, Serbian-Cyrilic, Slovak, +# Slovene, Spanish, Swedish, Ukrainian, and Vietnamese. + +OUTPUT_LANGUAGE = English + +# If the BRIEF_MEMBER_DESC tag is set to YES (the default) Doxygen will +# include brief member descriptions after the members that are listed in +# the file and class documentation (similar to JavaDoc). +# Set to NO to disable this. + +BRIEF_MEMBER_DESC = YES + +# If the REPEAT_BRIEF tag is set to YES (the default) Doxygen will prepend +# the brief description of a member or function before the detailed description. +# Note: if both HIDE_UNDOC_MEMBERS and BRIEF_MEMBER_DESC are set to NO, the +# brief descriptions will be completely suppressed. + +REPEAT_BRIEF = YES + +# This tag implements a quasi-intelligent brief description abbreviator +# that is used to form the text in various listings. Each string +# in this list, if found as the leading text of the brief description, will be +# stripped from the text and the result after processing the whole list, is +# used as the annotated text. Otherwise, the brief description is used as-is. +# If left blank, the following values are used ("$name" is automatically +# replaced with the name of the entity): "The $name class" "The $name widget" +# "The $name file" "is" "provides" "specifies" "contains" +# "represents" "a" "an" "the" + +ABBREVIATE_BRIEF = + +# If the ALWAYS_DETAILED_SEC and REPEAT_BRIEF tags are both set to YES then +# Doxygen will generate a detailed section even if there is only a brief +# description. + +ALWAYS_DETAILED_SEC = NO + +# If the INLINE_INHERITED_MEMB tag is set to YES, doxygen will show all +# inherited members of a class in the documentation of that class as if those +# members were ordinary class members. Constructors, destructors and assignment +# operators of the base classes will not be shown. + +INLINE_INHERITED_MEMB = NO + +# If the FULL_PATH_NAMES tag is set to YES then Doxygen will prepend the full +# path before files name in the file list and in the header files. If set +# to NO the shortest path that makes the file name unique will be used. + +FULL_PATH_NAMES = YES + +# If the FULL_PATH_NAMES tag is set to YES then the STRIP_FROM_PATH tag +# can be used to strip a user-defined part of the path. Stripping is +# only done if one of the specified strings matches the left-hand part of +# the path. The tag can be used to show relative paths in the file list. +# If left blank the directory from which doxygen is run is used as the +# path to strip. + +STRIP_FROM_PATH = . + +# The STRIP_FROM_INC_PATH tag can be used to strip a user-defined part of +# the path mentioned in the documentation of a class, which tells +# the reader which header file to include in order to use a class. +# If left blank only the name of the header file containing the class +# definition is used. Otherwise one should specify the include paths that +# are normally passed to the compiler using the -I flag. + +STRIP_FROM_INC_PATH = + +# If the SHORT_NAMES tag is set to YES, doxygen will generate much shorter +# (but less readable) file names. This can be useful is your file systems +# doesn't support long names like on DOS, Mac, or CD-ROM. + +SHORT_NAMES = NO + +# If the JAVADOC_AUTOBRIEF tag is set to YES then Doxygen +# will interpret the first line (until the first dot) of a JavaDoc-style +# comment as the brief description. If set to NO, the JavaDoc +# comments will behave just like regular Qt-style comments +# (thus requiring an explicit @brief command for a brief description.) + +JAVADOC_AUTOBRIEF = YES + +# If the QT_AUTOBRIEF tag is set to YES then Doxygen will +# interpret the first line (until the first dot) of a Qt-style +# comment as the brief description. If set to NO, the comments +# will behave just like regular Qt-style comments (thus requiring +# an explicit \brief command for a brief description.) + +QT_AUTOBRIEF = NO + +# The MULTILINE_CPP_IS_BRIEF tag can be set to YES to make Doxygen +# treat a multi-line C++ special comment block (i.e. a block of //! or /// +# comments) as a brief description. This used to be the default behaviour. +# The new default is to treat a multi-line C++ comment block as a detailed +# description. Set this tag to YES if you prefer the old behaviour instead. + +MULTILINE_CPP_IS_BRIEF = NO + +# If the INHERIT_DOCS tag is set to YES (the default) then an undocumented +# member inherits the documentation from any documented member that it +# re-implements. + +INHERIT_DOCS = YES + +# If the SEPARATE_MEMBER_PAGES tag is set to YES, then doxygen will produce +# a new page for each member. If set to NO, the documentation of a member will +# be part of the file/class/namespace that contains it. + +SEPARATE_MEMBER_PAGES = NO + +# The TAB_SIZE tag can be used to set the number of spaces in a tab. +# Doxygen uses this value to replace tabs by spaces in code fragments. + +TAB_SIZE = 8 + +# This tag can be used to specify a number of aliases that acts +# as commands in the documentation. An alias has the form "name=value". +# For example adding "sideeffect=\par Side Effects:\n" will allow you to +# put the command \sideeffect (or @sideeffect) in the documentation, which +# will result in a user-defined paragraph with heading "Side Effects:". +# You can put \n's in the value part of an alias to insert newlines. + +ALIASES = + +# Set the OPTIMIZE_OUTPUT_FOR_C tag to YES if your project consists of C +# sources only. Doxygen will then generate output that is more tailored for C. +# For instance, some of the names that are used will be different. The list +# of all members will be omitted, etc. + +OPTIMIZE_OUTPUT_FOR_C = YES + +# Set the OPTIMIZE_OUTPUT_JAVA tag to YES if your project consists of Java +# sources only. Doxygen will then generate output that is more tailored for +# Java. For instance, namespaces will be presented as packages, qualified +# scopes will look different, etc. + +OPTIMIZE_OUTPUT_JAVA = NO + +# Set the OPTIMIZE_FOR_FORTRAN tag to YES if your project consists of Fortran +# sources only. Doxygen will then generate output that is more tailored for +# Fortran. + +OPTIMIZE_FOR_FORTRAN = NO + +# Set the OPTIMIZE_OUTPUT_VHDL tag to YES if your project consists of VHDL +# sources. Doxygen will then generate output that is tailored for +# VHDL. + +OPTIMIZE_OUTPUT_VHDL = NO + +# Doxygen selects the parser to use depending on the extension of the files it +# parses. With this tag you can assign which parser to use for a given extension. +# Doxygen has a built-in mapping, but you can override or extend it using this +# tag. The format is ext=language, where ext is a file extension, and language +# is one of the parsers supported by doxygen: IDL, Java, Javascript, CSharp, C, +# C++, D, PHP, Objective-C, Python, Fortran, VHDL, C, C++. For instance to make +# doxygen treat .inc files as Fortran files (default is PHP), and .f files as C +# (default is Fortran), use: inc=Fortran f=C. Note that for custom extensions +# you also need to set FILE_PATTERNS otherwise the files are not read by doxygen. + +EXTENSION_MAPPING = + +# If you use STL classes (i.e. std::string, std::vector, etc.) but do not want +# to include (a tag file for) the STL sources as input, then you should +# set this tag to YES in order to let doxygen match functions declarations and +# definitions whose arguments contain STL classes (e.g. func(std::string); v.s. +# func(std::string) {}). This also make the inheritance and collaboration +# diagrams that involve STL classes more complete and accurate. + +BUILTIN_STL_SUPPORT = NO + +# If you use Microsoft's C++/CLI language, you should set this option to YES to +# enable parsing support. + +CPP_CLI_SUPPORT = NO + +# Set the SIP_SUPPORT tag to YES if your project consists of sip sources only. +# Doxygen will parse them like normal C++ but will assume all classes use public +# instead of private inheritance when no explicit protection keyword is present. + +SIP_SUPPORT = NO + +# For Microsoft's IDL there are propget and propput attributes to indicate getter +# and setter methods for a property. Setting this option to YES (the default) +# will make doxygen to replace the get and set methods by a property in the +# documentation. This will only work if the methods are indeed getting or +# setting a simple type. If this is not the case, or you want to show the +# methods anyway, you should set this option to NO. + +IDL_PROPERTY_SUPPORT = YES + +# If member grouping is used in the documentation and the DISTRIBUTE_GROUP_DOC +# tag is set to YES, then doxygen will reuse the documentation of the first +# member in the group (if any) for the other members of the group. By default +# all members of a group must be documented explicitly. + +DISTRIBUTE_GROUP_DOC = NO + +# Set the SUBGROUPING tag to YES (the default) to allow class member groups of +# the same type (for instance a group of public functions) to be put as a +# subgroup of that type (e.g. under the Public Functions section). Set it to +# NO to prevent subgrouping. Alternatively, this can be done per class using +# the \nosubgrouping command. + +SUBGROUPING = YES + +# When TYPEDEF_HIDES_STRUCT is enabled, a typedef of a struct, union, or enum +# is documented as struct, union, or enum with the name of the typedef. So +# typedef struct TypeS {} TypeT, will appear in the documentation as a struct +# with name TypeT. When disabled the typedef will appear as a member of a file, +# namespace, or class. And the struct will be named TypeS. This can typically +# be useful for C code in case the coding convention dictates that all compound +# types are typedef'ed and only the typedef is referenced, never the tag name. + +TYPEDEF_HIDES_STRUCT = NO + +# The SYMBOL_CACHE_SIZE determines the size of the internal cache use to +# determine which symbols to keep in memory and which to flush to disk. +# When the cache is full, less often used symbols will be written to disk. +# For small to medium size projects (<1000 input files) the default value is +# probably good enough. For larger projects a too small cache size can cause +# doxygen to be busy swapping symbols to and from disk most of the time +# causing a significant performance penality. +# If the system has enough physical memory increasing the cache will improve the +# performance by keeping more symbols in memory. Note that the value works on +# a logarithmic scale so increasing the size by one will roughly double the +# memory usage. The cache size is given by this formula: +# 2^(16+SYMBOL_CACHE_SIZE). The valid range is 0..9, the default is 0, +# corresponding to a cache size of 2^16 = 65536 symbols + +SYMBOL_CACHE_SIZE = 0 + +#--------------------------------------------------------------------------- +# Build related configuration options +#--------------------------------------------------------------------------- + +# If the EXTRACT_ALL tag is set to YES doxygen will assume all entities in +# documentation are documented, even if no documentation was available. +# Private class members and static file members will be hidden unless +# the EXTRACT_PRIVATE and EXTRACT_STATIC tags are set to YES + +EXTRACT_ALL = YES + +# If the EXTRACT_PRIVATE tag is set to YES all private members of a class +# will be included in the documentation. + +EXTRACT_PRIVATE = YES + +# If the EXTRACT_STATIC tag is set to YES all static members of a file +# will be included in the documentation. + +EXTRACT_STATIC = YES + +# If the EXTRACT_LOCAL_CLASSES tag is set to YES classes (and structs) +# defined locally in source files will be included in the documentation. +# If set to NO only classes defined in header files are included. + +EXTRACT_LOCAL_CLASSES = YES + +# This flag is only useful for Objective-C code. When set to YES local +# methods, which are defined in the implementation section but not in +# the interface are included in the documentation. +# If set to NO (the default) only methods in the interface are included. + +EXTRACT_LOCAL_METHODS = NO + +# If this flag is set to YES, the members of anonymous namespaces will be +# extracted and appear in the documentation as a namespace called +# 'anonymous_namespace{file}', where file will be replaced with the base +# name of the file that contains the anonymous namespace. By default +# anonymous namespace are hidden. + +EXTRACT_ANON_NSPACES = NO + +# If the HIDE_UNDOC_MEMBERS tag is set to YES, Doxygen will hide all +# undocumented members of documented classes, files or namespaces. +# If set to NO (the default) these members will be included in the +# various overviews, but no documentation section is generated. +# This option has no effect if EXTRACT_ALL is enabled. + +HIDE_UNDOC_MEMBERS = NO + +# If the HIDE_UNDOC_CLASSES tag is set to YES, Doxygen will hide all +# undocumented classes that are normally visible in the class hierarchy. +# If set to NO (the default) these classes will be included in the various +# overviews. This option has no effect if EXTRACT_ALL is enabled. + +HIDE_UNDOC_CLASSES = NO + +# If the HIDE_FRIEND_COMPOUNDS tag is set to YES, Doxygen will hide all +# friend (class|struct|union) declarations. +# If set to NO (the default) these declarations will be included in the +# documentation. + +HIDE_FRIEND_COMPOUNDS = NO + +# If the HIDE_IN_BODY_DOCS tag is set to YES, Doxygen will hide any +# documentation blocks found inside the body of a function. +# If set to NO (the default) these blocks will be appended to the +# function's detailed documentation block. + +HIDE_IN_BODY_DOCS = NO + +# The INTERNAL_DOCS tag determines if documentation +# that is typed after a \internal command is included. If the tag is set +# to NO (the default) then the documentation will be excluded. +# Set it to YES to include the internal documentation. + +INTERNAL_DOCS = NO + +# If the CASE_SENSE_NAMES tag is set to NO then Doxygen will only generate +# file names in lower-case letters. If set to YES upper-case letters are also +# allowed. This is useful if you have classes or files whose names only differ +# in case and if your file system supports case sensitive file names. Windows +# and Mac users are advised to set this option to NO. + +CASE_SENSE_NAMES = YES + +# If the HIDE_SCOPE_NAMES tag is set to NO (the default) then Doxygen +# will show members with their full class and namespace scopes in the +# documentation. If set to YES the scope will be hidden. + +HIDE_SCOPE_NAMES = NO + +# If the SHOW_INCLUDE_FILES tag is set to YES (the default) then Doxygen +# will put a list of the files that are included by a file in the documentation +# of that file. + +SHOW_INCLUDE_FILES = YES + +# If the FORCE_LOCAL_INCLUDES tag is set to YES then Doxygen +# will list include files with double quotes in the documentation +# rather than with sharp brackets. + +FORCE_LOCAL_INCLUDES = NO + +# If the INLINE_INFO tag is set to YES (the default) then a tag [inline] +# is inserted in the documentation for inline members. + +INLINE_INFO = YES + +# If the SORT_MEMBER_DOCS tag is set to YES (the default) then doxygen +# will sort the (detailed) documentation of file and class members +# alphabetically by member name. If set to NO the members will appear in +# declaration order. + +SORT_MEMBER_DOCS = YES + +# If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the +# brief documentation of file, namespace and class members alphabetically +# by member name. If set to NO (the default) the members will appear in +# declaration order. + +SORT_BRIEF_DOCS = NO + +# If the SORT_MEMBERS_CTORS_1ST tag is set to YES then doxygen +# will sort the (brief and detailed) documentation of class members so that +# constructors and destructors are listed first. If set to NO (the default) +# the constructors will appear in the respective orders defined by +# SORT_MEMBER_DOCS and SORT_BRIEF_DOCS. +# This tag will be ignored for brief docs if SORT_BRIEF_DOCS is set to NO +# and ignored for detailed docs if SORT_MEMBER_DOCS is set to NO. + +SORT_MEMBERS_CTORS_1ST = NO + +# If the SORT_GROUP_NAMES tag is set to YES then doxygen will sort the +# hierarchy of group names into alphabetical order. If set to NO (the default) +# the group names will appear in their defined order. + +SORT_GROUP_NAMES = NO + +# If the SORT_BY_SCOPE_NAME tag is set to YES, the class list will be +# sorted by fully-qualified names, including namespaces. If set to +# NO (the default), the class list will be sorted only by class name, +# not including the namespace part. +# Note: This option is not very useful if HIDE_SCOPE_NAMES is set to YES. +# Note: This option applies only to the class list, not to the +# alphabetical list. + +SORT_BY_SCOPE_NAME = NO + +# The GENERATE_TODOLIST tag can be used to enable (YES) or +# disable (NO) the todo list. This list is created by putting \todo +# commands in the documentation. + +GENERATE_TODOLIST = YES + +# The GENERATE_TESTLIST tag can be used to enable (YES) or +# disable (NO) the test list. This list is created by putting \test +# commands in the documentation. + +GENERATE_TESTLIST = YES + +# The GENERATE_BUGLIST tag can be used to enable (YES) or +# disable (NO) the bug list. This list is created by putting \bug +# commands in the documentation. + +GENERATE_BUGLIST = YES + +# The GENERATE_DEPRECATEDLIST tag can be used to enable (YES) or +# disable (NO) the deprecated list. This list is created by putting +# \deprecated commands in the documentation. + +GENERATE_DEPRECATEDLIST= YES + +# The ENABLED_SECTIONS tag can be used to enable conditional +# documentation sections, marked by \if sectionname ... \endif. + +ENABLED_SECTIONS = + +# The MAX_INITIALIZER_LINES tag determines the maximum number of lines +# the initial value of a variable or define consists of for it to appear in +# the documentation. If the initializer consists of more lines than specified +# here it will be hidden. Use a value of 0 to hide initializers completely. +# The appearance of the initializer of individual variables and defines in the +# documentation can be controlled using \showinitializer or \hideinitializer +# command in the documentation regardless of this setting. + +MAX_INITIALIZER_LINES = 30 + +# Set the SHOW_USED_FILES tag to NO to disable the list of files generated +# at the bottom of the documentation of classes and structs. If set to YES the +# list will mention the files that were used to generate the documentation. + +SHOW_USED_FILES = YES + +# Set the SHOW_FILES tag to NO to disable the generation of the Files page. +# This will remove the Files entry from the Quick Index and from the +# Folder Tree View (if specified). The default is YES. + +SHOW_FILES = YES + +# Set the SHOW_NAMESPACES tag to NO to disable the generation of the +# Namespaces page. +# This will remove the Namespaces entry from the Quick Index +# and from the Folder Tree View (if specified). The default is YES. + +SHOW_NAMESPACES = YES + +# The FILE_VERSION_FILTER tag can be used to specify a program or script that +# doxygen should invoke to get the current version for each file (typically from +# the version control system). Doxygen will invoke the program by executing (via +# popen()) the command <command> <input-file>, where <command> is the value of +# the FILE_VERSION_FILTER tag, and <input-file> is the name of an input file +# provided by doxygen. Whatever the program writes to standard output +# is used as the file version. See the manual for examples. + +FILE_VERSION_FILTER = + +# The LAYOUT_FILE tag can be used to specify a layout file which will be parsed +# by doxygen. The layout file controls the global structure of the generated +# output files in an output format independent way. The create the layout file +# that represents doxygen's defaults, run doxygen with the -l option. +# You can optionally specify a file name after the option, if omitted +# DoxygenLayout.xml will be used as the name of the layout file. + +LAYOUT_FILE = + +#--------------------------------------------------------------------------- +# configuration options related to warning and progress messages +#--------------------------------------------------------------------------- + +# The QUIET tag can be used to turn on/off the messages that are generated +# by doxygen. Possible values are YES and NO. If left blank NO is used. + +QUIET = YES + +# The WARNINGS tag can be used to turn on/off the warning messages that are +# generated by doxygen. Possible values are YES and NO. If left blank +# NO is used. + +WARNINGS = YES + +# If WARN_IF_UNDOCUMENTED is set to YES, then doxygen will generate warnings +# for undocumented members. If EXTRACT_ALL is set to YES then this flag will +# automatically be disabled. + +WARN_IF_UNDOCUMENTED = YES + +# If WARN_IF_DOC_ERROR is set to YES, doxygen will generate warnings for +# potential errors in the documentation, such as not documenting some +# parameters in a documented function, or documenting parameters that +# don't exist or using markup commands wrongly. + +WARN_IF_DOC_ERROR = YES + +# This WARN_NO_PARAMDOC option can be abled to get warnings for +# functions that are documented, but have no documentation for their parameters +# or return value. If set to NO (the default) doxygen will only warn about +# wrong or incomplete parameter documentation, but not about the absence of +# documentation. + +WARN_NO_PARAMDOC = NO + +# The WARN_FORMAT tag determines the format of the warning messages that +# doxygen can produce. The string should contain the $file, $line, and $text +# tags, which will be replaced by the file and line number from which the +# warning originated and the warning text. Optionally the format may contain +# $version, which will be replaced by the version of the file (if it could +# be obtained via FILE_VERSION_FILTER) + +WARN_FORMAT = "$file:$line: $text" + +# The WARN_LOGFILE tag can be used to specify a file to which warning +# and error messages should be written. If left blank the output is written +# to stderr. + +WARN_LOGFILE = + +#--------------------------------------------------------------------------- +# configuration options related to the input files +#--------------------------------------------------------------------------- + +# The INPUT tag can be used to specify the files and/or directories that contain +# documented source files. You may enter file names like "myfile.cpp" or +# directories like "/usr/src/myproject". Separate the files or directories +# with spaces. + +INPUT = + +# This tag can be used to specify the character encoding of the source files +# that doxygen parses. Internally doxygen uses the UTF-8 encoding, which is +# also the default input encoding. Doxygen uses libiconv (or the iconv built +# into libc) for the transcoding. See http://www.gnu.org/software/libiconv for +# the list of possible encodings. + +INPUT_ENCODING = UTF-8 + +# If the value of the INPUT tag contains directories, you can use the +# FILE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp +# and *.h) to filter out the source-files in the directories. If left +# blank the following patterns are tested: +# *.c *.cc *.cxx *.cpp *.c++ *.java *.ii *.ixx *.ipp *.i++ *.inl *.h *.hh *.hxx +# *.hpp *.h++ *.idl *.odl *.cs *.php *.php3 *.inc *.m *.mm *.py *.f90 + +FILE_PATTERNS = + +# The RECURSIVE tag can be used to turn specify whether or not subdirectories +# should be searched for input files as well. Possible values are YES and NO. +# If left blank NO is used. + +RECURSIVE = YES + +# The EXCLUDE tag can be used to specify files and/or directories that should +# excluded from the INPUT source files. This way you can easily exclude a +# subdirectory from a directory tree whose root is specified with the INPUT tag. + +EXCLUDE = + +# The EXCLUDE_SYMLINKS tag can be used select whether or not files or +# directories that are symbolic links (a Unix filesystem feature) are excluded +# from the input. + +EXCLUDE_SYMLINKS = NO + +# If the value of the INPUT tag contains directories, you can use the +# EXCLUDE_PATTERNS tag to specify one or more wildcard patterns to exclude +# certain files from those directories. Note that the wildcards are matched +# against the file with absolute path, so to exclude all test directories +# for example use the pattern */test/* + +EXCLUDE_PATTERNS = *.git \ + *.d + +# The EXCLUDE_SYMBOLS tag can be used to specify one or more symbol names +# (namespaces, classes, functions, etc.) that should be excluded from the +# output. The symbol name can be a fully qualified name, a word, or if the +# wildcard * is used, a substring. Examples: ANamespace, AClass, +# AClass::ANamespace, ANamespace::*Test + +EXCLUDE_SYMBOLS = + +# The EXAMPLE_PATH tag can be used to specify one or more files or +# directories that contain example code fragments that are included (see +# the \include command). + +EXAMPLE_PATH = doc/examples/ + +# If the value of the EXAMPLE_PATH tag contains directories, you can use the +# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp +# and *.h) to filter out the source-files in the directories. If left +# blank all files are included. + +EXAMPLE_PATTERNS = *.c + +# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be +# searched for input files to be used with the \include or \dontinclude +# commands irrespective of the value of the RECURSIVE tag. +# Possible values are YES and NO. If left blank NO is used. + +EXAMPLE_RECURSIVE = NO + +# The IMAGE_PATH tag can be used to specify one or more files or +# directories that contain image that are included in the documentation (see +# the \image command). + +IMAGE_PATH = + +# The INPUT_FILTER tag can be used to specify a program that doxygen should +# invoke to filter for each input file. Doxygen will invoke the filter program +# by executing (via popen()) the command <filter> <input-file>, where <filter> +# is the value of the INPUT_FILTER tag, and <input-file> is the name of an +# input file. Doxygen will then use the output that the filter program writes +# to standard output. +# If FILTER_PATTERNS is specified, this tag will be +# ignored. + +INPUT_FILTER = + +# The FILTER_PATTERNS tag can be used to specify filters on a per file pattern +# basis. +# Doxygen will compare the file name with each pattern and apply the +# filter if there is a match. +# The filters are a list of the form: +# pattern=filter (like *.cpp=my_cpp_filter). See INPUT_FILTER for further +# info on how filters are used. If FILTER_PATTERNS is empty, INPUT_FILTER +# is applied to all files. + +FILTER_PATTERNS = + +# If the FILTER_SOURCE_FILES tag is set to YES, the input filter (if set using +# INPUT_FILTER) will be used to filter the input files when producing source +# files to browse (i.e. when SOURCE_BROWSER is set to YES). + +FILTER_SOURCE_FILES = NO + +#--------------------------------------------------------------------------- +# configuration options related to source browsing +#--------------------------------------------------------------------------- + +# If the SOURCE_BROWSER tag is set to YES then a list of source files will +# be generated. Documented entities will be cross-referenced with these sources. +# Note: To get rid of all source code in the generated output, make sure also +# VERBATIM_HEADERS is set to NO. + +SOURCE_BROWSER = YES + +# Setting the INLINE_SOURCES tag to YES will include the body +# of functions and classes directly in the documentation. + +INLINE_SOURCES = NO + +# Setting the STRIP_CODE_COMMENTS tag to YES (the default) will instruct +# doxygen to hide any special comment blocks from generated source code +# fragments. Normal C and C++ comments will always remain visible. + +STRIP_CODE_COMMENTS = YES + +# If the REFERENCED_BY_RELATION tag is set to YES +# then for each documented function all documented +# functions referencing it will be listed. + +REFERENCED_BY_RELATION = YES + +# If the REFERENCES_RELATION tag is set to YES +# then for each documented function all documented entities +# called/used by that function will be listed. + +REFERENCES_RELATION = NO + +# If the REFERENCES_LINK_SOURCE tag is set to YES (the default) +# and SOURCE_BROWSER tag is set to YES, then the hyperlinks from +# functions in REFERENCES_RELATION and REFERENCED_BY_RELATION lists will +# link to the source code. +# Otherwise they will link to the documentation. + +REFERENCES_LINK_SOURCE = YES + +# If the USE_HTAGS tag is set to YES then the references to source code +# will point to the HTML generated by the htags(1) tool instead of doxygen +# built-in source browser. The htags tool is part of GNU's global source +# tagging system (see http://www.gnu.org/software/global/global.html). You +# will need version 4.8.6 or higher. + +USE_HTAGS = NO + +# If the VERBATIM_HEADERS tag is set to YES (the default) then Doxygen +# will generate a verbatim copy of the header file for each class for +# which an include is specified. Set to NO to disable this. + +VERBATIM_HEADERS = YES + +#--------------------------------------------------------------------------- +# configuration options related to the alphabetical class index +#--------------------------------------------------------------------------- + +# If the ALPHABETICAL_INDEX tag is set to YES, an alphabetical index +# of all compounds will be generated. Enable this if the project +# contains a lot of classes, structs, unions or interfaces. + +ALPHABETICAL_INDEX = YES + +# If the alphabetical index is enabled (see ALPHABETICAL_INDEX) then +# the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns +# in which this list will be split (can be a number in the range [1..20]) + +COLS_IN_ALPHA_INDEX = 2 + +# In case all classes in a project start with a common prefix, all +# classes will be put under the same header in the alphabetical index. +# The IGNORE_PREFIX tag can be used to specify one or more prefixes that +# should be ignored while generating the index headers. + +IGNORE_PREFIX = + +#--------------------------------------------------------------------------- +# configuration options related to the HTML output +#--------------------------------------------------------------------------- + +# If the GENERATE_HTML tag is set to YES (the default) Doxygen will +# generate HTML output. + +GENERATE_HTML = YES + +# The HTML_OUTPUT tag is used to specify where the HTML docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `html' will be used as the default path. + +HTML_OUTPUT = html + +# The HTML_FILE_EXTENSION tag can be used to specify the file extension for +# each generated HTML page (for example: .htm,.php,.asp). If it is left blank +# doxygen will generate files with .html extension. + +HTML_FILE_EXTENSION = .html + +# The HTML_HEADER tag can be used to specify a personal HTML header for +# each generated HTML page. If it is left blank doxygen will generate a +# standard header. + +#HTML_HEADER = doc/doxy/header.html + +# The HTML_FOOTER tag can be used to specify a personal HTML footer for +# each generated HTML page. If it is left blank doxygen will generate a +# standard footer. + +#HTML_FOOTER = doc/doxy/footer.html + +# The HTML_STYLESHEET tag can be used to specify a user-defined cascading +# style sheet that is used by each HTML page. It can be used to +# fine-tune the look of the HTML output. If the tag is left blank doxygen +# will generate a default style sheet. Note that doxygen will try to copy +# the style sheet file to the HTML output directory, so don't put your own +# stylesheet in the HTML output directory as well, or it will be erased! + +#HTML_STYLESHEET = doc/doxy/doxy_stylesheet.css + +# The HTML_COLORSTYLE_HUE tag controls the color of the HTML output. +# Doxygen will adjust the colors in the stylesheet and background images +# according to this color. Hue is specified as an angle on a colorwheel, +# see http://en.wikipedia.org/wiki/Hue for more information. +# For instance the value 0 represents red, 60 is yellow, 120 is green, +# 180 is cyan, 240 is blue, 300 purple, and 360 is red again. +# The allowed range is 0 to 359. + +#HTML_COLORSTYLE_HUE = 120 + +# The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of +# the colors in the HTML output. For a value of 0 the output will use +# grayscales only. A value of 255 will produce the most vivid colors. + +HTML_COLORSTYLE_SAT = 100 + +# The HTML_COLORSTYLE_GAMMA tag controls the gamma correction applied to +# the luminance component of the colors in the HTML output. Values below +# 100 gradually make the output lighter, whereas values above 100 make +# the output darker. The value divided by 100 is the actual gamma applied, +# so 80 represents a gamma of 0.8, The value 220 represents a gamma of 2.2, +# and 100 does not change the gamma. + +HTML_COLORSTYLE_GAMMA = 80 + +# If the HTML_TIMESTAMP tag is set to YES then the footer of each generated HTML +# page will contain the date and time when the page was generated. Setting +# this to NO can help when comparing the output of multiple runs. + +HTML_TIMESTAMP = YES + +# If the HTML_DYNAMIC_SECTIONS tag is set to YES then the generated HTML +# documentation will contain sections that can be hidden and shown after the +# page has loaded. For this to work a browser that supports +# JavaScript and DHTML is required (for instance Mozilla 1.0+, Firefox +# Netscape 6.0+, Internet explorer 5.0+, Konqueror, or Safari). + +HTML_DYNAMIC_SECTIONS = NO + +# If the GENERATE_DOCSET tag is set to YES, additional index files +# will be generated that can be used as input for Apple's Xcode 3 +# integrated development environment, introduced with OS X 10.5 (Leopard). +# To create a documentation set, doxygen will generate a Makefile in the +# HTML output directory. Running make will produce the docset in that +# directory and running "make install" will install the docset in +# ~/Library/Developer/Shared/Documentation/DocSets so that Xcode will find +# it at startup. +# See http://developer.apple.com/tools/creatingdocsetswithdoxygen.html +# for more information. + +GENERATE_DOCSET = NO + +# When GENERATE_DOCSET tag is set to YES, this tag determines the name of the +# feed. A documentation feed provides an umbrella under which multiple +# documentation sets from a single provider (such as a company or product suite) +# can be grouped. + +DOCSET_FEEDNAME = "Doxygen generated docs" + +# When GENERATE_DOCSET tag is set to YES, this tag specifies a string that +# should uniquely identify the documentation set bundle. This should be a +# reverse domain-name style string, e.g. com.mycompany.MyDocSet. Doxygen +# will append .docset to the name. + +DOCSET_BUNDLE_ID = org.doxygen.Project + +# When GENERATE_PUBLISHER_ID tag specifies a string that should uniquely identify +# the documentation publisher. This should be a reverse domain-name style +# string, e.g. com.mycompany.MyDocSet.documentation. + +DOCSET_PUBLISHER_ID = org.doxygen.Publisher + +# The GENERATE_PUBLISHER_NAME tag identifies the documentation publisher. + +DOCSET_PUBLISHER_NAME = Publisher + +# If the GENERATE_HTMLHELP tag is set to YES, additional index files +# will be generated that can be used as input for tools like the +# Microsoft HTML help workshop to generate a compiled HTML help file (.chm) +# of the generated HTML documentation. + +GENERATE_HTMLHELP = NO + +# If the GENERATE_HTMLHELP tag is set to YES, the CHM_FILE tag can +# be used to specify the file name of the resulting .chm file. You +# can add a path in front of the file if the result should not be +# written to the html output directory. + +CHM_FILE = + +# If the GENERATE_HTMLHELP tag is set to YES, the HHC_LOCATION tag can +# be used to specify the location (absolute path including file name) of +# the HTML help compiler (hhc.exe). If non-empty doxygen will try to run +# the HTML help compiler on the generated index.hhp. + +HHC_LOCATION = + +# If the GENERATE_HTMLHELP tag is set to YES, the GENERATE_CHI flag +# controls if a separate .chi index file is generated (YES) or that +# it should be included in the master .chm file (NO). + +GENERATE_CHI = NO + +# If the GENERATE_HTMLHELP tag is set to YES, the CHM_INDEX_ENCODING +# is used to encode HtmlHelp index (hhk), content (hhc) and project file +# content. + +CHM_INDEX_ENCODING = + +# If the GENERATE_HTMLHELP tag is set to YES, the BINARY_TOC flag +# controls whether a binary table of contents is generated (YES) or a +# normal table of contents (NO) in the .chm file. + +BINARY_TOC = NO + +# The TOC_EXPAND flag can be set to YES to add extra items for group members +# to the contents of the HTML help documentation and to the tree view. + +TOC_EXPAND = NO + +# If the GENERATE_QHP tag is set to YES and both QHP_NAMESPACE and +# QHP_VIRTUAL_FOLDER are set, an additional index file will be generated +# that can be used as input for Qt's qhelpgenerator to generate a +# Qt Compressed Help (.qch) of the generated HTML documentation. + +GENERATE_QHP = NO + +# If the QHG_LOCATION tag is specified, the QCH_FILE tag can +# be used to specify the file name of the resulting .qch file. +# The path specified is relative to the HTML output folder. + +QCH_FILE = + +# The QHP_NAMESPACE tag specifies the namespace to use when generating +# Qt Help Project output. For more information please see +# http://doc.trolltech.com/qthelpproject.html#namespace + +QHP_NAMESPACE = org.doxygen.Project + +# The QHP_VIRTUAL_FOLDER tag specifies the namespace to use when generating +# Qt Help Project output. For more information please see +# http://doc.trolltech.com/qthelpproject.html#virtual-folders + +QHP_VIRTUAL_FOLDER = doc + +# If QHP_CUST_FILTER_NAME is set, it specifies the name of a custom filter to +# add. For more information please see +# http://doc.trolltech.com/qthelpproject.html#custom-filters + +QHP_CUST_FILTER_NAME = + +# The QHP_CUST_FILT_ATTRS tag specifies the list of the attributes of the +# custom filter to add. For more information please see +# <a href="http://doc.trolltech.com/qthelpproject.html#custom-filters"> +# Qt Help Project / Custom Filters</a>. + +QHP_CUST_FILTER_ATTRS = + +# The QHP_SECT_FILTER_ATTRS tag specifies the list of the attributes this +# project's +# filter section matches. +# <a href="http://doc.trolltech.com/qthelpproject.html#filter-attributes"> +# Qt Help Project / Filter Attributes</a>. + +QHP_SECT_FILTER_ATTRS = + +# If the GENERATE_QHP tag is set to YES, the QHG_LOCATION tag can +# be used to specify the location of Qt's qhelpgenerator. +# If non-empty doxygen will try to run qhelpgenerator on the generated +# .qhp file. + +QHG_LOCATION = + +# If the GENERATE_ECLIPSEHELP tag is set to YES, additional index files +# will be generated, which together with the HTML files, form an Eclipse help +# plugin. To install this plugin and make it available under the help contents +# menu in Eclipse, the contents of the directory containing the HTML and XML +# files needs to be copied into the plugins directory of eclipse. The name of +# the directory within the plugins directory should be the same as +# the ECLIPSE_DOC_ID value. After copying Eclipse needs to be restarted before +# the help appears. + +GENERATE_ECLIPSEHELP = NO + +# A unique identifier for the eclipse help plugin. When installing the plugin +# the directory name containing the HTML and XML files should also have +# this name. + +ECLIPSE_DOC_ID = org.doxygen.Project + +# The DISABLE_INDEX tag can be used to turn on/off the condensed index at +# top of each HTML page. The value NO (the default) enables the index and +# the value YES disables it. + +DISABLE_INDEX = NO + +# This tag can be used to set the number of enum values (range [1..20]) +# that doxygen will group on one line in the generated HTML documentation. + +ENUM_VALUES_PER_LINE = 4 + +# The GENERATE_TREEVIEW tag is used to specify whether a tree-like index +# structure should be generated to display hierarchical information. +# If the tag value is set to YES, a side panel will be generated +# containing a tree-like index structure (just like the one that +# is generated for HTML Help). For this to work a browser that supports +# JavaScript, DHTML, CSS and frames is required (i.e. any modern browser). +# Windows users are probably better off using the HTML help feature. + +GENERATE_TREEVIEW = NO + +# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be +# used to set the initial width (in pixels) of the frame in which the tree +# is shown. + +TREEVIEW_WIDTH = 250 + +# When the EXT_LINKS_IN_WINDOW option is set to YES doxygen will open +# links to external symbols imported via tag files in a separate window. + +EXT_LINKS_IN_WINDOW = NO + +# Use this tag to change the font size of Latex formulas included +# as images in the HTML documentation. The default is 10. Note that +# when you change the font size after a successful doxygen run you need +# to manually remove any form_*.png images from the HTML output directory +# to force them to be regenerated. + +FORMULA_FONTSIZE = 10 + +# Use the FORMULA_TRANPARENT tag to determine whether or not the images +# generated for formulas are transparent PNGs. Transparent PNGs are +# not supported properly for IE 6.0, but are supported on all modern browsers. +# Note that when changing this option you need to delete any form_*.png files +# in the HTML output before the changes have effect. + +FORMULA_TRANSPARENT = YES + +# When the SEARCHENGINE tag is enabled doxygen will generate a search box +# for the HTML output. The underlying search engine uses javascript +# and DHTML and should work on any modern browser. Note that when using +# HTML help (GENERATE_HTMLHELP), Qt help (GENERATE_QHP), or docsets +# (GENERATE_DOCSET) there is already a search function so this one should +# typically be disabled. For large projects the javascript based search engine +# can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution. + +SEARCHENGINE = NO + +# When the SERVER_BASED_SEARCH tag is enabled the search engine will be +# implemented using a PHP enabled web server instead of at the web client +# using Javascript. Doxygen will generate the search PHP script and index +# file to put on the web server. The advantage of the server +# based approach is that it scales better to large projects and allows +# full text search. The disadvances is that it is more difficult to setup +# and does not have live searching capabilities. + +SERVER_BASED_SEARCH = NO + +#--------------------------------------------------------------------------- +# configuration options related to the LaTeX output +#--------------------------------------------------------------------------- + +# If the GENERATE_LATEX tag is set to YES (the default) Doxygen will +# generate Latex output. + +GENERATE_LATEX = NO + +# The LATEX_OUTPUT tag is used to specify where the LaTeX docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `latex' will be used as the default path. + +LATEX_OUTPUT = latex + +# The LATEX_CMD_NAME tag can be used to specify the LaTeX command name to be +# invoked. If left blank `latex' will be used as the default command name. +# Note that when enabling USE_PDFLATEX this option is only used for +# generating bitmaps for formulas in the HTML output, but not in the +# Makefile that is written to the output directory. + +LATEX_CMD_NAME = latex + +# The MAKEINDEX_CMD_NAME tag can be used to specify the command name to +# generate index for LaTeX. If left blank `makeindex' will be used as the +# default command name. + +MAKEINDEX_CMD_NAME = makeindex + +# If the COMPACT_LATEX tag is set to YES Doxygen generates more compact +# LaTeX documents. This may be useful for small projects and may help to +# save some trees in general. + +COMPACT_LATEX = NO + +# The PAPER_TYPE tag can be used to set the paper type that is used +# by the printer. Possible values are: a4, a4wide, letter, legal and +# executive. If left blank a4wide will be used. + +PAPER_TYPE = a4wide + +# The EXTRA_PACKAGES tag can be to specify one or more names of LaTeX +# packages that should be included in the LaTeX output. + +EXTRA_PACKAGES = + +# The LATEX_HEADER tag can be used to specify a personal LaTeX header for +# the generated latex document. The header should contain everything until +# the first chapter. If it is left blank doxygen will generate a +# standard header. Notice: only use this tag if you know what you are doing! + +LATEX_HEADER = + +# If the PDF_HYPERLINKS tag is set to YES, the LaTeX that is generated +# is prepared for conversion to pdf (using ps2pdf). The pdf file will +# contain links (just like the HTML output) instead of page references +# This makes the output suitable for online browsing using a pdf viewer. + +PDF_HYPERLINKS = NO + +# If the USE_PDFLATEX tag is set to YES, pdflatex will be used instead of +# plain latex in the generated Makefile. Set this option to YES to get a +# higher quality PDF documentation. + +USE_PDFLATEX = NO + +# If the LATEX_BATCHMODE tag is set to YES, doxygen will add the \\batchmode. +# command to the generated LaTeX files. This will instruct LaTeX to keep +# running if errors occur, instead of asking the user for help. +# This option is also used when generating formulas in HTML. + +LATEX_BATCHMODE = NO + +# If LATEX_HIDE_INDICES is set to YES then doxygen will not +# include the index chapters (such as File Index, Compound Index, etc.) +# in the output. + +LATEX_HIDE_INDICES = NO + +# If LATEX_SOURCE_CODE is set to YES then doxygen will include +# source code with syntax highlighting in the LaTeX output. +# Note that which sources are shown also depends on other settings +# such as SOURCE_BROWSER. + +LATEX_SOURCE_CODE = NO + +#--------------------------------------------------------------------------- +# configuration options related to the RTF output +#--------------------------------------------------------------------------- + +# If the GENERATE_RTF tag is set to YES Doxygen will generate RTF output +# The RTF output is optimized for Word 97 and may not look very pretty with +# other RTF readers or editors. + +GENERATE_RTF = NO + +# The RTF_OUTPUT tag is used to specify where the RTF docs will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `rtf' will be used as the default path. + +RTF_OUTPUT = rtf + +# If the COMPACT_RTF tag is set to YES Doxygen generates more compact +# RTF documents. This may be useful for small projects and may help to +# save some trees in general. + +COMPACT_RTF = NO + +# If the RTF_HYPERLINKS tag is set to YES, the RTF that is generated +# will contain hyperlink fields. The RTF file will +# contain links (just like the HTML output) instead of page references. +# This makes the output suitable for online browsing using WORD or other +# programs which support those fields. +# Note: wordpad (write) and others do not support links. + +RTF_HYPERLINKS = NO + +# Load stylesheet definitions from file. Syntax is similar to doxygen's +# config file, i.e. a series of assignments. You only have to provide +# replacements, missing definitions are set to their default value. + +RTF_STYLESHEET_FILE = + +# Set optional variables used in the generation of an rtf document. +# Syntax is similar to doxygen's config file. + +RTF_EXTENSIONS_FILE = + +#--------------------------------------------------------------------------- +# configuration options related to the man page output +#--------------------------------------------------------------------------- + +# If the GENERATE_MAN tag is set to YES (the default) Doxygen will +# generate man pages + +GENERATE_MAN = NO + +# The MAN_OUTPUT tag is used to specify where the man pages will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `man' will be used as the default path. + +MAN_OUTPUT = man + +# The MAN_EXTENSION tag determines the extension that is added to +# the generated man pages (default is the subroutine's section .3) + +MAN_EXTENSION = .3 + +# If the MAN_LINKS tag is set to YES and Doxygen generates man output, +# then it will generate one additional man file for each entity +# documented in the real man page(s). These additional files +# only source the real man page, but without them the man command +# would be unable to find the correct page. The default is NO. + +MAN_LINKS = NO + +#--------------------------------------------------------------------------- +# configuration options related to the XML output +#--------------------------------------------------------------------------- + +# If the GENERATE_XML tag is set to YES Doxygen will +# generate an XML file that captures the structure of +# the code including all documentation. + +GENERATE_XML = NO + +# The XML_OUTPUT tag is used to specify where the XML pages will be put. +# If a relative path is entered the value of OUTPUT_DIRECTORY will be +# put in front of it. If left blank `xml' will be used as the default path. + +XML_OUTPUT = xml + +# The XML_SCHEMA tag can be used to specify an XML schema, +# which can be used by a validating XML parser to check the +# syntax of the XML files. + +XML_SCHEMA = + +# The XML_DTD tag can be used to specify an XML DTD, +# which can be used by a validating XML parser to check the +# syntax of the XML files. + +XML_DTD = + +# If the XML_PROGRAMLISTING tag is set to YES Doxygen will +# dump the program listings (including syntax highlighting +# and cross-referencing information) to the XML output. Note that +# enabling this will significantly increase the size of the XML output. + +XML_PROGRAMLISTING = YES + +#--------------------------------------------------------------------------- +# configuration options for the AutoGen Definitions output +#--------------------------------------------------------------------------- + +# If the GENERATE_AUTOGEN_DEF tag is set to YES Doxygen will +# generate an AutoGen Definitions (see autogen.sf.net) file +# that captures the structure of the code including all +# documentation. Note that this feature is still experimental +# and incomplete at the moment. + +GENERATE_AUTOGEN_DEF = NO + +#--------------------------------------------------------------------------- +# configuration options related to the Perl module output +#--------------------------------------------------------------------------- + +# If the GENERATE_PERLMOD tag is set to YES Doxygen will +# generate a Perl module file that captures the structure of +# the code including all documentation. Note that this +# feature is still experimental and incomplete at the +# moment. + +GENERATE_PERLMOD = NO + +# If the PERLMOD_LATEX tag is set to YES Doxygen will generate +# the necessary Makefile rules, Perl scripts and LaTeX code to be able +# to generate PDF and DVI output from the Perl module output. + +PERLMOD_LATEX = NO + +# If the PERLMOD_PRETTY tag is set to YES the Perl module output will be +# nicely formatted so it can be parsed by a human reader. +# This is useful +# if you want to understand what is going on. +# On the other hand, if this +# tag is set to NO the size of the Perl module output will be much smaller +# and Perl will parse it just the same. + +PERLMOD_PRETTY = YES + +# The names of the make variables in the generated doxyrules.make file +# are prefixed with the string contained in PERLMOD_MAKEVAR_PREFIX. +# This is useful so different doxyrules.make files included by the same +# Makefile don't overwrite each other's variables. + +PERLMOD_MAKEVAR_PREFIX = + +#--------------------------------------------------------------------------- +# Configuration options related to the preprocessor +#--------------------------------------------------------------------------- + +# If the ENABLE_PREPROCESSING tag is set to YES (the default) Doxygen will +# evaluate all C-preprocessor directives found in the sources and include +# files. + +ENABLE_PREPROCESSING = YES + +# If the MACRO_EXPANSION tag is set to YES Doxygen will expand all macro +# names in the source code. If set to NO (the default) only conditional +# compilation will be performed. Macro expansion can be done in a controlled +# way by setting EXPAND_ONLY_PREDEF to YES. + +MACRO_EXPANSION = YES + +# If the EXPAND_ONLY_PREDEF and MACRO_EXPANSION tags are both set to YES +# then the macro expansion is limited to the macros specified with the +# PREDEFINED and EXPAND_AS_DEFINED tags. + +EXPAND_ONLY_PREDEF = YES + +# If the SEARCH_INCLUDES tag is set to YES (the default) the includes files +# in the INCLUDE_PATH (see below) will be search if a #include is found. + +SEARCH_INCLUDES = YES + +# The INCLUDE_PATH tag can be used to specify one or more directories that +# contain include files that are not input files but should be processed by +# the preprocessor. + +INCLUDE_PATH = + +# You can use the INCLUDE_FILE_PATTERNS tag to specify one or more wildcard +# patterns (like *.h and *.hpp) to filter out the header-files in the +# directories. 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When set to 0 (the default) doxygen will +# base this on the number of processors available in the system. You can set it +# explicitly to a value larger than 0 to get control over the balance +# between CPU load and processing speed. + +DOT_NUM_THREADS = 0 + +# By default doxygen will write a font called FreeSans.ttf to the output +# directory and reference it in all dot files that doxygen generates. This +# font does not include all possible unicode characters however, so when you need +# these (or just want a differently looking font) you can specify the font name +# using DOT_FONTNAME. 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The dependency relations are determined by the #include +# relations between the files in the directories. + +DIRECTORY_GRAPH = YES + +# The DOT_IMAGE_FORMAT tag can be used to set the image format of the images +# generated by dot. Possible values are png, jpg, or gif +# If left blank png will be used. + +DOT_IMAGE_FORMAT = png + +# The tag DOT_PATH can be used to specify the path where the dot tool can be +# found. If left blank, it is assumed the dot tool can be found in the path. + +DOT_PATH = + +# The DOTFILE_DIRS tag can be used to specify one or more directories that +# contain dot files that are included in the documentation (see the +# \dotfile command). + +DOTFILE_DIRS = + +# The DOT_GRAPH_MAX_NODES tag can be used to set the maximum number of +# nodes that will be shown in the graph. If the number of nodes in a graph +# becomes larger than this value, doxygen will truncate the graph, which is +# visualized by representing a node as a red box. Note that doxygen if the +# number of direct children of the root node in a graph is already larger than +# DOT_GRAPH_MAX_NODES then the graph will not be shown at all. Also note +# that the size of a graph can be further restricted by MAX_DOT_GRAPH_DEPTH. + +DOT_GRAPH_MAX_NODES = 50 + +# The MAX_DOT_GRAPH_DEPTH tag can be used to set the maximum depth of the +# graphs generated by dot. A depth value of 3 means that only nodes reachable +# from the root by following a path via at most 3 edges will be shown. Nodes +# that lay further from the root node will be omitted. Note that setting this +# option to 1 or 2 may greatly reduce the computation time needed for large +# code bases. Also note that the size of a graph can be further restricted by +# DOT_GRAPH_MAX_NODES. Using a depth of 0 means no depth restriction. + +MAX_DOT_GRAPH_DEPTH = 0 + +# Set the DOT_TRANSPARENT tag to YES to generate images with a transparent +# background. This is disabled by default, because dot on Windows does not +# seem to support this out of the box. Warning: Depending on the platform used, +# enabling this option may lead to badly anti-aliased labels on the edges of +# a graph (i.e. they become hard to read). + +DOT_TRANSPARENT = YES + +# Set the DOT_MULTI_TARGETS tag to YES allow dot to generate multiple output +# files in one run (i.e. multiple -o and -T options on the command line). This +# makes dot run faster, but since only newer versions of dot (>1.8.10) +# support this, this feature is disabled by default. + +DOT_MULTI_TARGETS = NO + +# If the GENERATE_LEGEND tag is set to YES (the default) Doxygen will +# generate a legend page explaining the meaning of the various boxes and +# arrows in the dot generated graphs. + +GENERATE_LEGEND = YES + +# If the DOT_CLEANUP tag is set to YES (the default) Doxygen will +# remove the intermediate dot files that are used to generate +# the various graphs. + +DOT_CLEANUP = YES diff --git a/ffmpeg1/doc/Makefile b/ffmpeg1/doc/Makefile new file mode 100644 index 0000000..a861655 --- /dev/null +++ b/ffmpeg1/doc/Makefile @@ -0,0 +1,103 @@ +LIBRARIES-$(CONFIG_AVUTIL) += libavutil +LIBRARIES-$(CONFIG_SWSCALE) += libswscale +LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample +LIBRARIES-$(CONFIG_AVCODEC) += libavcodec +LIBRARIES-$(CONFIG_AVFORMAT) += libavformat +LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice +LIBRARIES-$(CONFIG_AVFILTER) += libavfilter + +COMPONENTS-yes = $(PROGS-yes) +COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils +COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler +COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler +COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters +COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols +COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices +COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters + +MANPAGES = $(COMPONENTS-yes:%=doc/%.1) $(LIBRARIES-yes:%=doc/%.3) +PODPAGES = $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod) +HTMLPAGES = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \ + doc/developer.html \ + doc/faq.html \ + doc/fate.html \ + doc/general.html \ + doc/git-howto.html \ + doc/nut.html \ + doc/platform.html \ + +TXTPAGES = doc/fate.txt \ + + +DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES) +DOCS-$(CONFIG_PODPAGES) += $(PODPAGES) +DOCS-$(CONFIG_MANPAGES) += $(MANPAGES) +DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES) +DOCS = $(DOCS-yes) + +all-$(CONFIG_DOC): doc + +doc: documentation + +apidoc: doc/doxy/html +documentation: $(DOCS) + +TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d) + +doc/%.txt: TAG = TXT +doc/%.txt: doc/%.texi + $(Q)$(TEXIDEP) + $(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null + +GENTEXI = format codec +GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi) + +$(GENTEXI): TAG = GENTEXI +$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF) + $(M)doc/print_options $* > $@ + +doc/%.html: TAG = HTML +doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< + +doc/%.pod: TAG = POD +doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@ + +doc/%.1 doc/%.3: TAG = MAN +doc/%.1: doc/%.pod $(GENTEXI) + $(M)pod2man --section=1 --center=" " --release=" " $< > $@ +doc/%.3: doc/%.pod $(GENTEXI) + $(M)pod2man --section=3 --center=" " --release=" " $< > $@ + +$(DOCS) doc/doxy/html: | doc/ + +doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS) + $(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^ + +install-man: + +ifdef CONFIG_MANPAGES +install-progs-$(CONFIG_DOC): install-man + +install-man: $(MANPAGES) + $(Q)mkdir -p "$(MANDIR)/man1" + $(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1" +endif + +uninstall: uninstall-man + +uninstall-man: + $(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES)) + +clean:: docclean + +docclean: + $(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi + $(RM) -r doc/doxy/html + +-include $(wildcard $(DOCS:%=%.d)) + +.PHONY: apidoc doc documentation diff --git a/ffmpeg1/doc/RELEASE_NOTES b/ffmpeg1/doc/RELEASE_NOTES new file mode 100644 index 0000000..2faf40d --- /dev/null +++ b/ffmpeg1/doc/RELEASE_NOTES @@ -0,0 +1,16 @@ +Release Notes +============= + +* 1.2 "Magic" March, 2013 + + +General notes +------------- +See the Changelog file for a list of significant changes. Note, there +are many more new features and bugfixes than whats listed there. + +Bugreports against FFmpeg git master or the most recent FFmpeg release are +accepted. If you are experiencing issues with any formally released version of +FFmpeg, please try git master to check if the issue still exists. If it does, +make your report against the development code following the usual bug reporting +guidelines. diff --git a/ffmpeg1/doc/authors.texi b/ffmpeg1/doc/authors.texi new file mode 100644 index 0000000..6c8c1d7 --- /dev/null +++ b/ffmpeg1/doc/authors.texi @@ -0,0 +1,11 @@ +@chapter Authors + +The FFmpeg developers. + +For details about the authorship, see the Git history of the project +(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command +@command{git log} in the FFmpeg source directory, or browsing the +online repository at @url{http://source.ffmpeg.org}. + +Maintainers for the specific components are listed in the file +@file{MAINTAINERS} in the source code tree. diff --git a/ffmpeg1/doc/avtools-common-opts.texi b/ffmpeg1/doc/avtools-common-opts.texi new file mode 100644 index 0000000..d9d0bd0 --- /dev/null +++ b/ffmpeg1/doc/avtools-common-opts.texi @@ -0,0 +1,211 @@ +All the numerical options, if not specified otherwise, accept in input +a string representing a number, which may contain one of the +SI unit prefixes, for example 'K', 'M', 'G'. +If 'i' is appended after the prefix, binary prefixes are used, +which are based on powers of 1024 instead of powers of 1000. +The 'B' postfix multiplies the value by 8, and can be +appended after a unit prefix or used alone. This allows using for +example 'KB', 'MiB', 'G' and 'B' as number postfix. + +Options which do not take arguments are boolean options, and set the +corresponding value to true. They can be set to false by prefixing +with "no" the option name, for example using "-nofoo" in the +command line will set to false the boolean option with name "foo". + +@anchor{Stream specifiers} +@section Stream specifiers +Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers +are used to precisely specify which stream(s) does a given option belong to. + +A stream specifier is a string generally appended to the option name and +separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains +@code{a:1} stream specifier, which matches the second audio stream. Therefore it +would select the ac3 codec for the second audio stream. + +A stream specifier can match several streams, the option is then applied to all +of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio +streams. + +An empty stream specifier matches all streams, for example @code{-codec copy} +or @code{-codec: copy} would copy all the streams without reencoding. + +Possible forms of stream specifiers are: +@table @option +@item @var{stream_index} +Matches the stream with this index. E.g. @code{-threads:1 4} would set the +thread count for the second stream to 4. +@item @var{stream_type}[:@var{stream_index}] +@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle, +'d' for data and 't' for attachments. If @var{stream_index} is given, then +matches stream number @var{stream_index} of this type. Otherwise matches all +streams of this type. +@item p:@var{program_id}[:@var{stream_index}] +If @var{stream_index} is given, then matches stream number @var{stream_index} in +program with id @var{program_id}. Otherwise matches all streams in this program. +@item #@var{stream_id} +Matches the stream by format-specific ID. +@end table + +@section Generic options + +These options are shared amongst the av* tools. + +@table @option + +@item -L +Show license. + +@item -h, -?, -help, --help [@var{arg}] +Show help. An optional parameter may be specified to print help about a specific +item. + +Possible values of @var{arg} are: +@table @option +@item decoder=@var{decoder_name} +Print detailed information about the decoder named @var{decoder_name}. Use the +@option{-decoders} option to get a list of all decoders. + +@item encoder=@var{encoder_name} +Print detailed information about the encoder named @var{encoder_name}. Use the +@option{-encoders} option to get a list of all encoders. + +@item demuxer=@var{demuxer_name} +Print detailed information about the demuxer named @var{demuxer_name}. Use the +@option{-formats} option to get a list of all demuxers and muxers. + +@item muxer=@var{muxer_name} +Print detailed information about the muxer named @var{muxer_name}. Use the +@option{-formats} option to get a list of all muxers and demuxers. + +@end table + +@item -version +Show version. + +@item -formats +Show available formats. + +The fields preceding the format names have the following meanings: +@table @samp +@item D +Decoding available +@item E +Encoding available +@end table + +@item -codecs +Show all codecs known to libavcodec. + +Note that the term 'codec' is used throughout this documentation as a shortcut +for what is more correctly called a media bitstream format. + +@item -decoders +Show available decoders. + +@item -encoders +Show all available encoders. + +@item -bsfs +Show available bitstream filters. + +@item -protocols +Show available protocols. + +@item -filters +Show available libavfilter filters. + +@item -pix_fmts +Show available pixel formats. + +@item -sample_fmts +Show available sample formats. + +@item -layouts +Show channel names and standard channel layouts. + +@item -loglevel @var{loglevel} | -v @var{loglevel} +Set the logging level used by the library. +@var{loglevel} is a number or a string containing one of the following values: +@table @samp +@item quiet +@item panic +@item fatal +@item error +@item warning +@item info +@item verbose +@item debug +@end table + +By default the program logs to stderr, if coloring is supported by the +terminal, colors are used to mark errors and warnings. Log coloring +can be disabled setting the environment variable +@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting +the environment variable @env{AV_LOG_FORCE_COLOR}. +The use of the environment variable @env{NO_COLOR} is deprecated and +will be dropped in a following FFmpeg version. + +@item -report +Dump full command line and console output to a file named +@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current +directory. +This file can be useful for bug reports. +It also implies @code{-loglevel verbose}. + +Setting the environment variable @code{FFREPORT} to any value has the +same effect. If the value is a ':'-separated key=value sequence, these +options will affect the report; options values must be escaped if they +contain special characters or the options delimiter ':' (see the +``Quoting and escaping'' section in the ffmpeg-utils manual). The +following option is recognized: +@table @option +@item file +set the file name to use for the report; @code{%p} is expanded to the name +of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded +to a plain @code{%} +@end table + +Errors in parsing the environment variable are not fatal, and will not +appear in the report. + +@item -cpuflags flags (@emph{global}) +Allows setting and clearing cpu flags. This option is intended +for testing. Do not use it unless you know what you're doing. +@example +ffmpeg -cpuflags -sse+mmx ... +ffmpeg -cpuflags mmx ... +ffmpeg -cpuflags 0 ... +@end example + +@end table + +@section AVOptions + +These options are provided directly by the libavformat, libavdevice and +libavcodec libraries. To see the list of available AVOptions, use the +@option{-help} option. They are separated into two categories: +@table @option +@item generic +These options can be set for any container, codec or device. Generic options +are listed under AVFormatContext options for containers/devices and under +AVCodecContext options for codecs. +@item private +These options are specific to the given container, device or codec. Private +options are listed under their corresponding containers/devices/codecs. +@end table + +For example to write an ID3v2.3 header instead of a default ID3v2.4 to +an MP3 file, use the @option{id3v2_version} private option of the MP3 +muxer: +@example +ffmpeg -i input.flac -id3v2_version 3 out.mp3 +@end example + +All codec AVOptions are obviously per-stream, so the chapter on stream +specifiers applies to them + +Note @option{-nooption} syntax cannot be used for boolean AVOptions, +use @option{-option 0}/@option{-option 1}. + +Note2 old undocumented way of specifying per-stream AVOptions by prepending +v/a/s to the options name is now obsolete and will be removed soon. diff --git a/ffmpeg1/doc/avutil.txt b/ffmpeg1/doc/avutil.txt new file mode 100644 index 0000000..0847683 --- /dev/null +++ b/ffmpeg1/doc/avutil.txt @@ -0,0 +1,36 @@ +AVUtil +====== +libavutil is a small lightweight library of generally useful functions. +It is not a library for code needed by both libavcodec and libavformat. + + +Overview: +========= +adler32.c adler32 checksum +aes.c AES encryption and decryption +fifo.c resizeable first in first out buffer +intfloat_readwrite.c portable reading and writing of floating point values +log.c "printf" with context and level +md5.c MD5 Message-Digest Algorithm +rational.c code to perform exact calculations with rational numbers +tree.c generic AVL tree +crc.c generic CRC checksumming code +integer.c 128bit integer math +lls.c +mathematics.c greatest common divisor, integer sqrt, integer log2, ... +mem.c memory allocation routines with guaranteed alignment + +Headers: +bswap.h big/little/native-endian conversion code +x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code +avutil.h +common.h +intreadwrite.h reading and writing of unaligned big/little/native-endian integers + + +Goals: +====== +* Modular (few interdependencies and the possibility of disabling individual parts during ./configure) +* Small (source and object) +* Efficient (low CPU and memory usage) +* Useful (avoid useless features almost no one needs) diff --git a/ffmpeg1/doc/bitstream_filters.texi b/ffmpeg1/doc/bitstream_filters.texi new file mode 100644 index 0000000..2ee00c1 --- /dev/null +++ b/ffmpeg1/doc/bitstream_filters.texi @@ -0,0 +1,91 @@ +@chapter Bitstream Filters +@c man begin BITSTREAM FILTERS + +When you configure your FFmpeg build, all the supported bitstream +filters are enabled by default. You can list all available ones using +the configure option @code{--list-bsfs}. + +You can disable all the bitstream filters using the configure option +@code{--disable-bsfs}, and selectively enable any bitstream filter using +the option @code{--enable-bsf=BSF}, or you can disable a particular +bitstream filter using the option @code{--disable-bsf=BSF}. + +The option @code{-bsfs} of the ff* tools will display the list of +all the supported bitstream filters included in your build. + +Below is a description of the currently available bitstream filters. + +@section aac_adtstoasc + +@section chomp + +@section dump_extradata + +@section h264_mp4toannexb + +Convert an H.264 bitstream from length prefixed mode to start code +prefixed mode (as defined in the Annex B of the ITU-T H.264 +specification). + +This is required by some streaming formats, typically the MPEG-2 +transport stream format ("mpegts"). + +For example to remux an MP4 file containing an H.264 stream to mpegts +format with @command{ffmpeg}, you can use the command: + +@example +ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts +@end example + +@section imx_dump_header + +@section mjpeg2jpeg + +Convert MJPEG/AVI1 packets to full JPEG/JFIF packets. + +MJPEG is a video codec wherein each video frame is essentially a +JPEG image. The individual frames can be extracted without loss, +e.g. by + +@example +ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg +@end example + +Unfortunately, these chunks are incomplete JPEG images, because +they lack the DHT segment required for decoding. Quoting from +@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}: + +Avery Lee, writing in the rec.video.desktop newsgroup in 2001, +commented that "MJPEG, or at least the MJPEG in AVIs having the +MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- +Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, +and it must use basic Huffman encoding, not arithmetic or +progressive. . . . You can indeed extract the MJPEG frames and +decode them with a regular JPEG decoder, but you have to prepend +the DHT segment to them, or else the decoder won't have any idea +how to decompress the data. The exact table necessary is given in +the OpenDML spec." + +This bitstream filter patches the header of frames extracted from an MJPEG +stream (carrying the AVI1 header ID and lacking a DHT segment) to +produce fully qualified JPEG images. + +@example +ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg +exiftran -i -9 frame*.jpg +ffmpeg -i frame_%d.jpg -c:v copy rotated.avi +@end example + +@section mjpega_dump_header + +@section movsub + +@section mp3_header_compress + +@section mp3_header_decompress + +@section noise + +@section remove_extradata + +@c man end BITSTREAM FILTERS diff --git a/ffmpeg1/doc/build_system.txt b/ffmpeg1/doc/build_system.txt new file mode 100644 index 0000000..36c141e --- /dev/null +++ b/ffmpeg1/doc/build_system.txt @@ -0,0 +1,50 @@ +FFmpeg currently uses a custom build system, this text attempts to document +some of its obscure features and options. + +Makefile variables: + +V + Disable the default terse mode, the full command issued by make and its + output will be shown on the screen. + +DESTDIR + Destination directory for the install targets, useful to prepare packages + or install FFmpeg in cross-environments. + +Makefile targets: + +all + Default target, builds all the libraries and the executables. + +fate + Run the fate test suite, note you must have installed it + +fate-list + Will list all fate/regression test targets + +install + Install headers, libraries and programs. + +libavformat/output-example + Build the libavformat basic example. + +libavcodec/api-example + Build the libavcodec basic example. + +libswscale/swscale-test + Build the swscale self-test (useful also as example). + + +Useful standard make commands: +make -t <target> + Touch all files that otherwise would be build, this is useful to reduce + unneeded rebuilding when changing headers, but note you must force rebuilds + of files that actually need it by hand then. + +make -j<num> + rebuild with multiple jobs at the same time. Faster on multi processor systems + +make -k + continue build in case of errors, this is useful for the regression tests + sometimes but note it will still not run all reg tests. + diff --git a/ffmpeg1/doc/decoders.texi b/ffmpeg1/doc/decoders.texi new file mode 100644 index 0000000..2d812a2 --- /dev/null +++ b/ffmpeg1/doc/decoders.texi @@ -0,0 +1,89 @@ +@chapter Decoders +@c man begin DECODERS + +Decoders are configured elements in FFmpeg which allow the decoding of +multimedia streams. + +When you configure your FFmpeg build, all the supported native decoders +are enabled by default. Decoders requiring an external library must be enabled +manually via the corresponding @code{--enable-lib} option. You can list all +available decoders using the configure option @code{--list-decoders}. + +You can disable all the decoders with the configure option +@code{--disable-decoders} and selectively enable / disable single decoders +with the options @code{--enable-decoder=@var{DECODER}} / +@code{--disable-decoder=@var{DECODER}}. + +The option @code{-codecs} of the ff* tools will display the list of +enabled decoders. + +@c man end DECODERS + +@chapter Video Decoders +@c man begin VIDEO DECODERS + +A description of some of the currently available video decoders +follows. + +@section rawvideo + +Raw video decoder. + +This decoder decodes rawvideo streams. + +@subsection Options + +@table @option +@item top @var{top_field_first} +Specify the assumed field type of the input video. +@table @option +@item -1 +the video is assumed to be progressive (default) +@item 0 +bottom-field-first is assumed +@item 1 +top-field-first is assumed +@end table + +@end table + +@c man end VIDEO DECODERS + +@chapter Audio Decoders +@c man begin AUDIO DECODERS + +@section ffwavesynth + +Internal wave synthetizer. + +This decoder generates wave patterns according to predefined sequences. Its +use is purely internal and the format of the data it accepts is not publicly +documented. + +@c man end AUDIO DECODERS + +@chapter Subtitles Decoders +@c man begin SUBTILES DECODERS + +@section dvdsub + +This codec decodes the bitmap subtitles used in DVDs; the same subtitles can +also be found in VobSub file pairs and in some Matroska files. + +@subsection Options + +@table @option +@item palette +Specify the global palette used by the bitmaps. When stored in VobSub, the +palette is normally specified in the index file; in Matroska, the palette is +stored in the codec extra-data in the same format as in VobSub. In DVDs, the +palette is stored in the IFO file, and therefore not available when reading +from dumped VOB files. + +The format for this option is a string containing 16 24-bits hexadecimal +numbers (without 0x prefix) separated by comas, for example @code{0d00ee, +ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, +7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}. +@end table + +@c man end SUBTILES DECODERS diff --git a/ffmpeg1/doc/default.css b/ffmpeg1/doc/default.css new file mode 100644 index 0000000..77a3514 --- /dev/null +++ b/ffmpeg1/doc/default.css @@ -0,0 +1,149 @@ +a { + color: #2D6198; +} + +a:visited { + color: #884488; +} + +#banner { + background-color: white; + position: relative; + text-align: center; +} + +#banner img { + padding-bottom: 1px; + padding-top: 5px; +} + +#body { + margin-left: 1em; + margin-right: 1em; +} + +body { + background-color: #313131; + margin: 0; + text-align: justify; +} + +.center { + margin-left: auto; + margin-right: auto; + text-align: center; +} + +#container { + background-color: white; + color: #202020; + margin-left: 1em; + margin-right: 1em; +} + +#footer { + text-align: center; +} + +h1, h2, h3 { + padding-left: 0.4em; + border-radius: 4px; + padding-bottom: 0.2em; + padding-top: 0.2em; + border: 1px solid #6A996A; +} + +h1 { + background-color: #7BB37B; + color: #151515; + font-size: 1.2em; + padding-bottom: 0.3em; + padding-top: 0.3em; +} + +h2 { + color: #313131; + font-size: 0.9em; + background-color: #ABE3AB; +} + +h3 { + color: #313131; + font-size: 0.8em; + margin-bottom: -8px; + background-color: #BBF3BB; +} + +img { + border: 0; +} + +#navbar { + background-color: #738073; + border-bottom: 1px solid #5C665C; + border-top: 1px solid #5C665C; + margin-top: 12px; + padding: 0.3em; + position: relative; + text-align: center; +} + +#navbar a, #navbar_secondary a { + color: white; + padding: 0.3em; + text-decoration: none; +} + +#navbar a:hover, #navbar_secondary a:hover { + background-color: #313131; + color: white; + text-decoration: none; +} + +#navbar_secondary { + background-color: #738073; + border-bottom: 1px solid #5C665C; + border-left: 1px solid #5C665C; + border-right: 1px solid #5C665C; + padding: 0.3em; + position: relative; + text-align: center; +} + +p { + margin-left: 1em; + margin-right: 1em; +} + +pre { + margin-left: 3em; + margin-right: 3em; + padding: 0.3em; + border: 1px solid #bbb; + background-color: #f7f7f7; +} + +dl dt { + font-weight: bold; +} + +#proj_desc { + font-size: 1.2em; +} + +#repos { + margin-left: 1em; + margin-right: 1em; + border-collapse: collapse; + border: solid 1px #6A996A; +} + +#repos th { + background-color: #7BB37B; + border: solid 1px #6A996A; +} + +#repos td { + padding: 0.2em; + border: solid 1px #6A996A; +} diff --git a/ffmpeg1/doc/demuxers.texi b/ffmpeg1/doc/demuxers.texi new file mode 100644 index 0000000..fc50871 --- /dev/null +++ b/ffmpeg1/doc/demuxers.texi @@ -0,0 +1,311 @@ +@chapter Demuxers +@c man begin DEMUXERS + +Demuxers are configured elements in FFmpeg which allow to read the +multimedia streams from a particular type of file. + +When you configure your FFmpeg build, all the supported demuxers +are enabled by default. You can list all available ones using the +configure option @code{--list-demuxers}. + +You can disable all the demuxers using the configure option +@code{--disable-demuxers}, and selectively enable a single demuxer with +the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it +with the option @code{--disable-demuxer=@var{DEMUXER}}. + +The option @code{-formats} of the ff* tools will display the list of +enabled demuxers. + +The description of some of the currently available demuxers follows. + +@section applehttp + +Apple HTTP Live Streaming demuxer. + +This demuxer presents all AVStreams from all variant streams. +The id field is set to the bitrate variant index number. By setting +the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), +the caller can decide which variant streams to actually receive. +The total bitrate of the variant that the stream belongs to is +available in a metadata key named "variant_bitrate". + +@anchor{concat} +@section concat + +Virtual concatenation script demuxer. + +This demuxer reads a list of files and other directives from a text file and +demuxes them one after the other, as if all their packet had been muxed +together. + +The timestamps in the files are adjusted so that the first file starts at 0 +and each next file starts where the previous one finishes. Note that it is +done globally and may cause gaps if all streams do not have exactly the same +length. + +All files must have the same streams (same codecs, same time base, etc.). + +The duration of each file is used to adjust the timestamps of the next file: +if the duration is incorrect (because it was computed using the bit-rate or +because the file is truncated, for example), it can cause artifacts. The +@code{duration} directive can be used to override the duration stored in +each file. + +@subsection Syntax + +The script is a text file in extended-ASCII, with one directive per line. +Empty lines, leading spaces and lines starting with '#' are ignored. The +following directive is recognized: + +@table @option + +@item @code{file @var{path}} +Path to a file to read; special characters and spaces must be escaped with +backslash or single quotes. + +All subsequent directives apply to that file. + +@item @code{ffconcat version 1.0} +Identify the script type and version. It also sets the @option{safe} option +to 1 if it was to its default -1. + +To make FFmpeg recognize the format automatically, this directive must +appears exactly as is (no extra space or byte-order-mark) on the very first +line of the script. + +@item @code{duration @var{dur}} +Duration of the file. This information can be specified from the file; +specifying it here may be more efficient or help if the information from the +file is not available or accurate. + +If the duration is set for all files, then it is possible to seek in the +whole concatenated video. + +@end table + +@subsection Options + +This demuxer accepts the following option: + +@table @option + +@item safe +If set to 1, reject unsafe file paths. A file path is considered safe if it +does not contain a protocol specification and is relative and all components +only contain characters from the portable character set (letters, digits, +period, underscore and hyphen) and have no period at the beginning of a +component. + +If set to 0, any file name is accepted. + +The default is -1, it is equivalent to 1 if the format was automatically +probed and 0 otherwise. + +@end table + +@section image2 + +Image file demuxer. + +This demuxer reads from a list of image files specified by a pattern. +The syntax and meaning of the pattern is specified by the +option @var{pattern_type}. + +The pattern may contain a suffix which is used to automatically +determine the format of the images contained in the files. + +The size, the pixel format, and the format of each image must be the +same for all the files in the sequence. + +This demuxer accepts the following options: +@table @option +@item framerate +Set the framerate for the video stream. It defaults to 25. +@item loop +If set to 1, loop over the input. Default value is 0. +@item pattern_type +Select the pattern type used to interpret the provided filename. + +@var{pattern_type} accepts one of the following values. +@table @option +@item sequence +Select a sequence pattern type, used to specify a sequence of files +indexed by sequential numbers. + +A sequence pattern may contain the string "%d" or "%0@var{N}d", which +specifies the position of the characters representing a sequential +number in each filename matched by the pattern. If the form +"%d0@var{N}d" is used, the string representing the number in each +filename is 0-padded and @var{N} is the total number of 0-padded +digits representing the number. The literal character '%' can be +specified in the pattern with the string "%%". + +If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of +the file list specified by the pattern must contain a number +inclusively contained between @var{start_number} and +@var{start_number}+@var{start_number_range}-1, and all the following +numbers must be sequential. + +For example the pattern "img-%03d.bmp" will match a sequence of +filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., +@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a +sequence of filenames of the form @file{i%m%g-1.jpg}, +@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc. + +Note that the pattern must not necessarily contain "%d" or +"%0@var{N}d", for example to convert a single image file +@file{img.jpeg} you can employ the command: +@example +ffmpeg -i img.jpeg img.png +@end example + +@item glob +Select a glob wildcard pattern type. + +The pattern is interpreted like a @code{glob()} pattern. This is only +selectable if libavformat was compiled with globbing support. + +@item glob_sequence @emph{(deprecated, will be removed)} +Select a mixed glob wildcard/sequence pattern. + +If your version of libavformat was compiled with globbing support, and +the provided pattern contains at least one glob meta character among +@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is +interpreted like a @code{glob()} pattern, otherwise it is interpreted +like a sequence pattern. + +All glob special characters @code{%*?[]@{@}} must be prefixed +with "%". To escape a literal "%" you shall use "%%". + +For example the pattern @code{foo-%*.jpeg} will match all the +filenames prefixed by "foo-" and terminating with ".jpeg", and +@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with +"foo-", followed by a sequence of three characters, and terminating +with ".jpeg". + +This pattern type is deprecated in favor of @var{glob} and +@var{sequence}. +@end table + +Default value is @var{glob_sequence}. +@item pixel_format +Set the pixel format of the images to read. If not specified the pixel +format is guessed from the first image file in the sequence. +@item start_number +Set the index of the file matched by the image file pattern to start +to read from. Default value is 0. +@item start_number_range +Set the index interval range to check when looking for the first image +file in the sequence, starting from @var{start_number}. Default value +is 5. +@item video_size +Set the video size of the images to read. If not specified the video +size is guessed from the first image file in the sequence. +@end table + +@subsection Examples + +@itemize +@item +Use @command{ffmpeg} for creating a video from the images in the file +sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an +input frame rate of 10 frames per second: +@example +ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv +@end example + +@item +As above, but start by reading from a file with index 100 in the sequence: +@example +ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv +@end example + +@item +Read images matching the "*.png" glob pattern , that is all the files +terminating with the ".png" suffix: +@example +ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv +@end example +@end itemize + +@section rawvideo + +Raw video demuxer. + +This demuxer allows to read raw video data. Since there is no header +specifying the assumed video parameters, the user must specify them +in order to be able to decode the data correctly. + +This demuxer accepts the following options: +@table @option + +@item framerate +Set input video frame rate. Default value is 25. + +@item pixel_format +Set the input video pixel format. Default value is @code{yuv420p}. + +@item video_size +Set the input video size. This value must be specified explicitly. +@end table + +For example to read a rawvideo file @file{input.raw} with +@command{ffplay}, assuming a pixel format of @code{rgb24}, a video +size of @code{320x240}, and a frame rate of 10 images per second, use +the command: +@example +ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw +@end example + +@section sbg + +SBaGen script demuxer. + +This demuxer reads the script language used by SBaGen +@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG +script looks like that: +@example +-SE +a: 300-2.5/3 440+4.5/0 +b: 300-2.5/0 440+4.5/3 +off: - +NOW == a ++0:07:00 == b ++0:14:00 == a ++0:21:00 == b ++0:30:00 off +@end example + +A SBG script can mix absolute and relative timestamps. If the script uses +either only absolute timestamps (including the script start time) or only +relative ones, then its layout is fixed, and the conversion is +straightforward. On the other hand, if the script mixes both kind of +timestamps, then the @var{NOW} reference for relative timestamps will be +taken from the current time of day at the time the script is read, and the +script layout will be frozen according to that reference. That means that if +the script is directly played, the actual times will match the absolute +timestamps up to the sound controller's clock accuracy, but if the user +somehow pauses the playback or seeks, all times will be shifted accordingly. + +@section tedcaptions + +JSON captions used for @url{http://www.ted.com/, TED Talks}. + +TED does not provide links to the captions, but they can be guessed from the +page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree +contains a bookmarklet to expose them. + +This demuxer accepts the following option: +@table @option +@item start_time +Set the start time of the TED talk, in milliseconds. The default is 15000 +(15s). It is used to sync the captions with the downloadable videos, because +they include a 15s intro. +@end table + +Example: convert the captions to a format most players understand: +@example +ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt +@end example + +@c man end DEMUXERS diff --git a/ffmpeg1/doc/developer.texi b/ffmpeg1/doc/developer.texi new file mode 100644 index 0000000..bd3f7a7 --- /dev/null +++ b/ffmpeg1/doc/developer.texi @@ -0,0 +1,668 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Developer Documentation +@titlepage +@center @titlefont{Developer Documentation} +@end titlepage + +@top + +@contents + +@chapter Developers Guide + +@section API +@itemize @bullet +@item libavcodec is the library containing the codecs (both encoding and +decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use +it. + +@item libavformat is the library containing the file format handling (mux and +demux code for several formats). Look at @file{ffplay.c} to use it in a +player. See @file{doc/examples/muxing.c} to use it to generate audio or video +streams. + +@end itemize + +@section Integrating libavcodec or libavformat in your program + +You can integrate all the source code of the libraries to link them +statically to avoid any version problem. All you need is to provide a +'config.mak' and a 'config.h' in the parent directory. See the defines +generated by ./configure to understand what is needed. + +You can use libavcodec or libavformat in your commercial program, but +@emph{any patch you make must be published}. The best way to proceed is +to send your patches to the FFmpeg mailing list. + +@section Contributing + +There are 3 ways by which code gets into ffmpeg. +@itemize @bullet +@item Submitting Patches to the main developer mailing list + see @ref{Submitting patches} for details. +@item Directly committing changes to the main tree. +@item Committing changes to a git clone, for example on github.com or + gitorious.org. And asking us to merge these changes. +@end itemize + +Whichever way, changes should be reviewed by the maintainer of the code +before they are committed. And they should follow the @ref{Coding Rules}. +The developer making the commit and the author are responsible for their changes +and should try to fix issues their commit causes. + +@anchor{Coding Rules} +@section Coding Rules + +@subsection Code formatting conventions + +There are the following guidelines regarding the indentation in files: +@itemize @bullet +@item +Indent size is 4. +@item +The TAB character is forbidden outside of Makefiles as is any +form of trailing whitespace. Commits containing either will be +rejected by the git repository. +@item +You should try to limit your code lines to 80 characters; however, do so if +and only if this improves readability. +@end itemize +The presentation is one inspired by 'indent -i4 -kr -nut'. + +The main priority in FFmpeg is simplicity and small code size in order to +minimize the bug count. + +@subsection Comments +Use the JavaDoc/Doxygen format (see examples below) so that code documentation +can be generated automatically. All nontrivial functions should have a comment +above them explaining what the function does, even if it is just one sentence. +All structures and their member variables should be documented, too. + +Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace +@code{//!} with @code{///} and similar. Also @@ syntax should be employed +for markup commands, i.e. use @code{@@param} and not @code{\param}. + +@example +/** + * @@file + * MPEG codec. + * @@author ... + */ + +/** + * Summary sentence. + * more text ... + * ... + */ +typedef struct Foobar@{ + int var1; /**< var1 description */ + int var2; ///< var2 description + /** var3 description */ + int var3; +@} Foobar; + +/** + * Summary sentence. + * more text ... + * ... + * @@param my_parameter description of my_parameter + * @@return return value description + */ +int myfunc(int my_parameter) +... +@end example + +@subsection C language features + +FFmpeg is programmed in the ISO C90 language with a few additional +features from ISO C99, namely: +@itemize @bullet +@item +the @samp{inline} keyword; +@item +@samp{//} comments; +@item +designated struct initializers (@samp{struct s x = @{ .i = 17 @};}) +@item +compound literals (@samp{x = (struct s) @{ 17, 23 @};}) +@end itemize + +These features are supported by all compilers we care about, so we will not +accept patches to remove their use unless they absolutely do not impair +clarity and performance. + +All code must compile with recent versions of GCC and a number of other +currently supported compilers. To ensure compatibility, please do not use +additional C99 features or GCC extensions. Especially watch out for: +@itemize @bullet +@item +mixing statements and declarations; +@item +@samp{long long} (use @samp{int64_t} instead); +@item +@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar; +@item +GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). +@end itemize + +@subsection Naming conventions +All names should be composed with underscores (_), not CamelCase. For example, +@samp{avfilter_get_video_buffer} is an acceptable function name and +@samp{AVFilterGetVideo} is not. The exception from this are type names, like +for example structs and enums; they should always be in the CamelCase + +There are the following conventions for naming variables and functions: +@itemize @bullet +@item +For local variables no prefix is required. +@item +For variables and functions declared as @code{static} no prefix is required. +@item +For variables and functions used internally by a library an @code{ff_} +prefix should be used, e.g. @samp{ff_w64_demuxer}. +@item +For variables and functions used internally across multiple libraries, use +@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}. +@item +Each library has its own prefix for public symbols, in addition to the +commonly used @code{av_} (@code{avformat_} for libavformat, +@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc). +Check the existing code and choose names accordingly. +Note that some symbols without these prefixes are also exported for +retro-compatibility reasons. These exceptions are declared in the +@code{lib<name>/lib<name>.v} files. +@end itemize + +Furthermore, name space reserved for the system should not be invaded. +Identifiers ending in @code{_t} are reserved by +@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}. +Also avoid names starting with @code{__} or @code{_} followed by an uppercase +letter as they are reserved by the C standard. Names starting with @code{_} +are reserved at the file level and may not be used for externally visible +symbols. If in doubt, just avoid names starting with @code{_} altogether. + +@subsection Miscellaneous conventions +@itemize @bullet +@item +fprintf and printf are forbidden in libavformat and libavcodec, +please use av_log() instead. +@item +Casts should be used only when necessary. Unneeded parentheses +should also be avoided if they don't make the code easier to understand. +@end itemize + +@subsection Editor configuration +In order to configure Vim to follow FFmpeg formatting conventions, paste +the following snippet into your @file{.vimrc}: +@example +" indentation rules for FFmpeg: 4 spaces, no tabs +set expandtab +set shiftwidth=4 +set softtabstop=4 +set cindent +set cinoptions=(0 +" Allow tabs in Makefiles. +autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8 +" Trailing whitespace and tabs are forbidden, so highlight them. +highlight ForbiddenWhitespace ctermbg=red guibg=red +match ForbiddenWhitespace /\s\+$\|\t/ +" Do not highlight spaces at the end of line while typing on that line. +autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/ +@end example + +For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}: +@example +(c-add-style "ffmpeg" + '("k&r" + (c-basic-offset . 4) + (indent-tabs-mode . nil) + (show-trailing-whitespace . t) + (c-offsets-alist + (statement-cont . (c-lineup-assignments +))) + ) + ) +(setq c-default-style "ffmpeg") +@end example + +@section Development Policy + +@enumerate +@item + Contributions should be licensed under the + @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1}, + including an "or any later version" clause, or, if you prefer + a gift-style license, the + @uref{http://www.isc.org/software/license/, ISC} or + @uref{http://mit-license.org/, MIT} license. + @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including + an "or any later version" clause is also acceptable, but LGPL is + preferred. +@item + You must not commit code which breaks FFmpeg! (Meaning unfinished but + enabled code which breaks compilation or compiles but does not work or + breaks the regression tests) + You can commit unfinished stuff (for testing etc), but it must be disabled + (#ifdef etc) by default so it does not interfere with other developers' + work. +@item + The commit message should have a short first line in the form of + a @samp{topic: short description} as a header, separated by a newline + from the body consisting of an explanation of why the change is necessary. + If the commit fixes a known bug on the bug tracker, the commit message + should include its bug ID. Referring to the issue on the bug tracker does + not exempt you from writing an excerpt of the bug in the commit message. +@item + You do not have to over-test things. If it works for you, and you think it + should work for others, then commit. If your code has problems + (portability, triggers compiler bugs, unusual environment etc) they will be + reported and eventually fixed. +@item + Do not commit unrelated changes together, split them into self-contained + pieces. Also do not forget that if part B depends on part A, but A does not + depend on B, then A can and should be committed first and separate from B. + Keeping changes well split into self-contained parts makes reviewing and + understanding them on the commit log mailing list easier. This also helps + in case of debugging later on. + Also if you have doubts about splitting or not splitting, do not hesitate to + ask/discuss it on the developer mailing list. +@item + Do not change behavior of the programs (renaming options etc) or public + API or ABI without first discussing it on the ffmpeg-devel mailing list. + Do not remove functionality from the code. Just improve! + + Note: Redundant code can be removed. +@item + Do not commit changes to the build system (Makefiles, configure script) + which change behavior, defaults etc, without asking first. The same + applies to compiler warning fixes, trivial looking fixes and to code + maintained by other developers. We usually have a reason for doing things + the way we do. Send your changes as patches to the ffmpeg-devel mailing + list, and if the code maintainers say OK, you may commit. This does not + apply to files you wrote and/or maintain. +@item + We refuse source indentation and other cosmetic changes if they are mixed + with functional changes, such commits will be rejected and removed. Every + developer has his own indentation style, you should not change it. Of course + if you (re)write something, you can use your own style, even though we would + prefer if the indentation throughout FFmpeg was consistent (Many projects + force a given indentation style - we do not.). If you really need to make + indentation changes (try to avoid this), separate them strictly from real + changes. + + NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code, + then either do NOT change the indentation of the inner part within (do not + move it to the right)! or do so in a separate commit +@item + Always fill out the commit log message. Describe in a few lines what you + changed and why. You can refer to mailing list postings if you fix a + particular bug. Comments such as "fixed!" or "Changed it." are unacceptable. + Recommended format: + area changed: Short 1 line description + + details describing what and why and giving references. +@item + Make sure the author of the commit is set correctly. (see git commit --author) + If you apply a patch, send an + answer to ffmpeg-devel (or wherever you got the patch from) saying that + you applied the patch. +@item + When applying patches that have been discussed (at length) on the mailing + list, reference the thread in the log message. +@item + Do NOT commit to code actively maintained by others without permission. + Send a patch to ffmpeg-devel instead. If no one answers within a reasonable + timeframe (12h for build failures and security fixes, 3 days small changes, + 1 week for big patches) then commit your patch if you think it is OK. + Also note, the maintainer can simply ask for more time to review! +@item + Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits + are sent there and reviewed by all the other developers. Bugs and possible + improvements or general questions regarding commits are discussed there. We + expect you to react if problems with your code are uncovered. +@item + Update the documentation if you change behavior or add features. If you are + unsure how best to do this, send a patch to ffmpeg-devel, the documentation + maintainer(s) will review and commit your stuff. +@item + Try to keep important discussions and requests (also) on the public + developer mailing list, so that all developers can benefit from them. +@item + Never write to unallocated memory, never write over the end of arrays, + always check values read from some untrusted source before using them + as array index or other risky things. +@item + Remember to check if you need to bump versions for the specific libav* + parts (libavutil, libavcodec, libavformat) you are changing. You need + to change the version integer. + Incrementing the first component means no backward compatibility to + previous versions (e.g. removal of a function from the public API). + Incrementing the second component means backward compatible change + (e.g. addition of a function to the public API or extension of an + existing data structure). + Incrementing the third component means a noteworthy binary compatible + change (e.g. encoder bug fix that matters for the decoder). The third + component always starts at 100 to distinguish FFmpeg from Libav. +@item + Compiler warnings indicate potential bugs or code with bad style. If a type of + warning always points to correct and clean code, that warning should + be disabled, not the code changed. + Thus the remaining warnings can either be bugs or correct code. + If it is a bug, the bug has to be fixed. If it is not, the code should + be changed to not generate a warning unless that causes a slowdown + or obfuscates the code. +@item + If you add a new file, give it a proper license header. Do not copy and + paste it from a random place, use an existing file as template. +@end enumerate + +We think our rules are not too hard. If you have comments, contact us. + +@anchor{Submitting patches} +@section Submitting patches + +First, read the @ref{Coding Rules} above if you did not yet, in particular +the rules regarding patch submission. + +When you submit your patch, please use @code{git format-patch} or +@code{git send-email}. We cannot read other diffs :-) + +Also please do not submit a patch which contains several unrelated changes. +Split it into separate, self-contained pieces. This does not mean splitting +file by file. Instead, make the patch as small as possible while still +keeping it as a logical unit that contains an individual change, even +if it spans multiple files. This makes reviewing your patches much easier +for us and greatly increases your chances of getting your patch applied. + +Use the patcheck tool of FFmpeg to check your patch. +The tool is located in the tools directory. + +Run the @ref{Regression tests} before submitting a patch in order to verify +it does not cause unexpected problems. + +It also helps quite a bit if you tell us what the patch does (for example +'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant +and has no lrint()') + +Also please if you send several patches, send each patch as a separate mail, +do not attach several unrelated patches to the same mail. + +Patches should be posted to the +@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel} +mailing list. Use @code{git send-email} when possible since it will properly +send patches without requiring extra care. If you cannot, then send patches +as base64-encoded attachments, so your patch is not trashed during +transmission. + +Your patch will be reviewed on the mailing list. You will likely be asked +to make some changes and are expected to send in an improved version that +incorporates the requests from the review. This process may go through +several iterations. Once your patch is deemed good enough, some developer +will pick it up and commit it to the official FFmpeg tree. + +Give us a few days to react. But if some time passes without reaction, +send a reminder by email. Your patch should eventually be dealt with. + + +@section New codecs or formats checklist + +@enumerate +@item + Did you use av_cold for codec initialization and close functions? +@item + Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or + AVInputFormat/AVOutputFormat struct? +@item + Did you bump the minor version number (and reset the micro version + number) in @file{libavcodec/version.h} or @file{libavformat/version.h}? +@item + Did you register it in @file{allcodecs.c} or @file{allformats.c}? +@item + Did you add the AVCodecID to @file{avcodec.h}? + When adding new codec IDs, also add an entry to the codec descriptor + list in @file{libavcodec/codec_desc.c}. +@item + If it has a FourCC, did you add it to @file{libavformat/riff.c}, + even if it is only a decoder? +@item + Did you add a rule to compile the appropriate files in the Makefile? + Remember to do this even if you're just adding a format to a file that is + already being compiled by some other rule, like a raw demuxer. +@item + Did you add an entry to the table of supported formats or codecs in + @file{doc/general.texi}? +@item + Did you add an entry in the Changelog? +@item + If it depends on a parser or a library, did you add that dependency in + configure? +@item + Did you @code{git add} the appropriate files before committing? +@item + Did you make sure it compiles standalone, i.e. with + @code{configure --disable-everything --enable-decoder=foo} + (or @code{--enable-demuxer} or whatever your component is)? +@end enumerate + + +@section patch submission checklist + +@enumerate +@item + Does @code{make fate} pass with the patch applied? +@item + Was the patch generated with git format-patch or send-email? +@item + Did you sign off your patch? (git commit -s) + See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning + of sign off. +@item + Did you provide a clear git commit log message? +@item + Is the patch against latest FFmpeg git master branch? +@item + Are you subscribed to ffmpeg-devel? + (the list is subscribers only due to spam) +@item + Have you checked that the changes are minimal, so that the same cannot be + achieved with a smaller patch and/or simpler final code? +@item + If the change is to speed critical code, did you benchmark it? +@item + If you did any benchmarks, did you provide them in the mail? +@item + Have you checked that the patch does not introduce buffer overflows or + other security issues? +@item + Did you test your decoder or demuxer against damaged data? If no, see + tools/trasher, the noise bitstream filter, and + @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer + should not crash, end in a (near) infinite loop, or allocate ridiculous + amounts of memory when fed damaged data. +@item + Does the patch not mix functional and cosmetic changes? +@item + Did you add tabs or trailing whitespace to the code? Both are forbidden. +@item + Is the patch attached to the email you send? +@item + Is the mime type of the patch correct? It should be text/x-diff or + text/x-patch or at least text/plain and not application/octet-stream. +@item + If the patch fixes a bug, did you provide a verbose analysis of the bug? +@item + If the patch fixes a bug, did you provide enough information, including + a sample, so the bug can be reproduced and the fix can be verified? + Note please do not attach samples >100k to mails but rather provide a + URL, you can upload to ftp://upload.ffmpeg.org +@item + Did you provide a verbose summary about what the patch does change? +@item + Did you provide a verbose explanation why it changes things like it does? +@item + Did you provide a verbose summary of the user visible advantages and + disadvantages if the patch is applied? +@item + Did you provide an example so we can verify the new feature added by the + patch easily? +@item + If you added a new file, did you insert a license header? It should be + taken from FFmpeg, not randomly copied and pasted from somewhere else. +@item + You should maintain alphabetical order in alphabetically ordered lists as + long as doing so does not break API/ABI compatibility. +@item + Lines with similar content should be aligned vertically when doing so + improves readability. +@item + Consider to add a regression test for your code. +@item + If you added YASM code please check that things still work with --disable-yasm +@item + Make sure you check the return values of function and return appropriate + error codes. Especially memory allocation functions like @code{av_malloc()} + are notoriously left unchecked, which is a serious problem. +@item + Test your code with valgrind and or Address Sanitizer to ensure it's free + of leaks, out of array accesses, etc. +@end enumerate + +@section Patch review process + +All patches posted to ffmpeg-devel will be reviewed, unless they contain a +clear note that the patch is not for the git master branch. +Reviews and comments will be posted as replies to the patch on the +mailing list. The patch submitter then has to take care of every comment, +that can be by resubmitting a changed patch or by discussion. Resubmitted +patches will themselves be reviewed like any other patch. If at some point +a patch passes review with no comments then it is approved, that can for +simple and small patches happen immediately while large patches will generally +have to be changed and reviewed many times before they are approved. +After a patch is approved it will be committed to the repository. + +We will review all submitted patches, but sometimes we are quite busy so +especially for large patches this can take several weeks. + +If you feel that the review process is too slow and you are willing to try to +take over maintainership of the area of code you change then just clone +git master and maintain the area of code there. We will merge each area from +where its best maintained. + +When resubmitting patches, please do not make any significant changes +not related to the comments received during review. Such patches will +be rejected. Instead, submit significant changes or new features as +separate patches. + +@anchor{Regression tests} +@section Regression tests + +Before submitting a patch (or committing to the repository), you should at least +test that you did not break anything. + +Running 'make fate' accomplishes this, please see @url{fate.html} for details. + +[Of course, some patches may change the results of the regression tests. In +this case, the reference results of the regression tests shall be modified +accordingly]. + +@subsection Adding files to the fate-suite dataset + +When there is no muxer or encoder available to generate test media for a +specific test then the media has to be inlcuded in the fate-suite. +First please make sure that the sample file is as small as possible to test the +respective decoder or demuxer sufficiently. Large files increase network +bandwidth and disk space requirements. +Once you have a working fate test and fate sample, provide in the commit +message or introductionary message for the patch series that you post to +the ffmpeg-devel mailing list, a direct link to download the sample media. + + +@anchor{Release process} +@section Release process + +FFmpeg maintains a set of @strong{release branches}, which are the +recommended deliverable for system integrators and distributors (such as +Linux distributions, etc.). At regular times, a @strong{release +manager} prepares, tests and publishes tarballs on the +@url{http://ffmpeg.org} website. + +There are two kinds of releases: + +@enumerate +@item + @strong{Major releases} always include the latest and greatest + features and functionality. +@item + @strong{Point releases} are cut from @strong{release} branches, + which are named @code{release/X}, with @code{X} being the release + version number. +@end enumerate + +Note that we promise to our users that shared libraries from any FFmpeg +release never break programs that have been @strong{compiled} against +previous versions of @strong{the same release series} in any case! + +However, from time to time, we do make API changes that require adaptations +in applications. Such changes are only allowed in (new) major releases and +require further steps such as bumping library version numbers and/or +adjustments to the symbol versioning file. Please discuss such changes +on the @strong{ffmpeg-devel} mailing list in time to allow forward planning. + +@anchor{Criteria for Point Releases} +@subsection Criteria for Point Releases + +Changes that match the following criteria are valid candidates for +inclusion into a point release: + +@enumerate +@item + Fixes a security issue, preferably identified by a @strong{CVE + number} issued by @url{http://cve.mitre.org/}. +@item + Fixes a documented bug in @url{https://ffmpeg.org/trac/ffmpeg}. +@item + Improves the included documentation. +@item + Retains both source code and binary compatibility with previous + point releases of the same release branch. +@end enumerate + +The order for checking the rules is (1 OR 2 OR 3) AND 4. + + +@subsection Release Checklist + +The release process involves the following steps: + +@enumerate +@item + Ensure that the @file{RELEASE} file contains the version number for + the upcoming release. +@item + Add the release at @url{https://ffmpeg.org/trac/ffmpeg/admin/ticket/versions}. +@item + Announce the intent to do a release to the mailing list. +@item + Make sure all relevant security fixes have been backported. See + @url{https://ffmpeg.org/security.html}. +@item + Ensure that the FATE regression suite still passes in the release + branch on at least @strong{i386} and @strong{amd64} + (cf. @ref{Regression tests}). +@item + Prepare the release tarballs in @code{bz2} and @code{gz} formats, and + supplementing files that contain @code{gpg} signatures +@item + Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and + push an annotated tag in the form @code{nX}, with @code{X} + containing the version number. +@item + Propose and send a patch to the @strong{ffmpeg-devel} mailing list + with a news entry for the website. +@item + Publish the news entry. +@item + Send announcement to the mailing list. +@end enumerate + +@bye diff --git a/ffmpeg1/doc/doxy-wrapper.sh b/ffmpeg1/doc/doxy-wrapper.sh new file mode 100755 index 0000000..6650e38 --- /dev/null +++ b/ffmpeg1/doc/doxy-wrapper.sh @@ -0,0 +1,14 @@ +#!/bin/sh + +SRC_PATH="${1}" +DOXYFILE="${2}" + +shift 2 + +doxygen - <<EOF +@INCLUDE = ${DOXYFILE} +INPUT = $@ +HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html +HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html +HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css +EOF diff --git a/ffmpeg1/doc/doxy/doxy_stylesheet.css b/ffmpeg1/doc/doxy/doxy_stylesheet.css new file mode 100644 index 0000000..63238a2 --- /dev/null +++ b/ffmpeg1/doc/doxy/doxy_stylesheet.css @@ -0,0 +1,2019 @@ +/*! + * Bootstrap v2.1.1 + * + * Copyright 2012 Twitter, Inc + * Licensed under the Apache License v2.0 + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Designed and built with all the love in the world @twitter by @mdo and @fat. + */ + +html { + font-size: 100%; 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+} +ul, +ol { + padding: 0; + margin: 0 0 10px 25px; +} +ul ul, +ul ol, +ol ol, +ol ul { + margin-bottom: 0; +} +li { + line-height: 20px; +} +ul.unstyled, +ol.unstyled { + margin-left: 0; + list-style: none; +} +dl { + margin-bottom: 20px; +} +dt, +dd { + line-height: 20px; +} +dt { + font-weight: bold; +} +dd { + margin-left: 10px; +} +blockquote { + padding: 0 0 0 15px; + margin: 0 0 20px; + border-left: 5px solid #eeeeee; +} +blockquote p { + margin-bottom: 0; + font-size: 16px; + font-weight: 300; + line-height: 25px; +} +blockquote:before, +blockquote:after { + content: ""; +} +.fragment, +code, +pre { + padding: 0 3px 2px; + font-family: monospace; + font-size: 12px; + color: #333333; + -webkit-border-radius: 3px; + -moz-border-radius: 3px; + border-radius: 3px; +} +.fragment, +code { + padding: 2px 4px; + color: #d14; + background-color: #f7f7f9; + border: 1px solid #e1e1e8; +} +pre { + display: block; + padding: 9.5px; + margin: 0 0 10px; + font-size: 13px; + line-height: 20px; + word-break: break-all; + word-wrap: break-word; + white-space: pre; + white-space: pre-wrap; + background-color: #f5f5f5; + border: 1px solid #ccc; + border: 1px solid rgba(0, 0, 0, 0.15); + -webkit-border-radius: 4px; + -moz-border-radius: 4px; + border-radius: 4px; +} +pre code { + padding: 0; + color: inherit; + background-color: transparent; + border: 0; +} +.label, +.badge { + font-size: 11.844px; + font-weight: bold; + line-height: 14px; + color: #ffffff; + vertical-align: baseline; + white-space: nowrap; + text-shadow: 0 -1px 0 rgba(0, 0, 0, 0.25); + background-color: #999999; +} +.label { + padding: 1px 4px 2px; + -webkit-border-radius: 3px; + -moz-border-radius: 3px; + border-radius: 3px; +} +.badge { + padding: 1px 9px 2px; + -webkit-border-radius: 9px; + -moz-border-radius: 9px; + border-radius: 9px; +} +a.label:hover, +a.badge:hover { + color: #ffffff; + text-decoration: none; + cursor: pointer; +} +.label-important, +.badge-important { + background-color: #b94a48; +} +.label-important[href], +.badge-important[href] { + background-color: #953b39; +} +.label-warning, +.badge-warning { + background-color: #f89406; +} +.label-warning[href], +.badge-warning[href] { + background-color: #c67605; +} +.label-success, +.badge-success { + background-color: #468847; +} +.label-success[href], +.badge-success[href] { + background-color: #356635; +} +.label-info, +.badge-info { + background-color: #3a87ad; +} +.label-info[href], +.badge-info[href] { + background-color: #2d6987; +} +.label-inverse, +.badge-inverse { + background-color: #333333; +} +.label-inverse[href], +.badge-inverse[href] { + background-color: #1a1a1a; +} +table { + max-width: 100%; + background-color: transparent; + border-collapse: collapse; + border-spacing: 0; +} + +table [class*=span], +.row-fluid table [class*=span] { + display: table-cell; + float: none; + margin-left: 0; +} +fieldset { + padding: 0; + margin: 0; + border: 0; +} +legend { + display: block; + width: 100%; + padding: 0; + margin-bottom: 20px; + font-size: 21px; + line-height: 40px; + color: #333333; + border: 0; + border-bottom: 1px solid #e5e5e5; +} +legend small { + font-size: 15px; + color: #999999; +} +label, +input, +button, +select, +textarea { + font-size: 14px; + font-weight: normal; + line-height: 20px; +} +input, +button, +select, +textarea { + font-family: sans-serif; +} +label { + display: block; + margin-bottom: 5px; +} + +.tablist { + margin-left: 0; + margin-bottom: 20px; + list-style: none; +} +.tablist > li > a { + display: block; +} +.tablist > li > a:hover { + text-decoration: none; + background-color: #eeeeee; +} +.tablist > .pull-right { + float: right; +} +.tablist-header { + display: block; + padding: 3px 15px; + font-size: 11px; + font-weight: bold; + line-height: 20px; + color: #999999; + text-shadow: 0 1px 0 rgba(255, 255, 255, 0.5); + text-transform: uppercase; +} +.tablist li + .tablist-header { + margin-top: 9px; +} +.tablist-list { + padding-left: 15px; + padding-right: 15px; + margin-bottom: 0; +} +.tablist-list > li > a, +.tablist-list .tablist-header { + margin-left: -15px; + margin-right: -15px; + text-shadow: 0 1px 0 rgba(255, 255, 255, 0.5); +} +.tablist-list > li > a { + padding: 3px 15px; +} +.tablist-list > .current > a, +.tablist-list > .current > a:hover { + color: #ffffff; + text-shadow: 0 -1px 0 rgba(0, 0, 0, 0.2); + background-color: #0088cc; +} +.tablist-list [class^="icon-"] { + margin-right: 2px; +} +.tablist-list .divider { + *width: 100%; + height: 1px; + margin: 9px 1px; + *margin: -5px 0 5px; + overflow: hidden; + background-color: #e5e5e5; + border-bottom: 1px solid #ffffff; +} +.tablist-tabs, +.tablist { + *zoom: 1; +} +.tablist-tabs:before, +.tablist:before, +.tablist-tabs:after, +.tablist:after { + display: table; + content: ""; + line-height: 0; +} +.tablist-tabs:after, +.tablist:after { + clear: both; +} +.tablist-tabs > li, +.tablist > li { + float: left; +} +.tablist-tabs > li > a, +.tablist > li > a { + padding-right: 12px; + padding-left: 12px; + margin-right: 2px; + line-height: 14px; +} +.tablist-tabs { + border-bottom: 1px solid #ddd; +} +.tablist-tabs > li { + margin-bottom: -1px; +} +.tablist-tabs > li > a { + padding-top: 8px; + padding-bottom: 8px; + line-height: 20px; + border: 1px solid transparent; + -webkit-border-radius: 4px 4px 0 0; + -moz-border-radius: 4px 4px 0 0; + border-radius: 4px 4px 0 0; +} +.tablist-tabs > li > a:hover { + border-color: #eeeeee #eeeeee #dddddd; +} +.tablist-tabs > .current > a, +.tablist-tabs > .current > a:hover { + color: #555555; + background-color: #ffffff; + border: 1px solid #ddd; + border-bottom-color: transparent; + cursor: default; +} +.tablist > li > a { + padding-top: 8px; + padding-bottom: 8px; + margin-top: 2px; + margin-bottom: 2px; + -webkit-border-radius: 5px; + -moz-border-radius: 5px; + border-radius: 5px; +} +.tablist > .current > a, +.tablist > .current > a:hover { + color: #ffffff; + background-color: #0088cc; +} +.tablist-stacked > li { + float: none; +} +.tablist-stacked > li > a { + margin-right: 0; +} +.tablist-tabs.tablist-stacked { + border-bottom: 0; +} +.tablist-tabs.tablist-stacked > li > a { + border: 1px solid #ddd; + -webkit-border-radius: 0; + -moz-border-radius: 0; + border-radius: 0; +} +.tablist-tabs.tablist-stacked > li:first-child > a { + -webkit-border-top-right-radius: 4px; + -moz-border-radius-topright: 4px; + border-top-right-radius: 4px; + -webkit-border-top-left-radius: 4px; + -moz-border-radius-topleft: 4px; + border-top-left-radius: 4px; +} +.tablist-tabs.tablist-stacked > li:last-child > a { + -webkit-border-bottom-right-radius: 4px; + -moz-border-radius-bottomright: 4px; + border-bottom-right-radius: 4px; + -webkit-border-bottom-left-radius: 4px; + -moz-border-radius-bottomleft: 4px; + border-bottom-left-radius: 4px; +} +.tablist-tabs.tablist-stacked > li > a:hover { + border-color: #ddd; + z-index: 2; +} +.tablist.tablist-stacked > li > a { + margin-bottom: 3px; +} +.tablist.tablist-stacked > li:last-child > a { + margin-bottom: 1px; +} +.tablist-tabs .dropdown-menu { + -webkit-border-radius: 0 0 6px 6px; + -moz-border-radius: 0 0 6px 6px; + border-radius: 0 0 6px 6px; +} +.tablist .dropdown-menu { + -webkit-border-radius: 6px; + -moz-border-radius: 6px; + border-radius: 6px; +} +.tablist .dropdown-toggle .caret { + border-top-color: #0088cc; + border-bottom-color: #0088cc; + margin-top: 6px; +} +.tablist .dropdown-toggle:hover .caret { + border-top-color: #005580; + border-bottom-color: #005580; +} +/* move down carets for tabs */ +.tablist-tabs .dropdown-toggle .caret { + margin-top: 8px; +} +.tablist .current .dropdown-toggle .caret { + border-top-color: #fff; + border-bottom-color: #fff; +} +.tablist-tabs .current .dropdown-toggle .caret { + border-top-color: #555555; + border-bottom-color: #555555; +} +.tablist > .dropdown.current > a:hover { + cursor: pointer; +} +.tablist-tabs .open .dropdown-toggle, +.tablist .open .dropdown-toggle, +.tablist > li.dropdown.open.current > a:hover { + color: #ffffff; + background-color: #999999; + border-color: #999999; +} +.tablist li.dropdown.open .caret, +.tablist li.dropdown.open.current .caret, +.tablist li.dropdown.open a:hover .caret { + border-top-color: #ffffff; + border-bottom-color: #ffffff; + opacity: 1; + filter: alpha(opacity=100); +} +.tabs-stacked .open > a:hover { + border-color: #999999; +} +.tab-content > .tab-pane, +.pill-content > .pill-pane { + display: none; +} +.tab-content > .current, +.pill-content > .current { + display: block; +} +.tabs-below > .tablist-tabs { + border-top: 1px solid #ddd; +} +.tabs-below > .tablist-tabs > li { + margin-top: -1px; + margin-bottom: 0; +} +.tabs-below > .tablist-tabs > li > a { + -webkit-border-radius: 0 0 4px 4px; + -moz-border-radius: 0 0 4px 4px; + border-radius: 0 0 4px 4px; +} +.tabs-below > .tablist-tabs > li > a:hover { + border-bottom-color: transparent; + border-top-color: #ddd; +} +.tabs-below > .tablist-tabs > .current > a, +.tabs-below > .tablist-tabs > .current > a:hover { + border-color: transparent #ddd #ddd #ddd; +} +.tabs-left > .tablist-tabs > li, +.tabs-right > .tablist-tabs > li { + float: none; +} +.tabs-left > .tablist-tabs > li > a, +.tabs-right > .tablist-tabs > li > a { + min-width: 74px; + margin-right: 0; + margin-bottom: 3px; +} +.tabs-left > .tablist-tabs { + float: left; + margin-right: 19px; + border-right: 1px solid #ddd; +} +.tabs-left > .tablist-tabs > li > a { + margin-right: -1px; + -webkit-border-radius: 4px 0 0 4px; + -moz-border-radius: 4px 0 0 4px; + border-radius: 4px 0 0 4px; +} +.tabs-left > .tablist-tabs > li > a:hover { + border-color: #eeeeee #dddddd #eeeeee #eeeeee; +} +.tabs-left > .tablist-tabs .current > a, +.tabs-left > .tablist-tabs .current > a:hover { + border-color: #ddd transparent #ddd #ddd; + *border-right-color: #ffffff; +} +.tabs-right > .tablist-tabs { + float: right; + margin-left: 19px; + border-left: 1px solid #ddd; +} +.tabs-right > .tablist-tabs > li > a { + margin-left: -1px; + -webkit-border-radius: 0 4px 4px 0; + -moz-border-radius: 0 4px 4px 0; + border-radius: 0 4px 4px 0; +} +.tabs-right > .tablist-tabs > li > a:hover { + border-color: #eeeeee #eeeeee #eeeeee #dddddd; +} +.tabs-right > .tablist-tabs .current > a, +.tabs-right > .tablist-tabs .current > a:hover { + border-color: #ddd #ddd #ddd transparent; + *border-left-color: #ffffff; +} +.tablist > .disabled > a { + color: #999999; +} +.tablist > .disabled > a:hover { + text-decoration: none; + background-color: transparent; + cursor: default; +} +.tablistbar { + overflow: visible; + margin-bottom: 20px; + color: #ffffff; + *position: relative; + *z-index: 2; +} +.tablistbar-inner { + min-height: 40px; + padding-left: 20px; + padding-right: 20px; + background-color: #034c03; + background-image: -moz-linear-gradient(top, #024002, #045f04); + background-image: -webkit-gradient(linear, 0 0, 0 100%, from(#024002), to(#045f04)); + background-image: -webkit-linear-gradient(top, #024002, #045f04); + background-image: -o-linear-gradient(top, #024002, #045f04); + background-image: linear-gradient(to bottom, #024002, #045f04); + background-repeat: repeat-x; + filter: progid:DXImageTransform.Microsoft.gradient(startColorstr='#ff024002', endColorstr='#ff045f04', GradientType=0); + border: 1px solid #022402; + -webkit-border-radius: 4px; + -moz-border-radius: 4px; + border-radius: 4px; + -webkit-box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + -moz-box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + *zoom: 1; +} +.tablistbar-inner:before, +.tablistbar-inner:after { + display: table; + content: ""; + line-height: 0; +} +.tablistbar-inner:after { + clear: both; +} +.tablistbar .container { + width: auto; +} +.tablist-collapse.collapse { + height: auto; +} +.tablistbar .brand { + float: left; + display: block; + padding: 10px 20px 10px; + margin-left: -20px; + font-size: 20px; + font-weight: 200; + color: #ffffff; + text-shadow: 0 1px 0 #024002; +} +.tablistbar .brand:hover { + text-decoration: none; +} +.tablistbar-text { + margin-bottom: 0; + line-height: 40px; +} +.tablistbar-link { + color: #ffffff; +} +.tablistbar-link:hover { + color: #333333; +} +.tablistbar .tablist { + position: relative; + left: 0; + display: block; + float: left; + margin: 0 10px 0 0; +} +.tablistbar .tablist.pull-right { + float: right; + margin-right: 0; +} +.tablistbar .tablist > li { + float: left; +} +.tablistbar .tablist > li > a { + float: none; + padding: 10px 15px 10px; + color: #ffffff; + text-decoration: none; + text-shadow: 0 1px 0 #024002; +} +.tablistbar .tablist .dropdown-toggle .caret { + margin-top: 8px; +} +.tablistbar .tablist > li > a:focus, +.tablistbar .tablist > li > a:hover { + background-color: transparent; + color: white; + text-decoration: none; +} +.tablistbar .tablist > .current > a, +.tablistbar .tablist > .current > a:hover, +.tablistbar .tablist > .current > a:focus { + color: #555555; + text-decoration: none; + background-color: #034703; + -webkit-box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); + -moz-box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); + box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); +} +.tablistbar .btn-navbar { + display: none; + float: right; + padding: 7px 10px; + margin-left: 5px; + margin-right: 5px; + color: #ffffff; + text-shadow: 0 -1px 0 rgba(0, 0, 0, 0.25); + background-color: #023402; + background-image: -moz-linear-gradient(top, #012701, #034703); + background-image: -webkit-gradient(linear, 0 0, 0 100%, from(#012701), to(#034703)); + background-image: -webkit-linear-gradient(top, #012701, #034703); + background-image: -o-linear-gradient(top, #012701, #034703); + background-image: linear-gradient(to bottom, #012701, #034703); + background-repeat: repeat-x; + filter: progid:DXImageTransform.Microsoft.gradient(startColorstr='#ff012701', endColorstr='#ff034703', GradientType=0); + border-color: #034703 #034703 #000000; + border-color: rgba(0, 0, 0, 0.1) rgba(0, 0, 0, 0.1) rgba(0, 0, 0, 0.25); + *background-color: #034703; + /* Darken IE7 buttons by default so they stand out more given they won't have borders */ + + filter: progid:DXImageTransform.Microsoft.gradient(enabled = false); + -webkit-box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); + -moz-box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); + box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); +} +.tablistbar .tablist > li > .dropdown-menu:before { + content: ''; + display: inline-block; + border-left: 7px solid transparent; + border-right: 7px solid transparent; + border-bottom: 7px solid #ccc; + border-bottom-color: rgba(0, 0, 0, 0.2); + position: absolute; + top: -7px; + left: 9px; +} +.tablistbar .tablist > li > .dropdown-menu:after { + content: ''; + display: inline-block; + border-left: 6px solid transparent; + border-right: 6px solid transparent; + border-bottom: 6px solid #ffffff; + position: absolute; + top: -6px; + left: 10px; +} +.tablistbar .tablist li.dropdown.open > .dropdown-toggle, +.tablistbar .tablist li.dropdown.current > .dropdown-toggle, +.tablistbar .tablist li.dropdown.open.current > .dropdown-toggle { + background-color: #034703; + color: #555555; +} +.tablistbar .tablist li.dropdown > .dropdown-toggle .caret { + border-top-color: #ffffff; + border-bottom-color: #ffffff; +} +.tablistbar .tablist li.dropdown.open > .dropdown-toggle .caret, +.tablistbar .tablist li.dropdown.current > .dropdown-toggle .caret, +.tablistbar .tablist li.dropdown.open.current > .dropdown-toggle .caret { + border-top-color: #555555; + border-bottom-color: #555555; +} +.tablistbar .pull-right > li > .dropdown-menu, +.tablistbar .tablist > li > .dropdown-menu.pull-right { + left: auto; + right: 0; +} +.tablistbar .pull-right > li > .dropdown-menu:before, +.tablistbar .tablist > li > .dropdown-menu.pull-right:before { + left: auto; + right: 12px; +} +.tablistbar .pull-right > li > .dropdown-menu:after, +.tablistbar .tablist > li > .dropdown-menu.pull-right:after { + left: auto; + right: 13px; +} +.tablistbar .pull-right > li > .dropdown-menu .dropdown-menu, +.tablistbar .tablist > li > .dropdown-menu.pull-right .dropdown-menu { + left: auto; + right: 100%; + margin-left: 0; + margin-right: -1px; + -webkit-border-radius: 6px 0 6px 6px; + -moz-border-radius: 6px 0 6px 6px; + border-radius: 6px 0 6px 6px; +} +.breadcrumb { + padding: 8px 15px; + margin: 0 0 20px; + list-style: none; + background-color: #f5f5f5; + -webkit-border-radius: 4px; + -moz-border-radius: 4px; + border-radius: 4px; +} +.breadcrumb li { + display: inline-block; + *display: inline; + /* IE7 inline-block hack */ + + *zoom: 1; + text-shadow: 0 1px 0 #ffffff; +} +.breadcrumb .divider { + padding: 0 5px; + color: #ccc; +} +.breadcrumb .current { + color: #999999; +} +.pagination-right { + text-align: right; +} +.fade { + opacity: 0; + -webkit-transition: opacity 0.15s linear; + -moz-transition: opacity 0.15s linear; + -o-transition: opacity 0.15s linear; + transition: opacity 0.15s linear; +} +.fade.in { + opacity: 1; +} +.collapse { + position: relative; + height: 0; + overflow: hidden; + -webkit-transition: height 0.35s ease; + -moz-transition: height 0.35s ease; + -o-transition: height 0.35s ease; + transition: height 0.35s ease; +} +.collapse.in { + height: auto; +} +.hidden { + display: none; + visibility: hidden; +} +.visible-phone { + display: none !important; +} +.visible-tablet { + display: none !important; +} +.hidden-desktop { + display: none !important; +} +.visible-desktop { + display: inherit !important; +} +@media (min-width: 768px) and (max-width: 979px) { + .hidden-desktop { + display: inherit !important; + } + .visible-desktop { + display: none !important ; + } + .visible-tablet { + display: inherit !important; + } + .hidden-tablet { + display: none !important; + } +} +@media (max-width: 767px) { + .hidden-desktop { + display: inherit !important; + } + .visible-desktop { + display: none !important; + } + .visible-phone { + display: inherit !important; + } + .hidden-phone { + display: none !important; + } +} +@media (max-width: 767px) { + body { + padding-left: 20px; + padding-right: 20px; + } + .container { + width: auto; + } + .row, + .thumbnails { + margin-left: 0; + } +} +@media (max-width: 480px) { + .tablist-collapse { + -webkit-transform: translate3d(0, 0, 0); + } + .page-header h1 small { + display: block; + line-height: 20px; + } +} +@media (min-width: 768px) and (max-width: 979px) { + .row { + margin-left: -20px; + *zoom: 1; + } + .row:before, + .row:after { + display: table; + content: ""; + line-height: 0; + } + .row:after { + clear: both; + } + [class*="span"] { + float: left; + min-height: 1px; + margin-left: 20px; + } + .container { + width: 724px; + } +} +@media (min-width: 1200px) { + .row { + margin-left: -30px; + *zoom: 1; + } + .row:before, + .row:after { + display: table; + content: ""; + line-height: 0; + } + .row:after { + clear: both; + } + [class*="span"] { + float: left; + min-height: 1px; + margin-left: 30px; + } + .container { + width: 1070px; + } +} +@media (max-width: 979px) { + body { + padding-top: 0; + } +} +@media (min-width: 980px) { + .tablist-collapse.collapse { + height: auto !important; + overflow: visible !important; + } +} +.tablistbar .brand { + padding: 5px; + margin-left: 0; +} +.tablistbar .brand img { + width: 30px; + vertical-align: middle; +} + +h1 small { + font-size: 18px; +} + +h1 small, +h2 small, +h3 small, +h4 small, +h5 small, +h6 small, +.page-header small { + line-height: 0.8; + font-weight: normal; + color: #999999; + display:block; + vertical-align: middle; +} + +.page-header h1, h1:first-child { + font-size: 40px; + padding-bottom: 5px; +} + +.page-header h1 { + border-bottom: 1px solid #999999; + padding-bottom: 9px; +} + +.page-header img { + height: 80px; + padding-bottom: 5px; +} + +.page-header small { + line-height: 1.1; + font-size: 18px; +} + +h2, +h3, +h4, +div.ah, +.title { + border-color: #D6E9C6; + color: #468847; + border-style: solid; + border-width: 0 0 1px; + padding-left: 0.5em; +} + + +.google { + color: white; +} + +.breadcrumb { + font-size: 11px; + padding-top: 2px; + padding-bottom: 2px; +} + +h1 a, +h2 a, +h3 a, +h4 a { + color: inherit; +} + +.tablistbar-inner a { + font-weight: bold; +} + +.list-2panes:before, +.list-2panes:after { + display: table; + content: ""; + line-height: 0; +} + +.list-2panes:after { + clear:both; +} + +.list-2panes li { + width: 470px; + width: 470px; + float: left; + margin-left: 30px; + min-height: 1px; +} +/* The standard CSS for doxygen */ + +/* @group Heading Levels */ + + +dt { + font-weight: bold; +} + +div.multicol { + -moz-column-gap: 1em; + -webkit-column-gap: 1em; + -moz-column-count: 3; + -webkit-column-count: 3; +} + +p.startli, p.startdd, p.starttd { + margin-top: 2px; +} + +p.endli { + margin-bottom: 0px; +} + +p.enddd { + margin-bottom: 4px; +} + +p.endtd { + margin-bottom: 2px; +} + +/* @end */ + +caption { + font-weight: bold; +} + +span.legend { + font-size: 70%; + text-align: center; +} + +h3.version { + font-size: 90%; + text-align: center; +} + +div.qindex, div.tablisttab{ + background-color: #EBF6EB; + border: 1px solid #A3D7A3; + text-align: center; +} + +div.qindex, div.tablistpath { + width: 100%; + line-height: 140%; +} + +div.tablisttab { + margin-right: 15px; +} + +/* @group Link Styling */ + +a { + color: #3D8C3D; + font-weight: normal; + text-decoration: none; +} + +.contents a:visited { + color: #46A246; +} + +a:hover { + text-decoration: underline; +} + +a.qindex { + font-weight: bold; +} + +a.qindexHL { + font-weight: bold; + background-color: #9CD49C; + color: #ffffff; + border: 1px double #86CA86; +} + +.contents a.qindexHL:visited { + color: #ffffff; +} + +a.el { + font-weight: bold; +} + +a.elRef { +} + +a.code { + color: #4665A2; +} + +a.codeRef { + color: #4665A2; +} + +/* @end */ + +dl.el { + margin-left: -1cm; +} + +.fragment { + font-family: monospace, fixed; + font-size: 105%; +} + +pre.fragment { + border: 1px solid #C4E5C4; + background-color: #FBFDFB; + padding: 4px 6px; + margin: 4px 8px 4px 2px; + overflow: auto; + word-wrap: break-word; + font-size: 9pt; + line-height: 125%; +} + +div.groupHeader { + margin-left: 16px; + margin-top: 12px; + font-weight: bold; +} + +div.groupText { + margin-left: 16px; + font-style: italic; +} + +div.contents { + margin-top: 10px; + margin-left: 8px; + margin-right: 8px; +} + +td.indexkey { + white-space: nowrap; + vertical-align: top; +} + + +tr.memlist { + background-color: #EEF7EE; +} + +p.formulaDsp { + text-align: center; +} + +img.formulaDsp { + +} + +img.formulaInl { + vertical-align: middle; +} + +div.center { + text-align: center; + margin-top: 0px; + margin-bottom: 0px; + padding: 0px; +} + +div.center img { + border: 0px; +} + +#footer { + margin: -10px 1em 0; + padding-top: 20px; + text-align: center; + font-size: small; +} + +address.footer { + background-color: #ffffff; + text-align: center; +} + +img.footer { + border: 0px; + vertical-align: middle; +} + +/* @group Code Colorization */ + +span.keyword { + color: #008000 +} + +span.keywordtype { + color: #604020 +} + +span.keywordflow { + color: #e08000 +} + +span.comment { + color: #800000 +} + +span.preprocessor { + color: #806020 +} + +span.stringliteral { + color: #002080 +} + +span.charliteral { + color: #008080 +} + +span.vhdldigit { + color: #ff00ff +} + +span.vhdlchar { + color: #000000 +} + +span.vhdlkeyword { + color: #700070 +} + +span.vhdllogic { + color: #ff0000 +} + +/* @end */ + +/* +.search { + color: #003399; + font-weight: bold; +} + +form.search { + margin-bottom: 0px; + margin-top: 0px; +} + +input.search { + font-size: 75%; + color: #000080; + font-weight: normal; + background-color: #e8eef2; +} +*/ + +td.tiny { + font-size: 75%; +} + +.dirtab { + padding: 4px; + border-collapse: collapse; + border: 1px solid #A3D7A3; +} + +th.dirtab { + background: #EBF6EB; + font-weight: bold; +} + +hr { + height: 0px; + border: none; + border-top: 1px solid #4AAA4A; +} + +hr.footer { + height: 1px; +} + +/* @group Member Descriptions */ + +table.memberdecls { + border-spacing: 0px; + padding: 0px; +} + +.mdescLeft, .mdescRight, +.memItemLeft, .memItemRight, +.memTemplItemLeft, .memTemplItemRight, .memTemplParams { + background-color: #F9FCF9; + border: none; + margin: 4px; + padding: 1px 0 0 8px; +} + +.mdescLeft, .mdescRight { + padding: 0px 8px 4px 8px; + color: #555; +} + +.memItemLeft, .memItemRight, .memTemplParams { + border-top: 1px solid #C4E5C4; +} + +.memItemLeft, .memTemplItemLeft { + white-space: nowrap; +} + +.memItemRight { + width: 100%; +} + +.memTemplParams { + color: #46A246; + white-space: nowrap; +} + +/* @end */ + +/* @group Member Details */ + +/* Styles for detailed member documentation */ + +.memtemplate { + font-size: 80%; + color: #46A246; + font-weight: normal; + margin-left: 9px; +} + +.memnav { + background-color: #EBF6EB; + border: 1px solid #A3D7A3; + text-align: center; + margin: 2px; + margin-right: 15px; + padding: 2px; +} + +.mempage { + width: 100%; +} + +.memitem { + padding: 0; + margin-bottom: 10px; + margin-right: 5px; +} + +.memname { + white-space: nowrap; + font-weight: bold; + margin-left: 6px; +} + +.memproto, dl.reflist dt { + border-top: 1px solid #A8D9A8; + border-left: 1px solid #A8D9A8; + border-right: 1px solid #A8D9A8; + padding: 6px 0px 6px 0px; + color: #255525; + font-weight: bold; + text-shadow: 0px 1px 1px rgba(255, 255, 255, 0.9); + /* opera specific markup */ + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + border-top-right-radius: 8px; + border-top-left-radius: 8px; + /* firefox specific markup */ + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + -moz-border-radius-topright: 8px; + -moz-border-radius-topleft: 8px; + /* webkit specific markup */ + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + -webkit-border-top-right-radius: 8px; + -webkit-border-top-left-radius: 8px; + background-image:url('nav_f.png'); + background-repeat:repeat-x; + background-color: #E2F2E2; + +} + +.memdoc, dl.reflist dd { + border-bottom: 1px solid #A8D9A8; + border-left: 1px solid #A8D9A8; + border-right: 1px solid #A8D9A8; + padding: 2px 5px; + background-color: #FBFDFB; + border-top-width: 0; + /* opera specific markup */ + border-bottom-left-radius: 8px; + border-bottom-right-radius: 8px; + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + /* firefox specific markup */ + -moz-border-radius-bottomleft: 8px; + -moz-border-radius-bottomright: 8px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + background-image: -moz-linear-gradient(center top, #FFFFFF 0%, #FFFFFF 60%, #F7FBF7 95%, #EEF7EE); + /* webkit specific markup */ + -webkit-border-bottom-left-radius: 8px; + -webkit-border-bottom-right-radius: 8px; + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + background-image: -webkit-gradient(linear,center top,center bottom,from(#FFFFFF), color-stop(0.6,#FFFFFF), color-stop(0.60,#FFFFFF), color-stop(0.95,#F7FBF7), to(#EEF7EE)); +} + +dl.reflist dt { + padding: 5px; +} + +dl.reflist dd { + margin: 0px 0px 10px 0px; + padding: 5px; +} + +.paramkey { + text-align: right; +} + +.paramtype { + white-space: nowrap; +} + +.paramname { + color: #602020; + white-space: nowrap; +} +.paramname em { + font-style: normal; +} + +.params, .retval, .exception, .tparams { + border-spacing: 6px 2px; +} + +.params .paramname, .retval .paramname { + font-weight: bold; + vertical-align: top; +} + +.params .paramtype { + font-style: italic; + vertical-align: top; +} + +.params .paramdir { + font-family: "courier new",courier,monospace; + vertical-align: top; +} + + + + +/* @end */ + +/* @group Directory (tree) */ + +/* for the tree view */ + +.ftvtree { + font-family: sans-serif; + margin: 0px; +} + +/* these are for tree view when used as main index */ + +.directory { + font-size: 9pt; + font-weight: bold; + margin: 5px; +} + +.directory h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +/* +The following two styles can be used to replace the root node title +with an image of your choice. Simply uncomment the next two styles, +specify the name of your image and be sure to set 'height' to the +proper pixel height of your image. +*/ + +/* +.directory h3.swap { + height: 61px; + background-repeat: no-repeat; + background-image: url("yourimage.gif"); +} +.directory h3.swap span { + display: none; +} +*/ + +.directory > h3 { + margin-top: 0; +} + +.directory p { + margin: 0px; + white-space: nowrap; +} + +.directory div { + display: none; + margin: 0px; +} + +.directory img { + vertical-align: -30%; +} + +/* these are for tree view when not used as main index */ + +.directory-alt { + font-size: 100%; + font-weight: bold; +} + +.directory-alt h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +.directory-alt > h3 { + margin-top: 0; +} + +.directory-alt p { + margin: 0px; + white-space: nowrap; +} + +.directory-alt div { + display: none; + margin: 0px; +} + +.directory-alt img { + vertical-align: -30%; +} + +/* @end */ + +div.dynheader { + margin-top: 8px; +} + +address { + font-style: normal; + color: #2A612A; +} + +table.doxtable { + border-collapse:collapse; +} + +table.doxtable td, table.doxtable th { + border: 1px solid #2D682D; + padding: 3px 7px 2px; +} + +table.doxtable th { + background-color: #377F37; + color: #FFFFFF; + font-size: 110%; + padding-bottom: 4px; + padding-top: 5px; + text-align:left; +} + +table.fieldtable { + width: 100%; + margin-bottom: 10px; + border: 1px solid #A8D9A8; + border-spacing: 0px; + -moz-border-radius: 4px; + -webkit-border-radius: 4px; + border-radius: 4px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 2px 2px 2px; + -webkit-box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); + box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); +} + +.fieldtable td, .fieldtable th { + padding: 3px 7px 2px; +} + +.fieldtable td.fieldtype, .fieldtable td.fieldname { + white-space: nowrap; + border-right: 1px solid #A8D9A8; + border-bottom: 1px solid #A8D9A8; + vertical-align: top; +} + +.fieldtable td.fielddoc { + border-bottom: 1px solid #A8D9A8; + width: 100%; +} + +.fieldtable tr:last-child td { + border-bottom: none; +} + +.fieldtable th { + background-image:url('nav_f.png'); + background-repeat:repeat-x; + background-color: #E2F2E2; + font-size: 90%; + color: #255525; + padding-bottom: 4px; + padding-top: 5px; + text-align:left; + -moz-border-radius-topleft: 4px; + -moz-border-radius-topright: 4px; + -webkit-border-top-left-radius: 4px; + -webkit-border-top-right-radius: 4px; + border-top-left-radius: 4px; + border-top-right-radius: 4px; + border-bottom: 1px solid #A8D9A8; +} + + +.tabsearch { + top: 0px; + left: 10px; + height: 36px; + background-image: url('tab_b.png'); + z-index: 101; + overflow: hidden; + font-size: 13px; +} + +.tablistpath ul +{ + font-size: 11px; + background-image:url('tab_b.png'); + background-repeat:repeat-x; + height:30px; + line-height:30px; + color:#8ACC8A; + border:solid 1px #C2E4C2; + overflow:hidden; + margin:0px; + padding:0px; +} + +.tablistpath li +{ + list-style-type:none; + float:left; + padding-left:10px; + padding-right:15px; + background-image:url('bc_s.png'); + background-repeat:no-repeat; + background-position:right; + color:#367C36; +} + +.tablistpath li.tablistelem a +{ + height:32px; + display:block; + text-decoration: none; + outline: none; +} + +.tablistpath li.tablistelem a:hover +{ + color:#68BD68; +} + +.tablistpath li.footer +{ + list-style-type:none; + float:right; + padding-left:10px; + padding-right:15px; + background-image:none; + background-repeat:no-repeat; + background-position:right; + color:#367C36; + font-size: 8pt; +} + + +div.summary +{ + margin-top: 12px; + text-align: center; +} + +div.summary a +{ + white-space: nowrap; +} + +div.ingroups +{ + margin-left: 5px; + font-size: 8pt; + padding-left: 5px; + width: 50%; + text-align: left; +} + +div.ingroups a +{ + white-space: nowrap; +} + +div.headertitle +{ + padding: 5px 5px 5px 7px; +} + +dl +{ + padding: 0 0 0 10px; +} + +dl.note, dl.warning, dl.attention, dl.pre, dl.post, dl.invariant, dl.deprecated, dl.todo, dl.test, dl.bug +{ + border-left:4px solid; + padding: 0 0 0 6px; +} + +dl.note +{ + border-color: #D0C000; +} + +dl.warning, dl.attention +{ + border-color: #FF0000; +} + +dl.pre, dl.post, dl.invariant +{ + border-color: #00D000; +} + +dl.deprecated +{ + border-color: #505050; +} + +dl.todo +{ + border-color: #00C0E0; +} + +dl.test +{ + border-color: #3030E0; +} + +dl.bug +{ + border-color: #C08050; +} + +#projectlogo +{ + text-align: center; + vertical-align: bottom; + border-collapse: separate; +} + +#projectlogo img +{ + border: 0px none; +} + +#projectname +{ + font: 300% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 2px 0px; +} + +#projectbrief +{ + font: 120% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#projectnumber +{ + font: 50% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#titlearea +{ + padding: 0px; + margin: 0px; + width: 100%; + border-bottom: 1px solid #53B453; +} + +.image +{ + text-align: center; +} + +.dotgraph +{ + text-align: center; +} + +.mscgraph +{ + text-align: center; +} + +.caption +{ + font-weight: bold; +} + +div.zoom +{ + border: 1px solid #90CE90; +} + +dl.citelist { + margin-bottom:50px; +} + +dl.citelist dt { + color:#337533; + float:left; + font-weight:bold; + margin-right:10px; + padding:5px; +} + +dl.citelist dd { + margin:2px 0; + padding:5px 0; +} + +@media print +{ + #top { display: none; } + #side-nav { display: none; } + #nav-path { display: none; } + body { overflow:visible; } + h1, h2, h3, h4, h5, h6 { page-break-after: avoid; } + .summary { display: none; } + .memitem { page-break-inside: avoid; } + #doc-content + { + margin-left:0 !important; + height:auto !important; + width:auto !important; + overflow:inherit; + display:inline; + } + pre.fragment + { + overflow: visible; + text-wrap: unrestricted; + white-space: -moz-pre-wrap; /* Moz */ + white-space: -pre-wrap; /* Opera 4-6 */ + white-space: -o-pre-wrap; /* Opera 7 */ + white-space: pre-wrap; /* CSS3 */ + word-wrap: break-word; /* IE 5.5+ */ + } +} + +#proj_desc { + font-size: 1.2em; +} diff --git a/ffmpeg1/doc/doxy/footer.html b/ffmpeg1/doc/doxy/footer.html new file mode 100644 index 0000000..101e6fe --- /dev/null +++ b/ffmpeg1/doc/doxy/footer.html @@ -0,0 +1,9 @@ + + <footer class="footer pagination-right"> + <span class="label label-info"> + Generated on $datetime for $projectname by <a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion + </span> + </footer> +</div> +</body> +</html> diff --git a/ffmpeg1/doc/doxy/header.html b/ffmpeg1/doc/doxy/header.html new file mode 100644 index 0000000..312990c --- /dev/null +++ b/ffmpeg1/doc/doxy/header.html @@ -0,0 +1,16 @@ +<!DOCTYPE html> +<html> +<head> +<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/> +<meta http-equiv="X-UA-Compatible" content="IE=9"/> +<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME--> +<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME--> +<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" /> +<!--Header replace --> + +</head> + +<div class="container"> + +<!--Header replace --> +<div class="menu"> diff --git a/ffmpeg1/doc/encoders.texi b/ffmpeg1/doc/encoders.texi new file mode 100644 index 0000000..07343eb --- /dev/null +++ b/ffmpeg1/doc/encoders.texi @@ -0,0 +1,780 @@ +@chapter Encoders +@c man begin ENCODERS + +Encoders are configured elements in FFmpeg which allow the encoding of +multimedia streams. + +When you configure your FFmpeg build, all the supported native encoders +are enabled by default. Encoders requiring an external library must be enabled +manually via the corresponding @code{--enable-lib} option. You can list all +available encoders using the configure option @code{--list-encoders}. + +You can disable all the encoders with the configure option +@code{--disable-encoders} and selectively enable / disable single encoders +with the options @code{--enable-encoder=@var{ENCODER}} / +@code{--disable-encoder=@var{ENCODER}}. + +The option @code{-codecs} of the ff* tools will display the list of +enabled encoders. + +@c man end ENCODERS + +@chapter Audio Encoders +@c man begin AUDIO ENCODERS + +A description of some of the currently available audio encoders +follows. + +@section ac3 and ac3_fixed + +AC-3 audio encoders. + +These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as +the undocumented RealAudio 3 (a.k.a. dnet). + +The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed} +encoder only uses fixed-point integer math. This does not mean that one is +always faster, just that one or the other may be better suited to a +particular system. The floating-point encoder will generally produce better +quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the +default codec for any of the output formats, so it must be specified explicitly +using the option @code{-acodec ac3_fixed} in order to use it. + +@subsection AC-3 Metadata + +The AC-3 metadata options are used to set parameters that describe the audio, +but in most cases do not affect the audio encoding itself. Some of the options +do directly affect or influence the decoding and playback of the resulting +bitstream, while others are just for informational purposes. A few of the +options will add bits to the output stream that could otherwise be used for +audio data, and will thus affect the quality of the output. Those will be +indicated accordingly with a note in the option list below. + +These parameters are described in detail in several publicly-available +documents. +@itemize +@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard} +@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard} +@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide} +@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines} +@end itemize + +@subsubsection Metadata Control Options + +@table @option + +@item -per_frame_metadata @var{boolean} +Allow Per-Frame Metadata. Specifies if the encoder should check for changing +metadata for each frame. +@table @option +@item 0 +The metadata values set at initialization will be used for every frame in the +stream. (default) +@item 1 +Metadata values can be changed before encoding each frame. +@end table + +@end table + +@subsubsection Downmix Levels + +@table @option + +@item -center_mixlev @var{level} +Center Mix Level. The amount of gain the decoder should apply to the center +channel when downmixing to stereo. This field will only be written to the +bitstream if a center channel is present. The value is specified as a scale +factor. There are 3 valid values: +@table @option +@item 0.707 +Apply -3dB gain +@item 0.595 +Apply -4.5dB gain (default) +@item 0.500 +Apply -6dB gain +@end table + +@item -surround_mixlev @var{level} +Surround Mix Level. The amount of gain the decoder should apply to the surround +channel(s) when downmixing to stereo. This field will only be written to the +bitstream if one or more surround channels are present. The value is specified +as a scale factor. There are 3 valid values: +@table @option +@item 0.707 +Apply -3dB gain +@item 0.500 +Apply -6dB gain (default) +@item 0.000 +Silence Surround Channel(s) +@end table + +@end table + +@subsubsection Audio Production Information +Audio Production Information is optional information describing the mixing +environment. Either none or both of the fields are written to the bitstream. + +@table @option + +@item -mixing_level @var{number} +Mixing Level. Specifies peak sound pressure level (SPL) in the production +environment when the mix was mastered. Valid values are 80 to 111, or -1 for +unknown or not indicated. The default value is -1, but that value cannot be +used if the Audio Production Information is written to the bitstream. Therefore, +if the @code{room_type} option is not the default value, the @code{mixing_level} +option must not be -1. + +@item -room_type @var{type} +Room Type. Describes the equalization used during the final mixing session at +the studio or on the dubbing stage. A large room is a dubbing stage with the +industry standard X-curve equalization; a small room has flat equalization. +This field will not be written to the bitstream if both the @code{mixing_level} +option and the @code{room_type} option have the default values. +@table @option +@item 0 +@itemx notindicated +Not Indicated (default) +@item 1 +@itemx large +Large Room +@item 2 +@itemx small +Small Room +@end table + +@end table + +@subsubsection Other Metadata Options + +@table @option + +@item -copyright @var{boolean} +Copyright Indicator. Specifies whether a copyright exists for this audio. +@table @option +@item 0 +@itemx off +No Copyright Exists (default) +@item 1 +@itemx on +Copyright Exists +@end table + +@item -dialnorm @var{value} +Dialogue Normalization. Indicates how far the average dialogue level of the +program is below digital 100% full scale (0 dBFS). This parameter determines a +level shift during audio reproduction that sets the average volume of the +dialogue to a preset level. The goal is to match volume level between program +sources. A value of -31dB will result in no volume level change, relative to +the source volume, during audio reproduction. Valid values are whole numbers in +the range -31 to -1, with -31 being the default. + +@item -dsur_mode @var{mode} +Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround +(Pro Logic). This field will only be written to the bitstream if the audio +stream is stereo. Using this option does @b{NOT} mean the encoder will actually +apply Dolby Surround processing. +@table @option +@item 0 +@itemx notindicated +Not Indicated (default) +@item 1 +@itemx off +Not Dolby Surround Encoded +@item 2 +@itemx on +Dolby Surround Encoded +@end table + +@item -original @var{boolean} +Original Bit Stream Indicator. Specifies whether this audio is from the +original source and not a copy. +@table @option +@item 0 +@itemx off +Not Original Source +@item 1 +@itemx on +Original Source (default) +@end table + +@end table + +@subsection Extended Bitstream Information +The extended bitstream options are part of the Alternate Bit Stream Syntax as +specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts. +If any one parameter in a group is specified, all values in that group will be +written to the bitstream. Default values are used for those that are written +but have not been specified. If the mixing levels are written, the decoder +will use these values instead of the ones specified in the @code{center_mixlev} +and @code{surround_mixlev} options if it supports the Alternate Bit Stream +Syntax. + +@subsubsection Extended Bitstream Information - Part 1 + +@table @option + +@item -dmix_mode @var{mode} +Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt +(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode. +@table @option +@item 0 +@itemx notindicated +Not Indicated (default) +@item 1 +@itemx ltrt +Lt/Rt Downmix Preferred +@item 2 +@itemx loro +Lo/Ro Downmix Preferred +@end table + +@item -ltrt_cmixlev @var{level} +Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the +center channel when downmixing to stereo in Lt/Rt mode. +@table @option +@item 1.414 +Apply +3dB gain +@item 1.189 +Apply +1.5dB gain +@item 1.000 +Apply 0dB gain +@item 0.841 +Apply -1.5dB gain +@item 0.707 +Apply -3.0dB gain +@item 0.595 +Apply -4.5dB gain (default) +@item 0.500 +Apply -6.0dB gain +@item 0.000 +Silence Center Channel +@end table + +@item -ltrt_surmixlev @var{level} +Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the +surround channel(s) when downmixing to stereo in Lt/Rt mode. +@table @option +@item 0.841 +Apply -1.5dB gain +@item 0.707 +Apply -3.0dB gain +@item 0.595 +Apply -4.5dB gain +@item 0.500 +Apply -6.0dB gain (default) +@item 0.000 +Silence Surround Channel(s) +@end table + +@item -loro_cmixlev @var{level} +Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the +center channel when downmixing to stereo in Lo/Ro mode. +@table @option +@item 1.414 +Apply +3dB gain +@item 1.189 +Apply +1.5dB gain +@item 1.000 +Apply 0dB gain +@item 0.841 +Apply -1.5dB gain +@item 0.707 +Apply -3.0dB gain +@item 0.595 +Apply -4.5dB gain (default) +@item 0.500 +Apply -6.0dB gain +@item 0.000 +Silence Center Channel +@end table + +@item -loro_surmixlev @var{level} +Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the +surround channel(s) when downmixing to stereo in Lo/Ro mode. +@table @option +@item 0.841 +Apply -1.5dB gain +@item 0.707 +Apply -3.0dB gain +@item 0.595 +Apply -4.5dB gain +@item 0.500 +Apply -6.0dB gain (default) +@item 0.000 +Silence Surround Channel(s) +@end table + +@end table + +@subsubsection Extended Bitstream Information - Part 2 + +@table @option + +@item -dsurex_mode @var{mode} +Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX +(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually +apply Dolby Surround EX processing. +@table @option +@item 0 +@itemx notindicated +Not Indicated (default) +@item 1 +@itemx on +Dolby Surround EX Off +@item 2 +@itemx off +Dolby Surround EX On +@end table + +@item -dheadphone_mode @var{mode} +Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone +encoding (multi-channel matrixed to 2.0 for use with headphones). Using this +option does @b{NOT} mean the encoder will actually apply Dolby Headphone +processing. +@table @option +@item 0 +@itemx notindicated +Not Indicated (default) +@item 1 +@itemx on +Dolby Headphone Off +@item 2 +@itemx off +Dolby Headphone On +@end table + +@item -ad_conv_type @var{type} +A/D Converter Type. Indicates whether the audio has passed through HDCD A/D +conversion. +@table @option +@item 0 +@itemx standard +Standard A/D Converter (default) +@item 1 +@itemx hdcd +HDCD A/D Converter +@end table + +@end table + +@subsection Other AC-3 Encoding Options + +@table @option + +@item -stereo_rematrixing @var{boolean} +Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This +is an optional AC-3 feature that increases quality by selectively encoding +the left/right channels as mid/side. This option is enabled by default, and it +is highly recommended that it be left as enabled except for testing purposes. + +@end table + +@subsection Floating-Point-Only AC-3 Encoding Options + +These options are only valid for the floating-point encoder and do not exist +for the fixed-point encoder due to the corresponding features not being +implemented in fixed-point. + +@table @option + +@item -channel_coupling @var{boolean} +Enables/Disables use of channel coupling, which is an optional AC-3 feature +that increases quality by combining high frequency information from multiple +channels into a single channel. The per-channel high frequency information is +sent with less accuracy in both the frequency and time domains. This allows +more bits to be used for lower frequencies while preserving enough information +to reconstruct the high frequencies. This option is enabled by default for the +floating-point encoder and should generally be left as enabled except for +testing purposes or to increase encoding speed. +@table @option +@item -1 +@itemx auto +Selected by Encoder (default) +@item 0 +@itemx off +Disable Channel Coupling +@item 1 +@itemx on +Enable Channel Coupling +@end table + +@item -cpl_start_band @var{number} +Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a +value higher than the bandwidth is used, it will be reduced to 1 less than the +coupling end band. If @var{auto} is used, the start band will be determined by +the encoder based on the bit rate, sample rate, and channel layout. This option +has no effect if channel coupling is disabled. +@table @option +@item -1 +@itemx auto +Selected by Encoder (default) +@end table + +@end table + +@c man end AUDIO ENCODERS + +@chapter Video Encoders +@c man begin VIDEO ENCODERS + +A description of some of the currently available video encoders +follows. + +@section libtheora + +Theora format supported through libtheora. + +Requires the presence of the libtheora headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libtheora}. + +@subsection Options + +The following global options are mapped to internal libtheora options +which affect the quality and the bitrate of the encoded stream. + +@table @option +@item b +Set the video bitrate, only works if the @code{qscale} flag in +@option{flags} is not enabled. + +@item flags +Used to enable constant quality mode encoding through the +@option{qscale} flag, and to enable the @code{pass1} and @code{pass2} +modes. + +@item g +Set the GOP size. + +@item global_quality +Set the global quality in lambda units, only works if the +@code{qscale} flag in @option{flags} is enabled. The value is clipped +in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3 +to get a value in the native libtheora range [0-63]. A higher value +corresponds to a higher quality. + +For example, to set maximum constant quality encoding with +@command{ffmpeg}: +@example +ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg +@end example +@end table + +@section libvpx + +VP8 format supported through libvpx. + +Requires the presence of the libvpx headers and library during configuration. +You need to explicitly configure the build with @code{--enable-libvpx}. + +@subsection Options + +Mapping from FFmpeg to libvpx options with conversion notes in parentheses. + +@table @option + +@item threads +g_threads + +@item profile +g_profile + +@item vb +rc_target_bitrate + +@item g +kf_max_dist + +@item keyint_min +kf_min_dist + +@item qmin +rc_min_quantizer + +@item qmax +rc_max_quantizer + +@item bufsize, vb +rc_buf_sz +@code{(bufsize * 1000 / vb)} + +rc_buf_optimal_sz +@code{(bufsize * 1000 / vb * 5 / 6)} + +@item rc_init_occupancy, vb +rc_buf_initial_sz +@code{(rc_init_occupancy * 1000 / vb)} + +@item rc_buffer_aggressivity +rc_undershoot_pct + +@item skip_threshold +rc_dropframe_thresh + +@item qcomp +rc_2pass_vbr_bias_pct + +@item maxrate, vb +rc_2pass_vbr_maxsection_pct +@code{(maxrate * 100 / vb)} + +@item minrate, vb +rc_2pass_vbr_minsection_pct +@code{(minrate * 100 / vb)} + +@item minrate, maxrate, vb +@code{VPX_CBR} +@code{(minrate == maxrate == vb)} + +@item crf +@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL} + +@item quality +@table @option +@item @var{best} +@code{VPX_DL_BEST_QUALITY} +@item @var{good} +@code{VPX_DL_GOOD_QUALITY} +@item @var{realtime} +@code{VPX_DL_REALTIME} +@end table + +@item speed +@code{VP8E_SET_CPUUSED} + +@item nr +@code{VP8E_SET_NOISE_SENSITIVITY} + +@item mb_threshold +@code{VP8E_SET_STATIC_THRESHOLD} + +@item slices +@code{VP8E_SET_TOKEN_PARTITIONS} + +@item max-intra-rate +@code{VP8E_SET_MAX_INTRA_BITRATE_PCT} + +@item force_key_frames +@code{VPX_EFLAG_FORCE_KF} + +@item Alternate reference frame related +@table @option +@item vp8flags altref +@code{VP8E_SET_ENABLEAUTOALTREF} +@item @var{arnr_max_frames} +@code{VP8E_SET_ARNR_MAXFRAMES} +@item @var{arnr_type} +@code{VP8E_SET_ARNR_TYPE} +@item @var{arnr_strength} +@code{VP8E_SET_ARNR_STRENGTH} +@item @var{rc_lookahead} +g_lag_in_frames +@end table + +@item vp8flags error_resilient +g_error_resilient + +@end table + +For more information about libvpx see: +@url{http://www.webmproject.org/} + +@section libx264 + +x264 H.264/MPEG-4 AVC encoder wrapper + +Requires the presence of the libx264 headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libx264}. + +x264 supports an impressive number of features, including 8x8 and 4x4 adaptive +spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, +interlacing (MBAFF), lossless mode, psy optimizations for detail retention +(adaptive quantization, psy-RD, psy-trellis). + +The FFmpeg wrapper provides a mapping for most of them using global options +that match those of the encoders and provides private options for the unique +encoder options. Additionally an expert override is provided to directly pass +a list of key=value tuples as accepted by x264_param_parse. + +@subsection Option Mapping + +The following options are supported by the x264 wrapper, the x264-equivalent +options follow the FFmpeg ones. + +@multitable @columnfractions .2 .2 +@item b @tab bitrate +FFmpeg @code{b} option is expressed in bits/s, x264 @code{bitrate} in kilobits/s. +@item bf @tab bframes +Maximum number of B-frames. +@item g @tab keyint +Maximum GOP size. +@item qmin @tab qpmin +@item qmax @tab qpmax +@item qdiff @tab qpstep +@item qblur @tab qblur +@item qcomp @tab qcomp +@item refs @tab ref +@item sc_threshold @tab scenecut +@item trellis @tab trellis +@item nr @tab nr +Noise reduction. +@item me_range @tab merange +@item me_method @tab me +@item subq @tab subme +@item b_strategy @tab b-adapt +@item keyint_min @tab keyint-min +@item coder @tab cabac +Set coder to @code{ac} to use CABAC. +@item cmp @tab chroma-me +Set to @code{chroma} to use chroma motion estimation. +@item threads @tab threads +@item thread_type @tab sliced_threads +Set to @code{slice} to use sliced threading instead of frame threading. +@item flags -cgop @tab open-gop +Set @code{-cgop} to use recovery points to close GOPs. +@item rc_init_occupancy @tab vbv-init +Initial buffer occupancy. +@end multitable + +@subsection Private Options +@table @option +@item -preset @var{string} +Set the encoding preset (cf. x264 --fullhelp). +@item -tune @var{string} +Tune the encoding params (cf. x264 --fullhelp). +@item -profile @var{string} +Set profile restrictions (cf. x264 --fullhelp). +@item -fastfirstpass @var{integer} +Use fast settings when encoding first pass. +@item -crf @var{float} +Select the quality for constant quality mode. +@item -crf_max @var{float} +In CRF mode, prevents VBV from lowering quality beyond this point. +@item -qp @var{integer} +Constant quantization parameter rate control method. +@item -aq-mode @var{integer} +AQ method + +Possible values: +@table @samp +@item none + +@item variance +Variance AQ (complexity mask). +@item autovariance +Auto-variance AQ (experimental). +@end table +@item -aq-strength @var{float} +AQ strength, reduces blocking and blurring in flat and textured areas. +@item -psy @var{integer} +Use psychovisual optimizations. +@item -psy-rd @var{string} +Strength of psychovisual optimization, in <psy-rd>:<psy-trellis> format. +@item -rc-lookahead @var{integer} +Number of frames to look ahead for frametype and ratecontrol. +@item -weightb @var{integer} +Weighted prediction for B-frames. +@item -weightp @var{integer} +Weighted prediction analysis method. + +Possible values: +@table @samp +@item none + +@item simple + +@item smart + +@end table +@item -ssim @var{integer} +Calculate and print SSIM stats. +@item -intra-refresh @var{integer} +Use Periodic Intra Refresh instead of IDR frames. +@item -b-bias @var{integer} +Influences how often B-frames are used. +@item -b-pyramid @var{integer} +Keep some B-frames as references. + +Possible values: +@table @samp +@item none + +@item strict +Strictly hierarchical pyramid. +@item normal +Non-strict (not Blu-ray compatible). +@end table +@item -mixed-refs @var{integer} +One reference per partition, as opposed to one reference per macroblock. +@item -8x8dct @var{integer} +High profile 8x8 transform. +@item -fast-pskip @var{integer} +@item -aud @var{integer} +Use access unit delimiters. +@item -mbtree @var{integer} +Use macroblock tree ratecontrol. +@item -deblock @var{string} +Loop filter parameters, in <alpha:beta> form. +@item -cplxblur @var{float} +Reduce fluctuations in QP (before curve compression). +@item -partitions @var{string} +A comma-separated list of partitions to consider, possible values: p8x8, p4x4, b8x8, i8x8, i4x4, none, all. +@item -direct-pred @var{integer} +Direct MV prediction mode + +Possible values: +@table @samp +@item none + +@item spatial + +@item temporal + +@item auto + +@end table +@item -slice-max-size @var{integer} +Limit the size of each slice in bytes. +@item -stats @var{string} +Filename for 2 pass stats. +@item -nal-hrd @var{integer} +Signal HRD information (requires vbv-bufsize; cbr not allowed in .mp4). + +Possible values: +@table @samp +@item none + +@item vbr + +@item cbr + +@end table + +@item x264opts @var{options} +Allow to set any x264 option, see @code{x264 --fullhelp} for a list. + +@var{options} is a list of @var{key}=@var{value} couples separated by +":". In @var{filter} and @var{psy-rd} options that use ":" as a separator +themselves, use "," instead. They accept it as well since long ago but this +is kept undocumented for some reason. + +For example to specify libx264 encoding options with @command{ffmpeg}: +@example +ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv +@end example + +For more information about libx264 and the supported options see: +@url{http://www.videolan.org/developers/x264.html} + +@item -x264-params @var{string} +Override the x264 configuration using a :-separated list of key=value parameters. +@example +-x264-params level=30:bframes=0:weightp=0:cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 +@end example +@end table + +Encoding avpresets for common usages are provided so they can be used with the +general presets system (e.g. passing the @code{-pre} option). + +@c man end VIDEO ENCODERS diff --git a/ffmpeg1/doc/errno.txt b/ffmpeg1/doc/errno.txt new file mode 100644 index 0000000..31cab26 --- /dev/null +++ b/ffmpeg1/doc/errno.txt @@ -0,0 +1,174 @@ +The following table lists most error codes found in various operating +systems supported by FFmpeg. + + OS +Code Std F LBMWwb Text (YMMV) + +E2BIG POSIX ++++++ Argument list too long +EACCES POSIX ++++++ Permission denied +EADDRINUSE POSIX +++..+ Address in use +EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address +EADV +..... Advertise error +EAFNOSUPPORT POSIX +++..+ Address family not supported +EAGAIN POSIX + ++++++ Resource temporarily unavailable +EALREADY POSIX +++..+ Operation already in progress +EAUTH .++... Authentication error +EBADARCH ..+... Bad CPU type in executable +EBADE +..... Invalid exchange +EBADEXEC ..+... Bad executable +EBADF POSIX ++++++ Bad file descriptor +EBADFD +..... File descriptor in bad state +EBADMACHO ..+... Malformed Macho file +EBADMSG POSIX ++4... Bad message +EBADR +..... Invalid request descriptor +EBADRPC .++... RPC struct is bad +EBADRQC +..... Invalid request code +EBADSLT +..... Invalid slot +EBFONT +..... Bad font file format +EBUSY POSIX - ++++++ Device or resource busy +ECANCELED POSIX +++... Operation canceled +ECHILD POSIX ++++++ No child processes +ECHRNG +..... Channel number out of range +ECOMM +..... Communication error on send +ECONNABORTED POSIX +++..+ Software caused connection abort +ECONNREFUSED POSIX - +++ss+ Connection refused +ECONNRESET POSIX +++..+ Connection reset +EDEADLK POSIX ++++++ Resource deadlock avoided +EDEADLOCK +..++. File locking deadlock error +EDESTADDRREQ POSIX +++... Destination address required +EDEVERR ..+... Device error +EDOM C89 - ++++++ Numerical argument out of domain +EDOOFUS .F.... Programming error +EDOTDOT +..... RFS specific error +EDQUOT POSIX +++... Disc quota exceeded +EEXIST POSIX ++++++ File exists +EFAULT POSIX - ++++++ Bad address +EFBIG POSIX - ++++++ File too large +EFTYPE .++... Inappropriate file type or format +EHOSTDOWN +++... Host is down +EHOSTUNREACH POSIX +++..+ No route to host +EHWPOISON +..... Memory page has hardware error +EIDRM POSIX +++... Identifier removed +EILSEQ C99 ++++++ Illegal byte sequence +EINPROGRESS POSIX - +++ss+ Operation in progress +EINTR POSIX - ++++++ Interrupted system call +EINVAL POSIX + ++++++ Invalid argument +EIO POSIX + ++++++ I/O error +EISCONN POSIX +++..+ Socket is already connected +EISDIR POSIX ++++++ Is a directory +EISNAM +..... Is a named type file +EKEYEXPIRED +..... Key has expired +EKEYREJECTED +..... Key was rejected by service +EKEYREVOKED +..... Key has been revoked +EL2HLT +..... Level 2 halted +EL2NSYNC +..... Level 2 not synchronized +EL3HLT +..... Level 3 halted +EL3RST +..... Level 3 reset +ELIBACC +..... Can not access a needed shared library +ELIBBAD +..... Accessing a corrupted shared library +ELIBEXEC +..... Cannot exec a shared library directly +ELIBMAX +..... Too many shared libraries +ELIBSCN +..... .lib section in a.out corrupted +ELNRNG +..... Link number out of range +ELOOP POSIX +++..+ Too many levels of symbolic links +EMEDIUMTYPE +..... Wrong medium type +EMFILE POSIX ++++++ Too many open files +EMLINK POSIX ++++++ Too many links +EMSGSIZE POSIX +++..+ Message too long +EMULTIHOP POSIX ++4... Multihop attempted +ENAMETOOLONG POSIX - ++++++ Filen ame too long +ENAVAIL +..... No XENIX semaphores available +ENEEDAUTH .++... Need authenticator +ENETDOWN POSIX +++..+ Network is down +ENETRESET SUSv3 +++..+ Network dropped connection on reset +ENETUNREACH POSIX +++..+ Network unreachable +ENFILE POSIX ++++++ Too many open files in system +ENOANO +..... No anode +ENOATTR .++... Attribute not found +ENOBUFS POSIX - +++..+ No buffer space available +ENOCSI +..... No CSI structure available +ENODATA XSR +N4... No message available +ENODEV POSIX - ++++++ No such device +ENOENT POSIX - ++++++ No such file or directory +ENOEXEC POSIX ++++++ Exec format error +ENOFILE ...++. No such file or directory +ENOKEY +..... Required key not available +ENOLCK POSIX ++++++ No locks available +ENOLINK POSIX ++4... Link has been severed +ENOMEDIUM +..... No medium found +ENOMEM POSIX ++++++ Not enough space +ENOMSG POSIX +++..+ No message of desired type +ENONET +..... Machine is not on the network +ENOPKG +..... Package not installed +ENOPROTOOPT POSIX +++..+ Protocol not available +ENOSPC POSIX ++++++ No space left on device +ENOSR XSR +N4... No STREAM resources +ENOSTR XSR +N4... Not a STREAM +ENOSYS POSIX + ++++++ Function not implemented +ENOTBLK +++... Block device required +ENOTCONN POSIX +++..+ Socket is not connected +ENOTDIR POSIX ++++++ Not a directory +ENOTEMPTY POSIX ++++++ Directory not empty +ENOTNAM +..... Not a XENIX named type file +ENOTRECOVERABLE SUSv4 - +..... State not recoverable +ENOTSOCK POSIX +++..+ Socket operation on non-socket +ENOTSUP POSIX +++... Operation not supported +ENOTTY POSIX ++++++ Inappropriate I/O control operation +ENOTUNIQ +..... Name not unique on network +ENXIO POSIX ++++++ No such device or address +EOPNOTSUPP POSIX +++..+ Operation not supported (on socket) +EOVERFLOW POSIX +++..+ Value too large to be stored in data type +EOWNERDEAD SUSv4 +..... Owner died +EPERM POSIX - ++++++ Operation not permitted +EPFNOSUPPORT +++..+ Protocol family not supported +EPIPE POSIX - ++++++ Broken pipe +EPROCLIM .++... Too many processes +EPROCUNAVAIL .++... Bad procedure for program +EPROGMISMATCH .++... Program version wrong +EPROGUNAVAIL .++... RPC prog. not avail +EPROTO POSIX ++4... Protocol error +EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported +EPROTOTYPE POSIX +++..+ Protocol wrong type for socket +EPWROFF ..+... Device power is off +ERANGE C89 - ++++++ Result too large +EREMCHG +..... Remote address changed +EREMOTE +++... Object is remote +EREMOTEIO +..... Remote I/O error +ERESTART +..... Interrupted system call should be restarted +ERFKILL +..... Operation not possible due to RF-kill +EROFS POSIX ++++++ Read-only file system +ERPCMISMATCH .++... RPC version wrong +ESHLIBVERS ..+... Shared library version mismatch +ESHUTDOWN +++..+ Cannot send after socket shutdown +ESOCKTNOSUPPORT +++... Socket type not supported +ESPIPE POSIX ++++++ Illegal seek +ESRCH POSIX ++++++ No such process +ESRMNT +..... Srmount error +ESTALE POSIX +++..+ Stale NFS file handle +ESTRPIPE +..... Streams pipe error +ETIME XSR +N4... Stream ioctl timeout +ETIMEDOUT POSIX - +++ss+ Connection timed out +ETOOMANYREFS +++... Too many references: cannot splice +ETXTBSY POSIX +++... Text file busy +EUCLEAN +..... Structure needs cleaning +EUNATCH +..... Protocol driver not attached +EUSERS +++... Too many users +EWOULDBLOCK POSIX +++..+ Operation would block +EXDEV POSIX ++++++ Cross-device link +EXFULL +..... Exchange full + +Notations: + +F: used in FFmpeg (-: a few times, +: a lot) + +SUSv3: Single Unix Specification, version 3 +SUSv4: Single Unix Specification, version 4 +XSR: XSI STREAMS (obsolete) + +OS: availability on some supported operating systems +L: GNU/Linux +B: BSD (F: FreeBSD, N: NetBSD) +M: MacOS X +W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h) +w: Mingw32 (3.17) and Mingw64 (2.0.1) +b: BeOS diff --git a/ffmpeg1/doc/eval.texi b/ffmpeg1/doc/eval.texi new file mode 100644 index 0000000..e1a5c0a --- /dev/null +++ b/ffmpeg1/doc/eval.texi @@ -0,0 +1,299 @@ +@chapter Expression Evaluation +@c man begin EXPRESSION EVALUATION + +When evaluating an arithmetic expression, FFmpeg uses an internal +formula evaluator, implemented through the @file{libavutil/eval.h} +interface. + +An expression may contain unary, binary operators, constants, and +functions. + +Two expressions @var{expr1} and @var{expr2} can be combined to form +another expression "@var{expr1};@var{expr2}". +@var{expr1} and @var{expr2} are evaluated in turn, and the new +expression evaluates to the value of @var{expr2}. + +The following binary operators are available: @code{+}, @code{-}, +@code{*}, @code{/}, @code{^}. + +The following unary operators are available: @code{+}, @code{-}. + +The following functions are available: +@table @option +@item abs(x) +Compute absolute value of @var{x}. + +@item acos(x) +Compute arccosine of @var{x}. + +@item asin(x) +Compute arcsine of @var{x}. + +@item atan(x) +Compute arctangent of @var{x}. + +@item bitand(x, y) +@item bitor(x, y) +Compute bitwise and/or operation on @var{x} and @var{y}. + +The results of the evaluation of @var{x} and @var{y} are converted to +integers before executing the bitwise operation. + +Note that both the conversion to integer and the conversion back to +floating point can lose precision. Beware of unexpected results for +large numbers (usually 2^53 and larger). + +@item ceil(expr) +Round the value of expression @var{expr} upwards to the nearest +integer. For example, "ceil(1.5)" is "2.0". + +@item cos(x) +Compute cosine of @var{x}. + +@item cosh(x) +Compute hyperbolic cosine of @var{x}. + +@item eq(x, y) +Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise. + +@item exp(x) +Compute exponential of @var{x} (with base @code{e}, the Euler's number). + +@item floor(expr) +Round the value of expression @var{expr} downwards to the nearest +integer. For example, "floor(-1.5)" is "-2.0". + +@item gauss(x) +Compute Gauss function of @var{x}, corresponding to +@code{exp(-x*x/2) / sqrt(2*PI)}. + +@item gcd(x, y) +Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and +@var{y} are 0 or either or both are less than zero then behavior is undefined. + +@item gt(x, y) +Return 1 if @var{x} is greater than @var{y}, 0 otherwise. + +@item gte(x, y) +Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise. + +@item hypot(x, y) +This function is similar to the C function with the same name; it returns +"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a +right triangle with sides of length @var{x} and @var{y}, or the distance of the +point (@var{x}, @var{y}) from the origin. + +@item if(x, y) +Evaluate @var{x}, and if the result is non-zero return the result of +the evaluation of @var{y}, return 0 otherwise. + +@item if(x, y, z) +Evaluate @var{x}, and if the result is non-zero return the evaluation +result of @var{y}, otherwise the evaluation result of @var{z}. + +@item ifnot(x, y) +Evaluate @var{x}, and if the result is zero return the result of the +evaluation of @var{y}, return 0 otherwise. + +@item ifnot(x, y, z) +Evaluate @var{x}, and if the result is zero return the evaluation +result of @var{y}, otherwise the evaluation result of @var{z}. + +@item isinf(x) +Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise. + +@item isnan(x) +Return 1.0 if @var{x} is NAN, 0.0 otherwise. + +@item ld(var) +Allow to load the value of the internal variable with number +@var{var}, which was previously stored with st(@var{var}, @var{expr}). +The function returns the loaded value. + +@item log(x) +Compute natural logarithm of @var{x}. + +@item lt(x, y) +Return 1 if @var{x} is lesser than @var{y}, 0 otherwise. + +@item lte(x, y) +Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise. + +@item max(x, y) +Return the maximum between @var{x} and @var{y}. + +@item min(x, y) +Return the maximum between @var{x} and @var{y}. + +@item mod(x, y) +Compute the remainder of division of @var{x} by @var{y}. + +@item not(expr) +Return 1.0 if @var{expr} is zero, 0.0 otherwise. + +@item pow(x, y) +Compute the power of @var{x} elevated @var{y}, it is equivalent to +"(@var{x})^(@var{y})". + +@item print(t) +@item print(t, l) +Print the value of expression @var{t} with loglevel @var{l}. If +@var{l} is not specified then a default log level is used. +Returns the value of the expression printed. + +Prints t with loglevel l + +@item random(x) +Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the +internal variable which will be used to save the seed/state. + +@item root(expr, max) +Find an input value for which the function represented by @var{expr} +with argument @var{ld(0)} is 0 in the interval 0..@var{max}. + +The expression in @var{expr} must denote a continuous function or the +result is undefined. + +@var{ld(0)} is used to represent the function input value, which means +that the given expression will be evaluated multiple times with +various input values that the expression can access through +@code{ld(0)}. When the expression evaluates to 0 then the +corresponding input value will be returned. + +@item sin(x) +Compute sine of @var{x}. + +@item sinh(x) +Compute hyperbolic sine of @var{x}. + +@item sqrt(expr) +Compute the square root of @var{expr}. This is equivalent to +"(@var{expr})^.5". + +@item squish(x) +Compute expression @code{1/(1 + exp(4*x))}. + +@item st(var, expr) +Allow to store the value of the expression @var{expr} in an internal +variable. @var{var} specifies the number of the variable where to +store the value, and it is a value ranging from 0 to 9. The function +returns the value stored in the internal variable. +Note, Variables are currently not shared between expressions. + +@item tan(x) +Compute tangent of @var{x}. + +@item tanh(x) +Compute hyperbolic tangent of @var{x}. + +@item taylor(expr, x) +@item taylor(expr, x, id) +Evaluate a Taylor series at @var{x}, given an expression representing +the @code{ld(id)}-th derivative of a function at 0. + +When the series does not converge the result is undefined. + +@var{ld(id)} is used to represent the derivative order in @var{expr}, +which means that the given expression will be evaluated multiple times +with various input values that the expression can access through +@code{ld(id)}. If @var{id} is not specified then 0 is assumed. + +Note, when you have the derivatives at y instead of 0, +@code{taylor(expr, x-y)} can be used. + +@item time(0) +Return the current (wallclock) time in seconds. + +@item trunc(expr) +Round the value of expression @var{expr} towards zero to the nearest +integer. For example, "trunc(-1.5)" is "-1.0". + +@item while(cond, expr) +Evaluate expression @var{expr} while the expression @var{cond} is +non-zero, and returns the value of the last @var{expr} evaluation, or +NAN if @var{cond} was always false. +@end table + +The following constants are available: +@table @option +@item PI +area of the unit disc, approximately 3.14 +@item E +exp(1) (Euler's number), approximately 2.718 +@item PHI +golden ratio (1+sqrt(5))/2, approximately 1.618 +@end table + +Assuming that an expression is considered "true" if it has a non-zero +value, note that: + +@code{*} works like AND + +@code{+} works like OR + +For example the construct: +@example +if (A AND B) then C +@end example +is equivalent to: +@example +if(A*B, C) +@end example + +In your C code, you can extend the list of unary and binary functions, +and define recognized constants, so that they are available for your +expressions. + +The evaluator also recognizes the International System unit prefixes. +If 'i' is appended after the prefix, binary prefixes are used, which +are based on powers of 1024 instead of powers of 1000. +The 'B' postfix multiplies the value by 8, and can be appended after a +unit prefix or used alone. This allows using for example 'KB', 'MiB', +'G' and 'B' as number postfix. + +The list of available International System prefixes follows, with +indication of the corresponding powers of 10 and of 2. +@table @option +@item y +10^-24 / 2^-80 +@item z +10^-21 / 2^-70 +@item a +10^-18 / 2^-60 +@item f +10^-15 / 2^-50 +@item p +10^-12 / 2^-40 +@item n +10^-9 / 2^-30 +@item u +10^-6 / 2^-20 +@item m +10^-3 / 2^-10 +@item c +10^-2 +@item d +10^-1 +@item h +10^2 +@item k +10^3 / 2^10 +@item K +10^3 / 2^10 +@item M +10^6 / 2^20 +@item G +10^9 / 2^30 +@item T +10^12 / 2^40 +@item P +10^15 / 2^40 +@item E +10^18 / 2^50 +@item Z +10^21 / 2^60 +@item Y +10^24 / 2^70 +@end table + +@c man end diff --git a/ffmpeg1/doc/examples/Makefile b/ffmpeg1/doc/examples/Makefile new file mode 100644 index 0000000..c849daa --- /dev/null +++ b/ffmpeg1/doc/examples/Makefile @@ -0,0 +1,37 @@ +# use pkg-config for getting CFLAGS and LDLIBS +FFMPEG_LIBS= libavdevice \ + libavformat \ + libavfilter \ + libavcodec \ + libswresample \ + libswscale \ + libavutil \ + +CFLAGS += -Wall -O2 -g +CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) +LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) + +EXAMPLES= decoding_encoding \ + demuxing \ + filtering_video \ + filtering_audio \ + metadata \ + muxing \ + resampling_audio \ + scaling_video \ + +OBJS=$(addsuffix .o,$(EXAMPLES)) + +# the following examples make explicit use of the math library +decoding_encoding: LDLIBS += -lm +muxing: LDLIBS += -lm + +.phony: all clean-test clean + +all: $(OBJS) $(EXAMPLES) + +clean-test: + $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg + +clean: clean-test + $(RM) $(EXAMPLES) $(OBJS) diff --git a/ffmpeg1/doc/examples/README b/ffmpeg1/doc/examples/README new file mode 100644 index 0000000..a461813 --- /dev/null +++ b/ffmpeg1/doc/examples/README @@ -0,0 +1,18 @@ +FFmpeg examples README +---------------------- + +Both following use cases rely on pkg-config and make, thus make sure +that you have them installed and working on your system. + + +1) Build the installed examples in a generic read/write user directory + +Copy to a read/write user directory and just use "make", it will link +to the libraries on your system, assuming the PKG_CONFIG_PATH is +correctly configured. + +2) Build the examples in-tree + +Assuming you are in the source FFmpeg checkout directory, you need to build +FFmpeg (no need to make install in any prefix). Then you can go into the +doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make. diff --git a/ffmpeg1/doc/examples/decoding_encoding.c b/ffmpeg1/doc/examples/decoding_encoding.c new file mode 100644 index 0000000..ae1057c --- /dev/null +++ b/ffmpeg1/doc/examples/decoding_encoding.c @@ -0,0 +1,650 @@ +/* + * Copyright (c) 2001 Fabrice Bellard + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavcodec API use example. + * + * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...), + * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the + * format handling + * @example doc/examples/decoding_encoding.c + */ + +#include <math.h> + +#include <libavutil/opt.h> +#include <libavcodec/avcodec.h> +#include <libavutil/channel_layout.h> +#include <libavutil/common.h> +#include <libavutil/imgutils.h> +#include <libavutil/mathematics.h> +#include <libavutil/samplefmt.h> + +#define INBUF_SIZE 4096 +#define AUDIO_INBUF_SIZE 20480 +#define AUDIO_REFILL_THRESH 4096 + +/* check that a given sample format is supported by the encoder */ +static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) +{ + const enum AVSampleFormat *p = codec->sample_fmts; + + while (*p != AV_SAMPLE_FMT_NONE) { + if (*p == sample_fmt) + return 1; + p++; + } + return 0; +} + +/* just pick the highest supported samplerate */ +static int select_sample_rate(AVCodec *codec) +{ + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; +} + +/* select layout with the highest channel count */ +static int select_channel_layout(AVCodec *codec) +{ + const uint64_t *p; + uint64_t best_ch_layout = 0; + int best_nb_channells = 0; + + if (!codec->channel_layouts) + return AV_CH_LAYOUT_STEREO; + + p = codec->channel_layouts; + while (*p) { + int nb_channels = av_get_channel_layout_nb_channels(*p); + + if (nb_channels > best_nb_channells) { + best_ch_layout = *p; + best_nb_channells = nb_channels; + } + p++; + } + return best_ch_layout; +} + +/* + * Audio encoding example + */ +static void audio_encode_example(const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; + FILE *f; + uint16_t *samples; + float t, tincr; + + printf("Encode audio file %s\n", filename); + + /* find the MP2 encoder */ + codec = avcodec_find_encoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 64000; + + /* check that the encoder supports s16 pcm input */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { + fprintf(stderr, "Encoder does not support sample format %s", + av_get_sample_fmt_name(c->sample_fmt)); + exit(1); + } + + /* select other audio parameters supported by the encoder */ + c->sample_rate = select_sample_rate(codec); + c->channel_layout = select_channel_layout(codec); + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + /* frame containing input raw audio */ + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + + frame->nb_samples = c->frame_size; + frame->format = c->sample_fmt; + frame->channel_layout = c->channel_layout; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); + samples = av_malloc(buffer_size); + if (!samples) { + fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { + fprintf(stderr, "Could not setup audio frame\n"); + exit(1); + } + + /* encode a single tone sound */ + t = 0; + tincr = 2 * M_PI * 440.0 / c->sample_rate; + for(i=0;i<200;i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + for (j = 0; j < c->frame_size; j++) { + samples[2*j] = (int)(sin(t) * 10000); + + for (k = 1; k < c->channels; k++) + samples[2*j + k] = samples[2*j]; + t += tincr; + } + /* encode the samples */ + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + fclose(f); + + av_freep(&samples); + avcodec_free_frame(&frame); + avcodec_close(c); + av_free(c); +} + +/* + * Audio decoding. + */ +static void audio_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int len; + FILE *f, *outfile; + uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + AVFrame *decoded_frame = NULL; + + av_init_packet(&avpkt); + + printf("Decode audio file %s to %s\n", filename, outfilename); + + /* find the mpeg audio decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + outfile = fopen(outfilename, "wb"); + if (!outfile) { + av_free(c); + exit(1); + } + + /* decode until eof */ + avpkt.data = inbuf; + avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); + + while (avpkt.size > 0) { + int got_frame = 0; + + if (!decoded_frame) { + if (!(decoded_frame = avcodec_alloc_frame())) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + } else + avcodec_get_frame_defaults(decoded_frame); + + len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); + if (len < 0) { + fprintf(stderr, "Error while decoding\n"); + exit(1); + } + if (got_frame) { + /* if a frame has been decoded, output it */ + int data_size = av_samples_get_buffer_size(NULL, c->channels, + decoded_frame->nb_samples, + c->sample_fmt, 1); + fwrite(decoded_frame->data[0], 1, data_size, outfile); + } + avpkt.size -= len; + avpkt.data += len; + avpkt.dts = + avpkt.pts = AV_NOPTS_VALUE; + if (avpkt.size < AUDIO_REFILL_THRESH) { + /* Refill the input buffer, to avoid trying to decode + * incomplete frames. Instead of this, one could also use + * a parser, or use a proper container format through + * libavformat. */ + memmove(inbuf, avpkt.data, avpkt.size); + avpkt.data = inbuf; + len = fread(avpkt.data + avpkt.size, 1, + AUDIO_INBUF_SIZE - avpkt.size, f); + if (len > 0) + avpkt.size += len; + } + } + + fclose(outfile); + fclose(f); + + avcodec_close(c); + av_free(c); + avcodec_free_frame(&decoded_frame); +} + +/* + * Video encoding example + */ +static void video_encode_example(const char *filename, int codec_id) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int i, ret, x, y, got_output; + FILE *f; + AVFrame *frame; + AVPacket pkt; + uint8_t endcode[] = { 0, 0, 1, 0xb7 }; + + printf("Encode video file %s\n", filename); + + /* find the mpeg1 video encoder */ + codec = avcodec_find_encoder(codec_id); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 400000; + /* resolution must be a multiple of two */ + c->width = 352; + c->height = 288; + /* frames per second */ + c->time_base= (AVRational){1,25}; + c->gop_size = 10; /* emit one intra frame every ten frames */ + c->max_b_frames=1; + c->pix_fmt = AV_PIX_FMT_YUV420P; + + if(codec_id == AV_CODEC_ID_H264) + av_opt_set(c->priv_data, "preset", "slow", 0); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + frame->format = c->pix_fmt; + frame->width = c->width; + frame->height = c->height; + + /* the image can be allocated by any means and av_image_alloc() is + * just the most convenient way if av_malloc() is to be used */ + ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, + c->pix_fmt, 32); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw picture buffer\n"); + exit(1); + } + + /* encode 1 second of video */ + for(i=0;i<25;i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + fflush(stdout); + /* prepare a dummy image */ + /* Y */ + for(y=0;y<c->height;y++) { + for(x=0;x<c->width;x++) { + frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; + } + } + + /* Cb and Cr */ + for(y=0;y<c->height/2;y++) { + for(x=0;x<c->width/2;x++) { + frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; + frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; + } + } + + frame->pts = i; + + /* encode the image */ + ret = avcodec_encode_video2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + fflush(stdout); + + ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* add sequence end code to have a real mpeg file */ + fwrite(endcode, 1, sizeof(endcode), f); + fclose(f); + + avcodec_close(c); + av_free(c); + av_freep(&frame->data[0]); + avcodec_free_frame(&frame); + printf("\n"); +} + +/* + * Video decoding example + */ + +static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, + char *filename) +{ + FILE *f; + int i; + + f=fopen(filename,"w"); + fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255); + for(i=0;i<ysize;i++) + fwrite(buf + i * wrap,1,xsize,f); + fclose(f); +} + +static int decode_write_frame(const char *outfilename, AVCodecContext *avctx, + AVFrame *frame, int *frame_count, AVPacket *pkt, int last) +{ + int len, got_frame; + char buf[1024]; + + len = avcodec_decode_video2(avctx, frame, &got_frame, pkt); + if (len < 0) { + fprintf(stderr, "Error while decoding frame %d\n", *frame_count); + return len; + } + if (got_frame) { + printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count); + fflush(stdout); + + /* the picture is allocated by the decoder, no need to free it */ + snprintf(buf, sizeof(buf), outfilename, *frame_count); + pgm_save(frame->data[0], frame->linesize[0], + avctx->width, avctx->height, buf); + (*frame_count)++; + } + if (pkt->data) { + pkt->size -= len; + pkt->data += len; + } + return 0; +} + +static void video_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int frame_count; + FILE *f; + AVFrame *frame; + uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + + av_init_packet(&avpkt); + + /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */ + memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE); + + printf("Decode video file %s to %s\n", filename, outfilename); + + /* find the mpeg1 video decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + if(codec->capabilities&CODEC_CAP_TRUNCATED) + c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */ + + /* For some codecs, such as msmpeg4 and mpeg4, width and height + MUST be initialized there because this information is not + available in the bitstream. */ + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + + frame_count = 0; + for(;;) { + avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); + if (avpkt.size == 0) + break; + + /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio) + and this is the only method to use them because you cannot + know the compressed data size before analysing it. + + BUT some other codecs (msmpeg4, mpeg4) are inherently frame + based, so you must call them with all the data for one + frame exactly. You must also initialize 'width' and + 'height' before initializing them. */ + + /* NOTE2: some codecs allow the raw parameters (frame size, + sample rate) to be changed at any frame. We handle this, so + you should also take care of it */ + + /* here, we use a stream based decoder (mpeg1video), so we + feed decoder and see if it could decode a frame */ + avpkt.data = inbuf; + while (avpkt.size > 0) + if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) + exit(1); + } + + /* some codecs, such as MPEG, transmit the I and P frame with a + latency of one frame. You must do the following to have a + chance to get the last frame of the video */ + avpkt.data = NULL; + avpkt.size = 0; + decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); + + fclose(f); + + avcodec_close(c); + av_free(c); + avcodec_free_frame(&frame); + printf("\n"); +} + +int main(int argc, char **argv) +{ + const char *output_type; + + /* register all the codecs */ + avcodec_register_all(); + + if (argc < 2) { + printf("usage: %s output_type\n" + "API example program to decode/encode a media stream with libavcodec.\n" + "This program generates a synthetic stream and encodes it to a file\n" + "named test.h264, test.mp2 or test.mpg depending on output_type.\n" + "The encoded stream is then decoded and written to a raw data output.\n" + "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n", + argv[0]); + return 1; + } + output_type = argv[1]; + + if (!strcmp(output_type, "h264")) { + video_encode_example("test.h264", AV_CODEC_ID_H264); + } else if (!strcmp(output_type, "mp2")) { + audio_encode_example("test.mp2"); + audio_decode_example("test.sw", "test.mp2"); + } else if (!strcmp(output_type, "mpg")) { + video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); + video_decode_example("test%02d.pgm", "test.mpg"); + } else { + fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", + output_type); + return 1; + } + + return 0; +} diff --git a/ffmpeg1/doc/examples/demuxing.c b/ffmpeg1/doc/examples/demuxing.c new file mode 100644 index 0000000..8a1b69b --- /dev/null +++ b/ffmpeg1/doc/examples/demuxing.c @@ -0,0 +1,342 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat demuxing API use example. + * + * Show how to use the libavformat and libavcodec API to demux and + * decode audio and video data. + * @example doc/examples/demuxing.c + */ + +#include <libavutil/imgutils.h> +#include <libavutil/samplefmt.h> +#include <libavutil/timestamp.h> +#include <libavformat/avformat.h> + +static AVFormatContext *fmt_ctx = NULL; +static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx; +static AVStream *video_stream = NULL, *audio_stream = NULL; +static const char *src_filename = NULL; +static const char *video_dst_filename = NULL; +static const char *audio_dst_filename = NULL; +static FILE *video_dst_file = NULL; +static FILE *audio_dst_file = NULL; + +static uint8_t *video_dst_data[4] = {NULL}; +static int video_dst_linesize[4]; +static int video_dst_bufsize; + +static uint8_t **audio_dst_data = NULL; +static int audio_dst_linesize; +static int audio_dst_bufsize; + +static int video_stream_idx = -1, audio_stream_idx = -1; +static AVFrame *frame = NULL; +static AVPacket pkt; +static int video_frame_count = 0; +static int audio_frame_count = 0; + +static int decode_packet(int *got_frame, int cached) +{ + int ret = 0; + + if (pkt.stream_index == video_stream_idx) { + /* decode video frame */ + ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding video frame\n"); + return ret; + } + + if (*got_frame) { + printf("video_frame%s n:%d coded_n:%d pts:%s\n", + cached ? "(cached)" : "", + video_frame_count++, frame->coded_picture_number, + av_ts2timestr(frame->pts, &video_dec_ctx->time_base)); + + /* copy decoded frame to destination buffer: + * this is required since rawvideo expects non aligned data */ + av_image_copy(video_dst_data, video_dst_linesize, + (const uint8_t **)(frame->data), frame->linesize, + video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height); + + /* write to rawvideo file */ + fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); + } + } else if (pkt.stream_index == audio_stream_idx) { + /* decode audio frame */ + ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding audio frame\n"); + return ret; + } + + if (*got_frame) { + printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", + cached ? "(cached)" : "", + audio_frame_count++, frame->nb_samples, + av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); + + ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame), + frame->nb_samples, frame->format, 1); + if (ret < 0) { + fprintf(stderr, "Could not allocate audio buffer\n"); + return AVERROR(ENOMEM); + } + + /* TODO: extend return code of the av_samples_* functions so that this call is not needed */ + audio_dst_bufsize = + av_samples_get_buffer_size(NULL, av_frame_get_channels(frame), + frame->nb_samples, frame->format, 1); + + /* copy audio data to destination buffer: + * this is required since rawaudio expects non aligned data */ + av_samples_copy(audio_dst_data, frame->data, 0, 0, + frame->nb_samples, av_frame_get_channels(frame), frame->format); + + /* write to rawaudio file */ + fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file); + av_freep(&audio_dst_data[0]); + } + } + + return ret; +} + +static int open_codec_context(int *stream_idx, + AVFormatContext *fmt_ctx, enum AVMediaType type) +{ + int ret; + AVStream *st; + AVCodecContext *dec_ctx = NULL; + AVCodec *dec = NULL; + + ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); + if (ret < 0) { + fprintf(stderr, "Could not find %s stream in input file '%s'\n", + av_get_media_type_string(type), src_filename); + return ret; + } else { + *stream_idx = ret; + st = fmt_ctx->streams[*stream_idx]; + + /* find decoder for the stream */ + dec_ctx = st->codec; + dec = avcodec_find_decoder(dec_ctx->codec_id); + if (!dec) { + fprintf(stderr, "Failed to find %s codec\n", + av_get_media_type_string(type)); + return ret; + } + + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + fprintf(stderr, "Failed to open %s codec\n", + av_get_media_type_string(type)); + return ret; + } + } + + return 0; +} + +static int get_format_from_sample_fmt(const char **fmt, + enum AVSampleFormat sample_fmt) +{ + int i; + struct sample_fmt_entry { + enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; + } sample_fmt_entries[] = { + { AV_SAMPLE_FMT_U8, "u8", "u8" }, + { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, + { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, + { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, + { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, + }; + *fmt = NULL; + + for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { + struct sample_fmt_entry *entry = &sample_fmt_entries[i]; + if (sample_fmt == entry->sample_fmt) { + *fmt = AV_NE(entry->fmt_be, entry->fmt_le); + return 0; + } + } + + fprintf(stderr, + "sample format %s is not supported as output format\n", + av_get_sample_fmt_name(sample_fmt)); + return -1; +} + +int main (int argc, char **argv) +{ + int ret = 0, got_frame; + + if (argc != 4) { + fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n" + "API example program to show how to read frames from an input file.\n" + "This program reads frames from a file, decodes them, and writes decoded\n" + "video frames to a rawvideo file named video_output_file, and decoded\n" + "audio frames to a rawaudio file named audio_output_file.\n" + "\n", argv[0]); + exit(1); + } + src_filename = argv[1]; + video_dst_filename = argv[2]; + audio_dst_filename = argv[3]; + + /* register all formats and codecs */ + av_register_all(); + + /* open input file, and allocate format context */ + if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { + fprintf(stderr, "Could not open source file %s\n", src_filename); + exit(1); + } + + /* retrieve stream information */ + if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { + fprintf(stderr, "Could not find stream information\n"); + exit(1); + } + + if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { + video_stream = fmt_ctx->streams[video_stream_idx]; + video_dec_ctx = video_stream->codec; + + video_dst_file = fopen(video_dst_filename, "wb"); + if (!video_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); + ret = 1; + goto end; + } + + /* allocate image where the decoded image will be put */ + ret = av_image_alloc(video_dst_data, video_dst_linesize, + video_dec_ctx->width, video_dec_ctx->height, + video_dec_ctx->pix_fmt, 1); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw video buffer\n"); + goto end; + } + video_dst_bufsize = ret; + } + + if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) { + int nb_planes; + + audio_stream = fmt_ctx->streams[audio_stream_idx]; + audio_dec_ctx = audio_stream->codec; + audio_dst_file = fopen(audio_dst_filename, "wb"); + if (!audio_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); + ret = 1; + goto end; + } + + nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ? + audio_dec_ctx->channels : 1; + audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes); + if (!audio_dst_data) { + fprintf(stderr, "Could not allocate audio data buffers\n"); + ret = AVERROR(ENOMEM); + goto end; + } + } + + /* dump input information to stderr */ + av_dump_format(fmt_ctx, 0, src_filename, 0); + + if (!audio_stream && !video_stream) { + fprintf(stderr, "Could not find audio or video stream in the input, aborting\n"); + ret = 1; + goto end; + } + + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "Could not allocate frame\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + /* initialize packet, set data to NULL, let the demuxer fill it */ + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + if (video_stream) + printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename); + if (audio_stream) + printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename); + + /* read frames from the file */ + while (av_read_frame(fmt_ctx, &pkt) >= 0) { + decode_packet(&got_frame, 0); + av_free_packet(&pkt); + } + + /* flush cached frames */ + pkt.data = NULL; + pkt.size = 0; + do { + decode_packet(&got_frame, 1); + } while (got_frame); + + printf("Demuxing succeeded.\n"); + + if (video_stream) { + printf("Play the output video file with the command:\n" + "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", + av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height, + video_dst_filename); + } + + if (audio_stream) { + const char *fmt; + + if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0) + goto end; + printf("Play the output audio file with the command:\n" + "ffplay -f %s -ac %d -ar %d %s\n", + fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate, + audio_dst_filename); + } + +end: + if (video_dec_ctx) + avcodec_close(video_dec_ctx); + if (audio_dec_ctx) + avcodec_close(audio_dec_ctx); + avformat_close_input(&fmt_ctx); + if (video_dst_file) + fclose(video_dst_file); + if (audio_dst_file) + fclose(audio_dst_file); + av_free(frame); + av_free(video_dst_data[0]); + av_free(audio_dst_data); + + return ret < 0; +} diff --git a/ffmpeg1/doc/examples/filtering_audio.c b/ffmpeg1/doc/examples/filtering_audio.c new file mode 100644 index 0000000..456a1c9 --- /dev/null +++ b/ffmpeg1/doc/examples/filtering_audio.c @@ -0,0 +1,244 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2012 Clément Bœsch + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for audio decoding and filtering + * @example doc/examples/filtering_audio.c + */ + +#include <unistd.h> + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/avcodec.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> + +const char *filter_descr = "aresample=8000,aconvert=s16:mono"; +const char *player = "ffplay -f s16le -ar 8000 -ac 1 -"; + +static AVFormatContext *fmt_ctx; +static AVCodecContext *dec_ctx; +AVFilterContext *buffersink_ctx; +AVFilterContext *buffersrc_ctx; +AVFilterGraph *filter_graph; +static int audio_stream_index = -1; + +static int open_input_file(const char *filename) +{ + int ret; + AVCodec *dec; + + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + /* select the audio stream */ + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n"); + return ret; + } + audio_stream_index = ret; + dec_ctx = fmt_ctx->streams[audio_stream_index]->codec; + + /* init the audio decoder */ + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n"); + return ret; + } + + return 0; +} + +static int init_filters(const char *filters_descr) +{ + char args[512]; + int ret; + AVFilter *abuffersrc = avfilter_get_by_name("abuffer"); + AVFilter *abuffersink = avfilter_get_by_name("abuffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 }; + AVABufferSinkParams *abuffersink_params; + const AVFilterLink *outlink; + AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base; + + filter_graph = avfilter_graph_alloc(); + + /* buffer audio source: the decoded frames from the decoder will be inserted here. */ + if (!dec_ctx->channel_layout) + dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels); + snprintf(args, sizeof(args), + "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, + time_base.num, time_base.den, dec_ctx->sample_rate, + av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout); + ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); + return ret; + } + + /* buffer audio sink: to terminate the filter chain. */ + abuffersink_params = av_abuffersink_params_alloc(); + abuffersink_params->sample_fmts = sample_fmts; + ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out", + NULL, abuffersink_params, filter_graph); + av_free(abuffersink_params); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); + return ret; + } + + /* Endpoints for the filter graph. */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse(filter_graph, filters_descr, + &inputs, &outputs, NULL)) < 0) + return ret; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + return ret; + + /* Print summary of the sink buffer + * Note: args buffer is reused to store channel layout string */ + outlink = buffersink_ctx->inputs[0]; + av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout); + av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n", + (int)outlink->sample_rate, + (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"), + args); + + return 0; +} + +static void print_frame(const AVFrame *frame) +{ + const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame)); + const uint16_t *p = (uint16_t*)frame->data[0]; + const uint16_t *p_end = p + n; + + while (p < p_end) { + fputc(*p & 0xff, stdout); + fputc(*p>>8 & 0xff, stdout); + p++; + } + fflush(stdout); +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet; + AVFrame *frame = av_frame_alloc(); + AVFrame *filt_frame = av_frame_alloc(); + int got_frame; + + if (!frame || !filt_frame) { + perror("Could not allocate frame"); + exit(1); + } + if (argc != 2) { + fprintf(stderr, "Usage: %s file | %s\n", argv[0], player); + exit(1); + } + + avcodec_register_all(); + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = init_filters(filter_descr)) < 0) + goto end; + + /* read all packets */ + while (1) { + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) + break; + + if (packet.stream_index == audio_stream_index) { + avcodec_get_frame_defaults(frame); + got_frame = 0; + ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n"); + continue; + } + + if (got_frame) { + /* push the audio data from decoded frame into the filtergraph */ + if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n"); + break; + } + + /* pull filtered audio from the filtergraph */ + while (1) { + ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); + if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + break; + if(ret < 0) + goto end; + print_frame(filt_frame); + av_frame_unref(filt_frame); + } + } + } + av_free_packet(&packet); + } +end: + avfilter_graph_free(&filter_graph); + if (dec_ctx) + avcodec_close(dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + av_frame_free(&filt_frame); + + if (ret < 0 && ret != AVERROR_EOF) { + char buf[1024]; + av_strerror(ret, buf, sizeof(buf)); + fprintf(stderr, "Error occurred: %s\n", buf); + exit(1); + } + + exit(0); +} diff --git a/ffmpeg1/doc/examples/filtering_video.c b/ffmpeg1/doc/examples/filtering_video.c new file mode 100644 index 0000000..daa3966 --- /dev/null +++ b/ffmpeg1/doc/examples/filtering_video.c @@ -0,0 +1,251 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for decoding and filtering + * @example doc/examples/filtering_video.c + */ + +#define _XOPEN_SOURCE 600 /* for usleep */ +#include <unistd.h> + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/avcodec.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> + +const char *filter_descr = "scale=78:24"; + +static AVFormatContext *fmt_ctx; +static AVCodecContext *dec_ctx; +AVFilterContext *buffersink_ctx; +AVFilterContext *buffersrc_ctx; +AVFilterGraph *filter_graph; +static int video_stream_index = -1; +static int64_t last_pts = AV_NOPTS_VALUE; + +static int open_input_file(const char *filename) +{ + int ret; + AVCodec *dec; + + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + /* select the video stream */ + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n"); + return ret; + } + video_stream_index = ret; + dec_ctx = fmt_ctx->streams[video_stream_index]->codec; + + /* init the video decoder */ + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n"); + return ret; + } + + return 0; +} + +static int init_filters(const char *filters_descr) +{ + char args[512]; + int ret; + AVFilter *buffersrc = avfilter_get_by_name("buffer"); + AVFilter *buffersink = avfilter_get_by_name("buffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE }; + AVBufferSinkParams *buffersink_params; + + filter_graph = avfilter_graph_alloc(); + + /* buffer video source: the decoded frames from the decoder will be inserted here. */ + snprintf(args, sizeof(args), + "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", + dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt, + dec_ctx->time_base.num, dec_ctx->time_base.den, + dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den); + + ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n"); + return ret; + } + + /* buffer video sink: to terminate the filter chain. */ + buffersink_params = av_buffersink_params_alloc(); + buffersink_params->pixel_fmts = pix_fmts; + ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", + NULL, buffersink_params, filter_graph); + av_free(buffersink_params); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n"); + return ret; + } + + /* Endpoints for the filter graph. */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse(filter_graph, filters_descr, + &inputs, &outputs, NULL)) < 0) + return ret; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + return ret; + return 0; +} + +static void display_frame(const AVFrame *frame, AVRational time_base) +{ + int x, y; + uint8_t *p0, *p; + int64_t delay; + + if (frame->pts != AV_NOPTS_VALUE) { + if (last_pts != AV_NOPTS_VALUE) { + /* sleep roughly the right amount of time; + * usleep is in microseconds, just like AV_TIME_BASE. */ + delay = av_rescale_q(frame->pts - last_pts, + time_base, AV_TIME_BASE_Q); + if (delay > 0 && delay < 1000000) + usleep(delay); + } + last_pts = frame->pts; + } + + /* Trivial ASCII grayscale display. */ + p0 = frame->data[0]; + puts("\033c"); + for (y = 0; y < frame->height; y++) { + p = p0; + for (x = 0; x < frame->width; x++) + putchar(" .-+#"[*(p++) / 52]); + putchar('\n'); + p0 += frame->linesize[0]; + } + fflush(stdout); +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet; + AVFrame *frame = av_frame_alloc(); + AVFrame *filt_frame = av_frame_alloc(); + int got_frame; + + if (!frame || !filt_frame) { + perror("Could not allocate frame"); + exit(1); + } + if (argc != 2) { + fprintf(stderr, "Usage: %s file\n", argv[0]); + exit(1); + } + + avcodec_register_all(); + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = init_filters(filter_descr)) < 0) + goto end; + + /* read all packets */ + while (1) { + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) + break; + + if (packet.stream_index == video_stream_index) { + avcodec_get_frame_defaults(frame); + got_frame = 0; + ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error decoding video\n"); + break; + } + + if (got_frame) { + frame->pts = av_frame_get_best_effort_timestamp(frame); + + /* push the decoded frame into the filtergraph */ + if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n"); + break; + } + + /* pull filtered frames from the filtergraph */ + while (1) { + ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + break; + if (ret < 0) + goto end; + display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base); + av_frame_unref(filt_frame); + } + } + } + av_free_packet(&packet); + } +end: + avfilter_graph_free(&filter_graph); + if (dec_ctx) + avcodec_close(dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + av_frame_free(&filt_frame); + + if (ret < 0 && ret != AVERROR_EOF) { + char buf[1024]; + av_strerror(ret, buf, sizeof(buf)); + fprintf(stderr, "Error occurred: %s\n", buf); + exit(1); + } + + exit(0); +} diff --git a/ffmpeg1/doc/examples/metadata.c b/ffmpeg1/doc/examples/metadata.c new file mode 100644 index 0000000..9c1bcd7 --- /dev/null +++ b/ffmpeg1/doc/examples/metadata.c @@ -0,0 +1,56 @@ +/* + * Copyright (c) 2011 Reinhard Tartler + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * Shows how the metadata API can be used in application programs. + * @example doc/examples/metadata.c + */ + +#include <stdio.h> + +#include <libavformat/avformat.h> +#include <libavutil/dict.h> + +int main (int argc, char **argv) +{ + AVFormatContext *fmt_ctx = NULL; + AVDictionaryEntry *tag = NULL; + int ret; + + if (argc != 2) { + printf("usage: %s <input_file>\n" + "example program to demonstrate the use of the libavformat metadata API.\n" + "\n", argv[0]); + return 1; + } + + av_register_all(); + if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL))) + return ret; + + while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) + printf("%s=%s\n", tag->key, tag->value); + + avformat_close_input(&fmt_ctx); + return 0; +} diff --git a/ffmpeg1/doc/examples/muxing.c b/ffmpeg1/doc/examples/muxing.c new file mode 100644 index 0000000..7305cc6 --- /dev/null +++ b/ffmpeg1/doc/examples/muxing.c @@ -0,0 +1,508 @@ +/* + * Copyright (c) 2003 Fabrice Bellard + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat API example. + * + * Output a media file in any supported libavformat format. + * The default codecs are used. + * @example doc/examples/muxing.c + */ + +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include <math.h> + +#include <libavutil/mathematics.h> +#include <libavformat/avformat.h> +#include <libswscale/swscale.h> + +/* 5 seconds stream duration */ +#define STREAM_DURATION 200.0 +#define STREAM_FRAME_RATE 25 /* 25 images/s */ +#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE)) +#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */ + +static int sws_flags = SWS_BICUBIC; + +/**************************************************************/ +/* audio output */ + +static float t, tincr, tincr2; +static int16_t *samples; +static int audio_input_frame_size; + +/* Add an output stream. */ +static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec, + enum AVCodecID codec_id) +{ + AVCodecContext *c; + AVStream *st; + + /* find the encoder */ + *codec = avcodec_find_encoder(codec_id); + if (!(*codec)) { + fprintf(stderr, "Could not find encoder for '%s'\n", + avcodec_get_name(codec_id)); + exit(1); + } + + st = avformat_new_stream(oc, *codec); + if (!st) { + fprintf(stderr, "Could not allocate stream\n"); + exit(1); + } + st->id = oc->nb_streams-1; + c = st->codec; + + switch ((*codec)->type) { + case AVMEDIA_TYPE_AUDIO: + st->id = 1; + c->sample_fmt = AV_SAMPLE_FMT_S16; + c->bit_rate = 64000; + c->sample_rate = 44100; + c->channels = 2; + break; + + case AVMEDIA_TYPE_VIDEO: + c->codec_id = codec_id; + + c->bit_rate = 400000; + /* Resolution must be a multiple of two. */ + c->width = 352; + c->height = 288; + /* timebase: This is the fundamental unit of time (in seconds) in terms + * of which frame timestamps are represented. For fixed-fps content, + * timebase should be 1/framerate and timestamp increments should be + * identical to 1. */ + c->time_base.den = STREAM_FRAME_RATE; + c->time_base.num = 1; + c->gop_size = 12; /* emit one intra frame every twelve frames at most */ + c->pix_fmt = STREAM_PIX_FMT; + if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) { + /* just for testing, we also add B frames */ + c->max_b_frames = 2; + } + if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) { + /* Needed to avoid using macroblocks in which some coeffs overflow. + * This does not happen with normal video, it just happens here as + * the motion of the chroma plane does not match the luma plane. */ + c->mb_decision = 2; + } + break; + + default: + break; + } + + /* Some formats want stream headers to be separate. */ + if (oc->oformat->flags & AVFMT_GLOBALHEADER) + c->flags |= CODEC_FLAG_GLOBAL_HEADER; + + return st; +} + +/**************************************************************/ +/* audio output */ + +static float t, tincr, tincr2; +static int16_t *samples; +static int audio_input_frame_size; + +static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st) +{ + AVCodecContext *c; + int ret; + + c = st->codec; + + /* open it */ + ret = avcodec_open2(c, codec, NULL); + if (ret < 0) { + fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret)); + exit(1); + } + + /* init signal generator */ + t = 0; + tincr = 2 * M_PI * 110.0 / c->sample_rate; + /* increment frequency by 110 Hz per second */ + tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; + + if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) + audio_input_frame_size = 10000; + else + audio_input_frame_size = c->frame_size; + samples = av_malloc(audio_input_frame_size * + av_get_bytes_per_sample(c->sample_fmt) * + c->channels); + if (!samples) { + fprintf(stderr, "Could not allocate audio samples buffer\n"); + exit(1); + } +} + +/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and + * 'nb_channels' channels. */ +static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) +{ + int j, i, v; + int16_t *q; + + q = samples; + for (j = 0; j < frame_size; j++) { + v = (int)(sin(t) * 10000); + for (i = 0; i < nb_channels; i++) + *q++ = v; + t += tincr; + tincr += tincr2; + } +} + +static void write_audio_frame(AVFormatContext *oc, AVStream *st) +{ + AVCodecContext *c; + AVPacket pkt = { 0 }; // data and size must be 0; + AVFrame *frame = avcodec_alloc_frame(); + int got_packet, ret; + + av_init_packet(&pkt); + c = st->codec; + + get_audio_frame(samples, audio_input_frame_size, c->channels); + frame->nb_samples = audio_input_frame_size; + avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (uint8_t *)samples, + audio_input_frame_size * + av_get_bytes_per_sample(c->sample_fmt) * + c->channels, 1); + + ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet); + if (ret < 0) { + fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret)); + exit(1); + } + + if (!got_packet) + return; + + pkt.stream_index = st->index; + + /* Write the compressed frame to the media file. */ + ret = av_interleaved_write_frame(oc, &pkt); + if (ret != 0) { + fprintf(stderr, "Error while writing audio frame: %s\n", + av_err2str(ret)); + exit(1); + } + avcodec_free_frame(&frame); +} + +static void close_audio(AVFormatContext *oc, AVStream *st) +{ + avcodec_close(st->codec); + + av_free(samples); +} + +/**************************************************************/ +/* video output */ + +static AVFrame *frame; +static AVPicture src_picture, dst_picture; +static int frame_count; + +static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st) +{ + int ret; + AVCodecContext *c = st->codec; + + /* open the codec */ + ret = avcodec_open2(c, codec, NULL); + if (ret < 0) { + fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret)); + exit(1); + } + + /* allocate and init a re-usable frame */ + frame = avcodec_alloc_frame(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + + /* Allocate the encoded raw picture. */ + ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height); + if (ret < 0) { + fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret)); + exit(1); + } + + /* If the output format is not YUV420P, then a temporary YUV420P + * picture is needed too. It is then converted to the required + * output format. */ + if (c->pix_fmt != AV_PIX_FMT_YUV420P) { + ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height); + if (ret < 0) { + fprintf(stderr, "Could not allocate temporary picture: %s\n", + av_err2str(ret)); + exit(1); + } + } + + /* copy data and linesize picture pointers to frame */ + *((AVPicture *)frame) = dst_picture; +} + +/* Prepare a dummy image. */ +static void fill_yuv_image(AVPicture *pict, int frame_index, + int width, int height) +{ + int x, y, i; + + i = frame_index; + + /* Y */ + for (y = 0; y < height; y++) + for (x = 0; x < width; x++) + pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3; + + /* Cb and Cr */ + for (y = 0; y < height / 2; y++) { + for (x = 0; x < width / 2; x++) { + pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2; + pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5; + } + } +} + +static void write_video_frame(AVFormatContext *oc, AVStream *st) +{ + int ret; + static struct SwsContext *sws_ctx; + AVCodecContext *c = st->codec; + + if (frame_count >= STREAM_NB_FRAMES) { + /* No more frames to compress. The codec has a latency of a few + * frames if using B-frames, so we get the last frames by + * passing the same picture again. */ + } else { + if (c->pix_fmt != AV_PIX_FMT_YUV420P) { + /* as we only generate a YUV420P picture, we must convert it + * to the codec pixel format if needed */ + if (!sws_ctx) { + sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P, + c->width, c->height, c->pix_fmt, + sws_flags, NULL, NULL, NULL); + if (!sws_ctx) { + fprintf(stderr, + "Could not initialize the conversion context\n"); + exit(1); + } + } + fill_yuv_image(&src_picture, frame_count, c->width, c->height); + sws_scale(sws_ctx, + (const uint8_t * const *)src_picture.data, src_picture.linesize, + 0, c->height, dst_picture.data, dst_picture.linesize); + } else { + fill_yuv_image(&dst_picture, frame_count, c->width, c->height); + } + } + + if (oc->oformat->flags & AVFMT_RAWPICTURE) { + /* Raw video case - directly store the picture in the packet */ + AVPacket pkt; + av_init_packet(&pkt); + + pkt.flags |= AV_PKT_FLAG_KEY; + pkt.stream_index = st->index; + pkt.data = dst_picture.data[0]; + pkt.size = sizeof(AVPicture); + + ret = av_interleaved_write_frame(oc, &pkt); + } else { + AVPacket pkt = { 0 }; + int got_packet; + av_init_packet(&pkt); + + /* encode the image */ + ret = avcodec_encode_video2(c, &pkt, frame, &got_packet); + if (ret < 0) { + fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret)); + exit(1); + } + /* If size is zero, it means the image was buffered. */ + + if (!ret && got_packet && pkt.size) { + pkt.stream_index = st->index; + + /* Write the compressed frame to the media file. */ + ret = av_interleaved_write_frame(oc, &pkt); + } else { + ret = 0; + } + } + if (ret != 0) { + fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret)); + exit(1); + } + frame_count++; +} + +static void close_video(AVFormatContext *oc, AVStream *st) +{ + avcodec_close(st->codec); + av_free(src_picture.data[0]); + av_free(dst_picture.data[0]); + av_free(frame); +} + +/**************************************************************/ +/* media file output */ + +int main(int argc, char **argv) +{ + const char *filename; + AVOutputFormat *fmt; + AVFormatContext *oc; + AVStream *audio_st, *video_st; + AVCodec *audio_codec, *video_codec; + double audio_pts, video_pts; + int ret; + + /* Initialize libavcodec, and register all codecs and formats. */ + av_register_all(); + + if (argc != 2) { + printf("usage: %s output_file\n" + "API example program to output a media file with libavformat.\n" + "This program generates a synthetic audio and video stream, encodes and\n" + "muxes them into a file named output_file.\n" + "The output format is automatically guessed according to the file extension.\n" + "Raw images can also be output by using '%%d' in the filename.\n" + "\n", argv[0]); + return 1; + } + + filename = argv[1]; + + /* allocate the output media context */ + avformat_alloc_output_context2(&oc, NULL, NULL, filename); + if (!oc) { + printf("Could not deduce output format from file extension: using MPEG.\n"); + avformat_alloc_output_context2(&oc, NULL, "mpeg", filename); + } + if (!oc) { + return 1; + } + fmt = oc->oformat; + + /* Add the audio and video streams using the default format codecs + * and initialize the codecs. */ + video_st = NULL; + audio_st = NULL; + + if (fmt->video_codec != AV_CODEC_ID_NONE) { + video_st = add_stream(oc, &video_codec, fmt->video_codec); + } + if (fmt->audio_codec != AV_CODEC_ID_NONE) { + audio_st = add_stream(oc, &audio_codec, fmt->audio_codec); + } + + /* Now that all the parameters are set, we can open the audio and + * video codecs and allocate the necessary encode buffers. */ + if (video_st) + open_video(oc, video_codec, video_st); + if (audio_st) + open_audio(oc, audio_codec, audio_st); + + av_dump_format(oc, 0, filename, 1); + + /* open the output file, if needed */ + if (!(fmt->flags & AVFMT_NOFILE)) { + ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE); + if (ret < 0) { + fprintf(stderr, "Could not open '%s': %s\n", filename, + av_err2str(ret)); + return 1; + } + } + + /* Write the stream header, if any. */ + ret = avformat_write_header(oc, NULL); + if (ret < 0) { + fprintf(stderr, "Error occurred when opening output file: %s\n", + av_err2str(ret)); + return 1; + } + + if (frame) + frame->pts = 0; + for (;;) { + /* Compute current audio and video time. */ + if (audio_st) + audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den; + else + audio_pts = 0.0; + + if (video_st) + video_pts = (double)video_st->pts.val * video_st->time_base.num / + video_st->time_base.den; + else + video_pts = 0.0; + + if ((!audio_st || audio_pts >= STREAM_DURATION) && + (!video_st || video_pts >= STREAM_DURATION)) + break; + + /* write interleaved audio and video frames */ + if (!video_st || (video_st && audio_st && audio_pts < video_pts)) { + write_audio_frame(oc, audio_st); + } else { + write_video_frame(oc, video_st); + frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base); + } + } + + /* Write the trailer, if any. The trailer must be written before you + * close the CodecContexts open when you wrote the header; otherwise + * av_write_trailer() may try to use memory that was freed on + * av_codec_close(). */ + av_write_trailer(oc); + + /* Close each codec. */ + if (video_st) + close_video(oc, video_st); + if (audio_st) + close_audio(oc, audio_st); + + if (!(fmt->flags & AVFMT_NOFILE)) + /* Close the output file. */ + avio_close(oc->pb); + + /* free the stream */ + avformat_free_context(oc); + + return 0; +} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libavcodec.pc b/ffmpeg1/doc/examples/pc-uninstalled/libavcodec.pc new file mode 100644 index 0000000..787d687 --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libavcodec.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libavcodec +includedir=${pcfiledir}/../../.. + +Name: libavcodec +Description: FFmpeg codec library +Version: 55.1.100 +Requires: libavutil = 52.22.100 +Conflicts: +Libs: -L${libdir} -lavcodec +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libavdevice.pc b/ffmpeg1/doc/examples/pc-uninstalled/libavdevice.pc new file mode 100644 index 0000000..89ef046 --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libavdevice.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libavdevice +includedir=${pcfiledir}/../../.. + +Name: libavdevice +Description: FFmpeg device handling library +Version: 55.0.100 +Requires: libavfilter = 3.48.100, libavformat = 55.0.100 +Conflicts: +Libs: -L${libdir} -lavdevice +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libavfilter.pc b/ffmpeg1/doc/examples/pc-uninstalled/libavfilter.pc new file mode 100644 index 0000000..aacaf0a --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libavfilter.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libavfilter +includedir=${pcfiledir}/../../.. + +Name: libavfilter +Description: FFmpeg audio/video filtering library +Version: 3.48.100 +Requires: libpostproc = 52.2.100, libswresample = 0.17.102, libswscale = 2.2.100, libavformat = 55.0.100, libavcodec = 55.1.100, libavutil = 52.22.100 +Conflicts: +Libs: -L${libdir} -lavfilter +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libavformat.pc b/ffmpeg1/doc/examples/pc-uninstalled/libavformat.pc new file mode 100644 index 0000000..8f27151 --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libavformat.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libavformat +includedir=${pcfiledir}/../../.. + +Name: libavformat +Description: FFmpeg container format library +Version: 55.0.100 +Requires: libavcodec = 55.1.100 +Conflicts: +Libs: -L${libdir} -lavformat +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libavutil.pc b/ffmpeg1/doc/examples/pc-uninstalled/libavutil.pc new file mode 100644 index 0000000..8a95064 --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libavutil.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libavutil +includedir=${pcfiledir}/../../.. + +Name: libavutil +Description: FFmpeg utility library +Version: 52.22.100 +Requires: +Conflicts: +Libs: -L${libdir} -lavutil +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libpostproc.pc b/ffmpeg1/doc/examples/pc-uninstalled/libpostproc.pc new file mode 100644 index 0000000..5e87c13 --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libpostproc.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libpostproc +includedir=${pcfiledir}/../../.. + +Name: libpostproc +Description: FFmpeg postprocessing library +Version: 52.2.100 +Requires: libavutil = 52.22.100 +Conflicts: +Libs: -L${libdir} -lpostproc +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libswresample.pc b/ffmpeg1/doc/examples/pc-uninstalled/libswresample.pc new file mode 100644 index 0000000..873f39d --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libswresample.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libswresample +includedir=${pcfiledir}/../../.. + +Name: libswresample +Description: FFmpeg audio resampling library +Version: 0.17.102 +Requires: libavutil = 52.22.100 +Conflicts: +Libs: -L${libdir} -lswresample +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/pc-uninstalled/libswscale.pc b/ffmpeg1/doc/examples/pc-uninstalled/libswscale.pc new file mode 100644 index 0000000..764a10c --- /dev/null +++ b/ffmpeg1/doc/examples/pc-uninstalled/libswscale.pc @@ -0,0 +1,12 @@ +prefix= +exec_prefix= +libdir=${pcfiledir}/../../../libswscale +includedir=${pcfiledir}/../../.. + +Name: libswscale +Description: FFmpeg image rescaling library +Version: 2.2.100 +Requires: libavutil = 52.22.100 +Conflicts: +Libs: -L${libdir} -lswscale +Cflags: -I${includedir} diff --git a/ffmpeg1/doc/examples/resampling_audio.c b/ffmpeg1/doc/examples/resampling_audio.c new file mode 100644 index 0000000..dd128e8 --- /dev/null +++ b/ffmpeg1/doc/examples/resampling_audio.c @@ -0,0 +1,223 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @example doc/examples/resampling_audio.c + * libswresample API use example. + */ + +#include <libavutil/opt.h> +#include <libavutil/channel_layout.h> +#include <libavutil/samplefmt.h> +#include <libswresample/swresample.h> + +static int get_format_from_sample_fmt(const char **fmt, + enum AVSampleFormat sample_fmt) +{ + int i; + struct sample_fmt_entry { + enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; + } sample_fmt_entries[] = { + { AV_SAMPLE_FMT_U8, "u8", "u8" }, + { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, + { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, + { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, + { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, + }; + *fmt = NULL; + + for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { + struct sample_fmt_entry *entry = &sample_fmt_entries[i]; + if (sample_fmt == entry->sample_fmt) { + *fmt = AV_NE(entry->fmt_be, entry->fmt_le); + return 0; + } + } + + fprintf(stderr, + "Sample format %s not supported as output format\n", + av_get_sample_fmt_name(sample_fmt)); + return AVERROR(EINVAL); +} + +/** + * Fill dst buffer with nb_samples, generated starting from t. + */ +void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t) +{ + int i, j; + double tincr = 1.0 / sample_rate, *dstp = dst; + const double c = 2 * M_PI * 440.0; + + /* generate sin tone with 440Hz frequency and duplicated channels */ + for (i = 0; i < nb_samples; i++) { + *dstp = sin(c * *t); + for (j = 1; j < nb_channels; j++) + dstp[j] = dstp[0]; + dstp += nb_channels; + *t += tincr; + } +} + +int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels, + int nb_samples, enum AVSampleFormat sample_fmt, int align) +{ + int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1; + + *data = av_malloc(sizeof(*data) * nb_planes); + if (!*data) + return AVERROR(ENOMEM); + return av_samples_alloc(*data, linesize, nb_channels, + nb_samples, sample_fmt, align); +} + +int main(int argc, char **argv) +{ + int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND; + int src_rate = 48000, dst_rate = 44100; + uint8_t **src_data = NULL, **dst_data = NULL; + int src_nb_channels = 0, dst_nb_channels = 0; + int src_linesize, dst_linesize; + int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples; + enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16; + const char *dst_filename = NULL; + FILE *dst_file; + int dst_bufsize; + const char *fmt; + struct SwrContext *swr_ctx; + double t; + int ret; + + if (argc != 2) { + fprintf(stderr, "Usage: %s output_file\n" + "API example program to show how to resample an audio stream with libswresample.\n" + "This program generates a series of audio frames, resamples them to a specified " + "output format and rate and saves them to an output file named output_file.\n", + argv[0]); + exit(1); + } + dst_filename = argv[1]; + + dst_file = fopen(dst_filename, "wb"); + if (!dst_file) { + fprintf(stderr, "Could not open destination file %s\n", dst_filename); + exit(1); + } + + /* create resampler context */ + swr_ctx = swr_alloc(); + if (!swr_ctx) { + fprintf(stderr, "Could not allocate resampler context\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + /* set options */ + av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0); + av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0); + av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0); + + av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0); + av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0); + av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0); + + /* initialize the resampling context */ + if ((ret = swr_init(swr_ctx)) < 0) { + fprintf(stderr, "Failed to initialize the resampling context\n"); + goto end; + } + + /* allocate source and destination samples buffers */ + + src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout); + ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels, + src_nb_samples, src_sample_fmt, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate source samples\n"); + goto end; + } + + /* compute the number of converted samples: buffering is avoided + * ensuring that the output buffer will contain at least all the + * converted input samples */ + max_dst_nb_samples = dst_nb_samples = + av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); + + /* buffer is going to be directly written to a rawaudio file, no alignment */ + dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout); + ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels, + dst_nb_samples, dst_sample_fmt, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate destination samples\n"); + goto end; + } + + t = 0; + do { + /* generate synthetic audio */ + fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t); + + /* compute destination number of samples */ + dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) + + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); + if (dst_nb_samples > max_dst_nb_samples) { + av_free(dst_data[0]); + ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels, + dst_nb_samples, dst_sample_fmt, 1); + if (ret < 0) + break; + max_dst_nb_samples = dst_nb_samples; + } + + /* convert to destination format */ + ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples); + if (ret < 0) { + fprintf(stderr, "Error while converting\n"); + goto end; + } + dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, + ret, dst_sample_fmt, 1); + printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret); + fwrite(dst_data[0], 1, dst_bufsize, dst_file); + } while (t < 10); + + if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) + goto end; + fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n" + "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n", + fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename); + +end: + if (dst_file) + fclose(dst_file); + + if (src_data) + av_freep(&src_data[0]); + av_freep(&src_data); + + if (dst_data) + av_freep(&dst_data[0]); + av_freep(&dst_data); + + swr_free(&swr_ctx); + return ret < 0; +} diff --git a/ffmpeg1/doc/examples/scaling_video.c b/ffmpeg1/doc/examples/scaling_video.c new file mode 100644 index 0000000..be2c510 --- /dev/null +++ b/ffmpeg1/doc/examples/scaling_video.c @@ -0,0 +1,141 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libswscale API use example. + * @example doc/examples/scaling_video.c + */ + +#include <libavutil/imgutils.h> +#include <libavutil/parseutils.h> +#include <libswscale/swscale.h> + +static void fill_yuv_image(uint8_t *data[4], int linesize[4], + int width, int height, int frame_index) +{ + int x, y; + + /* Y */ + for (y = 0; y < height; y++) + for (x = 0; x < width; x++) + data[0][y * linesize[0] + x] = x + y + frame_index * 3; + + /* Cb and Cr */ + for (y = 0; y < height / 2; y++) { + for (x = 0; x < width / 2; x++) { + data[1][y * linesize[1] + x] = 128 + y + frame_index * 2; + data[2][y * linesize[2] + x] = 64 + x + frame_index * 5; + } + } +} + +int main(int argc, char **argv) +{ + uint8_t *src_data[4], *dst_data[4]; + int src_linesize[4], dst_linesize[4]; + int src_w = 320, src_h = 240, dst_w, dst_h; + enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24; + const char *dst_size = NULL; + const char *dst_filename = NULL; + FILE *dst_file; + int dst_bufsize; + struct SwsContext *sws_ctx; + int i, ret; + + if (argc != 3) { + fprintf(stderr, "Usage: %s output_file output_size\n" + "API example program to show how to scale an image with libswscale.\n" + "This program generates a series of pictures, rescales them to the given " + "output_size and saves them to an output file named output_file\n." + "\n", argv[0]); + exit(1); + } + dst_filename = argv[1]; + dst_size = argv[2]; + + if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) { + fprintf(stderr, + "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n", + dst_size); + exit(1); + } + + dst_file = fopen(dst_filename, "wb"); + if (!dst_file) { + fprintf(stderr, "Could not open destination file %s\n", dst_filename); + exit(1); + } + + /* create scaling context */ + sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt, + dst_w, dst_h, dst_pix_fmt, + SWS_BILINEAR, NULL, NULL, NULL); + if (!sws_ctx) { + fprintf(stderr, + "Impossible to create scale context for the conversion " + "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n", + av_get_pix_fmt_name(src_pix_fmt), src_w, src_h, + av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h); + ret = AVERROR(EINVAL); + goto end; + } + + /* allocate source and destination image buffers */ + if ((ret = av_image_alloc(src_data, src_linesize, + src_w, src_h, src_pix_fmt, 16)) < 0) { + fprintf(stderr, "Could not allocate source image\n"); + goto end; + } + + /* buffer is going to be written to rawvideo file, no alignment */ + if ((ret = av_image_alloc(dst_data, dst_linesize, + dst_w, dst_h, dst_pix_fmt, 1)) < 0) { + fprintf(stderr, "Could not allocate destination image\n"); + goto end; + } + dst_bufsize = ret; + + for (i = 0; i < 100; i++) { + /* generate synthetic video */ + fill_yuv_image(src_data, src_linesize, src_w, src_h, i); + + /* convert to destination format */ + sws_scale(sws_ctx, (const uint8_t * const*)src_data, + src_linesize, 0, src_h, dst_data, dst_linesize); + + /* write scaled image to file */ + fwrite(dst_data[0], 1, dst_bufsize, dst_file); + } + + fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n" + "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", + av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename); + +end: + if (dst_file) + fclose(dst_file); + av_freep(&src_data[0]); + av_freep(&dst_data[0]); + sws_freeContext(sws_ctx); + return ret < 0; +} diff --git a/ffmpeg1/doc/faq.texi b/ffmpeg1/doc/faq.texi new file mode 100644 index 0000000..ebf21f5 --- /dev/null +++ b/ffmpeg1/doc/faq.texi @@ -0,0 +1,558 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg FAQ +@titlepage +@center @titlefont{FFmpeg FAQ} +@end titlepage + +@top + +@contents + +@chapter General Questions + +@section Why doesn't FFmpeg support feature [xyz]? + +Because no one has taken on that task yet. FFmpeg development is +driven by the tasks that are important to the individual developers. +If there is a feature that is important to you, the best way to get +it implemented is to undertake the task yourself or sponsor a developer. + +@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it? + +No. Windows DLLs are not portable, bloated and often slow. +Moreover FFmpeg strives to support all codecs natively. +A DLL loader is not conducive to that goal. + +@section I cannot read this file although this format seems to be supported by ffmpeg. + +Even if ffmpeg can read the container format, it may not support all its +codecs. Please consult the supported codec list in the ffmpeg +documentation. + +@section Which codecs are supported by Windows? + +Windows does not support standard formats like MPEG very well, unless you +install some additional codecs. + +The following list of video codecs should work on most Windows systems: +@table @option +@item msmpeg4v2 +.avi/.asf +@item msmpeg4 +.asf only +@item wmv1 +.asf only +@item wmv2 +.asf only +@item mpeg4 +Only if you have some MPEG-4 codec like ffdshow or Xvid installed. +@item mpeg1video +.mpg only +@end table +Note, ASF files often have .wmv or .wma extensions in Windows. It should also +be mentioned that Microsoft claims a patent on the ASF format, and may sue +or threaten users who create ASF files with non-Microsoft software. It is +strongly advised to avoid ASF where possible. + +The following list of audio codecs should work on most Windows systems: +@table @option +@item adpcm_ima_wav +@item adpcm_ms +@item pcm_s16le +always +@item libmp3lame +If some MP3 codec like LAME is installed. +@end table + + +@chapter Compilation + +@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'} + +This is a bug in gcc. Do not report it to us. Instead, please report it to +the gcc developers. Note that we will not add workarounds for gcc bugs. + +Also note that (some of) the gcc developers believe this is not a bug or +not a bug they should fix: +@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}. +Then again, some of them do not know the difference between an undecidable +problem and an NP-hard problem... + +@section I have installed this library with my distro's package manager. Why does @command{configure} not see it? + +Distributions usually split libraries in several packages. The main package +contains the files necessary to run programs using the library. The +development package contains the files necessary to build programs using the +library. Sometimes, docs and/or data are in a separate package too. + +To build FFmpeg, you need to install the development package. It is usually +called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the +build is finished, but be sure to keep the main package. + +@chapter Usage + +@section ffmpeg does not work; what is wrong? + +Try a @code{make distclean} in the ffmpeg source directory before the build. +If this does not help see +(@url{http://ffmpeg.org/bugreports.html}). + +@section How do I encode single pictures into movies? + +First, rename your pictures to follow a numerical sequence. +For example, img1.jpg, img2.jpg, img3.jpg,... +Then you may run: + +@example + ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg +@end example + +Notice that @samp{%d} is replaced by the image number. + +@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc. + +Use the @option{-start_number} option to declare a starting number for +the sequence. This is useful if your sequence does not start with +@file{img001.jpg} but is still in a numerical order. The following +example will start with @file{img100.jpg}: + +@example + ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg +@end example + +If you have large number of pictures to rename, you can use the +following command to ease the burden. The command, using the bourne +shell syntax, symbolically links all files in the current directory +that match @code{*jpg} to the @file{/tmp} directory in the sequence of +@file{img001.jpg}, @file{img002.jpg} and so on. + +@example + x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done +@end example + +If you want to sequence them by oldest modified first, substitute +@code{$(ls -r -t *jpg)} in place of @code{*jpg}. + +Then run: + +@example + ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg +@end example + +The same logic is used for any image format that ffmpeg reads. + +You can also use @command{cat} to pipe images to ffmpeg: + +@example + cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg +@end example + +@section How do I encode movie to single pictures? + +Use: + +@example + ffmpeg -i movie.mpg movie%d.jpg +@end example + +The @file{movie.mpg} used as input will be converted to +@file{movie1.jpg}, @file{movie2.jpg}, etc... + +Instead of relying on file format self-recognition, you may also use +@table @option +@item -c:v ppm +@item -c:v png +@item -c:v mjpeg +@end table +to force the encoding. + +Applying that to the previous example: +@example + ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg +@end example + +Beware that there is no "jpeg" codec. Use "mjpeg" instead. + +@section Why do I see a slight quality degradation with multithreaded MPEG* encoding? + +For multithreaded MPEG* encoding, the encoded slices must be independent, +otherwise thread n would practically have to wait for n-1 to finish, so it's +quite logical that there is a small reduction of quality. This is not a bug. + +@section How can I read from the standard input or write to the standard output? + +Use @file{-} as file name. + +@section -f jpeg doesn't work. + +Try '-f image2 test%d.jpg'. + +@section Why can I not change the frame rate? + +Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates. +Choose a different codec with the -c:v command line option. + +@section How do I encode Xvid or DivX video with ffmpeg? + +Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4 +standard (note that there are many other coding formats that use this +same standard). Thus, use '-c:v mpeg4' to encode in these formats. The +default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want +a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will +force the fourcc 'xvid' to be stored as the video fourcc rather than the +default. + +@section Which are good parameters for encoding high quality MPEG-4? + +'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2', +things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'. + +@section Which are good parameters for encoding high quality MPEG-1/MPEG-2? + +'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2' +but beware the '-g 100' might cause problems with some decoders. +Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd. + +@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong? + +You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced +material, and try '-top 0/1' if the result looks really messed-up. + +@section How can I read DirectShow files? + +If you have built FFmpeg with @code{./configure --enable-avisynth} +(only possible on MinGW/Cygwin platforms), +then you may use any file that DirectShow can read as input. + +Just create an "input.avs" text file with this single line ... +@example + DirectShowSource("C:\path to your file\yourfile.asf") +@end example +... and then feed that text file to ffmpeg: +@example + ffmpeg -i input.avs +@end example + +For ANY other help on Avisynth, please visit the +@uref{http://www.avisynth.org/, Avisynth homepage}. + +@section How can I join video files? + +To "join" video files is quite ambiguous. The following list explains the +different kinds of "joining" and points out how those are addressed in +FFmpeg. To join video files may mean: + +@itemize + +@item +To put them one after the other: this is called to @emph{concatenate} them +(in short: concat) and is addressed +@ref{How can I concatenate video files, in this very faq}. + +@item +To put them together in the same file, to let the user choose between the +different versions (example: different audio languages): this is called to +@emph{multiplex} them together (in short: mux), and is done by simply +invoking ffmpeg with several @option{-i} options. + +@item +For audio, to put all channels together in a single stream (example: two +mono streams into one stereo stream): this is sometimes called to +@emph{merge} them, and can be done using the +@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter. + +@item +For audio, to play one on top of the other: this is called to @emph{mix} +them, and can be done by first merging them into a single stream and then +using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix +the channels at will. + +@item +For video, to display both together, side by side or one on top of a part of +the other; it can be done using the +@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter. + +@end itemize + +@anchor{How can I concatenate video files} +@section How can I concatenate video files? + +There are several solutions, depending on the exact circumstances. + +@subsection Concatenating using the concat @emph{filter} + +FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat, +@code{concat}} filter designed specifically for that, with examples in the +documentation. This operation is recommended if you need to re-encode. + +@subsection Concatenating using the concat @emph{demuxer} + +FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat, +@code{concat}} demuxer which you can use when you want to avoid a re-encode and +your format doesn't support file level concatenation. + +@subsection Concatenating using the concat @emph{protocol} (file level) + +FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat, +@code{concat}} protocol designed specifically for that, with examples in the +documentation. + +A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate +video by merely concatenating the files containing them. + +Hence you may concatenate your multimedia files by first transcoding them to +these privileged formats, then using the humble @code{cat} command (or the +equally humble @code{copy} under Windows), and finally transcoding back to your +format of choice. + +@example +ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg +ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg +cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg +ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi +@end example + +Additionally, you can use the @code{concat} protocol instead of @code{cat} or +@code{copy} which will avoid creation of a potentially huge intermediate file. + +@example +ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg +ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg +ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg +ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi +@end example + +Note that you may need to escape the character "|" which is special for many +shells. + +Another option is usage of named pipes, should your platform support it: + +@example +mkfifo intermediate1.mpg +mkfifo intermediate2.mpg +ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null & +ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null & +cat intermediate1.mpg intermediate2.mpg |\ +ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi +@end example + +@subsection Concatenating using raw audio and video + +Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also +allow concatenation, and the transcoding step is almost lossless. +When using multiple yuv4mpegpipe(s), the first line needs to be discarded +from all but the first stream. This can be accomplished by piping through +@code{tail} as seen below. Note that when piping through @code{tail} you +must use command grouping, @code{@{ ;@}}, to background properly. + +For example, let's say we want to concatenate two FLV files into an +output.flv file: + +@example +mkfifo temp1.a +mkfifo temp1.v +mkfifo temp2.a +mkfifo temp2.v +mkfifo all.a +mkfifo all.v +ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & +ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & +ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & +@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & +cat temp1.a temp2.a > all.a & +cat temp1.v temp2.v > all.v & +ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ + -f yuv4mpegpipe -i all.v \ + -y output.flv +rm temp[12].[av] all.[av] +@end example + +@section -profile option fails when encoding H.264 video with AAC audio + +@command{ffmpeg} prints an error like + +@example +Undefined constant or missing '(' in 'baseline' +Unable to parse option value "baseline" +Error setting option profile to value baseline. +@end example + +Short answer: write @option{-profile:v} instead of @option{-profile}. + +Long answer: this happens because the @option{-profile} option can apply to both +video and audio. Specifically the AAC encoder also defines some profiles, none +of which are named @var{baseline}. + +The solution is to apply the @option{-profile} option to the video stream only +by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}. +Appending @code{:v} to it will do exactly that. + +@section Using @option{-f lavfi}, audio becomes mono for no apparent reason. + +Use @option{-dumpgraph -} to find out exactly where the channel layout is +lost. + +Most likely, it is through @code{auto-inserted aconvert}. Try to understand +why the converting filter was needed at that place. + +Just before the output is a likely place, as @option{-f lavfi} currently +only support packed S16. + +Then insert the correct @code{aconvert} explicitly in the filter graph, +specifying the exact format. + +@example +aconvert=s16:stereo:packed +@end example + +@section Why does FFmpeg not see the subtitles in my VOB file? + +VOB and a few other formats do not have a global header that describes +everything present in the file. Instead, applications are supposed to scan +the file to see what it contains. Since VOB files are frequently large, only +the beginning is scanned. If the subtitles happen only later in the file, +they will not be initally detected. + +Some applications, including the @code{ffmpeg} command-line tool, can only +work with streams that were detected during the initial scan; streams that +are detected later are ignored. + +The size of the initial scan is controlled by two options: @code{probesize} +(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For +the subtitle stream to be detected, both values must be large enough. + +@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead? + +The @option{-sameq} option meant "same quantizer", and made sense only in a +very limited set of cases. Unfortunately, a lot of people mistook it for +"same quality" and used it in places where it did not make sense: it had +roughly the expected visible effect, but achieved it in a very inefficient +way. + +Each encoder has its own set of options to set the quality-vs-size balance, +use the options for the encoder you are using to set the quality level to a +point acceptable for your tastes. The most common options to do that are +@option{-qscale} and @option{-qmax}, but you should peruse the documentation +of the encoder you chose. + +@chapter Development + +@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat? + +Yes. Check the @file{doc/examples} directory in the source +repository, also available online at: +@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}. + +Examples are also installed by default, usually in +@code{$PREFIX/share/ffmpeg/examples}. + +Also you may read the Developers Guide of the FFmpeg documentation. Alternatively, +examine the source code for one of the many open source projects that +already incorporate FFmpeg at (@url{projects.html}). + +@section Can you support my C compiler XXX? + +It depends. If your compiler is C99-compliant, then patches to support +it are likely to be welcome if they do not pollute the source code +with @code{#ifdef}s related to the compiler. + +@section Is Microsoft Visual C++ supported? + +Yes. Please see the @uref{platform.html, Microsoft Visual C++} +section in the FFmpeg documentation. + +@section Can you add automake, libtool or autoconf support? + +No. These tools are too bloated and they complicate the build. + +@section Why not rewrite FFmpeg in object-oriented C++? + +FFmpeg is already organized in a highly modular manner and does not need to +be rewritten in a formal object language. Further, many of the developers +favor straight C; it works for them. For more arguments on this matter, +read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}. + +@section Why are the ffmpeg programs devoid of debugging symbols? + +The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug +information. Those binaries are stripped to create ffmpeg, ffplay, etc. If +you need the debug information, use the *_g versions. + +@section I do not like the LGPL, can I contribute code under the GPL instead? + +Yes, as long as the code is optional and can easily and cleanly be placed +under #if CONFIG_GPL without breaking anything. So, for example, a new codec +or filter would be OK under GPL while a bug fix to LGPL code would not. + +@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves. + +FFmpeg builds static libraries by default. In static libraries, dependencies +are not handled. That has two consequences. First, you must specify the +libraries in dependency order: @code{-lavdevice} must come before +@code{-lavformat}, @code{-lavutil} must come after everything else, etc. +Second, external libraries that are used in FFmpeg have to be specified too. + +An easy way to get the full list of required libraries in dependency order +is to use @code{pkg-config}. + +@example + c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec) +@end example + +See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for +more details. + +@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available. + +FFmpeg is a pure C project, so to use the libraries within your C++ application +you need to explicitly state that you are using a C library. You can do this by +encompassing your FFmpeg includes using @code{extern "C"}. + +See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3} + +@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope + +FFmpeg is a pure C project using C99 math features, in order to enable C++ +to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS + +@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat? + +You have to create a custom AVIOContext using @code{avio_alloc_context}, +see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources. + +@section Where can I find libav* headers for Pascal/Delphi? + +see @url{http://www.iversenit.dk/dev/ffmpeg-headers/} + +@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm? + +see @url{http://www.ffmpeg.org/~michael/} + +@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec? + +Even if peculiar since it is network oriented, RTP is a container like any +other. You have to @emph{demux} RTP before feeding the payload to libavcodec. +In this specific case please look at RFC 4629 to see how it should be done. + +@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate. + +r_frame_rate is NOT the average frame rate, it is the smallest frame rate +that can accurately represent all timestamps. So no, it is not +wrong if it is larger than the average! +For example, if you have mixed 25 and 30 fps content, then r_frame_rate +will be 150. + +@section Why is @code{make fate} not running all tests? + +Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable +or @code{FATE_SAMPLES} environment variable or the @code{--samples} +@command{configure} option is set to the right path. + +@section Why is @code{make fate} not finding the samples? + +Do you happen to have a @code{~} character in the samples path to indicate a +home directory? The value is used in ways where the shell cannot expand it, +causing FATE to not find files. Just replace @code{~} by the full path. + +@bye diff --git a/ffmpeg1/doc/fate.texi b/ffmpeg1/doc/fate.texi new file mode 100644 index 0000000..4c2ba4d --- /dev/null +++ b/ffmpeg1/doc/fate.texi @@ -0,0 +1,194 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Automated Testing Environment +@titlepage +@center @titlefont{FFmpeg Automated Testing Environment} +@end titlepage + +@node Top +@top + +@contents + +@chapter Introduction + + FATE is an extended regression suite on the client-side and a means +for results aggregation and presentation on the server-side. + + The first part of this document explains how you can use FATE from +your FFmpeg source directory to test your ffmpeg binary. The second +part describes how you can run FATE to submit the results to FFmpeg's +FATE server. + + In any way you can have a look at the publicly viewable FATE results +by visiting this website: + + @url{http://fate.ffmpeg.org/} + + This is especially recommended for all people contributing source +code to FFmpeg, as it can be seen if some test on some platform broke +with there recent contribution. This usually happens on the platforms +the developers could not test on. + + The second part of this document describes how you can run FATE to +submit your results to FFmpeg's FATE server. If you want to submit your +results be sure to check that your combination of CPU, OS and compiler +is not already listed on the above mentioned website. + + In the third part you can find a comprehensive listing of FATE makefile +targets and variables. + + +@chapter Using FATE from your FFmpeg source directory + + If you want to run FATE on your machine you need to have the samples +in place. You can get the samples via the build target fate-rsync. +Use this command from the top-level source directory: + +@example +make fate-rsync SAMPLES=fate-suite/ +make fate SAMPLES=fate-suite/ +@end example + + The above commands set the samples location by passing a makefile +variable via command line. It is also possible to set the samples +location at source configuration time by invoking configure with +`--samples=<path to the samples directory>'. Afterwards you can +invoke the makefile targets without setting the SAMPLES makefile +variable. This is illustrated by the following commands: + +@example +./configure --samples=fate-suite/ +make fate-rsync +make fate +@end example + + Yet another way to tell FATE about the location of the sample +directory is by making sure the environment variable FATE_SAMPLES +contains the path to your samples directory. This can be achieved +by e.g. putting that variable in your shell profile or by setting +it in your interactive session. + +@example +FATE_SAMPLES=fate-suite/ make fate +@end example + +@float NOTE +Do not put a '~' character in the samples path to indicate a home +directory. Because of shell nuances, this will cause FATE to fail. +@end float + +To use a custom wrapper to run the test, pass @option{--target-exec} to +@command{configure} or set the @var{TARGET_EXEC} Make variable. + + +@chapter Submitting the results to the FFmpeg result aggregation server + + To submit your results to the server you should run fate through the +shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs +to be invoked with a configuration file as its first argument. + +@example +tests/fate.sh /path/to/fate_config +@end example + + A configuration file template with comments describing the individual +configuration variables can be found at @file{doc/fate_config.sh.template}. + +@ifhtml + The mentioned configuration template is also available here: +@verbatiminclude fate_config.sh.template +@end ifhtml + + Create a configuration that suits your needs, based on the configuration +template. The `slot' configuration variable can be any string that is not +yet used, but it is suggested that you name it adhering to the following +pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file +itself will be sourced in a shell script, therefore all shell features may +be used. This enables you to setup the environment as you need it for your +build. + + For your first test runs the `fate_recv' variable should be empty or +commented out. This will run everything as normal except that it will omit +the submission of the results to the server. The following files should be +present in $workdir as specified in the configuration file: + +@itemize + @item configure.log + @item compile.log + @item test.log + @item report + @item version +@end itemize + + When you have everything working properly you can create an SSH key pair +and send the public key to the FATE server administrator who can be contacted +at the email address @email{fate-admin@@ffmpeg.org}. + + Configure your SSH client to use public key authentication with that key +when connecting to the FATE server. Also do not forget to check the identity +of the server and to accept its host key. This can usually be achieved by +running your SSH client manually and killing it after you accepted the key. +The FATE server's fingerprint is: + + b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92 + + If you have problems connecting to the FATE server, it may help to try out +the @command{ssh} command with one or more @option{-v} options. You should +get detailed output concerning your SSH configuration and the authentication +process. + + The only thing left is to automate the execution of the fate.sh script and +the synchronisation of the samples directory. + + +@chapter FATE makefile targets and variables + +@section Makefile targets + +@table @option +@item fate-rsync + Download/synchronize sample files to the configured samples directory. + +@item fate-list + Will list all fate/regression test targets. + +@item fate + Run the FATE test suite (requires the fate-suite dataset). +@end table + +@section Makefile variables + +@table @option +@item V + Verbosity level, can be set to 0, 1 or 2. + @itemize + @item 0: show just the test arguments + @item 1: show just the command used in the test + @item 2: show everything + @end itemize + +@item SAMPLES + Specify or override the path to the FATE samples at make time, it has a + meaning only while running the regression tests. + +@item THREADS + Specify how many threads to use while running regression tests, it is + quite useful to detect thread-related regressions. +@item THREAD_TYPE + Specify which threading strategy test, either @var{slice} or @var{frame}, + by default @var{slice+frame} +@item CPUFLAGS + Specify CPU flags. +@item TARGET_EXEC + Specify or override the wrapper used to run the tests. + The @var{TARGET_EXEC} option provides a way to run FATE wrapped in + @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets + through @command{ssh}. +@end table + +@section Examples + +@example +make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate +@end example diff --git a/ffmpeg1/doc/fate_config.sh.template b/ffmpeg1/doc/fate_config.sh.template new file mode 100644 index 0000000..f7bd625 --- /dev/null +++ b/ffmpeg1/doc/fate_config.sh.template @@ -0,0 +1,25 @@ +slot= # some unique identifier +repo=git://source.ffmpeg.org/ffmpeg.git # the source repository +samples= # path to samples directory +workdir= # directory in which to do all the work +#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report +comment= # optional description + +# the following are optional and map to configure options +arch= +cpu= +cross_prefix= +cc= +target_os= +sysroot= +target_exec= +target_path= +extra_cflags= +extra_ldflags= +extra_libs= +extra_conf= # extra configure options not covered above + +#make= # name of GNU make if not 'make' +makeopts= # extra options passed to 'make' +#tar= # command to create a tar archive from its arguments on stdout, + # defaults to 'tar c' diff --git a/ffmpeg1/doc/ffmpeg-bitstream-filters.texi b/ffmpeg1/doc/ffmpeg-bitstream-filters.texi new file mode 100644 index 0000000..e33e005 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-bitstream-filters.texi @@ -0,0 +1,45 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Bitstream Filters Documentation +@titlepage +@center @titlefont{FFmpeg Bitstream Filters Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the bitstream filters provided by the +libavcodec library. + +A bitstream filter operates on the encoded stream data, and performs +bitstream level modifications without performing decoding. + +@c man end DESCRIPTION + +@include bitstream_filters.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-bitstream-filters +@settitle FFmpeg bitstream filters + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-codecs.texi b/ffmpeg1/doc/ffmpeg-codecs.texi new file mode 100644 index 0000000..8f807c1 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-codecs.texi @@ -0,0 +1,1110 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Codecs Documentation +@titlepage +@center @titlefont{FFmpeg Codecs Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the codecs (decoders and encoders) provided by +the libavcodec library. + +@c man end DESCRIPTION + +@chapter Codec Options +@c man begin CODEC OPTIONS + +libavcodec provides some generic global options, which can be set on +all the encoders and decoders. In addition each codec may support +so-called private options, which are specific for a given codec. + +Sometimes, a global option may only affect a specific kind of codec, +and may be unsensical or ignored by another, so you need to be aware +of the meaning of the specified options. Also some options are +meant only for decoding or encoding. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the +@code{AVCodecContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +The list of supported options follow: + +@table @option +@item b @var{integer} (@emph{encoding,audio,video}) +Set bitrate in bits/s. Default value is 200K. + +@item ab @var{integer} (@emph{encoding,audio}) +Set audio bitrate (in bits/s). Default value is 128K. + +@item bt @var{integer} (@emph{encoding,video}) +Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate +tolerance specifies how far ratecontrol is willing to deviate from the +target average bitrate value. This is not related to min/max +bitrate. Lowering tolerance too much has an adverse effect on quality. + +@item flags @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) +Set generic flags. + +Possible values: +@table @samp +@item mv4 +Use four motion vector by macroblock (mpeg4). +@item qpel +Use 1/4 pel motion compensation. +@item loop +Use loop filter. +@item qscale +Use fixed qscale. +@item gmc +Use gmc. +@item mv0 +Always try a mb with mv=<0,0>. +@item input_preserved + +@item pass1 +Use internal 2pass ratecontrol in first pass mode. +@item pass2 +Use internal 2pass ratecontrol in second pass mode. +@item gray +Only decode/encode grayscale. +@item emu_edge +Do not draw edges. +@item psnr +Set error[?] variables during encoding. +@item truncated + +@item naq +Normalize adaptive quantization. +@item ildct +Use interlaced DCT. +@item low_delay +Force low delay. +@item global_header +Place global headers in extradata instead of every keyframe. +@item bitexact +Use only bitexact stuff (except (I)DCT). +@item aic +Apply H263 advanced intra coding / mpeg4 ac prediction. +@item cbp +Deprecated, use mpegvideo private options instead. +@item qprd +Deprecated, use mpegvideo private options instead. +@item ilme +Apply interlaced motion estimation. +@item cgop +Use closed gop. +@end table + +@item sub_id @var{integer} +Deprecated, currently unused. + +@item me_method @var{integer} (@emph{encoding,video}) +Set motion estimation method. + +Possible values: +@table @samp +@item zero +zero motion estimation (fastest) +@item full +full motion estimation (slowest) +@item epzs +EPZS motion estimation (default) +@item esa +esa motion estimation (alias for full) +@item tesa +tesa motion estimation +@item dia +dia motion estimation (alias for epzs) +@item log +log motion estimation +@item phods +phods motion estimation +@item x1 +X1 motion estimation +@item hex +hex motion estimation +@item umh +umh motion estimation +@item iter +iter motion estimation +@end table + +@item extradata_size @var{integer} +Set extradata size. + +@item time_base @var{rational number} +Set codec time base. + +It is the fundamental unit of time (in seconds) in terms of which +frame timestamps are represented. For fixed-fps content, timebase +should be 1/framerate and timestamp increments should be identically +1. + +@item g @var{integer} (@emph{encoding,video}) +Set the group of picture size. Default value is 12. + +@item ar @var{integer} (@emph{decoding/encoding,audio}) +Set audio sampling rate (in Hz). + +@item ac @var{integer} (@emph{decoding/encoding,audio}) +Set number of audio channels. + +@item cutoff @var{integer} (@emph{encoding,audio}) +Set cutoff bandwidth. + +@item frame_size @var{integer} (@emph{encoding,audio}) +Set audio frame size. + +Each submitted frame except the last must contain exactly frame_size +samples per channel. May be 0 when the codec has +CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not +restricted. It is set by some decoders to indicate constant frame +size. + +@item frame_number @var{integer} +Set the frame number. + +@item delay @var{integer} + +@item qcomp @var{float} (@emph{encoding,video}) +Set video quantizer scale compression (VBR). It is used as a constant +in the ratecontrol equation. Recommended range for default rc_eq: +0.0-1.0. + +@item qblur @var{float} (@emph{encoding,video}) +Set video quantizer scale blur (VBR). + +@item qmin @var{integer} (@emph{encoding,video}) +Set min video quantizer scale (VBR). Must be included between -1 and +69, default value is 2. + +@item qmax @var{integer} (@emph{encoding,video}) +Set max video quantizer scale (VBR). Must be included between -1 and +1024, default value is 31. + +@item qdiff @var{integer} (@emph{encoding,video}) +Set max difference between the quantizer scale (VBR). + +@item bf @var{integer} (@emph{encoding,video}) +Set max number of B frames. + +@item b_qfactor @var{float} (@emph{encoding,video}) +Set qp factor between P and B frames. + +@item rc_strategy @var{integer} (@emph{encoding,video}) +Set ratecontrol method. + +@item b_strategy @var{integer} (@emph{encoding,video}) +Set strategy to choose between I/P/B-frames. + +@item ps @var{integer} (@emph{encoding,video}) +Set RTP payload size in bytes. + +@item mv_bits @var{integer} +@item header_bits @var{integer} +@item i_tex_bits @var{integer} +@item p_tex_bits @var{integer} +@item i_count @var{integer} +@item p_count @var{integer} +@item skip_count @var{integer} +@item misc_bits @var{integer} +@item frame_bits @var{integer} +@item codec_tag @var{integer} +@item bug @var{flags} (@emph{decoding,video}) +Workaround not auto detected encoder bugs. + +Possible values: +@table @samp +@item autodetect + +@item old_msmpeg4 +some old lavc generated msmpeg4v3 files (no autodetection) +@item xvid_ilace +Xvid interlacing bug (autodetected if fourcc==XVIX) +@item ump4 +(autodetected if fourcc==UMP4) +@item no_padding +padding bug (autodetected) +@item amv + +@item ac_vlc +illegal vlc bug (autodetected per fourcc) +@item qpel_chroma + +@item std_qpel +old standard qpel (autodetected per fourcc/version) +@item qpel_chroma2 + +@item direct_blocksize +direct-qpel-blocksize bug (autodetected per fourcc/version) +@item edge +edge padding bug (autodetected per fourcc/version) +@item hpel_chroma + +@item dc_clip + +@item ms +Workaround various bugs in microsoft broken decoders. +@item trunc +trancated frames +@end table + +@item lelim @var{integer} (@emph{encoding,video}) +Set single coefficient elimination threshold for luminance (negative +values also consider DC coefficient). + +@item celim @var{integer} (@emph{encoding,video}) +Set single coefficient elimination threshold for chrominance (negative +values also consider dc coefficient) + +@item strict @var{integer} (@emph{decoding/encoding,audio,video}) +Specify how strictly to follow the standards. + +Possible values: +@table @samp +@item very +strictly conform to a older more strict version of the spec or reference software +@item strict +strictly conform to all the things in the spec no matter what consequences +@item normal + +@item unofficial +allow unofficial extensions +@item experimental +allow non standardized experimental things +@end table + +@item b_qoffset @var{float} (@emph{encoding,video}) +Set QP offset between P and B frames. + +@item err_detect @var{flags} (@emph{decoding,audio,video}) +Set error detection flags. + +Possible values: +@table @samp +@item crccheck +verify embedded CRCs +@item bitstream +detect bitstream specification deviations +@item buffer +detect improper bitstream length +@item explode +abort decoding on minor error detection +@item careful +consider things that violate the spec and have not been seen in the wild as errors +@item compliant +consider all spec non compliancies as errors +@item aggressive +consider things that a sane encoder should not do as an error +@end table + +@item has_b_frames @var{integer} + +@item block_align @var{integer} + +@item mpeg_quant @var{integer} (@emph{encoding,video}) +Use MPEG quantizers instead of H.263. + +@item qsquish @var{float} (@emph{encoding,video}) +How to keep quantizer between qmin and qmax (0 = clip, 1 = use +differentiable function). + +@item rc_qmod_amp @var{float} (@emph{encoding,video}) +Set experimental quantizer modulation. + +@item rc_qmod_freq @var{integer} (@emph{encoding,video}) +Set experimental quantizer modulation. + +@item rc_override_count @var{integer} + +@item rc_eq @var{string} (@emph{encoding,video}) +Set rate control equation. When computing the expression, besides the +standard functions defined in the section 'Expression Evaluation', the +following functions are available: bits2qp(bits), qp2bits(qp). Also +the following constants are available: iTex pTex tex mv fCode iCount +mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex +avgTex. + +@item maxrate @var{integer} (@emph{encoding,audio,video}) +Set max bitrate tolerance (in bits/s). Requires bufsize to be set. + +@item minrate @var{integer} (@emph{encoding,audio,video}) +Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR +encode. It is of little use elsewise. + +@item bufsize @var{integer} (@emph{encoding,audio,video}) +Set ratecontrol buffer size (in bits). + +@item rc_buf_aggressivity @var{float} (@emph{encoding,video}) +Currently useless. + +@item i_qfactor @var{float} (@emph{encoding,video}) +Set QP factor between P and I frames. + +@item i_qoffset @var{float} (@emph{encoding,video}) +Set QP offset between P and I frames. + +@item rc_init_cplx @var{float} (@emph{encoding,video}) +Set initial complexity for 1-pass encoding. + +@item dct @var{integer} (@emph{encoding,video}) +Set DCT algorithm. + +Possible values: +@table @samp +@item auto +autoselect a good one (default) +@item fastint +fast integer +@item int +accurate integer +@item mmx + +@item altivec + +@item faan +floating point AAN DCT +@end table + +@item lumi_mask @var{float} (@emph{encoding,video}) +Compress bright areas stronger than medium ones. + +@item tcplx_mask @var{float} (@emph{encoding,video}) +Set temporal complexity masking. + +@item scplx_mask @var{float} (@emph{encoding,video}) +Set spatial complexity masking. + +@item p_mask @var{float} (@emph{encoding,video}) +Set inter masking. + +@item dark_mask @var{float} (@emph{encoding,video}) +Compress dark areas stronger than medium ones. + +@item idct @var{integer} (@emph{decoding/encoding,video}) +Select IDCT implementation. + +Possible values: +@table @samp +@item auto + +@item int + +@item simple + +@item simplemmx + +@item libmpeg2mmx + +@item mmi + +@item arm + +@item altivec + +@item sh4 + +@item simplearm + +@item simplearmv5te + +@item simplearmv6 + +@item simpleneon + +@item simplealpha + +@item h264 + +@item vp3 + +@item ipp + +@item xvidmmx + +@item faani +floating point AAN IDCT +@end table + +@item slice_count @var{integer} + +@item ec @var{flags} (@emph{decoding,video}) +Set error concealment strategy. + +Possible values: +@table @samp +@item guess_mvs +iterative motion vector (MV) search (slow) +@item deblock +use strong deblock filter for damaged MBs +@end table + +@item bits_per_coded_sample @var{integer} + +@item pred @var{integer} (@emph{encoding,video}) +Set prediction method. + +Possible values: +@table @samp +@item left + +@item plane + +@item median + +@end table + +@item aspect @var{rational number} (@emph{encoding,video}) +Set sample aspect ratio. + +@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) +Print specific debug info. + +Possible values: +@table @samp +@item pict +picture info +@item rc +rate control +@item bitstream + +@item mb_type +macroblock (MB) type +@item qp +per-block quantization parameter (QP) +@item mv +motion vector +@item dct_coeff + +@item skip + +@item startcode + +@item pts + +@item er +error recognition +@item mmco +memory management control operations (H.264) +@item bugs + +@item vis_qp +visualize quantization parameter (QP), lower QP are tinted greener +@item vis_mb_type +visualize block types +@item buffers +picture buffer allocations +@item thread_ops +threading operations +@end table + +@item vismv @var{integer} (@emph{decoding,video}) +Visualize motion vectors (MVs). + +Possible values: +@table @samp +@item pf +forward predicted MVs of P-frames +@item bf +forward predicted MVs of B-frames +@item bb +backward predicted MVs of B-frames +@end table + +@item cmp @var{integer} (@emph{encoding,video}) +Set full pel me compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item subcmp @var{integer} (@emph{encoding,video}) +Set sub pel me compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item mbcmp @var{integer} (@emph{encoding,video}) +Set macroblock compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item ildctcmp @var{integer} (@emph{encoding,video}) +Set interlaced dct compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item dia_size @var{integer} (@emph{encoding,video}) +Set diamond type & size for motion estimation. + +@item last_pred @var{integer} (@emph{encoding,video}) +Set amount of motion predictors from the previous frame. + +@item preme @var{integer} (@emph{encoding,video}) +Set pre motion estimation. + +@item precmp @var{integer} (@emph{encoding,video}) +Set pre motion estimation compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item pre_dia_size @var{integer} (@emph{encoding,video}) +Set diamond type & size for motion estimation pre-pass. + +@item subq @var{integer} (@emph{encoding,video}) +Set sub pel motion estimation quality. + +@item dtg_active_format @var{integer} + +@item me_range @var{integer} (@emph{encoding,video}) +Set limit motion vectors range (1023 for DivX player). + +@item ibias @var{integer} (@emph{encoding,video}) +Set intra quant bias. + +@item pbias @var{integer} (@emph{encoding,video}) +Set inter quant bias. + +@item color_table_id @var{integer} + +@item global_quality @var{integer} (@emph{encoding,audio,video}) + +@item coder @var{integer} (@emph{encoding,video}) + +Possible values: +@table @samp +@item vlc +variable length coder / huffman coder +@item ac +arithmetic coder +@item raw +raw (no encoding) +@item rle +run-length coder +@item deflate +deflate-based coder +@end table + +@item context @var{integer} (@emph{encoding,video}) +Set context model. + +@item slice_flags @var{integer} + +@item xvmc_acceleration @var{integer} + +@item mbd @var{integer} (@emph{encoding,video}) +Set macroblock decision algorithm (high quality mode). + +Possible values: +@table @samp +@item simple +use mbcmp (default) +@item bits +use fewest bits +@item rd +use best rate distortion +@end table + +@item stream_codec_tag @var{integer} + +@item sc_threshold @var{integer} (@emph{encoding,video}) +Set scene change threshold. + +@item lmin @var{integer} (@emph{encoding,video}) +Set min lagrange factor (VBR). + +@item lmax @var{integer} (@emph{encoding,video}) +Set max lagrange factor (VBR). + +@item nr @var{integer} (@emph{encoding,video}) +Set noise reduction. + +@item rc_init_occupancy @var{integer} (@emph{encoding,video}) +Set number of bits which should be loaded into the rc buffer before +decoding starts. + +@item inter_threshold @var{integer} (@emph{encoding,video}) + +@item flags2 @var{flags} (@emph{decoding/encoding,audio,video}) + +Possible values: +@table @samp +@item fast +allow non spec compliant speedup tricks +@item sgop +Deprecated, use mpegvideo private options instead +@item noout +skip bitstream encoding +@item local_header +place global headers at every keyframe instead of in extradata +@item chunks +Frame data might be split into multiple chunks +@item showall +Show all frames before the first keyframe +@item skiprd +Deprecated, use mpegvideo private options instead +@end table + +@item error @var{integer} (@emph{encoding,video}) + +@item qns @var{integer} (@emph{encoding,video}) +Deprecated, use mpegvideo private options instead. + +@item threads @var{integer} (@emph{decoding/encoding,video}) + +Possible values: +@table @samp +@item auto +detect a good number of threads +@end table + +@item me_threshold @var{integer} (@emph{encoding,video}) +Set motion estimation threshold. + +@item mb_threshold @var{integer} (@emph{encoding,video}) +Set macroblock threshold. + +@item dc @var{integer} (@emph{encoding,video}) +Set intra_dc_precision. + +@item nssew @var{integer} (@emph{encoding,video}) +Set nsse weight. + +@item skip_top @var{integer} (@emph{decoding,video}) +Set number of macroblock rows at the top which are skipped. + +@item skip_bottom @var{integer} (@emph{decoding,video}) +Set number of macroblock rows at the bottom which are skipped. + +@item profile @var{integer} (@emph{encoding,audio,video}) + +Possible values: +@table @samp +@item unknown + +@item aac_main + +@item aac_low + +@item aac_ssr + +@item aac_ltp + +@item aac_he + +@item aac_he_v2 + +@item aac_ld + +@item aac_eld + +@item dts + +@item dts_es + +@item dts_96_24 + +@item dts_hd_hra + +@item dts_hd_ma + +@end table + +@item level @var{integer} (@emph{encoding,audio,video}) + +Possible values: +@table @samp +@item unknown + +@end table + +@item lowres @var{integer} (@emph{decoding,audio,video}) +Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions. + +@item skip_threshold @var{integer} (@emph{encoding,video}) +Set frame skip threshold. + +@item skip_factor @var{integer} (@emph{encoding,video}) +Set frame skip factor. + +@item skip_exp @var{integer} (@emph{encoding,video}) +Set frame skip exponent. + +@item skipcmp @var{integer} (@emph{encoding,video}) +Set frame skip compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item border_mask @var{float} (@emph{encoding,video}) +Increase the quantizer for macroblocks close to borders. + +@item mblmin @var{integer} (@emph{encoding,video}) +Set min macroblock lagrange factor (VBR). + +@item mblmax @var{integer} (@emph{encoding,video}) +Set max macroblock lagrange factor (VBR). + +@item mepc @var{integer} (@emph{encoding,video}) +Set motion estimation bitrate penalty compensation (1.0 = 256). + +@item skip_loop_filter @var{integer} (@emph{decoding,video}) +@item skip_idct @var{integer} (@emph{decoding,video}) +@item skip_frame @var{integer} (@emph{decoding,video}) + +Make decoder discard processing depending on the frame type selected +by the option value. + +@option{skip_loop_filter} skips frame loop filtering, @option{skip_idct} +skips frame IDCT/dequantization, @option{skip_frame} skips decoding. + +Possible values: +@table @samp +@item none +Discard no frame. + +@item default +Discard useless frames like 0-sized frames. + +@item noref +Discard all non-reference frames. + +@item bidir +Discard all bidirectional frames. + +@item nokey +Discard all frames excepts keyframes. + +@item all +Discard all frames. +@end table + +Default value is @samp{default}. + +@item bidir_refine @var{integer} (@emph{encoding,video}) +Refine the two motion vectors used in bidirectional macroblocks. + +@item brd_scale @var{integer} (@emph{encoding,video}) +Downscale frames for dynamic B-frame decision. + +@item keyint_min @var{integer} (@emph{encoding,video}) +Set minimum interval between IDR-frames. + +@item refs @var{integer} (@emph{encoding,video}) +Set reference frames to consider for motion compensation. + +@item chromaoffset @var{integer} (@emph{encoding,video}) +Set chroma qp offset from luma. + +@item trellis @var{integer} (@emph{encoding,audio,video}) +Set rate-distortion optimal quantization. + +@item sc_factor @var{integer} (@emph{encoding,video}) +Set value multiplied by qscale for each frame and added to +scene_change_score. + +@item mv0_threshold @var{integer} (@emph{encoding,video}) +@item b_sensitivity @var{integer} (@emph{encoding,video}) +Adjust sensitivity of b_frame_strategy 1. + +@item compression_level @var{integer} (@emph{encoding,audio,video}) +@item min_prediction_order @var{integer} (@emph{encoding,audio}) +@item max_prediction_order @var{integer} (@emph{encoding,audio}) +@item timecode_frame_start @var{integer} (@emph{encoding,video}) +Set GOP timecode frame start number, in non drop frame format. + +@item request_channels @var{integer} (@emph{decoding,audio}) +Set desired number of audio channels. + +@item bits_per_raw_sample @var{integer} +@item channel_layout @var{integer} (@emph{decoding/encoding,audio}) + +Possible values: +@table @samp +@end table +@item request_channel_layout @var{integer} (@emph{decoding,audio}) + +Possible values: +@table @samp +@end table +@item rc_max_vbv_use @var{float} (@emph{encoding,video}) +@item rc_min_vbv_use @var{float} (@emph{encoding,video}) +@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video}) +@item color_primaries @var{integer} (@emph{decoding/encoding,video}) +@item color_trc @var{integer} (@emph{decoding/encoding,video}) +@item colorspace @var{integer} (@emph{decoding/encoding,video}) +@item color_range @var{integer} (@emph{decoding/encoding,video}) +@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video}) + +@item log_level_offset @var{integer} +Set the log level offset. + +@item slices @var{integer} (@emph{encoding,video}) +Number of slices, used in parallelized encoding. + +@item thread_type @var{flags} (@emph{decoding/encoding,video}) +Select multithreading type. + +Possible values: +@table @samp +@item slice + +@item frame + +@end table +@item audio_service_type @var{integer} (@emph{encoding,audio}) +Set audio service type. + +Possible values: +@table @samp +@item ma +Main Audio Service +@item ef +Effects +@item vi +Visually Impaired +@item hi +Hearing Impaired +@item di +Dialogue +@item co +Commentary +@item em +Emergency +@item vo +Voice Over +@item ka +Karaoke +@end table + +@item request_sample_fmt @var{sample_fmt} (@emph{decoding,audio}) +Set sample format audio decoders should prefer. Default value is +@code{none}. + +@item pkt_timebase @var{rational number} + +@item sub_charenc @var{encoding} (@emph{decoding,subtitles}) +Set the input subtitles character encoding. +@end table + +@c man end CODEC OPTIONS + +@include decoders.texi +@include encoders.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-codecs +@settitle FFmpeg codecs + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-devices.texi b/ffmpeg1/doc/ffmpeg-devices.texi new file mode 100644 index 0000000..9e004d5 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-devices.texi @@ -0,0 +1,62 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Devices Documentation +@titlepage +@center @titlefont{FFmpeg Devices Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the input and output devices provided by the +libavdevice library. + +@c man end DESCRIPTION + +@chapter Device Options +@c man begin DEVICE OPTIONS + +The libavdevice library provides the same interface as +libavformat. Namely, an input device is considered like a demuxer, and +an output device like a muxer, and the interface and generic device +options are the same provided by libavformat (see the ffmpeg-formats +manual). + +In addition each input or output device may support so-called private +options, which are specific for that component. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the device +@code{AVFormatContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +@c man end DEVICE OPTIONS + +@include indevs.texi +@include outdevs.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavdevice.html,libavdevice} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-devices +@settitle FFmpeg devices + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-filters.texi b/ffmpeg1/doc/ffmpeg-filters.texi new file mode 100644 index 0000000..bb920ce --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-filters.texi @@ -0,0 +1,42 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Filters Documentation +@titlepage +@center @titlefont{FFmpeg Filters Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes filters, sources, and sinks provided by the +libavfilter library. + +@c man end DESCRIPTION + +@include filters.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavfilter.html,libavfilter} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-filters +@settitle FFmpeg filters + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-formats.texi b/ffmpeg1/doc/ffmpeg-formats.texi new file mode 100644 index 0000000..db9215c --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-formats.texi @@ -0,0 +1,182 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Formats Documentation +@titlepage +@center @titlefont{FFmpeg Formats Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the supported formats (muxers and demuxers) +provided by the libavformat library. + +@c man end DESCRIPTION + +@chapter Format Options +@c man begin FORMAT OPTIONS + +The libavformat library provides some generic global options, which +can be set on all the muxers and demuxers. In addition each muxer or +demuxer may support so-called private options, which are specific for +that component. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the +@code{AVFormatContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +The list of supported options follows: + +@table @option +@item avioflags @var{flags} (@emph{input/output}) +Possible values: +@table @samp +@item direct +Reduce buffering. +@end table + +@item probesize @var{integer} (@emph{input}) +Set probing size in bytes, i.e. the size of the data to analyze to get +stream information. A higher value will allow to detect more +information in case it is dispersed into the stream, but will increase +latency. Must be an integer not lesser than 32. It is 5000000 by default. + +@item packetsize @var{integer} (@emph{output}) +Set packet size. + +@item fflags @var{flags} (@emph{input/output}) +Set format flags. + +Possible values: +@table @samp +@item ignidx +Ignore index. +@item genpts +Generate PTS. +@item nofillin +Do not fill in missing values that can be exactly calculated. +@item noparse +Disable AVParsers, this needs @code{+nofillin} too. +@item igndts +Ignore DTS. +@item discardcorrupt +Discard corrupted frames. +@item sortdts +Try to interleave output packets by DTS. +@item keepside +Do not merge side data. +@item latm +Enable RTP MP4A-LATM payload. +@item nobuffer +Reduce the latency introduced by optional buffering +@end table + +@item analyzeduration @var{integer} (@emph{input}) +Specify how many microseconds are analyzed to probe the input. A +higher value will allow to detect more accurate information, but will +increase latency. It defaults to 5,000,000 microseconds = 5 seconds. + +@item cryptokey @var{hexadecimal string} (@emph{input}) +Set decryption key. + +@item indexmem @var{integer} (@emph{input}) +Set max memory used for timestamp index (per stream). + +@item rtbufsize @var{integer} (@emph{input}) +Set max memory used for buffering real-time frames. + +@item fdebug @var{flags} (@emph{input/output}) +Print specific debug info. + +Possible values: +@table @samp +@item ts +@end table + +@item max_delay @var{integer} (@emph{input/output}) +Set maximum muxing or demuxing delay in microseconds. + +@item fpsprobesize @var{integer} (@emph{input}) +Set number of frames used to probe fps. + +@item audio_preload @var{integer} (@emph{output}) +Set microseconds by which audio packets should be interleaved earlier. + +@item chunk_duration @var{integer} (@emph{output}) +Set microseconds for each chunk. + +@item chunk_size @var{integer} (@emph{output}) +Set size in bytes for each chunk. + +@item err_detect, f_err_detect @var{flags} (@emph{input}) +Set error detection flags. @code{f_err_detect} is deprecated and +should be used only via the @command{ffmpeg} tool. + +Possible values: +@table @samp +@item crccheck +Verify embedded CRCs. +@item bitstream +Detect bitstream specification deviations. +@item buffer +Detect improper bitstream length. +@item explode +Abort decoding on minor error detection. +@item careful +Consider things that violate the spec and have not been seen in the +wild as errors. +@item compliant +Consider all spec non compliancies as errors. +@item aggressive +Consider things that a sane encoder should not do as an error. +@end table + +@item use_wallclock_as_timestamps @var{integer} (@emph{input}) +Use wallclock as timestamps. + +@item avoid_negative_ts @var{integer} (@emph{output}) +Shift timestamps to make them positive. A value of 1 enables shifting, +a value of 0 disables it, the default value of -1 enables shifting +when required by the target format. + +When shifting is enabled, all output timestamps are shifted by the +same amount. Audio, video, and subtitles desynching and relative +timestamp differences are preserved compared to how they would have +been without shifting. + +Also note that this affects only leading negative timestamps, and not +non-monotonic negative timestamps. +@end table + +@c man end FORMAT OPTIONS + +@include demuxers.texi +@include muxers.texi +@include metadata.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-formats +@settitle FFmpeg formats + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-protocols.texi b/ffmpeg1/doc/ffmpeg-protocols.texi new file mode 100644 index 0000000..d992e75 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-protocols.texi @@ -0,0 +1,42 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Protocols Documentation +@titlepage +@center @titlefont{FFmpeg Protocols Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the input and output protocols provided by the +libavformat library. + +@c man end DESCRIPTION + +@include protocols.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-protocols +@settitle FFmpeg protocols + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-resampler.texi b/ffmpeg1/doc/ffmpeg-resampler.texi new file mode 100644 index 0000000..525907a --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-resampler.texi @@ -0,0 +1,265 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Resampler Documentation +@titlepage +@center @titlefont{FFmpeg Resampler Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The FFmpeg resampler provides an high-level interface to the +libswresample library audio resampling utilities. In particular it +allows to perform audio resampling, audio channel layout rematrixing, +and convert audio format and packing layout. + +@c man end DESCRIPTION + +@chapter Resampler Options +@c man begin RESAMPLER OPTIONS + +The audio resampler supports the following named options. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, @var{option}=@var{value} for the aresample filter, +by setting the value explicitly in the +@code{SwrContext} options or using the @file{libavutil/opt.h} API for +programmatic use. + +@table @option + +@item ich, in_channel_count +Set the number of input channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +@option{in_channel_layout} is set. + +@item och, out_channel_count +Set the number of output channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +@option{out_channel_layout} is set. + +@item uch, used_channel_count +Set the number of used input channels. Default value is 0. This option is +only used for special remapping. + +@item isr, in_sample_rate +Set the input sample rate. Default value is 0. + +@item osr, out_sample_rate +Set the output sample rate. Default value is 0. + +@item isf, in_sample_fmt +Specify the input sample format. It is set by default to @code{none}. + +@item osf, out_sample_fmt +Specify the output sample format. It is set by default to @code{none}. + +@item tsf, internal_sample_fmt +Set the internal sample format. Default value is @code{none}. +This will automatically be chosen when it is not explicitly set. + +@item icl, in_channel_layout +Set the input channel layout. + +@item ocl, out_channel_layout +Set the output channel layout. + +@item clev, center_mix_level +Set the center mix level. It is a value expressed in deciBel, and must be +in the interval [-32,32]. + +@item slev, surround_mix_level +Set the surround mix level. It is a value expressed in deciBel, and must +be in the interval [-32,32]. + +@item lfe_mix_level +Set LFE mix into non LFE level. It is used when there is a LFE input but no +LFE output. It is a value expressed in deciBel, and must +be in the interval [-32,32]. + +@item rmvol, rematrix_volume +Set rematrix volume. Default value is 1.0. + +@item flags, swr_flags +Set flags used by the converter. Default value is 0. + +It supports the following individual flags: +@table @option +@item res +force resampling, this flag forces resampling to be used even when the +input and output sample rates match. +@end table + +@item dither_scale +Set the dither scale. Default value is 1. + +@item dither_method +Set dither method. Default value is 0. + +Supported values: +@table @samp +@item rectangular +select rectangular dither +@item triangular +select triangular dither +@item triangular_hp +select triangular dither with high pass +@item lipshitz +select lipshitz noise shaping dither +@item shibata +select shibata noise shaping dither +@item low_shibata +select low shibata noise shaping dither +@item high_shibata +select high shibata noise shaping dither +@item f_weighted +select f-weighted noise shaping dither +@item modified_e_weighted +select modified-e-weighted noise shaping dither +@item improved_e_weighted +select improved-e-weighted noise shaping dither + +@end table + +@item resampler +Set resampling engine. Default value is swr. + +Supported values: +@table @samp +@item swr +select the native SW Resampler; filter options precision and cheby are not +applicable in this case. +@item soxr +select the SoX Resampler (where available); compensation, and filter options +filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this +case. +@end table + +@item filter_size +For swr only, set resampling filter size, default value is 32. + +@item phase_shift +For swr only, set resampling phase shift, default value is 10, and must be in +the interval [0,30]. + +@item linear_interp +Use Linear Interpolation if set to 1, default value is 0. + +@item cutoff +Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float +value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr +(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz). + +@item precision +For soxr only, the precision in bits to which the resampled signal will be +calculated. The default value of 20 (which, with suitable dithering, is +appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a +value of 28 gives SoX's 'Very High Quality'. + +@item cheby +For soxr only, selects passband rolloff none (Chebyshev) & higher-precision +approximation for 'irrational' ratios. Default value is 0. + +@item async +For swr only, simple 1 parameter audio sync to timestamps using stretching, +squeezing, filling and trimming. Setting this to 1 will enable filling and +trimming, larger values represent the maximum amount in samples that the data +may be stretched or squeezed for each second. +Default value is 0, thus no compensation is applied to make the samples match +the audio timestamps. + +@item first_pts +For swr only, assume the first pts should be this value. The time unit is 1 / sample rate. +This allows for padding/trimming at the start of stream. By default, no +assumption is made about the first frame's expected pts, so no padding or +trimming is done. For example, this could be set to 0 to pad the beginning with +silence if an audio stream starts after the video stream or to trim any samples +with a negative pts due to encoder delay. + +@item min_comp +For swr only, set the minimum difference between timestamps and audio data (in +seconds) to trigger stretching/squeezing/filling or trimming of the +data to make it match the timestamps. The default is that +stretching/squeezing/filling and trimming is disabled +(@option{min_comp} = @code{FLT_MAX}). + +@item min_hard_comp +For swr only, set the minimum difference between timestamps and audio data (in +seconds) to trigger adding/dropping samples to make it match the +timestamps. This option effectively is a threshold to select between +hard (trim/fill) and soft (squeeze/stretch) compensation. Note that +all compensation is by default disabled through @option{min_comp}. +The default is 0.1. + +@item comp_duration +For swr only, set duration (in seconds) over which data is stretched/squeezed +to make it match the timestamps. Must be a non-negative double float value, +default value is 1.0. + +@item max_soft_comp +For swr only, set maximum factor by which data is stretched/squeezed to make it +match the timestamps. Must be a non-negative double float value, default value +is 0. + +@item matrix_encoding +Select matrixed stereo encoding. + +It accepts the following values: +@table @samp +@item none +select none +@item dolby +select Dolby +@item dplii +select Dolby Pro Logic II +@end table + +Default value is @code{none}. + +@item filter_type +For swr only, select resampling filter type. This only affects resampling +operations. + +It accepts the following values: +@table @samp +@item cubic +select cubic +@item blackman_nuttall +select Blackman Nuttall Windowed Sinc +@item kaiser +select Kaiser Windowed Sinc +@end table + +@item kaiser_beta +For swr only, set Kaiser Window Beta value. Must be an integer in the +interval [2,16], default value is 9. + +@end table + +@c man end RESAMPLER OPTIONS + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libswresample.html,libswresample} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-resampler +@settitle FFmpeg Resampler + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-scaler.texi b/ffmpeg1/doc/ffmpeg-scaler.texi new file mode 100644 index 0000000..1110c69 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-scaler.texi @@ -0,0 +1,141 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Scaler Documentation +@titlepage +@center @titlefont{FFmpeg Scaler Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The FFmpeg rescaler provides an high-level interface to the libswscale +library image conversion utilities. In particular it allows to perform +image rescaling and pixel format conversion. + +@c man end DESCRIPTION + +@chapter Scaler Options +@c man begin SCALER OPTIONS + +The video scaler supports the following named options. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools. For programmatic use, they can be set explicitly in the +@code{SwsContext} options or through the @file{libavutil/opt.h} API. + +@table @option + +@item sws_flags +Set the scaler flags. This is also used to set the scaling +algorithm. Only a single algorithm should be selected. + +It accepts the following values: +@table @samp +@item fast_bilinear +Select fast bilinear scaling algorithm. + +@item bilinear +Select bilinear scaling algorithm. + +@item bicubic +Select bicubic scaling algorithm. + +@item experimental +Select experimental scaling algorithm. + +@item neighbor +Select nearest neighbor rescaling algorithm. + +@item area +Select averaging area rescaling algorithm. + +@item bicubiclin +Select bicubic scaling algorithm for the luma component, bilinear for +chroma components. + +@item gauss +Select Gaussian rescaling algorithm. + +@item sinc +Select sinc rescaling algorithm. + +@item lanczos +Select lanczos rescaling algorithm. + +@item spline +Select natural bicubic spline rescaling algorithm. + +@item print_info +Enable printing/debug logging. + +@item accurate_rnd +Enable accurate rounding. + +@item full_chroma_int +Enable full chroma interpolation. + +@item full_chroma_inp +Select full chroma input. + +@item bitexact +Enable bitexact output. +@end table + +@item srcw +Set source width. + +@item srch +Set source height. + +@item dstw +Set destination width. + +@item dsth +Set destination height. + +@item src_format +Set source pixel format (must be expressed as an integer). + +@item dst_format +Set destination pixel format (must be expressed as an integer). + +@item src_range +Select source range. + +@item dst_range +Select destination range. + +@item param0, param1 +Set scaling algorithm parameters. The specified values are specific of +some scaling algorithms and ignored by others. The specified values +are floating point number values. + +@end table + +@c man end SCALER OPTIONS + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libswscale.html,libswscale} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-scaler +@settitle FFmpeg video scaling and pixel format converter + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg-utils.texi b/ffmpeg1/doc/ffmpeg-utils.texi new file mode 100644 index 0000000..c5822a8 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg-utils.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- + +@settitle FFmpeg Utilities Documentation +@titlepage +@center @titlefont{FFmpeg Utilities Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes some generic features and utilities provided +by the libavutil library. + +@c man end DESCRIPTION + +@include syntax.texi +@include eval.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-utils +@settitle FFmpeg utilities + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg.texi b/ffmpeg1/doc/ffmpeg.texi new file mode 100644 index 0000000..ca5d652 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg.texi @@ -0,0 +1,1385 @@ +\input texinfo @c -*- texinfo -*- + +@settitle ffmpeg Documentation +@titlepage +@center @titlefont{ffmpeg Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ... + +@chapter Description +@c man begin DESCRIPTION + +ffmpeg is a very fast video and audio converter that can also grab from +a live audio/video source. It can also convert between arbitrary sample +rates and resize video on the fly with a high quality polyphase filter. + +ffmpeg reads from an arbitrary number of input "files" (which can be regular +files, pipes, network streams, grabbing devices, etc.), specified by the +@code{-i} option, and writes to an arbitrary number of output "files", which are +specified by a plain output filename. Anything found on the command line which +cannot be interpreted as an option is considered to be an output filename. + +Each input or output file can in principle contain any number of streams of +different types (video/audio/subtitle/attachment/data). Allowed number and/or +types of streams can be limited by the container format. Selecting, which +streams from which inputs go into output, is done either automatically or with +the @code{-map} option (see the Stream selection chapter). + +To refer to input files in options, you must use their indices (0-based). E.g. +the first input file is @code{0}, the second is @code{1} etc. Similarly, streams +within a file are referred to by their indices. E.g. @code{2:3} refers to the +fourth stream in the third input file. See also the Stream specifiers chapter. + +As a general rule, options are applied to the next specified +file. Therefore, order is important, and you can have the same +option on the command line multiple times. Each occurrence is +then applied to the next input or output file. +Exceptions from this rule are the global options (e.g. verbosity level), +which should be specified first. + +Do not mix input and output files -- first specify all input files, then all +output files. Also do not mix options which belong to different files. All +options apply ONLY to the next input or output file and are reset between files. + +@itemize +@item +To set the video bitrate of the output file to 64kbit/s: +@example +ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi +@end example + +@item +To force the frame rate of the output file to 24 fps: +@example +ffmpeg -i input.avi -r 24 output.avi +@end example + +@item +To force the frame rate of the input file (valid for raw formats only) +to 1 fps and the frame rate of the output file to 24 fps: +@example +ffmpeg -r 1 -i input.m2v -r 24 output.avi +@end example +@end itemize + +The format option may be needed for raw input files. + +@c man end DESCRIPTION + +@chapter Detailed description +@c man begin DETAILED DESCRIPTION + +The transcoding process in @command{ffmpeg} for each output can be described by +the following diagram: + +@example + _______ ______________ _________ ______________ ________ +| | | | | | | | | | +| input | demuxer | encoded data | decoder | decoded | encoder | encoded data | muxer | output | +| file | ---------> | packets | ---------> | frames | ---------> | packets | -------> | file | +|_______| |______________| |_________| |______________| |________| + +@end example + +@command{ffmpeg} calls the libavformat library (containing demuxers) to read +input files and get packets containing encoded data from them. When there are +multiple input files, @command{ffmpeg} tries to keep them synchronized by +tracking lowest timestamp on any active input stream. + +Encoded packets are then passed to the decoder (unless streamcopy is selected +for the stream, see further for a description). The decoder produces +uncompressed frames (raw video/PCM audio/...) which can be processed further by +filtering (see next section). After filtering the frames are passed to the +encoder, which encodes them and outputs encoded packets again. Finally those are +passed to the muxer, which writes the encoded packets to the output file. + +@section Filtering +Before encoding, @command{ffmpeg} can process raw audio and video frames using +filters from the libavfilter library. Several chained filters form a filter +graph. @command{ffmpeg} distinguishes between two types of filtergraphs - +simple and complex. + +@subsection Simple filtergraphs +Simple filtergraphs are those that have exactly one input and output, both of +the same type. In the above diagram they can be represented by simply inserting +an additional step between decoding and encoding: + +@example + _________ __________ ______________ +| | | | | | +| decoded | simple filtergraph | filtered | encoder | encoded data | +| frames | -------------------> | frames | ---------> | packets | +|_________| |__________| |______________| + +@end example + +Simple filtergraphs are configured with the per-stream @option{-filter} option +(with @option{-vf} and @option{-af} aliases for video and audio respectively). +A simple filtergraph for video can look for example like this: + +@example + _______ _____________ _______ _____ ________ +| | | | | | | | | | +| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output | +|_______| |_____________| |_______| |_____| |________| + +@end example + +Note that some filters change frame properties but not frame contents. E.g. the +@code{fps} filter in the example above changes number of frames, but does not +touch the frame contents. Another example is the @code{setpts} filter, which +only sets timestamps and otherwise passes the frames unchanged. + +@subsection Complex filtergraphs +Complex filtergraphs are those which cannot be described as simply a linear +processing chain applied to one stream. This is the case e.g. when the graph has +more than one input and/or output, or when output stream type is different from +input. They can be represented with the following diagram: + +@example + _________ +| | +| input 0 |\ __________ +|_________| \ | | + \ _________ /| output 0 | + \ | | / |__________| + _________ \| complex | / +| | | |/ +| input 1 |---->| filter |\ +|_________| | | \ __________ + /| graph | \ | | + / | | \| output 1 | + _________ / |_________| |__________| +| | / +| input 2 |/ +|_________| + +@end example + +Complex filtergraphs are configured with the @option{-filter_complex} option. +Note that this option is global, since a complex filtergraph by its nature +cannot be unambiguously associated with a single stream or file. + +The @option{-lavfi} option is equivalent to @option{-filter_complex}. + +A trivial example of a complex filtergraph is the @code{overlay} filter, which +has two video inputs and one video output, containing one video overlaid on top +of the other. Its audio counterpart is the @code{amix} filter. + +@section Stream copy +Stream copy is a mode selected by supplying the @code{copy} parameter to the +@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding +step for the specified stream, so it does only demuxing and muxing. It is useful +for changing the container format or modifying container-level metadata. The +diagram above will in this case simplify to this: + +@example + _______ ______________ ________ +| | | | | | +| input | demuxer | encoded data | muxer | output | +| file | ---------> | packets | -------> | file | +|_______| |______________| |________| + +@end example + +Since there is no decoding or encoding, it is very fast and there is no quality +loss. However it might not work in some cases because of many factors. Applying +filters is obviously also impossible, since filters work on uncompressed data. + +@c man end DETAILED DESCRIPTION + +@chapter Stream selection +@c man begin STREAM SELECTION + +By default ffmpeg includes only one stream of each type (video, audio, subtitle) +present in the input files and adds them to each output file. It picks the +"best" of each based upon the following criteria; for video it is the stream +with the highest resolution, for audio the stream with the most channels, for +subtitle it's the first subtitle stream. In the case where several streams of +the same type rate equally, the lowest numbered stream is chosen. + +You can disable some of those defaults by using @code{-vn/-an/-sn} options. For +full manual control, use the @code{-map} option, which disables the defaults just +described. + +@c man end STREAM SELECTION + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option + +@item -f @var{fmt} (@emph{input/output}) +Force input or output file format. The format is normally auto detected for input +files and guessed from file extension for output files, so this option is not +needed in most cases. + +@item -i @var{filename} (@emph{input}) +input file name + +@item -y (@emph{global}) +Overwrite output files without asking. + +@item -n (@emph{global}) +Do not overwrite output files but exit if file exists. + +@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream}) +@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream}) +Select an encoder (when used before an output file) or a decoder (when used +before an input file) for one or more streams. @var{codec} is the name of a +decoder/encoder or a special value @code{copy} (output only) to indicate that +the stream is not to be re-encoded. + +For example +@example +ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT +@end example +encodes all video streams with libx264 and copies all audio streams. + +For each stream, the last matching @code{c} option is applied, so +@example +ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT +@end example +will copy all the streams except the second video, which will be encoded with +libx264, and the 138th audio, which will be encoded with libvorbis. + +@item -t @var{duration} (@emph{output}) +Stop writing the output after its duration reaches @var{duration}. +@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form. + +-to and -t are mutually exclusive and -t has priority. + +@item -to @var{position} (@emph{output}) +Stop writing the output at @var{position}. +@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form. + +-to and -t are mutually exclusive and -t has priority. + +@item -fs @var{limit_size} (@emph{output}) +Set the file size limit, expressed in bytes. + +@item -ss @var{position} (@emph{input/output}) +When used as an input option (before @code{-i}), seeks in this input file to +@var{position}. When used as an output option (before an output filename), +decodes but discards input until the timestamps reach @var{position}. This is +slower, but more accurate. + +@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form. + +@item -itsoffset @var{offset} (@emph{input}) +Set the input time offset in seconds. +@code{[-]hh:mm:ss[.xxx]} syntax is also supported. +The offset is added to the timestamps of the input files. +Specifying a positive offset means that the corresponding +streams are delayed by @var{offset} seconds. + +@item -timestamp @var{time} (@emph{output}) +Set the recording timestamp in the container. +The syntax for @var{time} is: +@example +now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z]) +@end example +If the value is "now" it takes the current time. +Time is local time unless 'Z' or 'z' is appended, in which case it is +interpreted as UTC. +If the year-month-day part is not specified it takes the current +year-month-day. + +@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata}) +Set a metadata key/value pair. + +An optional @var{metadata_specifier} may be given to set metadata +on streams or chapters. See @code{-map_metadata} documentation for +details. + +This option overrides metadata set with @code{-map_metadata}. It is +also possible to delete metadata by using an empty value. + +For example, for setting the title in the output file: +@example +ffmpeg -i in.avi -metadata title="my title" out.flv +@end example + +To set the language of the first audio stream: +@example +ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT +@end example + +@item -target @var{type} (@emph{output}) +Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv}, +@code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or +@code{film-} to use the corresponding standard. All the format options +(bitrate, codecs, buffer sizes) are then set automatically. You can just type: + +@example +ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg +@end example + +Nevertheless you can specify additional options as long as you know +they do not conflict with the standard, as in: + +@example +ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg +@end example + +@item -dframes @var{number} (@emph{output}) +Set the number of data frames to record. This is an alias for @code{-frames:d}. + +@item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream}) +Stop writing to the stream after @var{framecount} frames. + +@item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) +@itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) +Use fixed quality scale (VBR). The meaning of @var{q} is +codec-dependent. + +@anchor{filter_option} +@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream}) +Create the filter graph specified by @var{filter_graph} and use it to +filter the stream. + +@var{filter_graph} is a description of the filter graph to apply to +the stream, and must have a single input and a single output of the +same type of the stream. In the filter graph, the input is associated +to the label @code{in}, and the output to the label @code{out}. See +the ffmpeg-filters manual for more information about the filtergraph +syntax. + +See the @ref{filter_complex_option,,-filter_complex option} if you +want to create filter graphs with multiple inputs and/or outputs. + +@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream}) +Specify the preset for matching stream(s). + +@item -stats (@emph{global}) +Print encoding progress/statistics. It is on by default, to explicitly +disable it you need to specify @code{-nostats}. + +@item -progress @var{url} (@emph{global}) +Send program-friendly progress information to @var{url}. + +Progress information is written approximately every second and at the end of +the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key} +consists of only alphanumeric characters. The last key of a sequence of +progress information is always "progress". + +@item -stdin +Enable interaction on standard input. On by default unless standard input is +used as an input. To explicitly disable interaction you need to specify +@code{-nostdin}. + +Disabling interaction on standard input is useful, for example, if +ffmpeg is in the background process group. Roughly the same result can +be achieved with @code{ffmpeg ... < /dev/null} but it requires a +shell. + +@item -debug_ts (@emph{global}) +Print timestamp information. It is off by default. This option is +mostly useful for testing and debugging purposes, and the output +format may change from one version to another, so it should not be +employed by portable scripts. + +See also the option @code{-fdebug ts}. + +@item -attach @var{filename} (@emph{output}) +Add an attachment to the output file. This is supported by a few formats +like Matroska for e.g. fonts used in rendering subtitles. Attachments +are implemented as a specific type of stream, so this option will add +a new stream to the file. It is then possible to use per-stream options +on this stream in the usual way. Attachment streams created with this +option will be created after all the other streams (i.e. those created +with @code{-map} or automatic mappings). + +Note that for Matroska you also have to set the mimetype metadata tag: +@example +ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv +@end example +(assuming that the attachment stream will be third in the output file). + +@item -dump_attachment[:@var{stream_specifier}] @var{filename} (@emph{input,per-stream}) +Extract the matching attachment stream into a file named @var{filename}. If +@var{filename} is empty, then the value of the @code{filename} metadata tag +will be used. + +E.g. to extract the first attachment to a file named 'out.ttf': +@example +ffmpeg -dump_attachment:t:0 out.ttf -i INPUT +@end example +To extract all attachments to files determined by the @code{filename} tag: +@example +ffmpeg -dump_attachment:t "" -i INPUT +@end example + +Technical note -- attachments are implemented as codec extradata, so this +option can actually be used to extract extradata from any stream, not just +attachments. + +@end table + +@section Video Options + +@table @option +@item -vframes @var{number} (@emph{output}) +Set the number of video frames to record. This is an alias for @code{-frames:v}. +@item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream}) +Set frame rate (Hz value, fraction or abbreviation). + +As an input option, ignore any timestamps stored in the file and instead +generate timestamps assuming constant frame rate @var{fps}. + +As an output option, duplicate or drop input frames to achieve constant output +frame rate @var{fps}. + +@item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream}) +Set frame size. + +As an input option, this is a shortcut for the @option{video_size} private +option, recognized by some demuxers for which the frame size is either not +stored in the file or is configurable -- e.g. raw video or video grabbers. + +As an output option, this inserts the @code{scale} video filter to the +@emph{end} of the corresponding filtergraph. Please use the @code{scale} filter +directly to insert it at the beginning or some other place. + +The format is @samp{wxh} (default - same as source). + +@item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream}) +Set the video display aspect ratio specified by @var{aspect}. + +@var{aspect} can be a floating point number string, or a string of the +form @var{num}:@var{den}, where @var{num} and @var{den} are the +numerator and denominator of the aspect ratio. For example "4:3", +"16:9", "1.3333", and "1.7777" are valid argument values. + +@item -vn (@emph{output}) +Disable video recording. + +@item -vcodec @var{codec} (@emph{output}) +Set the video codec. This is an alias for @code{-codec:v}. + +@item -pass[:@var{stream_specifier}] @var{n} (@emph{output,per-stream}) +Select the pass number (1 or 2). It is used to do two-pass +video encoding. The statistics of the video are recorded in the first +pass into a log file (see also the option -passlogfile), +and in the second pass that log file is used to generate the video +at the exact requested bitrate. +On pass 1, you may just deactivate audio and set output to null, +examples for Windows and Unix: +@example +ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL +ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null +@end example + +@item -passlogfile[:@var{stream_specifier}] @var{prefix} (@emph{output,per-stream}) +Set two-pass log file name prefix to @var{prefix}, the default file name +prefix is ``ffmpeg2pass''. The complete file name will be +@file{PREFIX-N.log}, where N is a number specific to the output +stream + +@item -vlang @var{code} +Set the ISO 639 language code (3 letters) of the current video stream. + +@item -vf @var{filter_graph} (@emph{output}) +Create the filter graph specified by @var{filter_graph} and use it to +filter the stream. + +This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}. +@end table + +@section Advanced Video Options + +@table @option +@item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream}) +Set pixel format. Use @code{-pix_fmts} to show all the supported +pixel formats. +If the selected pixel format can not be selected, ffmpeg will print a +warning and select the best pixel format supported by the encoder. +If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error +if the requested pixel format can not be selected, and automatic conversions +inside filter graphs are disabled. +If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format +as the input (or graph output) and automatic conversions are disabled. + +@item -sws_flags @var{flags} (@emph{input/output}) +Set SwScaler flags. +@item -vdt @var{n} +Discard threshold. + +@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream}) +Rate control override for specific intervals, formatted as "int,int,int" +list separated with slashes. Two first values are the beginning and +end frame numbers, last one is quantizer to use if positive, or quality +factor if negative. + +@item -deinterlace +Deinterlace pictures. +This option is deprecated since the deinterlacing is very low quality. +Use the yadif filter with @code{-filter:v yadif}. +@item -ilme +Force interlacing support in encoder (MPEG-2 and MPEG-4 only). +Use this option if your input file is interlaced and you want +to keep the interlaced format for minimum losses. +The alternative is to deinterlace the input stream with +@option{-deinterlace}, but deinterlacing introduces losses. +@item -psnr +Calculate PSNR of compressed frames. +@item -vstats +Dump video coding statistics to @file{vstats_HHMMSS.log}. +@item -vstats_file @var{file} +Dump video coding statistics to @var{file}. +@item -top[:@var{stream_specifier}] @var{n} (@emph{output,per-stream}) +top=1/bottom=0/auto=-1 field first +@item -dc @var{precision} +Intra_dc_precision. +@item -vtag @var{fourcc/tag} (@emph{output}) +Force video tag/fourcc. This is an alias for @code{-tag:v}. +@item -qphist (@emph{global}) +Show QP histogram +@item -vbsf @var{bitstream_filter} +Deprecated see -bsf + +@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream}) +@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream}) +Force key frames at the specified timestamps, more precisely at the first +frames after each specified time. + +If the argument is prefixed with @code{expr:}, the string @var{expr} +is interpreted like an expression and is evaluated for each frame. A +key frame is forced in case the evaluation is non-zero. + +If one of the times is "@code{chapters}[@var{delta}]", it is expanded into +the time of the beginning of all chapters in the file, shifted by +@var{delta}, expressed as a time in seconds. +This option can be useful to ensure that a seek point is present at a +chapter mark or any other designated place in the output file. + +For example, to insert a key frame at 5 minutes, plus key frames 0.1 second +before the beginning of every chapter: +@example +-force_key_frames 0:05:00,chapters-0.1 +@end example + +The expression in @var{expr} can contain the following constants: +@table @option +@item n +the number of current processed frame, starting from 0 +@item n_forced +the number of forced frames +@item prev_forced_n +the number of the previous forced frame, it is @code{NAN} when no +keyframe was forced yet +@item prev_forced_t +the time of the previous forced frame, it is @code{NAN} when no +keyframe was forced yet +@item t +the time of the current processed frame +@end table + +For example to force a key frame every 5 seconds, you can specify: +@example +-force_key_frames expr:gte(t,n_forced*5) +@end example + +To force a key frame 5 seconds after the time of the last forced one, +starting from second 13: +@example +-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5)) +@end example + +Note that forcing too many keyframes is very harmful for the lookahead +algorithms of certain encoders: using fixed-GOP options or similar +would be more efficient. + +@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream}) +When doing stream copy, copy also non-key frames found at the +beginning. +@end table + +@section Audio Options + +@table @option +@item -aframes @var{number} (@emph{output}) +Set the number of audio frames to record. This is an alias for @code{-frames:a}. +@item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream}) +Set the audio sampling frequency. For output streams it is set by +default to the frequency of the corresponding input stream. For input +streams this option only makes sense for audio grabbing devices and raw +demuxers and is mapped to the corresponding demuxer options. +@item -aq @var{q} (@emph{output}) +Set the audio quality (codec-specific, VBR). This is an alias for -q:a. +@item -ac[:@var{stream_specifier}] @var{channels} (@emph{input/output,per-stream}) +Set the number of audio channels. For output streams it is set by +default to the number of input audio channels. For input streams +this option only makes sense for audio grabbing devices and raw demuxers +and is mapped to the corresponding demuxer options. +@item -an (@emph{output}) +Disable audio recording. +@item -acodec @var{codec} (@emph{input/output}) +Set the audio codec. This is an alias for @code{-codec:a}. +@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream}) +Set the audio sample format. Use @code{-sample_fmts} to get a list +of supported sample formats. + +@item -af @var{filter_graph} (@emph{output}) +Create the filter graph specified by @var{filter_graph} and use it to +filter the stream. + +This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}. +@end table + +@section Advanced Audio options: + +@table @option +@item -atag @var{fourcc/tag} (@emph{output}) +Force audio tag/fourcc. This is an alias for @code{-tag:a}. +@item -absf @var{bitstream_filter} +Deprecated, see -bsf +@item -guess_layout_max @var{channels} (@emph{input,per-stream}) +If some input channel layout is not known, try to guess only if it +corresponds to at most the specified number of channels. For example, 2 +tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as +stereo but not 6 channels as 5.1. The default is to always try to guess. Use +0 to disable all guessing. +@end table + +@section Subtitle options: + +@table @option +@item -slang @var{code} +Set the ISO 639 language code (3 letters) of the current subtitle stream. +@item -scodec @var{codec} (@emph{input/output}) +Set the subtitle codec. This is an alias for @code{-codec:s}. +@item -sn (@emph{output}) +Disable subtitle recording. +@item -sbsf @var{bitstream_filter} +Deprecated, see -bsf +@end table + +@section Advanced Subtitle options: + +@table @option + +@item -fix_sub_duration +Fix subtitles durations. For each subtitle, wait for the next packet in the +same stream and adjust the duration of the first to avoid overlap. This is +necessary with some subtitles codecs, especially DVB subtitles, because the +duration in the original packet is only a rough estimate and the end is +actually marked by an empty subtitle frame. Failing to use this option when +necessary can result in exaggerated durations or muxing failures due to +non-monotonic timestamps. + +Note that this option will delay the output of all data until the next +subtitle packet is decoded: it may increase memory consumption and latency a +lot. + +@item -canvas_size @var{size} +Set the size of the canvas used to render subtitles. + +@end table + +@section Advanced options + +@table @option +@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output}) + +Designate one or more input streams as a source for the output file. Each input +stream is identified by the input file index @var{input_file_id} and +the input stream index @var{input_stream_id} within the input +file. Both indices start at 0. If specified, +@var{sync_file_id}:@var{stream_specifier} sets which input stream +is used as a presentation sync reference. + +The first @code{-map} option on the command line specifies the +source for output stream 0, the second @code{-map} option specifies +the source for output stream 1, etc. + +A @code{-} character before the stream identifier creates a "negative" mapping. +It disables matching streams from already created mappings. + +An alternative @var{[linklabel]} form will map outputs from complex filter +graphs (see the @option{-filter_complex} option) to the output file. +@var{linklabel} must correspond to a defined output link label in the graph. + +For example, to map ALL streams from the first input file to output +@example +ffmpeg -i INPUT -map 0 output +@end example + +For example, if you have two audio streams in the first input file, +these streams are identified by "0:0" and "0:1". You can use +@code{-map} to select which streams to place in an output file. For +example: +@example +ffmpeg -i INPUT -map 0:1 out.wav +@end example +will map the input stream in @file{INPUT} identified by "0:1" to +the (single) output stream in @file{out.wav}. + +For example, to select the stream with index 2 from input file +@file{a.mov} (specified by the identifier "0:2"), and stream with +index 6 from input @file{b.mov} (specified by the identifier "1:6"), +and copy them to the output file @file{out.mov}: +@example +ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov +@end example + +To select all video and the third audio stream from an input file: +@example +ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT +@end example + +To map all the streams except the second audio, use negative mappings +@example +ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT +@end example + +Note that using this option disables the default mappings for this output file. + +@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}] +Map an audio channel from a given input to an output. If +@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will +be mapped on all the audio streams. + +Using "-1" instead of +@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted +channel. + +For example, assuming @var{INPUT} is a stereo audio file, you can switch the +two audio channels with the following command: +@example +ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT +@end example + +If you want to mute the first channel and keep the second: +@example +ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT +@end example + +The order of the "-map_channel" option specifies the order of the channels in +the output stream. The output channel layout is guessed from the number of +channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" +in combination of "-map_channel" makes the channel gain levels to be updated if +input and output channel layouts don't match (for instance two "-map_channel" +options and "-ac 6"). + +You can also extract each channel of an input to specific outputs; the following +command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0) +to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs: +@example +ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1 +@end example + +The following example splits the channels of a stereo input into two separate +streams, which are put into the same output file: +@example +ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg +@end example + +Note that currently each output stream can only contain channels from a single +input stream; you can't for example use "-map_channel" to pick multiple input +audio channels contained in different streams (from the same or different files) +and merge them into a single output stream. It is therefore not currently +possible, for example, to turn two separate mono streams into a single stereo +stream. However splitting a stereo stream into two single channel mono streams +is possible. + +If you need this feature, a possible workaround is to use the @emph{amerge} +filter. For example, if you need to merge a media (here @file{input.mkv}) with 2 +mono audio streams into one single stereo channel audio stream (and keep the +video stream), you can use the following command: +@example +ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv +@end example + +@item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata}) +Set metadata information of the next output file from @var{infile}. Note that +those are file indices (zero-based), not filenames. +Optional @var{metadata_spec_in/out} parameters specify, which metadata to copy. +A metadata specifier can have the following forms: +@table @option +@item @var{g} +global metadata, i.e. metadata that applies to the whole file + +@item @var{s}[:@var{stream_spec}] +per-stream metadata. @var{stream_spec} is a stream specifier as described +in the @ref{Stream specifiers} chapter. In an input metadata specifier, the first +matching stream is copied from. In an output metadata specifier, all matching +streams are copied to. + +@item @var{c}:@var{chapter_index} +per-chapter metadata. @var{chapter_index} is the zero-based chapter index. + +@item @var{p}:@var{program_index} +per-program metadata. @var{program_index} is the zero-based program index. +@end table +If metadata specifier is omitted, it defaults to global. + +By default, global metadata is copied from the first input file, +per-stream and per-chapter metadata is copied along with streams/chapters. These +default mappings are disabled by creating any mapping of the relevant type. A negative +file index can be used to create a dummy mapping that just disables automatic copying. + +For example to copy metadata from the first stream of the input file to global metadata +of the output file: +@example +ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3 +@end example + +To do the reverse, i.e. copy global metadata to all audio streams: +@example +ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv +@end example +Note that simple @code{0} would work as well in this example, since global +metadata is assumed by default. + +@item -map_chapters @var{input_file_index} (@emph{output}) +Copy chapters from input file with index @var{input_file_index} to the next +output file. If no chapter mapping is specified, then chapters are copied from +the first input file with at least one chapter. Use a negative file index to +disable any chapter copying. + +@item -benchmark (@emph{global}) +Show benchmarking information at the end of an encode. +Shows CPU time used and maximum memory consumption. +Maximum memory consumption is not supported on all systems, +it will usually display as 0 if not supported. +@item -benchmark_all (@emph{global}) +Show benchmarking information during the encode. +Shows CPU time used in various steps (audio/video encode/decode). +@item -timelimit @var{duration} (@emph{global}) +Exit after ffmpeg has been running for @var{duration} seconds. +@item -dump (@emph{global}) +Dump each input packet to stderr. +@item -hex (@emph{global}) +When dumping packets, also dump the payload. +@item -re (@emph{input}) +Read input at native frame rate. Mainly used to simulate a grab device. +By default @command{ffmpeg} attempts to read the input(s) as fast as possible. +This option will slow down the reading of the input(s) to the native frame rate +of the input(s). It is useful for real-time output (e.g. live streaming). If +your input(s) is coming from some other live streaming source (through HTTP or +UDP for example) the server might already be in real-time, thus the option will +likely not be required. On the other hand, this is meaningful if your input(s) +is a file you are trying to push in real-time. +@item -loop_input +Loop over the input stream. Currently it works only for image +streams. This option is used for automatic FFserver testing. +This option is deprecated, use -loop 1. +@item -loop_output @var{number_of_times} +Repeatedly loop output for formats that support looping such as animated GIF +(0 will loop the output infinitely). +This option is deprecated, use -loop. +@item -vsync @var{parameter} +Video sync method. +For compatibility reasons old values can be specified as numbers. +Newly added values will have to be specified as strings always. + +@table @option +@item 0, passthrough +Each frame is passed with its timestamp from the demuxer to the muxer. +@item 1, cfr +Frames will be duplicated and dropped to achieve exactly the requested +constant framerate. +@item 2, vfr +Frames are passed through with their timestamp or dropped so as to +prevent 2 frames from having the same timestamp. +@item drop +As passthrough but destroys all timestamps, making the muxer generate +fresh timestamps based on frame-rate. +@item -1, auto +Chooses between 1 and 2 depending on muxer capabilities. This is the +default method. +@end table + +Note that the timestamps may be further modified by the muxer, after this. +For example, in the case that the format option @option{avoid_negative_ts} +is enabled. + +With -map you can select from which stream the timestamps should be +taken. You can leave either video or audio unchanged and sync the +remaining stream(s) to the unchanged one. + +@item -async @var{samples_per_second} +Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, +the parameter is the maximum samples per second by which the audio is changed. +-async 1 is a special case where only the start of the audio stream is corrected +without any later correction. + +Note that the timestamps may be further modified by the muxer, after this. +For example, in the case that the format option @option{avoid_negative_ts} +is enabled. + +This option has been deprecated. Use the @code{aresample} audio filter instead. + +@item -copyts +Do not process input timestamps, but keep their values without trying +to sanitize them. In particular, do not remove the initial start time +offset value. + +Note that, depending on the @option{vsync} option or on specific muxer +processing (e.g. in case the format option @option{avoid_negative_ts} +is enabled) the output timestamps may mismatch with the input +timestamps even when this option is selected. + +@item -copytb @var{mode} +Specify how to set the encoder timebase when stream copying. @var{mode} is an +integer numeric value, and can assume one of the following values: + +@table @option +@item 1 +Use the demuxer timebase. + +The time base is copied to the output encoder from the corresponding input +demuxer. This is sometimes required to avoid non monotonically increasing +timestamps when copying video streams with variable frame rate. + +@item 0 +Use the decoder timebase. + +The time base is copied to the output encoder from the corresponding input +decoder. + +@item -1 +Try to make the choice automatically, in order to generate a sane output. +@end table + +Default value is -1. + +@item -shortest (@emph{output}) +Finish encoding when the shortest input stream ends. +@item -dts_delta_threshold +Timestamp discontinuity delta threshold. +@item -muxdelay @var{seconds} (@emph{input}) +Set the maximum demux-decode delay. +@item -muxpreload @var{seconds} (@emph{input}) +Set the initial demux-decode delay. +@item -streamid @var{output-stream-index}:@var{new-value} (@emph{output}) +Assign a new stream-id value to an output stream. This option should be +specified prior to the output filename to which it applies. +For the situation where multiple output files exist, a streamid +may be reassigned to a different value. + +For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for +an output mpegts file: +@example +ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts +@end example + +@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream}) +Set bitstream filters for matching streams. @var{bitstream_filters} is +a comma-separated list of bitstream filters. Use the @code{-bsfs} option +to get the list of bitstream filters. +@example +ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 +@end example +@example +ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt +@end example + +@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream}) +Force a tag/fourcc for matching streams. + +@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff} +Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';' +(or '.') for drop. +@example +ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg +@end example + +@anchor{filter_complex_option} +@item -filter_complex @var{filtergraph} (@emph{global}) +Define a complex filter graph, i.e. one with arbitrary number of inputs and/or +outputs. For simple graphs -- those with one input and one output of the same +type -- see the @option{-filter} options. @var{filtergraph} is a description of +the filter graph, as described in the ``Filtergraph syntax'' section of the +ffmpeg-filters manual. + +Input link labels must refer to input streams using the +@code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map} +uses). If @var{stream_specifier} matches multiple streams, the first one will be +used. An unlabeled input will be connected to the first unused input stream of +the matching type. + +Output link labels are referred to with @option{-map}. Unlabeled outputs are +added to the first output file. + +Note that with this option it is possible to use only lavfi sources without +normal input files. + +For example, to overlay an image over video +@example +ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map +'[out]' out.mkv +@end example +Here @code{[0:v]} refers to the first video stream in the first input file, +which is linked to the first (main) input of the overlay filter. Similarly the +first video stream in the second input is linked to the second (overlay) input +of overlay. + +Assuming there is only one video stream in each input file, we can omit input +labels, so the above is equivalent to +@example +ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map +'[out]' out.mkv +@end example + +Furthermore we can omit the output label and the single output from the filter +graph will be added to the output file automatically, so we can simply write +@example +ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv +@end example + +To generate 5 seconds of pure red video using lavfi @code{color} source: +@example +ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv +@end example + +@item -lavfi @var{filtergraph} (@emph{global}) +Define a complex filter graph, i.e. one with arbitrary number of inputs and/or +outputs. Equivalent to @option{-filter_complex}. + +@end table + +As a special exception, you can use a bitmap subtitle stream as input: it +will be converted into a video with the same size as the largest video in +the file, or 720x576 if no video is present. Note that this is an +experimental and temporary solution. It will be removed once libavfilter has +proper support for subtitles. + +For example, to hardcode subtitles on top of a DVB-T recording stored in +MPEG-TS format, delaying the subtitles by 1 second: +@example +ffmpeg -i input.ts -filter_complex \ + '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \ + -sn -map '#0x2dc' output.mkv +@end example +(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, +audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too) + +@section Preset files +A preset file contains a sequence of @var{option}=@var{value} pairs, +one for each line, specifying a sequence of options which would be +awkward to specify on the command line. Lines starting with the hash +('#') character are ignored and are used to provide comments. Check +the @file{presets} directory in the FFmpeg source tree for examples. + +Preset files are specified with the @code{vpre}, @code{apre}, +@code{spre}, and @code{fpre} options. The @code{fpre} option takes the +filename of the preset instead of a preset name as input and can be +used for any kind of codec. For the @code{vpre}, @code{apre}, and +@code{spre} options, the options specified in a preset file are +applied to the currently selected codec of the same type as the preset +option. + +The argument passed to the @code{vpre}, @code{apre}, and @code{spre} +preset options identifies the preset file to use according to the +following rules: + +First ffmpeg searches for a file named @var{arg}.ffpreset in the +directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in +the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg}) +or in a @file{ffpresets} folder along the executable on win32, +in that order. For example, if the argument is @code{libvpx-1080p}, it will +search for the file @file{libvpx-1080p.ffpreset}. + +If no such file is found, then ffmpeg will search for a file named +@var{codec_name}-@var{arg}.ffpreset in the above-mentioned +directories, where @var{codec_name} is the name of the codec to which +the preset file options will be applied. For example, if you select +the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p}, +then it will search for the file @file{libvpx-1080p.ffpreset}. +@c man end OPTIONS + +@chapter Tips +@c man begin TIPS + +@itemize +@item +For streaming at very low bitrate application, use a low frame rate +and a small GOP size. This is especially true for RealVideo where +the Linux player does not seem to be very fast, so it can miss +frames. An example is: + +@example +ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm +@end example + +@item +The parameter 'q' which is displayed while encoding is the current +quantizer. The value 1 indicates that a very good quality could +be achieved. The value 31 indicates the worst quality. If q=31 appears +too often, it means that the encoder cannot compress enough to meet +your bitrate. You must either increase the bitrate, decrease the +frame rate or decrease the frame size. + +@item +If your computer is not fast enough, you can speed up the +compression at the expense of the compression ratio. You can use +'-me zero' to speed up motion estimation, and '-g 0' to disable +motion estimation completely (you have only I-frames, which means it +is about as good as JPEG compression). + +@item +To have very low audio bitrates, reduce the sampling frequency +(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3). + +@item +To have a constant quality (but a variable bitrate), use the option +'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst +quality). + +@end itemize +@c man end TIPS + +@chapter Examples +@c man begin EXAMPLES + +@section Preset files + +A preset file contains a sequence of @var{option=value} pairs, one for +each line, specifying a sequence of options which can be specified also on +the command line. Lines starting with the hash ('#') character are ignored and +are used to provide comments. Empty lines are also ignored. Check the +@file{presets} directory in the FFmpeg source tree for examples. + +Preset files are specified with the @code{pre} option, this option takes a +preset name as input. FFmpeg searches for a file named @var{preset_name}.avpreset in +the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in +the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg}) +in that order. For example, if the argument is @code{libx264-max}, it will +search for the file @file{libx264-max.avpreset}. + +@section Video and Audio grabbing + +If you specify the input format and device then ffmpeg can grab video +and audio directly. + +@example +ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg +@end example + +Or with an ALSA audio source (mono input, card id 1) instead of OSS: +@example +ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg +@end example + +Note that you must activate the right video source and channel before +launching ffmpeg with any TV viewer such as +@uref{http://linux.bytesex.org/xawtv/, xawtv} by Gerd Knorr. You also +have to set the audio recording levels correctly with a +standard mixer. + +@section X11 grabbing + +Grab the X11 display with ffmpeg via + +@example +ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg +@end example + +0.0 is display.screen number of your X11 server, same as +the DISPLAY environment variable. + +@example +ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg +@end example + +0.0 is display.screen number of your X11 server, same as the DISPLAY environment +variable. 10 is the x-offset and 20 the y-offset for the grabbing. + +@section Video and Audio file format conversion + +Any supported file format and protocol can serve as input to ffmpeg: + +Examples: +@itemize +@item +You can use YUV files as input: + +@example +ffmpeg -i /tmp/test%d.Y /tmp/out.mpg +@end example + +It will use the files: +@example +/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, +/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... +@end example + +The Y files use twice the resolution of the U and V files. They are +raw files, without header. They can be generated by all decent video +decoders. You must specify the size of the image with the @option{-s} option +if ffmpeg cannot guess it. + +@item +You can input from a raw YUV420P file: + +@example +ffmpeg -i /tmp/test.yuv /tmp/out.avi +@end example + +test.yuv is a file containing raw YUV planar data. Each frame is composed +of the Y plane followed by the U and V planes at half vertical and +horizontal resolution. + +@item +You can output to a raw YUV420P file: + +@example +ffmpeg -i mydivx.avi hugefile.yuv +@end example + +@item +You can set several input files and output files: + +@example +ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg +@end example + +Converts the audio file a.wav and the raw YUV video file a.yuv +to MPEG file a.mpg. + +@item +You can also do audio and video conversions at the same time: + +@example +ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 +@end example + +Converts a.wav to MPEG audio at 22050 Hz sample rate. + +@item +You can encode to several formats at the same time and define a +mapping from input stream to output streams: + +@example +ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2 +@end example + +Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map +file:index' specifies which input stream is used for each output +stream, in the order of the definition of output streams. + +@item +You can transcode decrypted VOBs: + +@example +ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi +@end example + +This is a typical DVD ripping example; the input is a VOB file, the +output an AVI file with MPEG-4 video and MP3 audio. Note that in this +command we use B-frames so the MPEG-4 stream is DivX5 compatible, and +GOP size is 300 which means one intra frame every 10 seconds for 29.97fps +input video. Furthermore, the audio stream is MP3-encoded so you need +to enable LAME support by passing @code{--enable-libmp3lame} to configure. +The mapping is particularly useful for DVD transcoding +to get the desired audio language. + +NOTE: To see the supported input formats, use @code{ffmpeg -formats}. + +@item +You can extract images from a video, or create a video from many images: + +For extracting images from a video: +@example +ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg +@end example + +This will extract one video frame per second from the video and will +output them in files named @file{foo-001.jpeg}, @file{foo-002.jpeg}, +etc. Images will be rescaled to fit the new WxH values. + +If you want to extract just a limited number of frames, you can use the +above command in combination with the -vframes or -t option, or in +combination with -ss to start extracting from a certain point in time. + +For creating a video from many images: +@example +ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi +@end example + +The syntax @code{foo-%03d.jpeg} specifies to use a decimal number +composed of three digits padded with zeroes to express the sequence +number. It is the same syntax supported by the C printf function, but +only formats accepting a normal integer are suitable. + +When importing an image sequence, -i also supports expanding +shell-like wildcard patterns (globbing) internally, by selecting the +image2-specific @code{-pattern_type glob} option. + +For example, for creating a video from filenames matching the glob pattern +@code{foo-*.jpeg}: +@example +ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi +@end example + +@item +You can put many streams of the same type in the output: + +@example +ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut +@end example + +The resulting output file @file{test12.avi} will contain first four streams from +the input file in reverse order. + +@item +To force CBR video output: +@example +ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v +@end example + +@item +The four options lmin, lmax, mblmin and mblmax use 'lambda' units, +but you may use the QP2LAMBDA constant to easily convert from 'q' units: +@example +ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext +@end example + +@end itemize +@c man end EXAMPLES + +@chapter See Also + +@ifhtml +@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg +@settitle ffmpeg video converter + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffmpeg.txt b/ffmpeg1/doc/ffmpeg.txt new file mode 100644 index 0000000..a028ca2 --- /dev/null +++ b/ffmpeg1/doc/ffmpeg.txt @@ -0,0 +1,47 @@ + : + ffmpeg.c : libav* + ======== : ====== + : + : + --------------------------------:---> AVStream... + InputStream input_streams[] / : + / : + InputFile input_files[] +==========================+ / ^ : + ------> 0 | : st ---:-----------:--/ : : + ^ +------+-----------+-----+ / +--------------------------+ : : + : | :ist_index--:-----:---------/ 1 | : st : | : : + : +------+-----------+-----+ +==========================+ : : + nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : : + : +------+-----------+-----+ +--------------------------+ : nb_input_streams : + : | :ist_index : | 3 | ... | : : + v +------+-----------+-----+ +--------------------------+ : : + --> 4 | | : : + | +--------------------------+ : : + | 5 | | : : + | +==========================+ v : + | : + | : + | : + | : + --------- --------------------------------:---> AVStream... + \ / : + OutputStream output_streams[] / : + \ / : + +======\======================/======+ ^ : + ------> 0 | : source_index : st-:--- | : : + OutputFile output_files[] / +------------------------------------+ : : + / 1 | : : : | : : + ^ +------+------------+-----+ / +------------------------------------+ : : + : | : ost_index -:-----:------/ 2 | : : : | : : + nb_output_files : +------+------------+-----+ +====================================+ : : + : | : ost_index -:-----|-----------------> 3 | : : : | : : + : +------+------------+-----+ +------------------------------------+ : nb_output_streams : + : | : : | 4 | | : : + : +------+------------+-----+ +------------------------------------+ : : + : | : : | 5 | | : : + v +------+------------+-----+ +------------------------------------+ : : + 6 | | : : + +------------------------------------+ : : + 7 | | : : + +====================================+ v : + : diff --git a/ffmpeg1/doc/ffplay.texi b/ffmpeg1/doc/ffplay.texi new file mode 100644 index 0000000..ee160a0 --- /dev/null +++ b/ffmpeg1/doc/ffplay.texi @@ -0,0 +1,235 @@ +\input texinfo @c -*- texinfo -*- + +@settitle ffplay Documentation +@titlepage +@center @titlefont{ffplay Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffplay [@var{options}] [@file{input_file}] + +@chapter Description +@c man begin DESCRIPTION + +FFplay is a very simple and portable media player using the FFmpeg +libraries and the SDL library. It is mostly used as a testbed for the +various FFmpeg APIs. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option +@item -x @var{width} +Force displayed width. +@item -y @var{height} +Force displayed height. +@item -s @var{size} +Set frame size (WxH or abbreviation), needed for videos which do +not contain a header with the frame size like raw YUV. This option +has been deprecated in favor of private options, try -video_size. +@item -an +Disable audio. +@item -vn +Disable video. +@item -ss @var{pos} +Seek to a given position in seconds. +@item -t @var{duration} +play <duration> seconds of audio/video +@item -bytes +Seek by bytes. +@item -nodisp +Disable graphical display. +@item -f @var{fmt} +Force format. +@item -window_title @var{title} +Set window title (default is the input filename). +@item -loop @var{number} +Loops movie playback <number> times. 0 means forever. +@item -showmode @var{mode} +Set the show mode to use. +Available values for @var{mode} are: +@table @samp +@item 0, video +show video +@item 1, waves +show audio waves +@item 2, rdft +show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform) +@end table + +Default value is "video", if video is not present or cannot be played +"rdft" is automatically selected. + +You can interactively cycle through the available show modes by +pressing the key @key{w}. + +@item -vf @var{filter_graph} +Create the filter graph specified by @var{filter_graph} and use it to +filter the video stream. + +@var{filter_graph} is a description of the filter graph to apply to +the stream, and must have a single video input and a single video +output. In the filter graph, the input is associated to the label +@code{in}, and the output to the label @code{out}. See the +ffmpeg-filters manual for more information about the filtergraph +syntax. + +@item -af @var{filter_graph} +@var{filter_graph} is a description of the filter graph to apply to +the input audio. +Use the option "-filters" to show all the available filters (including +sources and sinks). + +@item -i @var{input_file} +Read @var{input_file}. +@end table + +@section Advanced options +@table @option +@item -pix_fmt @var{format} +Set pixel format. +This option has been deprecated in favor of private options, try -pixel_format. + +@item -stats +Print several playback statistics, in particular show the stream +duration, the codec parameters, the current position in the stream and +the audio/video synchronisation drift. It is on by default, to +explicitly disable it you need to specify @code{-nostats}. + +@item -bug +Work around bugs. +@item -fast +Non-spec-compliant optimizations. +@item -genpts +Generate pts. +@item -rtp_tcp +Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful +if you are streaming with the RTSP protocol. +@item -sync @var{type} +Set the master clock to audio (@code{type=audio}), video +(@code{type=video}) or external (@code{type=ext}). Default is audio. The +master clock is used to control audio-video synchronization. Most media +players use audio as master clock, but in some cases (streaming or high +quality broadcast) it is necessary to change that. This option is mainly +used for debugging purposes. +@item -threads @var{count} +Set the thread count. +@item -ast @var{audio_stream_number} +Select the desired audio stream number, counting from 0. The number +refers to the list of all the input audio streams. If it is greater +than the number of audio streams minus one, then the last one is +selected, if it is negative the audio playback is disabled. +@item -vst @var{video_stream_number} +Select the desired video stream number, counting from 0. The number +refers to the list of all the input video streams. If it is greater +than the number of video streams minus one, then the last one is +selected, if it is negative the video playback is disabled. +@item -sst @var{subtitle_stream_number} +Select the desired subtitle stream number, counting from 0. The number +refers to the list of all the input subtitle streams. If it is greater +than the number of subtitle streams minus one, then the last one is +selected, if it is negative the subtitle rendering is disabled. +@item -autoexit +Exit when video is done playing. +@item -exitonkeydown +Exit if any key is pressed. +@item -exitonmousedown +Exit if any mouse button is pressed. + +@item -codec:@var{media_specifier} @var{codec_name} +Force a specific decoder implementation for the stream identified by +@var{media_specifier}, which can assume the values @code{a} (audio), +@code{v} (video), and @code{s} subtitle. + +@item -acodec @var{codec_name} +Force a specific audio decoder. + +@item -vcodec @var{codec_name} +Force a specific video decoder. + +@item -scodec @var{codec_name} +Force a specific subtitle decoder. +@end table + +@section While playing + +@table @key +@item q, ESC +Quit. + +@item f +Toggle full screen. + +@item p, SPC +Pause. + +@item a +Cycle audio channel. + +@item v +Cycle video channel. + +@item t +Cycle subtitle channel. + +@item w +Show audio waves. + +@item left/right +Seek backward/forward 10 seconds. + +@item down/up +Seek backward/forward 1 minute. + +@item page down/page up +Seek backward/forward 10 minutes. + +@item mouse click +Seek to percentage in file corresponding to fraction of width. + +@end table + +@c man end + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffplay +@settitle FFplay media player + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffprobe.texi b/ffmpeg1/doc/ffprobe.texi new file mode 100644 index 0000000..6e30b2f --- /dev/null +++ b/ffmpeg1/doc/ffprobe.texi @@ -0,0 +1,521 @@ +\input texinfo @c -*- texinfo -*- + +@settitle ffprobe Documentation +@titlepage +@center @titlefont{ffprobe Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffprobe [@var{options}] [@file{input_file}] + +@chapter Description +@c man begin DESCRIPTION + +ffprobe gathers information from multimedia streams and prints it in +human- and machine-readable fashion. + +For example it can be used to check the format of the container used +by a multimedia stream and the format and type of each media stream +contained in it. + +If a filename is specified in input, ffprobe will try to open and +probe the file content. If the file cannot be opened or recognized as +a multimedia file, a positive exit code is returned. + +ffprobe may be employed both as a standalone application or in +combination with a textual filter, which may perform more +sophisticated processing, e.g. statistical processing or plotting. + +Options are used to list some of the formats supported by ffprobe or +for specifying which information to display, and for setting how +ffprobe will show it. + +ffprobe output is designed to be easily parsable by a textual filter, +and consists of one or more sections of a form defined by the selected +writer, which is specified by the @option{print_format} option. + +Sections may contain other nested sections, and are identified by a +name (which may be shared by other sections), and an unique +name. See the output of @option{sections}. + +Metadata tags stored in the container or in the streams are recognized +and printed in the corresponding "FORMAT" or "STREAM" section. + +@c man end + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option + +@item -f @var{format} +Force format to use. + +@item -unit +Show the unit of the displayed values. + +@item -prefix +Use SI prefixes for the displayed values. +Unless the "-byte_binary_prefix" option is used all the prefixes +are decimal. + +@item -byte_binary_prefix +Force the use of binary prefixes for byte values. + +@item -sexagesimal +Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. + +@item -pretty +Prettify the format of the displayed values, it corresponds to the +options "-unit -prefix -byte_binary_prefix -sexagesimal". + +@item -of, -print_format @var{writer_name}[=@var{writer_options}] +Set the output printing format. + +@var{writer_name} specifies the name of the writer, and +@var{writer_options} specifies the options to be passed to the writer. + +For example for printing the output in JSON format, specify: +@example +-print_format json +@end example + +For more details on the available output printing formats, see the +Writers section below. + +@item -sections +Print sections structure and section information, and exit. The output +is not meant to be parsed by a machine. + +@item -select_streams @var{stream_specifier} +Select only the streams specified by @var{stream_specifier}. This +option affects only the options related to streams +(e.g. @code{show_streams}, @code{show_packets}, etc.). + +For example to show only audio streams, you can use the command: +@example +ffprobe -show_streams -select_streams a INPUT +@end example + +To show only video packets belonging to the video stream with index 1: +@example +ffprobe -show_packets -select_streams v:1 INPUT +@end example + +@item -show_data +Show payload data, as an hexadecimal and ASCII dump. Coupled with +@option{-show_packets}, it will dump the packets' data. Coupled with +@option{-show_streams}, it will dump the codec extradata. + +The dump is printed as the "data" field. It may contain newlines. + +@item -show_error +Show information about the error found when trying to probe the input. + +The error information is printed within a section with name "ERROR". + +@item -show_format +Show information about the container format of the input multimedia +stream. + +All the container format information is printed within a section with +name "FORMAT". + +@item -show_format_entry @var{name} +Like @option{-show_format}, but only prints the specified entry of the +container format information, rather than all. This option may be given more +than once, then all specified entries will be shown. + +This option is deprecated, use @code{show_entries} instead. + +@item -show_entries @var{section_entries} +Set list of entries to show. + +Entries are specified according to the following +syntax. @var{section_entries} contains a list of section entries +separated by @code{:}. Each section entry is composed by a section +name (or unique name), optionally followed by a list of entries local +to that section, separated by @code{,}. + +If section name is specified but is followed by no @code{=}, all +entries are printed to output, together with all the contained +sections. Otherwise only the entries specified in the local section +entries list are printed. In particular, if @code{=} is specified but +the list of local entries is empty, then no entries will be shown for +that section. + +Note that the order of specification of the local section entries is +not honored in the output, and the usual display order will be +retained. + +The formal syntax is given by: +@example +@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}] +@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]] +@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}] +@end example + +For example, to show only the index and type of each stream, and the PTS +time, duration time, and stream index of the packets, you can specify +the argument: +@example +packet=pts_time,duration_time,stream_index : stream=index,codec_type +@end example + +To show all the entries in the section "format", but only the codec +type in the section "stream", specify the argument: +@example +format : stream=codec_type +@end example + +To show all the tags in the stream and format sections: +@example +format_tags : format_tags +@end example + +To show only the @code{title} tag (if available) in the stream +sections: +@example +stream_tags=title +@end example + +@item -show_packets +Show information about each packet contained in the input multimedia +stream. + +The information for each single packet is printed within a dedicated +section with name "PACKET". + +@item -show_frames +Show information about each frame contained in the input multimedia +stream. + +The information for each single frame is printed within a dedicated +section with name "FRAME". + +@item -show_streams +Show information about each media stream contained in the input +multimedia stream. + +Each media stream information is printed within a dedicated section +with name "STREAM". + +@item -count_frames +Count the number of frames per stream and report it in the +corresponding stream section. + +@item -count_packets +Count the number of packets per stream and report it in the +corresponding stream section. + +@item -show_private_data, -private +Show private data, that is data depending on the format of the +particular shown element. +This option is enabled by default, but you may need to disable it +for specific uses, for example when creating XSD-compliant XML output. + +@item -show_program_version +Show information related to program version. + +Version information is printed within a section with name +"PROGRAM_VERSION". + +@item -show_library_versions +Show information related to library versions. + +Version information for each library is printed within a section with +name "LIBRARY_VERSION". + +@item -show_versions +Show information related to program and library versions. This is the +equivalent of setting both @option{-show_program_version} and +@option{-show_library_versions} options. + +@item -bitexact +Force bitexact output, useful to produce output which is not dependent +on the specific build. + +@item -i @var{input_file} +Read @var{input_file}. + +@end table +@c man end + +@chapter Writers +@c man begin WRITERS + +A writer defines the output format adopted by @command{ffprobe}, and will be +used for printing all the parts of the output. + +A writer may accept one or more arguments, which specify the options +to adopt. The options are specified as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the currently available writers follows. + +@section default +Default format. + +Print each section in the form: +@example +[SECTION] +key1=val1 +... +keyN=valN +[/SECTION] +@end example + +Metadata tags are printed as a line in the corresponding FORMAT or +STREAM section, and are prefixed by the string "TAG:". + +A description of the accepted options follows. + +@table @option + +@item nokey, nk +If set to 1 specify not to print the key of each field. Default value +is 0. + +@item noprint_wrappers, nw +If set to 1 specify not to print the section header and footer. +Default value is 0. +@end table + +@section compact, csv +Compact and CSV format. + +The @code{csv} writer is equivalent to @code{compact}, but supports +different defaults. + +Each section is printed on a single line. +If no option is specifid, the output has the form: +@example +section|key1=val1| ... |keyN=valN +@end example + +Metadata tags are printed in the corresponding "format" or "stream" +section. A metadata tag key, if printed, is prefixed by the string +"tag:". + +The description of the accepted options follows. + +@table @option + +@item item_sep, s +Specify the character to use for separating fields in the output line. +It must be a single printable character, it is "|" by default ("," for +the @code{csv} writer). + +@item nokey, nk +If set to 1 specify not to print the key of each field. Its default +value is 0 (1 for the @code{csv} writer). + +@item escape, e +Set the escape mode to use, default to "c" ("csv" for the @code{csv} +writer). + +It can assume one of the following values: +@table @option +@item c +Perform C-like escaping. Strings containing a newline ('\n'), carriage +return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping +character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned +escaping, so that a newline is converted to the sequence "\n", a +carriage return to "\r", '\' to "\\" and the separator @var{SEP} is +converted to "\@var{SEP}". + +@item csv +Perform CSV-like escaping, as described in RFC4180. Strings +containing a newline ('\n'), a carriage return ('\r'), a double quote +('"'), or @var{SEP} are enclosed in double-quotes. + +@item none +Perform no escaping. +@end table + +@item print_section, p +Print the section name at the begin of each line if the value is +@code{1}, disable it with value set to @code{0}. Default value is +@code{1}. + +@end table + +@section flat +Flat format. + +A free-form output where each line contains an explicit key=value, such as +"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be +directly embedded in sh scripts as long as the separator character is an +alphanumeric character or an underscore (see @var{sep_char} option). + +The description of the accepted options follows. + +@table @option +@item sep_char, s +Separator character used to separate the chapter, the section name, IDs and +potential tags in the printed field key. + +Default value is '.'. + +@item hierarchical, h +Specify if the section name specification should be hierarchical. If +set to 1, and if there is more than one section in the current +chapter, the section name will be prefixed by the name of the +chapter. A value of 0 will disable this behavior. + +Default value is 1. +@end table + +@section ini +INI format output. + +Print output in an INI based format. + +The following conventions are adopted: + +@itemize +@item +all key and values are UTF-8 +@item +'.' is the subgroup separator +@item +newline, '\t', '\f', '\b' and the following characters are escaped +@item +'\' is the escape character +@item +'#' is the comment indicator +@item +'=' is the key/value separator +@item +':' is not used but usually parsed as key/value separator +@end itemize + +This writer accepts options as a list of @var{key}=@var{value} pairs, +separated by ":". + +The description of the accepted options follows. + +@table @option +@item hierarchical, h +Specify if the section name specification should be hierarchical. If +set to 1, and if there is more than one section in the current +chapter, the section name will be prefixed by the name of the +chapter. A value of 0 will disable this behavior. + +Default value is 1. +@end table + +@section json +JSON based format. + +Each section is printed using JSON notation. + +The description of the accepted options follows. + +@table @option + +@item compact, c +If set to 1 enable compact output, that is each section will be +printed on a single line. Default value is 0. +@end table + +For more information about JSON, see @url{http://www.json.org/}. + +@section xml +XML based format. + +The XML output is described in the XML schema description file +@file{ffprobe.xsd} installed in the FFmpeg datadir. + +An updated version of the schema can be retrieved at the url +@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the +latest schema committed into the FFmpeg development source code tree. + +Note that the output issued will be compliant to the +@file{ffprobe.xsd} schema only when no special global output options +(@option{unit}, @option{prefix}, @option{byte_binary_prefix}, +@option{sexagesimal} etc.) are specified. + +The description of the accepted options follows. + +@table @option + +@item fully_qualified, q +If set to 1 specify if the output should be fully qualified. Default +value is 0. +This is required for generating an XML file which can be validated +through an XSD file. + +@item xsd_compliant, x +If set to 1 perform more checks for ensuring that the output is XSD +compliant. Default value is 0. +This option automatically sets @option{fully_qualified} to 1. +@end table + +For more information about the XML format, see +@url{http://www.w3.org/XML/}. +@c man end WRITERS + +@chapter Timecode +@c man begin TIMECODE + +@command{ffprobe} supports Timecode extraction: + +@itemize + +@item +MPEG1/2 timecode is extracted from the GOP, and is available in the video +stream details (@option{-show_streams}, see @var{timecode}). + +@item +MOV timecode is extracted from tmcd track, so is available in the tmcd +stream metadata (@option{-show_streams}, see @var{TAG:timecode}). + +@item +DV, GXF and AVI timecodes are available in format metadata +(@option{-show_format}, see @var{TAG:timecode}). + +@end itemize +@c man end TIMECODE + +@chapter See Also + +@ifhtml +@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffprobe +@settitle ffprobe media prober + +@end ignore + +@bye diff --git a/ffmpeg1/doc/ffprobe.xsd b/ffmpeg1/doc/ffprobe.xsd new file mode 100644 index 0000000..eab97fb --- /dev/null +++ b/ffmpeg1/doc/ffprobe.xsd @@ -0,0 +1,198 @@ +<?xml version="1.0" encoding="UTF-8"?> + +<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema" + targetNamespace="http://www.ffmpeg.org/schema/ffprobe" + xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe"> + + <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/> + + <xsd:complexType name="ffprobeType"> + <xsd:sequence> + <xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" /> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="packetsType"> + <xsd:sequence> + <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="framesType"> + <xsd:sequence> + <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="packetType"> + <xsd:attribute name="codec_type" type="xsd:string" use="required" /> + <xsd:attribute name="stream_index" type="xsd:int" use="required" /> + <xsd:attribute name="pts" type="xsd:long" /> + <xsd:attribute name="pts_time" type="xsd:float" /> + <xsd:attribute name="dts" type="xsd:long" /> + <xsd:attribute name="dts_time" type="xsd:float" /> + <xsd:attribute name="duration" type="xsd:long" /> + <xsd:attribute name="duration_time" type="xsd:float" /> + <xsd:attribute name="convergence_duration" type="xsd:long" /> + <xsd:attribute name="convergence_duration_time" type="xsd:float" /> + <xsd:attribute name="size" type="xsd:long" use="required" /> + <xsd:attribute name="pos" type="xsd:long" /> + <xsd:attribute name="flags" type="xsd:string" use="required" /> + <xsd:attribute name="data" type="xsd:string" /> + </xsd:complexType> + + <xsd:complexType name="frameType"> + <xsd:attribute name="media_type" type="xsd:string" use="required"/> + <xsd:attribute name="key_frame" type="xsd:int" use="required"/> + <xsd:attribute name="pts" type="xsd:long" /> + <xsd:attribute name="pts_time" type="xsd:float"/> + <xsd:attribute name="pkt_pts" type="xsd:long" /> + <xsd:attribute name="pkt_pts_time" type="xsd:float"/> + <xsd:attribute name="pkt_dts" type="xsd:long" /> + <xsd:attribute name="pkt_dts_time" type="xsd:float"/> + <xsd:attribute name="pkt_duration" type="xsd:long" /> + <xsd:attribute name="pkt_duration_time" type="xsd:float"/> + <xsd:attribute name="pkt_pos" type="xsd:long" /> + <xsd:attribute name="pkt_size" type="xsd:int" /> + + <!-- audio attributes --> + <xsd:attribute name="sample_fmt" type="xsd:string"/> + <xsd:attribute name="nb_samples" type="xsd:long" /> + <xsd:attribute name="channels" type="xsd:int" /> + <xsd:attribute name="channel_layout" type="xsd:string"/> + + <!-- video attributes --> + <xsd:attribute name="width" type="xsd:long" /> + <xsd:attribute name="height" type="xsd:long" /> + <xsd:attribute name="pix_fmt" type="xsd:string"/> + <xsd:attribute name="sample_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="pict_type" type="xsd:string"/> + <xsd:attribute name="coded_picture_number" type="xsd:long" /> + <xsd:attribute name="display_picture_number" type="xsd:long" /> + <xsd:attribute name="interlaced_frame" type="xsd:int" /> + <xsd:attribute name="top_field_first" type="xsd:int" /> + <xsd:attribute name="repeat_pict" type="xsd:int" /> + </xsd:complexType> + + <xsd:complexType name="streamsType"> + <xsd:sequence> + <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="streamDispositionType"> + <xsd:attribute name="default" type="xsd:int" use="required" /> + <xsd:attribute name="dub" type="xsd:int" use="required" /> + <xsd:attribute name="original" type="xsd:int" use="required" /> + <xsd:attribute name="comment" type="xsd:int" use="required" /> + <xsd:attribute name="lyrics" type="xsd:int" use="required" /> + <xsd:attribute name="karaoke" type="xsd:int" use="required" /> + <xsd:attribute name="forced" type="xsd:int" use="required" /> + <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" /> + <xsd:attribute name="visual_impaired" type="xsd:int" use="required" /> + <xsd:attribute name="clean_effects" type="xsd:int" use="required" /> + <xsd:attribute name="attached_pic" type="xsd:int" use="required" /> + </xsd:complexType> + + <xsd:complexType name="streamType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/> + </xsd:sequence> + + <xsd:attribute name="index" type="xsd:int" use="required"/> + <xsd:attribute name="codec_name" type="xsd:string" /> + <xsd:attribute name="codec_long_name" type="xsd:string" /> + <xsd:attribute name="profile" type="xsd:string" /> + <xsd:attribute name="codec_type" type="xsd:string" /> + <xsd:attribute name="codec_time_base" type="xsd:string" use="required"/> + <xsd:attribute name="codec_tag" type="xsd:string" use="required"/> + <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/> + <xsd:attribute name="extradata" type="xsd:string" /> + + <!-- video attributes --> + <xsd:attribute name="width" type="xsd:int"/> + <xsd:attribute name="height" type="xsd:int"/> + <xsd:attribute name="has_b_frames" type="xsd:int"/> + <xsd:attribute name="sample_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="display_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="pix_fmt" type="xsd:string"/> + <xsd:attribute name="level" type="xsd:int"/> + <xsd:attribute name="timecode" type="xsd:string"/> + + <!-- audio attributes --> + <xsd:attribute name="sample_fmt" type="xsd:string"/> + <xsd:attribute name="sample_rate" type="xsd:int"/> + <xsd:attribute name="channels" type="xsd:int"/> + <xsd:attribute name="bits_per_sample" type="xsd:int"/> + + <xsd:attribute name="id" type="xsd:string"/> + <xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/> + <xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/> + <xsd:attribute name="time_base" type="xsd:string" use="required"/> + <xsd:attribute name="start_pts" type="xsd:long"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="duration_ts" type="xsd:long"/> + <xsd:attribute name="duration" type="xsd:float"/> + <xsd:attribute name="bit_rate" type="xsd:int"/> + <xsd:attribute name="nb_frames" type="xsd:int"/> + <xsd:attribute name="nb_read_frames" type="xsd:int"/> + <xsd:attribute name="nb_read_packets" type="xsd:int"/> + </xsd:complexType> + + <xsd:complexType name="formatType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + + <xsd:attribute name="filename" type="xsd:string" use="required"/> + <xsd:attribute name="nb_streams" type="xsd:int" use="required"/> + <xsd:attribute name="format_name" type="xsd:string" use="required"/> + <xsd:attribute name="format_long_name" type="xsd:string"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="duration" type="xsd:float"/> + <xsd:attribute name="size" type="xsd:long"/> + <xsd:attribute name="bit_rate" type="xsd:long"/> + </xsd:complexType> + + <xsd:complexType name="tagType"> + <xsd:attribute name="key" type="xsd:string" use="required"/> + <xsd:attribute name="value" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="errorType"> + <xsd:attribute name="code" type="xsd:int" use="required"/> + <xsd:attribute name="string" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="programVersionType"> + <xsd:attribute name="version" type="xsd:string" use="required"/> + <xsd:attribute name="copyright" type="xsd:string" use="required"/> + <xsd:attribute name="build_date" type="xsd:string" use="required"/> + <xsd:attribute name="build_time" type="xsd:string" use="required"/> + <xsd:attribute name="compiler_type" type="xsd:string" use="required"/> + <xsd:attribute name="compiler_version" type="xsd:string" use="required"/> + <xsd:attribute name="configuration" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="libraryVersionType"> + <xsd:attribute name="name" type="xsd:string" use="required"/> + <xsd:attribute name="major" type="xsd:int" use="required"/> + <xsd:attribute name="minor" type="xsd:int" use="required"/> + <xsd:attribute name="micro" type="xsd:int" use="required"/> + <xsd:attribute name="version" type="xsd:int" use="required"/> + <xsd:attribute name="ident" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="libraryVersionsType"> + <xsd:sequence> + <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> +</xsd:schema> diff --git a/ffmpeg1/doc/ffserver.conf b/ffmpeg1/doc/ffserver.conf new file mode 100644 index 0000000..0f5922c --- /dev/null +++ b/ffmpeg1/doc/ffserver.conf @@ -0,0 +1,371 @@ +# Port on which the server is listening. You must select a different +# port from your standard HTTP web server if it is running on the same +# computer. +Port 8090 + +# Address on which the server is bound. Only useful if you have +# several network interfaces. +BindAddress 0.0.0.0 + +# Number of simultaneous HTTP connections that can be handled. It has +# to be defined *before* the MaxClients parameter, since it defines the +# MaxClients maximum limit. +MaxHTTPConnections 2000 + +# Number of simultaneous requests that can be handled. Since FFServer +# is very fast, it is more likely that you will want to leave this high +# and use MaxBandwidth, below. +MaxClients 1000 + +# This the maximum amount of kbit/sec that you are prepared to +# consume when streaming to clients. +MaxBandwidth 1000 + +# Access log file (uses standard Apache log file format) +# '-' is the standard output. +CustomLog - + +################################################################## +# Definition of the live feeds. Each live feed contains one video +# and/or audio sequence coming from an ffmpeg encoder or another +# ffserver. This sequence may be encoded simultaneously with several +# codecs at several resolutions. + +<Feed feed1.ffm> + +# You must use 'ffmpeg' to send a live feed to ffserver. In this +# example, you can type: +# +# ffmpeg http://localhost:8090/feed1.ffm + +# ffserver can also do time shifting. It means that it can stream any +# previously recorded live stream. The request should contain: +# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify +# a path where the feed is stored on disk. You also specify the +# maximum size of the feed, where zero means unlimited. Default: +# File=/tmp/feed_name.ffm FileMaxSize=5M +File /tmp/feed1.ffm +FileMaxSize 200K + +# You could specify +# ReadOnlyFile /saved/specialvideo.ffm +# This marks the file as readonly and it will not be deleted or updated. + +# Specify launch in order to start ffmpeg automatically. +# First ffmpeg must be defined with an appropriate path if needed, +# after that options can follow, but avoid adding the http:// field +#Launch ffmpeg + +# Only allow connections from localhost to the feed. +ACL allow 127.0.0.1 + +</Feed> + + +################################################################## +# Now you can define each stream which will be generated from the +# original audio and video stream. Each format has a filename (here +# 'test1.mpg'). FFServer will send this stream when answering a +# request containing this filename. + +<Stream test1.mpg> + +# coming from live feed 'feed1' +Feed feed1.ffm + +# Format of the stream : you can choose among: +# mpeg : MPEG-1 multiplexed video and audio +# mpegvideo : only MPEG-1 video +# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) +# ogg : Ogg format (Vorbis audio codec) +# rm : RealNetworks-compatible stream. Multiplexed audio and video. +# ra : RealNetworks-compatible stream. Audio only. +# mpjpeg : Multipart JPEG (works with Netscape without any plugin) +# jpeg : Generate a single JPEG image. +# asf : ASF compatible streaming (Windows Media Player format). +# swf : Macromedia Flash compatible stream +# avi : AVI format (MPEG-4 video, MPEG audio sound) +Format mpeg + +# Bitrate for the audio stream. Codecs usually support only a few +# different bitrates. +AudioBitRate 32 + +# Number of audio channels: 1 = mono, 2 = stereo +AudioChannels 1 + +# Sampling frequency for audio. When using low bitrates, you should +# lower this frequency to 22050 or 11025. The supported frequencies +# depend on the selected audio codec. +AudioSampleRate 44100 + +# Bitrate for the video stream +VideoBitRate 64 + +# Ratecontrol buffer size +VideoBufferSize 40 + +# Number of frames per second +VideoFrameRate 3 + +# Size of the video frame: WxH (default: 160x128) +# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga, +# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga, +# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720, +# hd1080 +VideoSize 160x128 + +# Transmit only intra frames (useful for low bitrates, but kills frame rate). +#VideoIntraOnly + +# If non-intra only, an intra frame is transmitted every VideoGopSize +# frames. Video synchronization can only begin at an intra frame. +VideoGopSize 12 + +# More MPEG-4 parameters +# VideoHighQuality +# Video4MotionVector + +# Choose your codecs: +#AudioCodec mp2 +#VideoCodec mpeg1video + +# Suppress audio +#NoAudio + +# Suppress video +#NoVideo + +#VideoQMin 3 +#VideoQMax 31 + +# Set this to the number of seconds backwards in time to start. Note that +# most players will buffer 5-10 seconds of video, and also you need to allow +# for a keyframe to appear in the data stream. +#Preroll 15 + +# ACL: + +# You can allow ranges of addresses (or single addresses) +#ACL ALLOW <first address> <last address> + +# You can deny ranges of addresses (or single addresses) +#ACL DENY <first address> <last address> + +# You can repeat the ACL allow/deny as often as you like. It is on a per +# stream basis. The first match defines the action. If there are no matches, +# then the default is the inverse of the last ACL statement. +# +# Thus 'ACL allow localhost' only allows access from localhost. +# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and +# allow everybody else. + +</Stream> + + +################################################################## +# Example streams + + +# Multipart JPEG + +#<Stream test.mjpg> +#Feed feed1.ffm +#Format mpjpeg +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#Strict -1 +#</Stream> + + +# Single JPEG + +#<Stream test.jpg> +#Feed feed1.ffm +#Format jpeg +#VideoFrameRate 2 +#VideoIntraOnly +##VideoSize 352x240 +#NoAudio +#Strict -1 +#</Stream> + + +# Flash + +#<Stream test.swf> +#Feed feed1.ffm +#Format swf +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#</Stream> + + +# ASF compatible + +<Stream test.asf> +Feed feed1.ffm +Format asf +VideoFrameRate 15 +VideoSize 352x240 +VideoBitRate 256 +VideoBufferSize 40 +VideoGopSize 30 +AudioBitRate 64 +StartSendOnKey +</Stream> + + +# MP3 audio + +#<Stream test.mp3> +#Feed feed1.ffm +#Format mp2 +#AudioCodec mp3 +#AudioBitRate 64 +#AudioChannels 1 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Ogg Vorbis audio + +#<Stream test.ogg> +#Feed feed1.ffm +#Title "Stream title" +#AudioBitRate 64 +#AudioChannels 2 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Real with audio only at 32 kbits + +#<Stream test.ra> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#NoVideo +#NoAudio +#</Stream> + + +# Real with audio and video at 64 kbits + +#<Stream test.rm> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#VideoBitRate 128 +#VideoFrameRate 25 +#VideoGopSize 25 +#NoAudio +#</Stream> + + +################################################################## +# A stream coming from a file: you only need to set the input +# filename and optionally a new format. Supported conversions: +# AVI -> ASF + +#<Stream file.rm> +#File "/usr/local/httpd/htdocs/tlive.rm" +#NoAudio +#</Stream> + +#<Stream file.asf> +#File "/usr/local/httpd/htdocs/test.asf" +#NoAudio +#Author "Me" +#Copyright "Super MegaCorp" +#Title "Test stream from disk" +#Comment "Test comment" +#</Stream> + + +################################################################## +# RTSP examples +# +# You can access this stream with the RTSP URL: +# rtsp://localhost:5454/test1-rtsp.mpg +# +# A non-standard RTSP redirector is also created. Its URL is: +# http://localhost:8090/test1-rtsp.rtsp + +#<Stream test1-rtsp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#</Stream> + + +# Transcode an incoming live feed to another live feed, +# using libx264 and video presets + +#<Stream live.h264> +#Format rtp +#Feed feed1.ffm +#VideoCodec libx264 +#VideoFrameRate 24 +#VideoBitRate 100 +#VideoSize 480x272 +#AVPresetVideo default +#AVPresetVideo baseline +#AVOptionVideo flags +global_header +# +#AudioCodec libfaac +#AudioBitRate 32 +#AudioChannels 2 +#AudioSampleRate 22050 +#AVOptionAudio flags +global_header +#</Stream> + +################################################################## +# SDP/multicast examples +# +# If you want to send your stream in multicast, you must set the +# multicast address with MulticastAddress. The port and the TTL can +# also be set. +# +# An SDP file is automatically generated by ffserver by adding the +# 'sdp' extension to the stream name (here +# http://localhost:8090/test1-sdp.sdp). You should usually give this +# file to your player to play the stream. +# +# The 'NoLoop' option can be used to avoid looping when the stream is +# terminated. + +#<Stream test1-sdp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#MulticastAddress 224.124.0.1 +#MulticastPort 5000 +#MulticastTTL 16 +#NoLoop +#</Stream> + + +################################################################## +# Special streams + +# Server status + +<Stream stat.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 + +#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico +</Stream> + + +# Redirect index.html to the appropriate site + +<Redirect index.html> +URL http://www.ffmpeg.org/ +</Redirect> diff --git a/ffmpeg1/doc/ffserver.texi b/ffmpeg1/doc/ffserver.texi new file mode 100644 index 0000000..f1b7599 --- /dev/null +++ b/ffmpeg1/doc/ffserver.texi @@ -0,0 +1,281 @@ +\input texinfo @c -*- texinfo -*- + +@settitle ffserver Documentation +@titlepage +@center @titlefont{ffserver Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffserver [@var{options}] + +@chapter Description +@c man begin DESCRIPTION + +@command{ffserver} is a streaming server for both audio and video. It +supports several live feeds, streaming from files and time shifting on +live feeds (you can seek to positions in the past on each live feed, +provided you specify a big enough feed storage in +@file{ffserver.conf}). + +@command{ffserver} receives prerecorded files or FFM streams from some +@command{ffmpeg} instance as input, then streams them over +RTP/RTSP/HTTP. + +An @command{ffserver} instance will listen on some port as specified +in the configuration file. You can launch one or more instances of +@command{ffmpeg} and send one or more FFM streams to the port where +ffserver is expecting to receive them. Alternately, you can make +@command{ffserver} launch such @command{ffmpeg} instances at startup. + +Input streams are called feeds, and each one is specified by a +@code{<Feed>} section in the configuration file. + +For each feed you can have different output streams in various +formats, each one specified by a @code{<Stream>} section in the +configuration file. + +@section Status stream + +ffserver supports an HTTP interface which exposes the current status +of the server. + +Simply point your browser to the address of the special status stream +specified in the configuration file. + +For example if you have: +@example +<Stream status.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 +</Stream> +@end example + +then the server will post a page with the status information when +the special stream @file{status.html} is requested. + +@section What can this do? + +When properly configured and running, you can capture video and audio in real +time from a suitable capture card, and stream it out over the Internet to +either Windows Media Player or RealAudio player (with some restrictions). + +It can also stream from files, though that is currently broken. Very often, a +web server can be used to serve up the files just as well. + +It can stream prerecorded video from .ffm files, though it is somewhat tricky +to make it work correctly. + +@section How do I make it work? + +First, build the kit. It *really* helps to have installed LAME first. Then when +you run the ffserver ./configure, make sure that you have the +@code{--enable-libmp3lame} flag turned on. + +LAME is important as it allows for streaming audio to Windows Media Player. +Don't ask why the other audio types do not work. + +As a simple test, just run the following two command lines where INPUTFILE +is some file which you can decode with ffmpeg: + +@example +ffserver -f doc/ffserver.conf & +ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm +@end example + +At this point you should be able to go to your Windows machine and fire up +Windows Media Player (WMP). Go to Open URL and enter + +@example + http://<linuxbox>:8090/test.asf +@end example + +You should (after a short delay) see video and hear audio. + +WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to +transfer the entire file before starting to play. +The same is true of AVI files. + +@section What happens next? + +You should edit the ffserver.conf file to suit your needs (in terms of +frame rates etc). Then install ffserver and ffmpeg, write a script to start +them up, and off you go. + +@section Troubleshooting + +@subsection I don't hear any audio, but video is fine. + +Maybe you didn't install LAME, or got your ./configure statement wrong. Check +the ffmpeg output to see if a line referring to MP3 is present. If not, then +your configuration was incorrect. If it is, then maybe your wiring is not +set up correctly. Maybe the sound card is not getting data from the right +input source. Maybe you have a really awful audio interface (like I do) +that only captures in stereo and also requires that one channel be flipped. +If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before +starting ffmpeg. + +@subsection The audio and video lose sync after a while. + +Yes, they do. + +@subsection After a long while, the video update rate goes way down in WMP. + +Yes, it does. Who knows why? + +@subsection WMP 6.4 behaves differently to WMP 7. + +Yes, it does. Any thoughts on this would be gratefully received. These +differences extend to embedding WMP into a web page. [There are two +object IDs that you can use: The old one, which does not play well, and +the new one, which does (both tested on the same system). However, +I suspect that the new one is not available unless you have installed WMP 7]. + +@section What else can it do? + +You can replay video from .ffm files that was recorded earlier. +However, there are a number of caveats, including the fact that the +ffserver parameters must match the original parameters used to record the +file. If they do not, then ffserver deletes the file before recording into it. +(Now that I write this, it seems broken). + +You can fiddle with many of the codec choices and encoding parameters, and +there are a bunch more parameters that you cannot control. Post a message +to the mailing list if there are some 'must have' parameters. Look in +ffserver.conf for a list of the currently available controls. + +It will automatically generate the ASX or RAM files that are often used +in browsers. These files are actually redirections to the underlying ASF +or RM file. The reason for this is that the browser often fetches the +entire file before starting up the external viewer. The redirection files +are very small and can be transferred quickly. [The stream itself is +often 'infinite' and thus the browser tries to download it and never +finishes.] + +@section Tips + +* When you connect to a live stream, most players (WMP, RA, etc) want to +buffer a certain number of seconds of material so that they can display the +signal continuously. However, ffserver (by default) starts sending data +in realtime. This means that there is a pause of a few seconds while the +buffering is being done by the player. The good news is that this can be +cured by adding a '?buffer=5' to the end of the URL. This means that the +stream should start 5 seconds in the past -- and so the first 5 seconds +of the stream are sent as fast as the network will allow. It will then +slow down to real time. This noticeably improves the startup experience. + +You can also add a 'Preroll 15' statement into the ffserver.conf that will +add the 15 second prebuffering on all requests that do not otherwise +specify a time. In addition, ffserver will skip frames until a key_frame +is found. This further reduces the startup delay by not transferring data +that will be discarded. + +* You may want to adjust the MaxBandwidth in the ffserver.conf to limit +the amount of bandwidth consumed by live streams. + +@section Why does the ?buffer / Preroll stop working after a time? + +It turns out that (on my machine at least) the number of frames successfully +grabbed is marginally less than the number that ought to be grabbed. This +means that the timestamp in the encoded data stream gets behind realtime. +This means that if you say 'Preroll 10', then when the stream gets 10 +or more seconds behind, there is no Preroll left. + +Fixing this requires a change in the internals of how timestamps are +handled. + +@section Does the @code{?date=} stuff work. + +Yes (subject to the limitation outlined above). Also note that whenever you +start ffserver, it deletes the ffm file (if any parameters have changed), +thus wiping out what you had recorded before. + +The format of the @code{?date=xxxxxx} is fairly flexible. You should use one +of the following formats (the 'T' is literal): + +@example +* YYYY-MM-DDTHH:MM:SS (localtime) +* YYYY-MM-DDTHH:MM:SSZ (UTC) +@end example + +You can omit the YYYY-MM-DD, and then it refers to the current day. However +note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this +may be in the future and so is unlikely to be useful. + +You use this by adding the ?date= to the end of the URL for the stream. +For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. +@c man end + +@section What is FFM, FFM2 + +FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of +video and audio streams and encoding options, and can store a moving time segment +of an infinite movie or a whole movie. + +FFM is version specific, and there is limited compatibility of FFM files +generated by one version of ffmpeg/ffserver and another version of +ffmpeg/ffserver. It may work but it is not guaranteed to work. + +FFM2 is extensible while maintaining compatibility and should work between +differing versions of tools. FFM2 is the default. + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option +@item -f @var{configfile} +Use @file{configfile} instead of @file{/etc/ffserver.conf}. +@item -n +Enable no-launch mode. This option disables all the Launch directives +within the various <Stream> sections. Since ffserver will not launch +any ffmpeg instances, you will have to launch them manually. +@item -d +Enable debug mode. This option increases log verbosity, directs log +messages to stdout. +@end table +@c man end + +@chapter See Also + +@ifhtml +The @file{doc/ffserver.conf} example, +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffserver +@settitle ffserver video server + +@end ignore + +@bye diff --git a/ffmpeg1/doc/filter_design.txt b/ffmpeg1/doc/filter_design.txt new file mode 100644 index 0000000..772ca9d --- /dev/null +++ b/ffmpeg1/doc/filter_design.txt @@ -0,0 +1,265 @@ +Filter design +============= + +This document explains guidelines that should be observed (or ignored with +good reason) when writing filters for libavfilter. + +In this document, the word “frame” indicates either a video frame or a group +of audio samples, as stored in an AVFilterBuffer structure. + + +Format negotiation +================== + + The query_formats method should set, for each input and each output links, + the list of supported formats. + + For video links, that means pixel format. For audio links, that means + channel layout, sample format (the sample packing is implied by the sample + format) and sample rate. + + The lists are not just lists, they are references to shared objects. When + the negotiation mechanism computes the intersection of the formats + supported at each end of a link, all references to both lists are replaced + with a reference to the intersection. And when a single format is + eventually chosen for a link amongst the remaining list, again, all + references to the list are updated. + + That means that if a filter requires that its input and output have the + same format amongst a supported list, all it has to do is use a reference + to the same list of formats. + + +Buffer references ownership and permissions +=========================================== + + Principle + --------- + + Audio and video data are voluminous; the buffer and buffer reference + mechanism is intended to avoid, as much as possible, expensive copies of + that data while still allowing the filters to produce correct results. + + The data is stored in buffers represented by AVFilterBuffer structures. + They must not be accessed directly, but through references stored in + AVFilterBufferRef structures. Several references can point to the + same buffer; the buffer is automatically deallocated once all + corresponding references have been destroyed. + + The characteristics of the data (resolution, sample rate, etc.) are + stored in the reference; different references for the same buffer can + show different characteristics. In particular, a video reference can + point to only a part of a video buffer. + + A reference is usually obtained as input to the start_frame or + filter_frame method or requested using the ff_get_video_buffer or + ff_get_audio_buffer functions. A new reference on an existing buffer can + be created with the avfilter_ref_buffer. A reference is destroyed using + the avfilter_unref_bufferp function. + + Reference ownership + ------------------- + + At any time, a reference “belongs” to a particular piece of code, + usually a filter. With a few caveats that will be explained below, only + that piece of code is allowed to access it. It is also responsible for + destroying it, although this is sometimes done automatically (see the + section on link reference fields). + + Here are the (fairly obvious) rules for reference ownership: + + * A reference received by the filter_frame method (or its start_frame + deprecated version) belongs to the corresponding filter. + + Special exception: for video references: the reference may be used + internally for automatic copying and must not be destroyed before + end_frame; it can be given away to ff_start_frame. + + * A reference passed to ff_filter_frame (or the deprecated + ff_start_frame) is given away and must no longer be used. + + * A reference created with avfilter_ref_buffer belongs to the code that + created it. + + * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer + belongs to the code that requested it. + + * A reference given as return value by the get_video_buffer or + get_audio_buffer method is given away and must no longer be used. + + Link reference fields + --------------------- + + The AVFilterLink structure has a few AVFilterBufferRef fields. The + cur_buf and out_buf were used with the deprecated + start_frame/draw_slice/end_frame API and should no longer be used. + src_buf, cur_buf_copy and partial_buf are used by libavfilter internally + and must not be accessed by filters. + + Reference permissions + --------------------- + + The AVFilterBufferRef structure has a perms field that describes what + the code that owns the reference is allowed to do to the buffer data. + Different references for the same buffer can have different permissions. + + For video filters that implement the deprecated + start_frame/draw_slice/end_frame API, the permissions only apply to the + parts of the buffer that have already been covered by the draw_slice + method. + + The value is a binary OR of the following constants: + + * AV_PERM_READ: the owner can read the buffer data; this is essentially + always true and is there for self-documentation. + + * AV_PERM_WRITE: the owner can modify the buffer data. + + * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data + will not be modified by previous filters. + + * AV_PERM_REUSE: the owner can output the buffer several times, without + modifying the data in between. + + * AV_PERM_REUSE2: the owner can output the buffer several times and + modify the data in between (useless without the WRITE permissions). + + * AV_PERM_ALIGN: the owner can access the data using fast operations + that require data alignment. + + The READ, WRITE and PRESERVE permissions are about sharing the same + buffer between several filters to avoid expensive copies without them + doing conflicting changes on the data. + + The REUSE and REUSE2 permissions are about special memory for direct + rendering. For example a buffer directly allocated in video memory must + not modified once it is displayed on screen, or it will cause tearing; + it will therefore not have the REUSE2 permission. + + The ALIGN permission is about extracting part of the buffer, for + copy-less padding or cropping for example. + + + References received on input pads are guaranteed to have all the + permissions stated in the min_perms field and none of the permissions + stated in the rej_perms. + + References obtained by ff_get_video_buffer and ff_get_audio_buffer are + guaranteed to have at least all the permissions requested as argument. + + References created by avfilter_ref_buffer have the same permissions as + the original reference minus the ones explicitly masked; the mask is + usually ~0 to keep the same permissions. + + Filters should remove permissions on reference they give to output + whenever necessary. It can be automatically done by setting the + rej_perms field on the output pad. + + Here are a few guidelines corresponding to common situations: + + * Filters that modify and forward their frame (like drawtext) need the + WRITE permission. + + * Filters that read their input to produce a new frame on output (like + scale) need the READ permission on input and and must request a buffer + with the WRITE permission. + + * Filters that intend to keep a reference after the filtering process + is finished (after filter_frame returns) must have the PRESERVE + permission on it and remove the WRITE permission if they create a new + reference to give it away. + + * Filters that intend to modify a reference they have kept after the end + of the filtering process need the REUSE2 permission and must remove + the PRESERVE permission if they create a new reference to give it + away. + + +Frame scheduling +================ + + The purpose of these rules is to ensure that frames flow in the filter + graph without getting stuck and accumulating somewhere. + + Simple filters that output one frame for each input frame should not have + to worry about it. + + filter_frame + ------------ + + This method is called when a frame is pushed to the filter's input. It + can be called at any time except in a reentrant way. + + If the input frame is enough to produce output, then the filter should + push the output frames on the output link immediately. + + As an exception to the previous rule, if the input frame is enough to + produce several output frames, then the filter needs output only at + least one per link. The additional frames can be left buffered in the + filter; these buffered frames must be flushed immediately if a new input + produces new output. + + (Example: framerate-doubling filter: filter_frame must (1) flush the + second copy of the previous frame, if it is still there, (2) push the + first copy of the incoming frame, (3) keep the second copy for later.) + + If the input frame is not enough to produce output, the filter must not + call request_frame to get more. It must just process the frame or queue + it. The task of requesting more frames is left to the filter's + request_frame method or the application. + + If a filter has several inputs, the filter must be ready for frames + arriving randomly on any input. Therefore, any filter with several inputs + will most likely require some kind of queuing mechanism. It is perfectly + acceptable to have a limited queue and to drop frames when the inputs + are too unbalanced. + + request_frame + ------------- + + This method is called when a frame is wanted on an output. + + For an input, it should directly call filter_frame on the corresponding + output. + + For a filter, if there are queued frames already ready, one of these + frames should be pushed. If not, the filter should request a frame on + one of its inputs, repeatedly until at least one frame has been pushed. + + Return values: + if request_frame could produce a frame, it should return 0; + if it could not for temporary reasons, it should return AVERROR(EAGAIN); + if it could not because there are no more frames, it should return + AVERROR_EOF. + + The typical implementation of request_frame for a filter with several + inputs will look like that: + + if (frames_queued) { + push_one_frame(); + return 0; + } + while (!frame_pushed) { + input = input_where_a_frame_is_most_needed(); + ret = ff_request_frame(input); + if (ret == AVERROR_EOF) { + process_eof_on_input(); + } else if (ret < 0) { + return ret; + } + } + return 0; + + Note that, except for filters that can have queued frames, request_frame + does not push frames: it requests them to its input, and as a reaction, + the filter_frame method will be called and do the work. + +Legacy API +========== + + Until libavfilter 3.23, the filter_frame method was split: + + - for video filters, it was made of start_frame, draw_slice (that could be + called several times on distinct parts of the frame) and end_frame; + + - for audio filters, it was called filter_samples. diff --git a/ffmpeg1/doc/filters.texi b/ffmpeg1/doc/filters.texi new file mode 100644 index 0000000..74a682a --- /dev/null +++ b/ffmpeg1/doc/filters.texi @@ -0,0 +1,7034 @@ +@chapter Filtering Introduction +@c man begin FILTERING INTRODUCTION + +Filtering in FFmpeg is enabled through the libavfilter library. + +In libavfilter, it is possible for filters to have multiple inputs and +multiple outputs. +To illustrate the sorts of things that are possible, we can +use a complex filter graph. For example, the following one: + +@example +input --> split ---------------------> overlay --> output + | ^ + | | + +-----> crop --> vflip -------+ +@end example + +splits the stream in two streams, sends one stream through the crop filter +and the vflip filter before merging it back with the other stream by +overlaying it on top. You can use the following command to achieve this: + +@example +ffmpeg -i input -vf "[in] split [T1], [T2] overlay=0:H/2 [out]; [T1] crop=iw:ih/2:0:ih/2, vflip [T2]" output +@end example + +The result will be that in output the top half of the video is mirrored +onto the bottom half. + +Filters are loaded using the @var{-vf} or @var{-af} option passed to +@command{ffmpeg} or to @command{ffplay}. Filters in the same linear +chain are separated by commas. In our example, @var{split, +overlay} are in one linear chain, and @var{crop, vflip} are in +another. The points where the linear chains join are labeled by names +enclosed in square brackets. In our example, that is @var{[T1]} and +@var{[T2]}. The special labels @var{[in]} and @var{[out]} are the points +where video is input and output. + +Some filters take in input a list of parameters: they are specified +after the filter name and an equal sign, and are separated from each other +by a colon. + +There exist so-called @var{source filters} that do not have an +audio/video input, and @var{sink filters} that will not have audio/video +output. + +@c man end FILTERING INTRODUCTION + +@chapter graph2dot +@c man begin GRAPH2DOT + +The @file{graph2dot} program included in the FFmpeg @file{tools} +directory can be used to parse a filter graph description and issue a +corresponding textual representation in the dot language. + +Invoke the command: +@example +graph2dot -h +@end example + +to see how to use @file{graph2dot}. + +You can then pass the dot description to the @file{dot} program (from +the graphviz suite of programs) and obtain a graphical representation +of the filter graph. + +For example the sequence of commands: +@example +echo @var{GRAPH_DESCRIPTION} | \ +tools/graph2dot -o graph.tmp && \ +dot -Tpng graph.tmp -o graph.png && \ +display graph.png +@end example + +can be used to create and display an image representing the graph +described by the @var{GRAPH_DESCRIPTION} string. Note that this string must be +a complete self-contained graph, with its inputs and outputs explicitly defined. +For example if your command line is of the form: +@example +ffmpeg -i infile -vf scale=640:360 outfile +@end example +your @var{GRAPH_DESCRIPTION} string will need to be of the form: +@example +nullsrc,scale=640:360,nullsink +@end example +you may also need to set the @var{nullsrc} parameters and add a @var{format} +filter in order to simulate a specific input file. + +@c man end GRAPH2DOT + +@chapter Filtergraph description +@c man begin FILTERGRAPH DESCRIPTION + +A filtergraph is a directed graph of connected filters. It can contain +cycles, and there can be multiple links between a pair of +filters. Each link has one input pad on one side connecting it to one +filter from which it takes its input, and one output pad on the other +side connecting it to the one filter accepting its output. + +Each filter in a filtergraph is an instance of a filter class +registered in the application, which defines the features and the +number of input and output pads of the filter. + +A filter with no input pads is called a "source", a filter with no +output pads is called a "sink". + +@anchor{Filtergraph syntax} +@section Filtergraph syntax + +A filtergraph can be represented using a textual representation, which is +recognized by the @option{-filter}/@option{-vf} and @option{-filter_complex} +options in @command{ffmpeg} and @option{-vf} in @command{ffplay}, and by the +@code{avfilter_graph_parse()}/@code{avfilter_graph_parse2()} function defined in +@file{libavfilter/avfiltergraph.h}. + +A filterchain consists of a sequence of connected filters, each one +connected to the previous one in the sequence. A filterchain is +represented by a list of ","-separated filter descriptions. + +A filtergraph consists of a sequence of filterchains. A sequence of +filterchains is represented by a list of ";"-separated filterchain +descriptions. + +A filter is represented by a string of the form: +[@var{in_link_1}]...[@var{in_link_N}]@var{filter_name}=@var{arguments}[@var{out_link_1}]...[@var{out_link_M}] + +@var{filter_name} is the name of the filter class of which the +described filter is an instance of, and has to be the name of one of +the filter classes registered in the program. +The name of the filter class is optionally followed by a string +"=@var{arguments}". + +@var{arguments} is a string which contains the parameters used to +initialize the filter instance, and are described in the filter +descriptions below. + +The list of arguments can be quoted using the character "'" as initial +and ending mark, and the character '\' for escaping the characters +within the quoted text; otherwise the argument string is considered +terminated when the next special character (belonging to the set +"[]=;,") is encountered. + +The name and arguments of the filter are optionally preceded and +followed by a list of link labels. +A link label allows to name a link and associate it to a filter output +or input pad. The preceding labels @var{in_link_1} +... @var{in_link_N}, are associated to the filter input pads, +the following labels @var{out_link_1} ... @var{out_link_M}, are +associated to the output pads. + +When two link labels with the same name are found in the +filtergraph, a link between the corresponding input and output pad is +created. + +If an output pad is not labelled, it is linked by default to the first +unlabelled input pad of the next filter in the filterchain. +For example in the filterchain: +@example +nullsrc, split[L1], [L2]overlay, nullsink +@end example +the split filter instance has two output pads, and the overlay filter +instance two input pads. The first output pad of split is labelled +"L1", the first input pad of overlay is labelled "L2", and the second +output pad of split is linked to the second input pad of overlay, +which are both unlabelled. + +In a complete filterchain all the unlabelled filter input and output +pads must be connected. A filtergraph is considered valid if all the +filter input and output pads of all the filterchains are connected. + +Libavfilter will automatically insert scale filters where format +conversion is required. It is possible to specify swscale flags +for those automatically inserted scalers by prepending +@code{sws_flags=@var{flags};} +to the filtergraph description. + +Follows a BNF description for the filtergraph syntax: +@example +@var{NAME} ::= sequence of alphanumeric characters and '_' +@var{LINKLABEL} ::= "[" @var{NAME} "]" +@var{LINKLABELS} ::= @var{LINKLABEL} [@var{LINKLABELS}] +@var{FILTER_ARGUMENTS} ::= sequence of chars (eventually quoted) +@var{FILTER} ::= [@var{LINKLABELS}] @var{NAME} ["=" @var{FILTER_ARGUMENTS}] [@var{LINKLABELS}] +@var{FILTERCHAIN} ::= @var{FILTER} [,@var{FILTERCHAIN}] +@var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}] +@end example + +@section Notes on filtergraph escaping + +Some filter arguments require the use of special characters, typically +@code{:} to separate key=value pairs in a named options list. In this +case the user should perform a first level escaping when specifying +the filter arguments. For example, consider the following literal +string to be embedded in the @ref{drawtext} filter arguments: +@example +this is a 'string': may contain one, or more, special characters +@end example + +Since @code{:} is special for the filter arguments syntax, it needs to +be escaped, so you get: +@example +text=this is a \'string\'\: may contain one, or more, special characters +@end example + +A second level of escaping is required when embedding the filter +arguments in a filtergraph description, in order to escape all the +filtergraph special characters. Thus the example above becomes: +@example +drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters +@end example + +Finally an additional level of escaping may be needed when writing the +filtergraph description in a shell command, which depends on the +escaping rules of the adopted shell. For example, assuming that +@code{\} is special and needs to be escaped with another @code{\}, the +previous string will finally result in: +@example +-vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters" +@end example + +Sometimes, it might be more convenient to employ quoting in place of +escaping. For example the string: +@example +Caesar: tu quoque, Brute, fili mi +@end example + +Can be quoted in the filter arguments as: +@example +text='Caesar: tu quoque, Brute, fili mi' +@end example + +And finally inserted in a filtergraph like: +@example +drawtext=text=\'Caesar: tu quoque\, Brute\, fili mi\' +@end example + +See the ``Quoting and escaping'' section in the ffmpeg-utils manual +for more information about the escaping and quoting rules adopted by +FFmpeg. + +@c man end FILTERGRAPH DESCRIPTION + +@chapter Audio Filters +@c man begin AUDIO FILTERS + +When you configure your FFmpeg build, you can disable any of the +existing filters using @code{--disable-filters}. +The configure output will show the audio filters included in your +build. + +Below is a description of the currently available audio filters. + +@section aconvert + +Convert the input audio format to the specified formats. + +The filter accepts a string of the form: +"@var{sample_format}:@var{channel_layout}". + +@var{sample_format} specifies the sample format, and can be a string or the +corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p' +suffix for a planar sample format. + +@var{channel_layout} specifies the channel layout, and can be a string +or the corresponding number value defined in @file{libavutil/channel_layout.h}. + +The special parameter "auto", signifies that the filter will +automatically select the output format depending on the output filter. + +@subsection Examples + +@itemize +@item +Convert input to float, planar, stereo: +@example +aconvert=fltp:stereo +@end example + +@item +Convert input to unsigned 8-bit, automatically select out channel layout: +@example +aconvert=u8:auto +@end example +@end itemize + +@section allpass + +Apply a two-pole all-pass filter with central frequency (in Hz) +@var{frequency}, and filter-width @var{width}. +An all-pass filter changes the audio's frequency to phase relationship +without changing its frequency to amplitude relationship. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set frequency in Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@section highpass + +Apply a high-pass filter with 3dB point frequency. +The filter can be either single-pole, or double-pole (the default). +The filter roll off at 6dB per pole per octave (20dB per pole per decade). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set frequency in Hz. Default is 3000. + +@item poles, p +Set number of poles. Default is 2. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +Applies only to double-pole filter. +The default is 0.707q and gives a Butterworth response. +@end table + +@section lowpass + +Apply a low-pass filter with 3dB point frequency. +The filter can be either single-pole or double-pole (the default). +The filter roll off at 6dB per pole per octave (20dB per pole per decade). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set frequency in Hz. Default is 500. + +@item poles, p +Set number of poles. Default is 2. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +Applies only to double-pole filter. +The default is 0.707q and gives a Butterworth response. +@end table + +@section bass + +Boost or cut the bass (lower) frequencies of the audio using a two-pole +shelving filter with a response similar to that of a standard +hi-fi's tone-controls. This is also known as shelving equalisation (EQ). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item gain, g +Give the gain at 0 Hz. Its useful range is about -20 +(for a large cut) to +20 (for a large boost). +Beware of clipping when using a positive gain. + +@item frequency, f +Set the filter's central frequency and so can be used +to extend or reduce the frequency range to be boosted or cut. +The default value is @code{100} Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Determine how steep is the filter's shelf transition. +@end table + +@section treble + +Boost or cut treble (upper) frequencies of the audio using a two-pole +shelving filter with a response similar to that of a standard +hi-fi's tone-controls. This is also known as shelving equalisation (EQ). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item gain, g +Give the gain at whichever is the lower of ~22 kHz and the +Nyquist frequency. Its useful range is about -20 (for a large cut) +to +20 (for a large boost). Beware of clipping when using a positive gain. + +@item frequency, f +Set the filter's central frequency and so can be used +to extend or reduce the frequency range to be boosted or cut. +The default value is @code{3000} Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Determine how steep is the filter's shelf transition. +@end table + +@section bandpass + +Apply a two-pole Butterworth band-pass filter with central +frequency @var{frequency}, and (3dB-point) band-width width. +The @var{csg} option selects a constant skirt gain (peak gain = Q) +instead of the default: constant 0dB peak gain. +The filter roll off at 6dB per octave (20dB per decade). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set the filter's central frequency. Default is @code{3000}. + +@item csg +Constant skirt gain if set to 1. Defaults to 0. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@section bandreject + +Apply a two-pole Butterworth band-reject filter with central +frequency @var{frequency}, and (3dB-point) band-width @var{width}. +The filter roll off at 6dB per octave (20dB per decade). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set the filter's central frequency. Default is @code{3000}. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@section biquad + +Apply a biquad IIR filter with the given coefficients. +Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2} +are the numerator and denominator coefficients respectively. + +@section equalizer + +Apply a two-pole peaking equalisation (EQ) filter. With this +filter, the signal-level at and around a selected frequency can +be increased or decreased, whilst (unlike bandpass and bandreject +filters) that at all other frequencies is unchanged. + +In order to produce complex equalisation curves, this filter can +be given several times, each with a different central frequency. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item frequency, f +Set the filter's central frequency in Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. + +@item gain, g +Set the required gain or attenuation in dB. +Beware of clipping when using a positive gain. +@end table + +@section afade + +Apply fade-in/out effect to input audio. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item type, t +Specify the effect type, can be either @code{in} for fade-in, or +@code{out} for a fade-out effect. Default is @code{in}. + +@item start_sample, ss +Specify the number of the start sample for starting to apply the fade +effect. Default is 0. + +@item nb_samples, ns +Specify the number of samples for which the fade effect has to last. At +the end of the fade-in effect the output audio will have the same +volume as the input audio, at the end of the fade-out transition +the output audio will be silence. Default is 44100. + +@item start_time, st +Specify time in seconds for starting to apply the fade +effect. Default is 0. +If set this option is used instead of @var{start_sample} one. + +@item duration, d +Specify the number of seconds for which the fade effect has to last. At +the end of the fade-in effect the output audio will have the same +volume as the input audio, at the end of the fade-out transition +the output audio will be silence. Default is 0. +If set this option is used instead of @var{nb_samples} one. + +@item curve +Set curve for fade transition. + +It accepts the following values: +@table @option +@item tri +select triangular, linear slope (default) +@item qsin +select quarter of sine wave +@item hsin +select half of sine wave +@item esin +select exponential sine wave +@item log +select logarithmic +@item par +select inverted parabola +@item qua +select quadratic +@item cub +select cubic +@item squ +select square root +@item cbr +select cubic root +@end table +@end table + +@subsection Examples + +@itemize +@item +Fade in first 15 seconds of audio: +@example +afade=t=in:ss=0:d=15 +@end example + +@item +Fade out last 25 seconds of a 900 seconds audio: +@example +afade=t=out:ss=875:d=25 +@end example +@end itemize + +@anchor{aformat} +@section aformat + +Set output format constraints for the input audio. The framework will +negotiate the most appropriate format to minimize conversions. + +The filter accepts the following named parameters: +@table @option + +@item sample_fmts +A comma-separated list of requested sample formats. + +@item sample_rates +A comma-separated list of requested sample rates. + +@item channel_layouts +A comma-separated list of requested channel layouts. + +@end table + +If a parameter is omitted, all values are allowed. + +For example to force the output to either unsigned 8-bit or signed 16-bit stereo: +@example +aformat='sample_fmts=u8,s16:channel_layouts=stereo' +@end example + +@section amerge + +Merge two or more audio streams into a single multi-channel stream. + +The filter accepts the following named options: + +@table @option + +@item inputs +Set the number of inputs. Default is 2. + +@end table + +If the channel layouts of the inputs are disjoint, and therefore compatible, +the channel layout of the output will be set accordingly and the channels +will be reordered as necessary. If the channel layouts of the inputs are not +disjoint, the output will have all the channels of the first input then all +the channels of the second input, in that order, and the channel layout of +the output will be the default value corresponding to the total number of +channels. + +For example, if the first input is in 2.1 (FL+FR+LF) and the second input +is FC+BL+BR, then the output will be in 5.1, with the channels in the +following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the +first input, b1 is the first channel of the second input). + +On the other hand, if both input are in stereo, the output channels will be +in the default order: a1, a2, b1, b2, and the channel layout will be +arbitrarily set to 4.0, which may or may not be the expected value. + +All inputs must have the same sample rate, and format. + +If inputs do not have the same duration, the output will stop with the +shortest. + +@subsection Examples + +@itemize +@item +Merge two mono files into a stereo stream: +@example +amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge +@end example + +@item +Multiple merges: +@example +ffmpeg -f lavfi -i " +amovie=input.mkv:si=0 [a0]; +amovie=input.mkv:si=1 [a1]; +amovie=input.mkv:si=2 [a2]; +amovie=input.mkv:si=3 [a3]; +amovie=input.mkv:si=4 [a4]; +amovie=input.mkv:si=5 [a5]; +[a0][a1][a2][a3][a4][a5] amerge=inputs=6" -c:a pcm_s16le output.mkv +@end example +@end itemize + +@section amix + +Mixes multiple audio inputs into a single output. + +For example +@example +ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT +@end example +will mix 3 input audio streams to a single output with the same duration as the +first input and a dropout transition time of 3 seconds. + +The filter accepts the following named parameters: +@table @option + +@item inputs +Number of inputs. If unspecified, it defaults to 2. + +@item duration +How to determine the end-of-stream. +@table @option + +@item longest +Duration of longest input. (default) + +@item shortest +Duration of shortest input. + +@item first +Duration of first input. + +@end table + +@item dropout_transition +Transition time, in seconds, for volume renormalization when an input +stream ends. The default value is 2 seconds. + +@end table + +@section anull + +Pass the audio source unchanged to the output. + +@section apad + +Pad the end of a audio stream with silence, this can be used together with +-shortest to extend audio streams to the same length as the video stream. + +@anchor{aresample} +@section aresample + +Resample the input audio to the specified parameters, using the +libswresample library. If none are specified then the filter will +automatically convert between its input and output. + +This filter is also able to stretch/squeeze the audio data to make it match +the timestamps or to inject silence / cut out audio to make it match the +timestamps, do a combination of both or do neither. + +The filter accepts the syntax +[@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate} +expresses a sample rate and @var{resampler_options} is a list of +@var{key}=@var{value} pairs, separated by ":". See the +ffmpeg-resampler manual for the complete list of supported options. + +@subsection Examples + +@itemize +@item +Resample the input audio to 44100Hz: +@example +aresample=44100 +@end example + +@item +Stretch/squeeze samples to the given timestamps, with a maximum of 1000 +samples per second compensation: +@example +aresample=async=1000 +@end example +@end itemize + +@section asetnsamples + +Set the number of samples per each output audio frame. + +The last output packet may contain a different number of samples, as +the filter will flush all the remaining samples when the input audio +signal its end. + +The filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +@table @option + +@item nb_out_samples, n +Set the number of frames per each output audio frame. The number is +intended as the number of samples @emph{per each channel}. +Default value is 1024. + +@item pad, p +If set to 1, the filter will pad the last audio frame with zeroes, so +that the last frame will contain the same number of samples as the +previous ones. Default value is 1. +@end table + +For example, to set the number of per-frame samples to 1234 and +disable padding for the last frame, use: +@example +asetnsamples=n=1234:p=0 +@end example + +@section ashowinfo + +Show a line containing various information for each input audio frame. +The input audio is not modified. + +The shown line contains a sequence of key/value pairs of the form +@var{key}:@var{value}. + +A description of each shown parameter follows: + +@table @option +@item n +sequential number of the input frame, starting from 0 + +@item pts +Presentation timestamp of the input frame, in time base units; the time base +depends on the filter input pad, and is usually 1/@var{sample_rate}. + +@item pts_time +presentation timestamp of the input frame in seconds + +@item pos +position of the frame in the input stream, -1 if this information in +unavailable and/or meaningless (for example in case of synthetic audio) + +@item fmt +sample format + +@item chlayout +channel layout + +@item rate +sample rate for the audio frame + +@item nb_samples +number of samples (per channel) in the frame + +@item checksum +Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio +the data is treated as if all the planes were concatenated. + +@item plane_checksums +A list of Adler-32 checksums for each data plane. +@end table + +@section asplit + +Split input audio into several identical outputs. + +The filter accepts a single parameter which specifies the number of outputs. If +unspecified, it defaults to 2. + +For example: +@example +[in] asplit [out0][out1] +@end example + +will create two separate outputs from the same input. + +To create 3 or more outputs, you need to specify the number of +outputs, like in: +@example +[in] asplit=3 [out0][out1][out2] +@end example + +@example +ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT +@end example +will create 5 copies of the input audio. + + +@section astreamsync + +Forward two audio streams and control the order the buffers are forwarded. + +The argument to the filter is an expression deciding which stream should be +forwarded next: if the result is negative, the first stream is forwarded; if +the result is positive or zero, the second stream is forwarded. It can use +the following variables: + +@table @var +@item b1 b2 +number of buffers forwarded so far on each stream +@item s1 s2 +number of samples forwarded so far on each stream +@item t1 t2 +current timestamp of each stream +@end table + +The default value is @code{t1-t2}, which means to always forward the stream +that has a smaller timestamp. + +Example: stress-test @code{amerge} by randomly sending buffers on the wrong +input, while avoiding too much of a desynchronization: +@example +amovie=file.ogg [a] ; amovie=file.mp3 [b] ; +[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ; +[a2] [b2] amerge +@end example + +@section atempo + +Adjust audio tempo. + +The filter accepts exactly one parameter, the audio tempo. If not +specified then the filter will assume nominal 1.0 tempo. Tempo must +be in the [0.5, 2.0] range. + +@subsection Examples + +@itemize +@item +Slow down audio to 80% tempo: +@example +atempo=0.8 +@end example + +@item +To speed up audio to 125% tempo: +@example +atempo=1.25 +@end example +@end itemize + +@section earwax + +Make audio easier to listen to on headphones. + +This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio +so that when listened to on headphones the stereo image is moved from +inside your head (standard for headphones) to outside and in front of +the listener (standard for speakers). + +Ported from SoX. + +@section pan + +Mix channels with specific gain levels. The filter accepts the output +channel layout followed by a set of channels definitions. + +This filter is also designed to remap efficiently the channels of an audio +stream. + +The filter accepts parameters of the form: +"@var{l}:@var{outdef}:@var{outdef}:..." + +@table @option +@item l +output channel layout or number of channels + +@item outdef +output channel specification, of the form: +"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]" + +@item out_name +output channel to define, either a channel name (FL, FR, etc.) or a channel +number (c0, c1, etc.) + +@item gain +multiplicative coefficient for the channel, 1 leaving the volume unchanged + +@item in_name +input channel to use, see out_name for details; it is not possible to mix +named and numbered input channels +@end table + +If the `=' in a channel specification is replaced by `<', then the gains for +that specification will be renormalized so that the total is 1, thus +avoiding clipping noise. + +@subsection Mixing examples + +For example, if you want to down-mix from stereo to mono, but with a bigger +factor for the left channel: +@example +pan=1:c0=0.9*c0+0.1*c1 +@end example + +A customized down-mix to stereo that works automatically for 3-, 4-, 5- and +7-channels surround: +@example +pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR +@end example + +Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system +that should be preferred (see "-ac" option) unless you have very specific +needs. + +@subsection Remapping examples + +The channel remapping will be effective if, and only if: + +@itemize +@item gain coefficients are zeroes or ones, +@item only one input per channel output, +@end itemize + +If all these conditions are satisfied, the filter will notify the user ("Pure +channel mapping detected"), and use an optimized and lossless method to do the +remapping. + +For example, if you have a 5.1 source and want a stereo audio stream by +dropping the extra channels: +@example +pan="stereo: c0=FL : c1=FR" +@end example + +Given the same source, you can also switch front left and front right channels +and keep the input channel layout: +@example +pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5" +@end example + +If the input is a stereo audio stream, you can mute the front left channel (and +still keep the stereo channel layout) with: +@example +pan="stereo:c1=c1" +@end example + +Still with a stereo audio stream input, you can copy the right channel in both +front left and right: +@example +pan="stereo: c0=FR : c1=FR" +@end example + +@section silencedetect + +Detect silence in an audio stream. + +This filter logs a message when it detects that the input audio volume is less +or equal to a noise tolerance value for a duration greater or equal to the +minimum detected noise duration. + +The printed times and duration are expressed in seconds. + +@table @option +@item duration, d +Set silence duration until notification (default is 2 seconds). + +@item noise, n +Set noise tolerance. Can be specified in dB (in case "dB" is appended to the +specified value) or amplitude ratio. Default is -60dB, or 0.001. +@end table + +@subsection Examples + +@itemize +@item +Detect 5 seconds of silence with -50dB noise tolerance: +@example +silencedetect=n=-50dB:d=5 +@end example + +@item +Complete example with @command{ffmpeg} to detect silence with 0.0001 noise +tolerance in @file{silence.mp3}: +@example +ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null - +@end example +@end itemize + +@section asyncts +Synchronize audio data with timestamps by squeezing/stretching it and/or +dropping samples/adding silence when needed. + +This filter is not built by default, please use @ref{aresample} to do squeezing/stretching. + +The filter accepts the following named parameters: +@table @option + +@item compensate +Enable stretching/squeezing the data to make it match the timestamps. Disabled +by default. When disabled, time gaps are covered with silence. + +@item min_delta +Minimum difference between timestamps and audio data (in seconds) to trigger +adding/dropping samples. Default value is 0.1. If you get non-perfect sync with +this filter, try setting this parameter to 0. + +@item max_comp +Maximum compensation in samples per second. Relevant only with compensate=1. +Default value 500. + +@item first_pts +Assume the first pts should be this value. The time base is 1 / sample rate. +This allows for padding/trimming at the start of stream. By default, no +assumption is made about the first frame's expected pts, so no padding or +trimming is done. For example, this could be set to 0 to pad the beginning with +silence if an audio stream starts after the video stream or to trim any samples +with a negative pts due to encoder delay. + +@end table + +@section channelsplit +Split each channel in input audio stream into a separate output stream. + +This filter accepts the following named parameters: +@table @option +@item channel_layout +Channel layout of the input stream. Default is "stereo". +@end table + +For example, assuming a stereo input MP3 file +@example +ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv +@end example +will create an output Matroska file with two audio streams, one containing only +the left channel and the other the right channel. + +To split a 5.1 WAV file into per-channel files +@example +ffmpeg -i in.wav -filter_complex +'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' +-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' +front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' +side_right.wav +@end example + +@section channelmap +Remap input channels to new locations. + +This filter accepts the following named parameters: +@table @option +@item channel_layout +Channel layout of the output stream. + +@item map +Map channels from input to output. The argument is a comma-separated list of +mappings, each in the @code{@var{in_channel}-@var{out_channel}} or +@var{in_channel} form. @var{in_channel} can be either the name of the input +channel (e.g. FL for front left) or its index in the input channel layout. +@var{out_channel} is the name of the output channel or its index in the output +channel layout. If @var{out_channel} is not given then it is implicitly an +index, starting with zero and increasing by one for each mapping. +@end table + +If no mapping is present, the filter will implicitly map input channels to +output channels preserving index. + +For example, assuming a 5.1+downmix input MOV file +@example +ffmpeg -i in.mov -filter 'channelmap=map=DL-FL\,DR-FR' out.wav +@end example +will create an output WAV file tagged as stereo from the downmix channels of +the input. + +To fix a 5.1 WAV improperly encoded in AAC's native channel order +@example +ffmpeg -i in.wav -filter 'channelmap=1\,2\,0\,5\,3\,4:channel_layout=5.1' out.wav +@end example + +@section join +Join multiple input streams into one multi-channel stream. + +The filter accepts the following named parameters: +@table @option + +@item inputs +Number of input streams. Defaults to 2. + +@item channel_layout +Desired output channel layout. Defaults to stereo. + +@item map +Map channels from inputs to output. The argument is a comma-separated list of +mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}} +form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel} +can be either the name of the input channel (e.g. FL for front left) or its +index in the specified input stream. @var{out_channel} is the name of the output +channel. +@end table + +The filter will attempt to guess the mappings when those are not specified +explicitly. It does so by first trying to find an unused matching input channel +and if that fails it picks the first unused input channel. + +E.g. to join 3 inputs (with properly set channel layouts) +@example +ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT +@end example + +To build a 5.1 output from 6 single-channel streams: +@example +ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex +'join=inputs=6:channel_layout=5.1:map=0.0-FL\,1.0-FR\,2.0-FC\,3.0-SL\,4.0-SR\,5.0-LFE' +out +@end example + +@section resample +Convert the audio sample format, sample rate and channel layout. This filter is +not meant to be used directly. + +@section volume + +Adjust the input audio volume. + +The filter accepts the following named parameters. If the key of the +first options is omitted, the arguments are interpreted according to +the following syntax: +@example +volume=@var{volume}:@var{precision} +@end example + +@table @option + +@item volume +Expresses how the audio volume will be increased or decreased. + +Output values are clipped to the maximum value. + +The output audio volume is given by the relation: +@example +@var{output_volume} = @var{volume} * @var{input_volume} +@end example + +Default value for @var{volume} is 1.0. + +@item precision +Set the mathematical precision. + +This determines which input sample formats will be allowed, which affects the +precision of the volume scaling. + +@table @option +@item fixed +8-bit fixed-point; limits input sample format to U8, S16, and S32. +@item float +32-bit floating-point; limits input sample format to FLT. (default) +@item double +64-bit floating-point; limits input sample format to DBL. +@end table +@end table + +@subsection Examples + +@itemize +@item +Halve the input audio volume: +@example +volume=volume=0.5 +volume=volume=1/2 +volume=volume=-6.0206dB +@end example + +In all the above example the named key for @option{volume} can be +omitted, for example like in: +@example +volume=0.5 +@end example + +@item +Increase input audio power by 6 decibels using fixed-point precision: +@example +volume=volume=6dB:precision=fixed +@end example +@end itemize + +@section volumedetect + +Detect the volume of the input video. + +The filter has no parameters. The input is not modified. Statistics about +the volume will be printed in the log when the input stream end is reached. + +In particular it will show the mean volume (root mean square), maximum +volume (on a per-sample basis), and the beginning of an histogram of the +registered volume values (from the maximum value to a cumulated 1/1000 of +the samples). + +All volumes are in decibels relative to the maximum PCM value. + +@subsection Examples + +Here is an excerpt of the output: +@example +[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB +[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB +[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6 +[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62 +[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286 +[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042 +[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551 +[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609 +[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409 +@end example + +It means that: +@itemize +@item +The mean square energy is approximately -27 dB, or 10^-2.7. +@item +The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB. +@item +There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc. +@end itemize + +In other words, raising the volume by +4 dB does not cause any clipping, +raising it by +5 dB causes clipping for 6 samples, etc. + +@c man end AUDIO FILTERS + +@chapter Audio Sources +@c man begin AUDIO SOURCES + +Below is a description of the currently available audio sources. + +@section abuffer + +Buffer audio frames, and make them available to the filter chain. + +This source is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/asrc_abuffer.h}. + +It accepts the following mandatory parameters: +@var{sample_rate}:@var{sample_fmt}:@var{channel_layout} + +@table @option + +@item sample_rate +The sample rate of the incoming audio buffers. + +@item sample_fmt +The sample format of the incoming audio buffers. +Either a sample format name or its corresponging integer representation from +the enum AVSampleFormat in @file{libavutil/samplefmt.h} + +@item channel_layout +The channel layout of the incoming audio buffers. +Either a channel layout name from channel_layout_map in +@file{libavutil/channel_layout.c} or its corresponding integer representation +from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h} + +@item channels +The number of channels of the incoming audio buffers. +If both @var{channels} and @var{channel_layout} are specified, then they +must be consistent. + +@end table + +@subsection Examples + +@example +abuffer=44100:s16p:stereo +@end example + +will instruct the source to accept planar 16bit signed stereo at 44100Hz. +Since the sample format with name "s16p" corresponds to the number +6 and the "stereo" channel layout corresponds to the value 0x3, this is +equivalent to: +@example +abuffer=44100:6:0x3 +@end example + +@section aevalsrc + +Generate an audio signal specified by an expression. + +This source accepts in input one or more expressions (one for each +channel), which are evaluated and used to generate a corresponding +audio signal. + +It accepts the syntax: @var{exprs}[::@var{options}]. +@var{exprs} is a list of expressions separated by ":", one for each +separate channel. In case the @var{channel_layout} is not +specified, the selected channel layout depends on the number of +provided expressions. + +@var{options} is an optional sequence of @var{key}=@var{value} pairs, +separated by ":". + +The description of the accepted options follows. + +@table @option + +@item channel_layout, c +Set the channel layout. The number of channels in the specified layout +must be equal to the number of specified expressions. + +@item duration, d +Set the minimum duration of the sourced audio. See the function +@code{av_parse_time()} for the accepted format. +Note that the resulting duration may be greater than the specified +duration, as the generated audio is always cut at the end of a +complete frame. + +If not specified, or the expressed duration is negative, the audio is +supposed to be generated forever. + +@item nb_samples, n +Set the number of samples per channel per each output frame, +default to 1024. + +@item sample_rate, s +Specify the sample rate, default to 44100. +@end table + +Each expression in @var{exprs} can contain the following constants: + +@table @option +@item n +number of the evaluated sample, starting from 0 + +@item t +time of the evaluated sample expressed in seconds, starting from 0 + +@item s +sample rate + +@end table + +@subsection Examples + +@itemize +@item +Generate silence: +@example +aevalsrc=0 +@end example + +@item +Generate a sin signal with frequency of 440 Hz, set sample rate to +8000 Hz: +@example +aevalsrc="sin(440*2*PI*t)::s=8000" +@end example + +@item +Generate a two channels signal, specify the channel layout (Front +Center + Back Center) explicitly: +@example +aevalsrc="sin(420*2*PI*t):cos(430*2*PI*t)::c=FC|BC" +@end example + +@item +Generate white noise: +@example +aevalsrc="-2+random(0)" +@end example + +@item +Generate an amplitude modulated signal: +@example +aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" +@end example + +@item +Generate 2.5 Hz binaural beats on a 360 Hz carrier: +@example +aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) : 0.1*sin(2*PI*(360+2.5/2)*t)" +@end example + +@end itemize + +@section anullsrc + +Null audio source, return unprocessed audio frames. It is mainly useful +as a template and to be employed in analysis / debugging tools, or as +the source for filters which ignore the input data (for example the sox +synth filter). + +It accepts an optional sequence of @var{key}=@var{value} pairs, +separated by ":". + +The description of the accepted options follows. + +@table @option + +@item sample_rate, s +Specify the sample rate, and defaults to 44100. + +@item channel_layout, cl + +Specify the channel layout, and can be either an integer or a string +representing a channel layout. The default value of @var{channel_layout} +is "stereo". + +Check the channel_layout_map definition in +@file{libavutil/channel_layout.c} for the mapping between strings and +channel layout values. + +@item nb_samples, n +Set the number of samples per requested frames. + +@end table + +@subsection Examples + +@itemize +@item +Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO. +@example +anullsrc=r=48000:cl=4 +@end example + +@item +Do the same operation with a more obvious syntax: +@example +anullsrc=r=48000:cl=mono +@end example +@end itemize + +@section abuffer +Buffer audio frames, and make them available to the filter chain. + +This source is not intended to be part of user-supplied graph descriptions but +for insertion by calling programs through the interface defined in +@file{libavfilter/buffersrc.h}. + +It accepts the following named parameters: +@table @option + +@item time_base +Timebase which will be used for timestamps of submitted frames. It must be +either a floating-point number or in @var{numerator}/@var{denominator} form. + +@item sample_rate +Audio sample rate. + +@item sample_fmt +Name of the sample format, as returned by @code{av_get_sample_fmt_name()}. + +@item channel_layout +Channel layout of the audio data, in the form that can be accepted by +@code{av_get_channel_layout()}. +@end table + +All the parameters need to be explicitly defined. + +@section flite + +Synthesize a voice utterance using the libflite library. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libflite}. + +Note that the flite library is not thread-safe. + +The source accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +The description of the accepted parameters follows. + +@table @option + +@item list_voices +If set to 1, list the names of the available voices and exit +immediately. Default value is 0. + +@item nb_samples, n +Set the maximum number of samples per frame. Default value is 512. + +@item textfile +Set the filename containing the text to speak. + +@item text +Set the text to speak. + +@item voice, v +Set the voice to use for the speech synthesis. Default value is +@code{kal}. See also the @var{list_voices} option. +@end table + +@subsection Examples + +@itemize +@item +Read from file @file{speech.txt}, and synthetize the text using the +standard flite voice: +@example +flite=textfile=speech.txt +@end example + +@item +Read the specified text selecting the @code{slt} voice: +@example +flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt +@end example + +@item +Input text to ffmpeg: +@example +ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt +@end example + +@item +Make @file{ffplay} speak the specified text, using @code{flite} and +the @code{lavfi} device: +@example +ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.' +@end example +@end itemize + +For more information about libflite, check: +@url{http://www.speech.cs.cmu.edu/flite/} + +@section sine + +Generate an audio signal made of a sine wave with amplitude 1/8. + +The audio signal is bit-exact. + +It accepts a list of options in the form of @var{key}=@var{value} pairs +separated by ":". If the option name is omitted, the first option is the +frequency and the second option is the beep factor. + +The supported options are: + +@table @option + +@item frequency, f +Set the carrier frequency. Default is 440 Hz. + +@item beep_factor, b +Enable a periodic beep every second with frequency @var{beep_factor} times +the carrier frequency. Default is 0, meaning the beep is disabled. + +@item sample_rate, s +Specify the sample rate, default is 44100. + +@item duration, d +Specify the duration of the generated audio stream. + +@item samples_per_frame +Set the number of samples per output frame, default is 1024. +@end table + +@subsection Examples + +@itemize + +@item +Generate a simple 440 Hz sine wave: +@example +sine +@end example + +@item +Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds: +@example +sine=220:4:d=5 +sine=f=220:b=4:d=5 +sine=frequency=220:beep_factor=4:duration=5 +@end example + +@end itemize + +@c man end AUDIO SOURCES + +@chapter Audio Sinks +@c man begin AUDIO SINKS + +Below is a description of the currently available audio sinks. + +@section abuffersink + +Buffer audio frames, and make them available to the end of filter chain. + +This sink is mainly intended for programmatic use, in particular +through the interface defined in @file{libavfilter/buffersink.h}. + +It requires a pointer to an AVABufferSinkContext structure, which +defines the incoming buffers' formats, to be passed as the opaque +parameter to @code{avfilter_init_filter} for initialization. + +@section anullsink + +Null audio sink, do absolutely nothing with the input audio. It is +mainly useful as a template and to be employed in analysis / debugging +tools. + +@section abuffersink +This sink is intended for programmatic use. Frames that arrive on this sink can +be retrieved by the calling program using the interface defined in +@file{libavfilter/buffersink.h}. + +This filter accepts no parameters. + +@c man end AUDIO SINKS + +@chapter Video Filters +@c man begin VIDEO FILTERS + +When you configure your FFmpeg build, you can disable any of the +existing filters using @code{--disable-filters}. +The configure output will show the video filters included in your +build. + +Below is a description of the currently available video filters. + +@section alphaextract + +Extract the alpha component from the input as a grayscale video. This +is especially useful with the @var{alphamerge} filter. + +@section alphamerge + +Add or replace the alpha component of the primary input with the +grayscale value of a second input. This is intended for use with +@var{alphaextract} to allow the transmission or storage of frame +sequences that have alpha in a format that doesn't support an alpha +channel. + +For example, to reconstruct full frames from a normal YUV-encoded video +and a separate video created with @var{alphaextract}, you might use: +@example +movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] +@end example + +Since this filter is designed for reconstruction, it operates on frame +sequences without considering timestamps, and terminates when either +input reaches end of stream. This will cause problems if your encoding +pipeline drops frames. If you're trying to apply an image as an +overlay to a video stream, consider the @var{overlay} filter instead. + +@section ass + +Same as the @ref{subtitles} filter, except that it doesn't require libavcodec +and libavformat to work. On the other hand, it is limited to ASS (Advanced +Substation Alpha) subtitles files. + +@section bbox + +Compute the bounding box for the non-black pixels in the input frame +luminance plane. + +This filter computes the bounding box containing all the pixels with a +luminance value greater than the minimum allowed value. +The parameters describing the bounding box are printed on the filter +log. + +@section blackdetect + +Detect video intervals that are (almost) completely black. Can be +useful to detect chapter transitions, commercials, or invalid +recordings. Output lines contains the time for the start, end and +duration of the detected black interval expressed in seconds. + +In order to display the output lines, you need to set the loglevel at +least to the AV_LOG_INFO value. + +This filter accepts a list of options in the form of +@var{key}=@var{value} pairs separated by ":". A description of the +accepted options follows. + +@table @option +@item black_min_duration, d +Set the minimum detected black duration expressed in seconds. It must +be a non-negative floating point number. + +Default value is 2.0. + +@item picture_black_ratio_th, pic_th +Set the threshold for considering a picture "black". +Express the minimum value for the ratio: +@example +@var{nb_black_pixels} / @var{nb_pixels} +@end example + +for which a picture is considered black. +Default value is 0.98. + +@item pixel_black_th, pix_th +Set the threshold for considering a pixel "black". + +The threshold expresses the maximum pixel luminance value for which a +pixel is considered "black". The provided value is scaled according to +the following equation: +@example +@var{absolute_threshold} = @var{luminance_minimum_value} + @var{pixel_black_th} * @var{luminance_range_size} +@end example + +@var{luminance_range_size} and @var{luminance_minimum_value} depend on +the input video format, the range is [0-255] for YUV full-range +formats and [16-235] for YUV non full-range formats. + +Default value is 0.10. +@end table + +The following example sets the maximum pixel threshold to the minimum +value, and detects only black intervals of 2 or more seconds: +@example +blackdetect=d=2:pix_th=0.00 +@end example + +@section blackframe + +Detect frames that are (almost) completely black. Can be useful to +detect chapter transitions or commercials. Output lines consist of +the frame number of the detected frame, the percentage of blackness, +the position in the file if known or -1 and the timestamp in seconds. + +In order to display the output lines, you need to set the loglevel at +least to the AV_LOG_INFO value. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +blackframe[=@var{amount}[:@var{threshold}]]. + +A description of the accepted options follows. + +@table @option +@item amount +Set the percentage of pixels that have to be below the +threshold to enable black detection. Default value is 98. + +@item threshold +Set the threshold below which a pixel value is considered +black. Default value is 32. +@end table + +@section blend + +Blend two video frames into each other. + +It takes two input streams and outputs one stream, the first input is the +"top" layer and second input is "bottom" layer. +Output terminates when shortest input terminates. + +This filter accepts a list of options in the form of @var{key}=@var{value} +pairs separated by ":". A description of the accepted options follows. + +@table @option +@item c0_mode +@item c1_mode +@item c2_mode +@item c3_mode +@item all_mode +Set blend mode for specific pixel component or all pixel components in case +of @var{all_mode}. Default value is @code{normal}. + +Available values for component modes are: +@table @samp +@item addition +@item and +@item average +@item burn +@item darken +@item difference +@item divide +@item dodge +@item exclusion +@item hardlight +@item lighten +@item multiply +@item negation +@item normal +@item or +@item overlay +@item phoenix +@item pinlight +@item reflect +@item screen +@item softlight +@item subtract +@item vividlight +@item xor +@end table + +@item c0_opacity +@item c1_opacity +@item c2_opacity +@item c3_opacity +@item all_opacity +Set blend opacity for specific pixel component or all pixel components in case +of @var{all_expr}. Only used in combination with pixel component blend modes. + +@item c0_expr +@item c1_expr +@item c2_expr +@item c3_expr +@item all_expr +Set blend expression for specific pixel component or all pixel components in case +of @var{all_expr}. Note that related mode options will be ignored if those are set. + +The expressions can use the following variables: + +@table @option +@item X +@item Y +the coordinates of the current sample + +@item W +@item H +the width and height of currently filtered plane + +@item SW +@item SH +Width and height scale depending on the currently filtered plane. It is the +ratio between the corresponding luma plane number of pixels and the current +plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and +@code{0.5,0.5} for chroma planes. + +@item T +Time of the current frame, expressed in seconds. + +@item TOP, A +Value of pixel component at current location for first video frame (top layer). + +@item BOTTOM, B +Value of pixel component at current location for second video frame (bottom layer). +@end table +@end table + +@subsection Examples + +@itemize +@item +Apply transition from bottom layer to top layer in first 10 seconds: +@example +blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))' +@end example + +@item +Apply 1x1 checkerboard effect: +@example +blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)' +@end example +@end itemize + +@section boxblur + +Apply boxblur algorithm to the input video. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +@option{luma_radius}:@option{luma_power}:@option{chroma_radius}:@option{chroma_power}:@option{alpha_radius}:@option{alpha_power}. + +A description of the accepted options follows. + +@table @option +@item luma_radius, lr +@item chroma_radius, cr +@item alpha_radius, ar +Set an expression for the box radius in pixels used for blurring the +corresponding input plane. + +The radius value must be a non-negative number, and must not be +greater than the value of the expression @code{min(w,h)/2} for the +luma and alpha planes, and of @code{min(cw,ch)/2} for the chroma +planes. + +Default value for @option{luma_radius} is "2". If not specified, +@option{chroma_radius} and @option{alpha_radius} default to the +corresponding value set for @option{luma_radius}. + +The expressions can contain the following constants: +@table @option +@item w, h +the input width and height in pixels + +@item cw, ch +the input chroma image width and height in pixels + +@item hsub, vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@end table + +@item luma_power, lp +@item chroma_power, cp +@item alpha_power, ap +Specify how many times the boxblur filter is applied to the +corresponding plane. + +Default value for @option{luma_power} is 2. If not specified, +@option{chroma_power} and @option{alpha_power} default to the +corresponding value set for @option{luma_power}. + +A value of 0 will disable the effect. +@end table + +@subsection Examples + +@itemize +@item +Apply a boxblur filter with luma, chroma, and alpha radius +set to 2: +@example +boxblur=2:1 +@end example + +@item +Set luma radius to 2, alpha and chroma radius to 0: +@example +boxblur=2:1:cr=0:ar=0 +@end example + +@item +Set luma and chroma radius to a fraction of the video dimension: +@example +boxblur=min(h\,w)/10:1:min(cw\,ch)/10:1 +@end example +@end itemize + +@section colormatrix + +The colormatrix filter allows conversion between any of the following color +space: BT.709 (@var{bt709}), BT.601 (@var{bt601}), SMPTE-240M (@var{smpte240m}) +and FCC (@var{fcc}). + +The syntax of the parameters is @var{source}:@var{destination}: + +@example +colormatrix=bt601:smpte240m +@end example + +@section copy + +Copy the input source unchanged to the output. Mainly useful for +testing purposes. + +@section crop + +Crop the input video. + +This filter accepts a list of @var{key}=@var{value} pairs as argument, +separated by ':'. If the key of the first options is omitted, the +arguments are interpreted according to the syntax +@var{out_w}:@var{out_h}:@var{x}:@var{y}:@var{keep_aspect}. + +A description of the accepted options follows: +@table @option +@item w, out_w +Set the crop area width. It defaults to @code{iw}. +This expression is evaluated only once during the filter +configuration. + +@item h, out_h +Set the crop area width. It defaults to @code{ih}. +This expression is evaluated only once during the filter +configuration. + +@item x +Set the expression for the x top-left coordinate of the cropped area. +It defaults to @code{(in_w-out_w)/2}. +This expression is evaluated per-frame. + +@item y +Set the expression for the y top-left coordinate of the cropped area. +It defaults to @code{(in_h-out_h)/2}. +This expression is evaluated per-frame. + +@item keep_aspect +If set to 1 will force the output display aspect ratio +to be the same of the input, by changing the output sample aspect +ratio. It defaults to 0. +@end table + +The @var{out_w}, @var{out_h}, @var{x}, @var{y} parameters are +expressions containing the following constants: + +@table @option +@item x, y +the computed values for @var{x} and @var{y}. They are evaluated for +each new frame. + +@item in_w, in_h +the input width and height + +@item iw, ih +same as @var{in_w} and @var{in_h} + +@item out_w, out_h +the output (cropped) width and height + +@item ow, oh +same as @var{out_w} and @var{out_h} + +@item a +same as @var{iw} / @var{ih} + +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} + +@item hsub, vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item n +the number of input frame, starting from 0 + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@end table + +The expression for @var{out_w} may depend on the value of @var{out_h}, +and the expression for @var{out_h} may depend on @var{out_w}, but they +cannot depend on @var{x} and @var{y}, as @var{x} and @var{y} are +evaluated after @var{out_w} and @var{out_h}. + +The @var{x} and @var{y} parameters specify the expressions for the +position of the top-left corner of the output (non-cropped) area. They +are evaluated for each frame. If the evaluated value is not valid, it +is approximated to the nearest valid value. + +The expression for @var{x} may depend on @var{y}, and the expression +for @var{y} may depend on @var{x}. + +@subsection Examples + +@itemize +@item +Crop area with size 100x100 at position (12,34). +@example +crop=100:100:12:34 +@end example + +Using named options, the example above becomes: +@example +crop=w=100:h=100:x=12:y=34 +@end example + +@item +Crop the central input area with size 100x100: +@example +crop=100:100 +@end example + +@item +Crop the central input area with size 2/3 of the input video: +@example +crop=2/3*in_w:2/3*in_h +@end example + +@item +Crop the input video central square: +@example +crop=in_h +@end example + +@item +Delimit the rectangle with the top-left corner placed at position +100:100 and the right-bottom corner corresponding to the right-bottom +corner of the input image: +@example +crop=in_w-100:in_h-100:100:100 +@end example + +@item +Crop 10 pixels from the left and right borders, and 20 pixels from +the top and bottom borders +@example +crop=in_w-2*10:in_h-2*20 +@end example + +@item +Keep only the bottom right quarter of the input image: +@example +crop=in_w/2:in_h/2:in_w/2:in_h/2 +@end example + +@item +Crop height for getting Greek harmony: +@example +crop=in_w:1/PHI*in_w +@end example + +@item +Appply trembling effect: +@example +crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7) +@end example + +@item +Apply erratic camera effect depending on timestamp: +@example +crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" +@end example + +@item +Set x depending on the value of y: +@example +crop=in_w/2:in_h/2:y:10+10*sin(n/10) +@end example +@end itemize + +@section cropdetect + +Auto-detect crop size. + +Calculate necessary cropping parameters and prints the recommended +parameters through the logging system. The detected dimensions +correspond to the non-black area of the input video. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +[@option{limit}[:@option{round}[:@option{reset}]]]. + +A description of the accepted options follows. + +@table @option + +@item limit +Set higher black value threshold, which can be optionally specified +from nothing (0) to everything (255). An intensity value greater +to the set value is considered non-black. Default value is 24. + +@item round +Set the value for which the width/height should be divisible by. The +offset is automatically adjusted to center the video. Use 2 to get +only even dimensions (needed for 4:2:2 video). 16 is best when +encoding to most video codecs. Default value is 16. + +@item reset +Set the counter that determines after how many frames cropdetect will +reset the previously detected largest video area and start over to +detect the current optimal crop area. Default value is 0. + +This can be useful when channel logos distort the video area. 0 +indicates never reset and return the largest area encountered during +playback. +@end table + +@section curves + +Apply color adjustments using curves. + +This filter is similar to the Adobe Photoshop and GIMP curves tools. Each +component (red, green and blue) has its values defined by @var{N} key points +tied from each other using a smooth curve. The x-axis represents the pixel +values from the input frame, and the y-axis the new pixel values to be set for +the output frame. + +By default, a component curve is defined by the two points @var{(0;0)} and +@var{(1;1)}. This creates a straight line where each original pixel value is +"adjusted" to its own value, which means no change to the image. + +The filter allows you to redefine these two points and add some more. A new +curve (using a natural cubic spline interpolation) will be define to pass +smoothly through all these new coordinates. The new defined points needs to be +strictly increasing over the x-axis, and their @var{x} and @var{y} values must +be in the @var{[0;1]} interval. If the computed curves happened to go outside +the vector spaces, the values will be clipped accordingly. + +If there is no key point defined in @code{x=0}, the filter will automatically +insert a @var{(0;0)} point. In the same way, if there is no key point defined +in @code{x=1}, the filter will automatically insert a @var{(1;1)} point. + +The filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +A description of the accepted parameters follows. + +@table @option +@item red, r +Set the key points for the red component. +@item green, g +Set the key points for the green component. +@item blue, b +Set the key points for the blue component. +@end table + +To avoid some filtergraph syntax conflicts, each key points list need to be +defined using the following syntax: @code{x0/y0 x1/y1 x2/y2 ...}. + +@subsection Examples + +@itemize +@item +Increase slightly the middle level of blue: +@example +curves=blue='0.5/0.58' +@end example + +@item +Vintage effect: +@example +curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':b='0/0.22 .49/.44 1/0.8' +@end example +Here we obtain the following coordinates for each components: +@table @var +@item red +@code{(0;0.11) (0.42;0.51) (1;0.95)} +@item green +@code{(0;0) (0.50;0.48) (1;1)} +@item blue +@code{(0;0.22) (0.49;0.44) (1;0.80)} +@end table +@end itemize + +@section decimate + +Drop frames that do not differ greatly from the previous frame in +order to reduce framerate. + +The main use of this filter is for very-low-bitrate encoding +(e.g. streaming over dialup modem), but it could in theory be used for +fixing movies that were inverse-telecined incorrectly. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax: +@option{max}:@option{hi}:@option{lo}:@option{frac}. + +A description of the accepted options follows. + +@table @option +@item max +Set the maximum number of consecutive frames which can be dropped (if +positive), or the minimum interval between dropped frames (if +negative). If the value is 0, the frame is dropped unregarding the +number of previous sequentially dropped frames. + +Default value is 0. + +@item hi +@item lo +@item frac +Set the dropping threshold values. + +Values for @option{hi} and @option{lo} are for 8x8 pixel blocks and +represent actual pixel value differences, so a threshold of 64 +corresponds to 1 unit of difference for each pixel, or the same spread +out differently over the block. + +A frame is a candidate for dropping if no 8x8 blocks differ by more +than a threshold of @option{hi}, and if no more than @option{frac} blocks (1 +meaning the whole image) differ by more than a threshold of @option{lo}. + +Default value for @option{hi} is 64*12, default value for @option{lo} is +64*5, and default value for @option{frac} is 0.33. +@end table + +@section delogo + +Suppress a TV station logo by a simple interpolation of the surrounding +pixels. Just set a rectangle covering the logo and watch it disappear +(and sometimes something even uglier appear - your mileage may vary). + +The filter accepts parameters as a string of the form +"@var{x}:@var{y}:@var{w}:@var{h}:@var{band}", or as a list of +@var{key}=@var{value} pairs, separated by ":". + +The description of the accepted parameters follows. + +@table @option + +@item x, y +Specify the top left corner coordinates of the logo. They must be +specified. + +@item w, h +Specify the width and height of the logo to clear. They must be +specified. + +@item band, t +Specify the thickness of the fuzzy edge of the rectangle (added to +@var{w} and @var{h}). The default value is 4. + +@item show +When set to 1, a green rectangle is drawn on the screen to simplify +finding the right @var{x}, @var{y}, @var{w}, @var{h} parameters, and +@var{band} is set to 4. The default value is 0. + +@end table + +@subsection Examples + +@itemize +@item +Set a rectangle covering the area with top left corner coordinates 0,0 +and size 100x77, setting a band of size 10: +@example +delogo=0:0:100:77:10 +@end example + +@item +As the previous example, but use named options: +@example +delogo=x=0:y=0:w=100:h=77:band=10 +@end example + +@end itemize + +@section deshake + +Attempt to fix small changes in horizontal and/or vertical shift. This +filter helps remove camera shake from hand-holding a camera, bumping a +tripod, moving on a vehicle, etc. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +@var{x}:@var{y}:@var{w}:@var{h}:@var{rx}:@var{ry}:@var{edge}:@var{blocksize}:@var{contrast}:@var{search}:@var{filename}. + +A description of the accepted parameters follows. + +@table @option + +@item x, y, w, h +Specify a rectangular area where to limit the search for motion +vectors. +If desired the search for motion vectors can be limited to a +rectangular area of the frame defined by its top left corner, width +and height. These parameters have the same meaning as the drawbox +filter which can be used to visualise the position of the bounding +box. + +This is useful when simultaneous movement of subjects within the frame +might be confused for camera motion by the motion vector search. + +If any or all of @var{x}, @var{y}, @var{w} and @var{h} are set to -1 +then the full frame is used. This allows later options to be set +without specifying the bounding box for the motion vector search. + +Default - search the whole frame. + +@item rx, ry +Specify the maximum extent of movement in x and y directions in the +range 0-64 pixels. Default 16. + +@item edge +Specify how to generate pixels to fill blanks at the edge of the +frame. Available values are: +@table @samp +@item blank, 0 +Fill zeroes at blank locations +@item original, 1 +Original image at blank locations +@item clamp, 2 +Extruded edge value at blank locations +@item mirror, 3 +Mirrored edge at blank locations +@end table +Default value is @samp{mirror}. + +@item blocksize +Specify the blocksize to use for motion search. Range 4-128 pixels, +default 8. + +@item contrast +Specify the contrast threshold for blocks. Only blocks with more than +the specified contrast (difference between darkest and lightest +pixels) will be considered. Range 1-255, default 125. + +@item search +Specify the search strategy. Available values are: +@table @samp +@item exhaustive, 0 +Set exhaustive search +@item less, 1 +Set less exhaustive search. +@end table +Default value is @samp{exhaustive}. + +@item filename +If set then a detailed log of the motion search is written to the +specified file. + +@end table + +@section drawbox + +Draw a colored box on the input image. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +@option{x}:@option{y}:@option{width}:@option{height}:@option{color}:@option{thickness}. + +A description of the accepted options follows. + +@table @option +@item x, y +Specify the top left corner coordinates of the box. Default to 0. + +@item width, w +@item height, h +Specify the width and height of the box, if 0 they are interpreted as +the input width and height. Default to 0. + +@item color, c +Specify the color of the box to write, it can be the name of a color +(case insensitive match) or a 0xRRGGBB[AA] sequence. If the special +value @code{invert} is used, the box edge color is the same as the +video with inverted luma. + +@item thickness, t +Set the thickness of the box edge. Default value is @code{4}. +@end table + +@subsection Examples + +@itemize +@item +Draw a black box around the edge of the input image: +@example +drawbox +@end example + +@item +Draw a box with color red and an opacity of 50%: +@example +drawbox=10:20:200:60:red@@0.5 +@end example + +The previous example can be specified as: +@example +drawbox=x=10:y=20:w=200:h=60:color=red@@0.5 +@end example + +@item +Fill the box with pink color: +@example +drawbox=x=10:y=10:w=100:h=100:color=pink@@0.5:t=max +@end example +@end itemize + +@anchor{drawtext} +@section drawtext + +Draw text string or text from specified file on top of video using the +libfreetype library. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libfreetype}. + +@subsection Syntax + +The filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +The description of the accepted parameters follows. + +@table @option + +@item box +Used to draw a box around text using background color. +Value should be either 1 (enable) or 0 (disable). +The default value of @var{box} is 0. + +@item boxcolor +The color to be used for drawing box around text. +Either a string (e.g. "yellow") or in 0xRRGGBB[AA] format +(e.g. "0xff00ff"), possibly followed by an alpha specifier. +The default value of @var{boxcolor} is "white". + +@item draw +Set an expression which specifies if the text should be drawn. If the +expression evaluates to 0, the text is not drawn. This is useful for +specifying that the text should be drawn only when specific conditions +are met. + +Default value is "1". + +See below for the list of accepted constants and functions. + +@item expansion +Select how the @var{text} is expanded. Can be either @code{none}, +@code{strftime} (deprecated) or +@code{normal} (default). See the @ref{drawtext_expansion, Text expansion} section +below for details. + +@item fix_bounds +If true, check and fix text coords to avoid clipping. + +@item fontcolor +The color to be used for drawing fonts. +Either a string (e.g. "red") or in 0xRRGGBB[AA] format +(e.g. "0xff000033"), possibly followed by an alpha specifier. +The default value of @var{fontcolor} is "black". + +@item fontfile +The font file to be used for drawing text. Path must be included. +This parameter is mandatory. + +@item fontsize +The font size to be used for drawing text. +The default value of @var{fontsize} is 16. + +@item ft_load_flags +Flags to be used for loading the fonts. + +The flags map the corresponding flags supported by libfreetype, and are +a combination of the following values: +@table @var +@item default +@item no_scale +@item no_hinting +@item render +@item no_bitmap +@item vertical_layout +@item force_autohint +@item crop_bitmap +@item pedantic +@item ignore_global_advance_width +@item no_recurse +@item ignore_transform +@item monochrome +@item linear_design +@item no_autohint +@item end table +@end table + +Default value is "render". + +For more information consult the documentation for the FT_LOAD_* +libfreetype flags. + +@item shadowcolor +The color to be used for drawing a shadow behind the drawn text. It +can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA] +form (e.g. "0xff00ff"), possibly followed by an alpha specifier. +The default value of @var{shadowcolor} is "black". + +@item shadowx, shadowy +The x and y offsets for the text shadow position with respect to the +position of the text. They can be either positive or negative +values. Default value for both is "0". + +@item tabsize +The size in number of spaces to use for rendering the tab. +Default value is 4. + +@item timecode +Set the initial timecode representation in "hh:mm:ss[:;.]ff" +format. It can be used with or without text parameter. @var{timecode_rate} +option must be specified. + +@item timecode_rate, rate, r +Set the timecode frame rate (timecode only). + +@item text +The text string to be drawn. The text must be a sequence of UTF-8 +encoded characters. +This parameter is mandatory if no file is specified with the parameter +@var{textfile}. + +@item textfile +A text file containing text to be drawn. The text must be a sequence +of UTF-8 encoded characters. + +This parameter is mandatory if no text string is specified with the +parameter @var{text}. + +If both @var{text} and @var{textfile} are specified, an error is thrown. + +@item reload +If set to 1, the @var{textfile} will be reloaded before each frame. +Be sure to update it atomically, or it may be read partially, or even fail. + +@item x, y +The expressions which specify the offsets where text will be drawn +within the video frame. They are relative to the top/left border of the +output image. + +The default value of @var{x} and @var{y} is "0". + +See below for the list of accepted constants and functions. +@end table + +The parameters for @var{x} and @var{y} are expressions containing the +following constants and functions: + +@table @option +@item dar +input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar} + +@item hsub, vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item line_h, lh +the height of each text line + +@item main_h, h, H +the input height + +@item main_w, w, W +the input width + +@item max_glyph_a, ascent +the maximum distance from the baseline to the highest/upper grid +coordinate used to place a glyph outline point, for all the rendered +glyphs. +It is a positive value, due to the grid's orientation with the Y axis +upwards. + +@item max_glyph_d, descent +the maximum distance from the baseline to the lowest grid coordinate +used to place a glyph outline point, for all the rendered glyphs. +This is a negative value, due to the grid's orientation, with the Y axis +upwards. + +@item max_glyph_h +maximum glyph height, that is the maximum height for all the glyphs +contained in the rendered text, it is equivalent to @var{ascent} - +@var{descent}. + +@item max_glyph_w +maximum glyph width, that is the maximum width for all the glyphs +contained in the rendered text + +@item n +the number of input frame, starting from 0 + +@item rand(min, max) +return a random number included between @var{min} and @var{max} + +@item sar +input sample aspect ratio + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@item text_h, th +the height of the rendered text + +@item text_w, tw +the width of the rendered text + +@item x, y +the x and y offset coordinates where the text is drawn. + +These parameters allow the @var{x} and @var{y} expressions to refer +each other, so you can for example specify @code{y=x/dar}. +@end table + +If libavfilter was built with @code{--enable-fontconfig}, then +@option{fontfile} can be a fontconfig pattern or omitted. + +@anchor{drawtext_expansion} +@subsection Text expansion + +If @option{expansion} is set to @code{strftime}, +the filter recognizes strftime() sequences in the provided text and +expands them accordingly. Check the documentation of strftime(). This +feature is deprecated. + +If @option{expansion} is set to @code{none}, the text is printed verbatim. + +If @option{expansion} is set to @code{normal} (which is the default), +the following expansion mechanism is used. + +The backslash character '\', followed by any character, always expands to +the second character. + +Sequence of the form @code{%@{...@}} are expanded. The text between the +braces is a function name, possibly followed by arguments separated by ':'. +If the arguments contain special characters or delimiters (':' or '@}'), +they should be escaped. + +Note that they probably must also be escaped as the value for the +@option{text} option in the filter argument string and as the filter +argument in the filter graph description, and possibly also for the shell, +that makes up to four levels of escaping; using a text file avoids these +problems. + +The following functions are available: + +@table @command + +@item expr, e +The expression evaluation result. + +It must take one argument specifying the expression to be evaluated, +which accepts the same constants and functions as the @var{x} and +@var{y} values. Note that not all constants should be used, for +example the text size is not known when evaluating the expression, so +the constants @var{text_w} and @var{text_h} will have an undefined +value. + +@item gmtime +The time at which the filter is running, expressed in UTC. +It can accept an argument: a strftime() format string. + +@item localtime +The time at which the filter is running, expressed in the local time zone. +It can accept an argument: a strftime() format string. + +@item n, frame_num +The frame number, starting from 0. + +@item pts +The timestamp of the current frame, in seconds, with microsecond accuracy. + +@end table + +@subsection Examples + +@itemize +@item +Draw "Test Text" with font FreeSerif, using the default values for the +optional parameters. + +@example +drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" +@end example + +@item +Draw 'Test Text' with font FreeSerif of size 24 at position x=100 +and y=50 (counting from the top-left corner of the screen), text is +yellow with a red box around it. Both the text and the box have an +opacity of 20%. + +@example +drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\ + x=100: y=50: fontsize=24: fontcolor=yellow@@0.2: box=1: boxcolor=red@@0.2" +@end example + +Note that the double quotes are not necessary if spaces are not used +within the parameter list. + +@item +Show the text at the center of the video frame: +@example +drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2" +@end example + +@item +Show a text line sliding from right to left in the last row of the video +frame. The file @file{LONG_LINE} is assumed to contain a single line +with no newlines. +@example +drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" +@end example + +@item +Show the content of file @file{CREDITS} off the bottom of the frame and scroll up. +@example +drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" +@end example + +@item +Draw a single green letter "g", at the center of the input video. +The glyph baseline is placed at half screen height. +@example +drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" +@end example + +@item +Show text for 1 second every 3 seconds: +@example +drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:draw=lt(mod(t\,3)\,1):text='blink'" +@end example + +@item +Use fontconfig to set the font. Note that the colons need to be escaped. +@example +drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg' +@end example + +@item +Print the date of a real-time encoding (see strftime(3)): +@example +drawtext='fontfile=FreeSans.ttf:text=%@{localtime:%a %b %d %Y@}' +@end example + +@end itemize + +For more information about libfreetype, check: +@url{http://www.freetype.org/}. + +For more information about fontconfig, check: +@url{http://freedesktop.org/software/fontconfig/fontconfig-user.html}. + +@section edgedetect + +Detect and draw edges. The filter uses the Canny Edge Detection algorithm. + +This filter accepts the following optional named parameters: + +@table @option +@item low, high +Set low and high threshold values used by the Canny thresholding +algorithm. + +The high threshold selects the "strong" edge pixels, which are then +connected through 8-connectivity with the "weak" edge pixels selected +by the low threshold. + +@var{low} and @var{high} threshold values must be choosen in the range +[0,1], and @var{low} should be lesser or equal to @var{high}. + +Default value for @var{low} is @code{20/255}, and default value for @var{high} +is @code{50/255}. +@end table + +Example: +@example +edgedetect=low=0.1:high=0.4 +@end example + +@section fade + +Apply fade-in/out effect to input video. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the syntax +@var{type}:@var{start_frame}:@var{nb_frames}. + +A description of the accepted parameters follows. + +@table @option +@item type, t +Specify if the effect type, can be either @code{in} for fade-in, or +@code{out} for a fade-out effect. Default is @code{in}. + +@item start_frame, s +Specify the number of the start frame for starting to apply the fade +effect. Default is 0. + +@item nb_frames, n +Specify the number of frames for which the fade effect has to last. At +the end of the fade-in effect the output video will have the same +intensity as the input video, at the end of the fade-out transition +the output video will be completely black. Default is 25. + +@item alpha +If set to 1, fade only alpha channel, if one exists on the input. +Default value is 0. +@end table + +@subsection Examples + +@itemize +@item +Fade in first 30 frames of video: +@example +fade=in:0:30 +@end example + +The command above is equivalent to: +@example +fade=t=in:s=0:n=30 +@end example + +@item +Fade out last 45 frames of a 200-frame video: +@example +fade=out:155:45 +@end example + +@item +Fade in first 25 frames and fade out last 25 frames of a 1000-frame video: +@example +fade=in:0:25, fade=out:975:25 +@end example + +@item +Make first 5 frames black, then fade in from frame 5-24: +@example +fade=in:5:20 +@end example + +@item +Fade in alpha over first 25 frames of video: +@example +fade=in:0:25:alpha=1 +@end example +@end itemize + +@section field + +Extract a single field from an interlaced image using stride +arithmetic to avoid wasting CPU time. The output frames are marked as +non-interlaced. + +This filter accepts the following named options: +@table @option +@item type +Specify whether to extract the top (if the value is @code{0} or +@code{top}) or the bottom field (if the value is @code{1} or +@code{bottom}). +@end table + +If the option key is not specified, the first value sets the @var{type} +option. For example: +@example +field=bottom +@end example + +is equivalent to: +@example +field=type=bottom +@end example + +@section fieldorder + +Transform the field order of the input video. + +This filter accepts the named option @option{order} which +specifies the required field order that the input interlaced video +will be transformed to. The option name can be omitted. + +The option @option{order} can assume one of the following values: +@table @samp +@item bff +output bottom field first +@item tff +output top field first +@end table + +Default value is @samp{tff}. + +Transformation is achieved by shifting the picture content up or down +by one line, and filling the remaining line with appropriate picture content. +This method is consistent with most broadcast field order converters. + +If the input video is not flagged as being interlaced, or it is already +flagged as being of the required output field order then this filter does +not alter the incoming video. + +This filter is very useful when converting to or from PAL DV material, +which is bottom field first. + +For example: +@example +ffmpeg -i in.vob -vf "fieldorder=bff" out.dv +@end example + +@section fifo + +Buffer input images and send them when they are requested. + +This filter is mainly useful when auto-inserted by the libavfilter +framework. + +The filter does not take parameters. + +@anchor{format} +@section format + +Convert the input video to one of the specified pixel formats. +Libavfilter will try to pick one that is supported for the input to +the next filter. + +The filter accepts a list of pixel format names, separated by ":", +for example "yuv420p:monow:rgb24". + +@subsection Examples + +@itemize +@item +Convert the input video to the format @var{yuv420p} +@example +format=yuv420p +@end example + +Convert the input video to any of the formats in the list +@example +format=yuv420p:yuv444p:yuv410p +@end example +@end itemize + +@section fps + +Convert the video to specified constant framerate by duplicating or dropping +frames as necessary. + +This filter accepts the following named parameters: +@table @option + +@item fps +Desired output framerate. The default is @code{25}. + +@item round +Rounding method. + +Possible values are: +@table @option +@item zero +zero round towards 0 +@item inf +round away from 0 +@item down +round towards -infinity +@item up +round towards +infinity +@item near +round to nearest +@end table +The default is @code{near}. + +@end table + +Alternatively, the options can be specified as a flat string: +@var{fps}[:@var{round}]. + +See also the @ref{setpts} filter. + +@section framestep + +Select one frame every N. + +This filter accepts in input a string representing a positive +integer. Default argument is @code{1}. + +@anchor{frei0r} +@section frei0r + +Apply a frei0r effect to the input video. + +To enable compilation of this filter you need to install the frei0r +header and configure FFmpeg with @code{--enable-frei0r}. + +The filter supports the syntax: +@example +@var{filter_name}[@{:|=@}@var{param1}:@var{param2}:...:@var{paramN}] +@end example + +@var{filter_name} is the name of the frei0r effect to load. If the +environment variable @env{FREI0R_PATH} is defined, the frei0r effect +is searched in each one of the directories specified by the colon (or +semicolon on Windows platforms) separated list in @env{FREIOR_PATH}, +otherwise in the standard frei0r paths, which are in this order: +@file{HOME/.frei0r-1/lib/}, @file{/usr/local/lib/frei0r-1/}, +@file{/usr/lib/frei0r-1/}. + +@var{param1}, @var{param2}, ... , @var{paramN} specify the parameters +for the frei0r effect. + +A frei0r effect parameter can be a boolean (whose values are specified +with "y" and "n"), a double, a color (specified by the syntax +@var{R}/@var{G}/@var{B}, @var{R}, @var{G}, and @var{B} being float +numbers from 0.0 to 1.0) or by an @code{av_parse_color()} color +description), a position (specified by the syntax @var{X}/@var{Y}, +@var{X} and @var{Y} being float numbers) and a string. + +The number and kind of parameters depend on the loaded effect. If an +effect parameter is not specified the default value is set. + +@subsection Examples + +@itemize +@item +Apply the distort0r effect, set the first two double parameters: +@example +frei0r=distort0r:0.5:0.01 +@end example + +@item +Apply the colordistance effect, take a color as first parameter: +@example +frei0r=colordistance:0.2/0.3/0.4 +frei0r=colordistance:violet +frei0r=colordistance:0x112233 +@end example + +@item +Apply the perspective effect, specify the top left and top right image +positions: +@example +frei0r=perspective:0.2/0.2:0.8/0.2 +@end example +@end itemize + +For more information see: +@url{http://frei0r.dyne.org} + +@section geq + +The filter takes one, two, three or four equations as parameter, separated by ':'. +The first equation is mandatory and applies to the luma plane. The two +following are respectively for chroma blue and chroma red planes. + +The filter syntax allows named parameters: + +@table @option +@item lum_expr +the luminance expression +@item cb_expr +the chrominance blue expression +@item cr_expr +the chrominance red expression +@item alpha_expr +the alpha expression +@end table + +If one of the chrominance expression is not defined, it falls back on the other +one. If no alpha expression is specified it will evaluate to opaque value. +If none of chrominance expressions are +specified, they will evaluate the luminance expression. + +The expressions can use the following variables and functions: + +@table @option +@item N +The sequential number of the filtered frame, starting from @code{0}. + +@item X, Y +The coordinates of the current sample. + +@item W, H +The width and height of the image. + +@item SW, SH +Width and height scale depending on the currently filtered plane. It is the +ratio between the corresponding luma plane number of pixels and the current +plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and +@code{0.5,0.5} for chroma planes. + +@item T +Time of the current frame, expressed in seconds. + +@item p(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the current +plane. + +@item lum(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the luminance +plane. + +@item cb(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the +blue-difference chroma plane. Returns 0 if there is no such plane. + +@item cr(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the +red-difference chroma plane. Returns 0 if there is no such plane. + +@item alpha(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the alpha +plane. Returns 0 if there is no such plane. +@end table + +For functions, if @var{x} and @var{y} are outside the area, the value will be +automatically clipped to the closer edge. + +@subsection Examples + +@itemize +@item +Flip the image horizontally: +@example +geq=p(W-X\,Y) +@end example + +@item +Generate a bidimensional sine wave, with angle @code{PI/3} and a +wavelength of 100 pixels: +@example +geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128 +@end example + +@item +Generate a fancy enigmatic moving light: +@example +nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128 +@end example +@end itemize + +@section gradfun + +Fix the banding artifacts that are sometimes introduced into nearly flat +regions by truncation to 8bit color depth. +Interpolate the gradients that should go where the bands are, and +dither them. + +This filter is designed for playback only. Do not use it prior to +lossy compression, because compression tends to lose the dither and +bring back the bands. + +The filter accepts a list of options in the form of @var{key}=@var{value} pairs +separated by ":". A description of the accepted options follows. + +@table @option + +@item strength +The maximum amount by which the filter will change +any one pixel. Also the threshold for detecting nearly flat +regions. Acceptable values range from @code{0.51} to @code{64}, default value +is @code{1.2}. + +@item radius +The neighborhood to fit the gradient to. A larger +radius makes for smoother gradients, but also prevents the filter from +modifying the pixels near detailed regions. Acceptable values are +@code{8-32}, default value is @code{16}. + +@end table + +Alternatively, the options can be specified as a flat string: +@var{strength}[:@var{radius}] + +@subsection Examples + +@itemize +@item +Apply the filter with a @code{3.5} strength and radius of @code{8}: +@example +gradfun=3.5:8 +@end example + +@item +Specify radius, omitting the strength (which will fall-back to the default +value): +@example +gradfun=radius=8 +@end example + +@end itemize + +@section hflip + +Flip the input video horizontally. + +For example to horizontally flip the input video with @command{ffmpeg}: +@example +ffmpeg -i in.avi -vf "hflip" out.avi +@end example + +@section histeq +This filter applies a global color histogram equalization on a +per-frame basis. + +It can be used to correct video that has a compressed range of pixel +intensities. The filter redistributes the pixel intensities to +equalize their distribution across the intensity range. It may be +viewed as an "automatically adjusting contrast filter". This filter is +useful only for correcting degraded or poorly captured source +video. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to syntax +@var{strength}:@var{intensity}:@var{antibanding}. + +This filter accepts the following named options: + +@table @option +@item strength +Determine the amount of equalization to be applied. As the strength +is reduced, the distribution of pixel intensities more-and-more +approaches that of the input frame. The value must be a float number +in the range [0,1] and defaults to 0.200. + +@item intensity +Set the maximum intensity that can generated and scale the output +values appropriately. The strength should be set as desired and then +the intensity can be limited if needed to avoid washing-out. The value +must be a float number in the range [0,1] and defaults to 0.210. + +@item antibanding +Set the antibanding level. If enabled the filter will randomly vary +the luminance of output pixels by a small amount to avoid banding of +the histogram. Possible values are @code{none}, @code{weak} or +@code{strong}. It defaults to @code{none}. +@end table + +@section histogram + +Compute and draw a color distribution histogram for the input video. + +The computed histogram is a representation of distribution of color components +in an image. + +The filter accepts the following named parameters: + +@table @option +@item mode +Set histogram mode. + +It accepts the following values: +@table @samp +@item levels +standard histogram that display color components distribution in an image. +Displays color graph for each color component. Shows distribution +of the Y, U, V, A or G, B, R components, depending on input format, +in current frame. Bellow each graph is color component scale meter. + +@item color +chroma values in vectorscope, if brighter more such chroma values are +distributed in an image. +Displays chroma values (U/V color placement) in two dimensional graph +(which is called a vectorscope). It can be used to read of the hue and +saturation of the current frame. At a same time it is a histogram. +The whiter a pixel in the vectorscope, the more pixels of the input frame +correspond to that pixel (that is the more pixels have this chroma value). +The V component is displayed on the horizontal (X) axis, with the leftmost +side being V = 0 and the rightmost side being V = 255. +The U component is displayed on the vertical (Y) axis, with the top +representing U = 0 and the bottom representing U = 255. + +The position of a white pixel in the graph corresponds to the chroma value +of a pixel of the input clip. So the graph can be used to read of the +hue (color flavor) and the saturation (the dominance of the hue in the color). +As the hue of a color changes, it moves around the square. At the center of +the square, the saturation is zero, which means that the corresponding pixel +has no color. If you increase the amount of a specific color, while leaving +the other colors unchanged, the saturation increases, and you move towards +the edge of the square. + +@item color2 +chroma values in vectorscope, similar as @code{color} but actual chroma values +are displayed. + +@item waveform +per row/column color component graph. In row mode graph in the left side represents +color component value 0 and right side represents value = 255. In column mode top +side represents color component value = 0 and bottom side represents value = 255. +@end table +Default value is @code{levels}. + +@item level_height +Set height of level in @code{levels}. Default value is @code{200}. +Allowed range is [50, 2048]. + +@item scale_height +Set height of color scale in @code{levels}. Default value is @code{12}. +Allowed range is [0, 40]. + +@item step +Set step for @code{waveform} mode. Smaller values are useful to find out how much +of same luminance values across input rows/columns are distributed. +Default value is @code{10}. Allowed range is [1, 255]. + +@item waveform_mode +Set mode for @code{waveform}. Can be either @code{row}, or @code{column}. +Default is @code{row}. + +@item display_mode +Set display mode for @code{waveform} and @code{levels}. +It accepts the following values: +@table @samp +@item parade +Display separate graph for the color components side by side in +@code{row} waveform mode or one below other in @code{column} waveform mode +for @code{waveform} histogram mode. For @code{levels} histogram mode +per color component graphs are placed one bellow other. + +This display mode in @code{waveform} histogram mode makes it easy to spot +color casts in the highlights and shadows of an image, by comparing the +contours of the top and the bottom of each waveform. +Since whites, grays, and blacks are characterized by +exactly equal amounts of red, green, and blue, neutral areas of the +picture should display three waveforms of roughly equal width/height. +If not, the correction is easy to make by making adjustments to level the +three waveforms. + +@item overlay +Presents information that's identical to that in the @code{parade}, except +that the graphs representing color components are superimposed directly +over one another. + +This display mode in @code{waveform} histogram mode can make it easier to spot +the relative differences or similarities in overlapping areas of the color +components that are supposed to be identical, such as neutral whites, grays, +or blacks. +@end table +Default is @code{parade}. +@end table + +@subsection Examples + +@itemize + +@item +Calculate and draw histogram: +@example +ffplay -i input -vf histogram +@end example + +@end itemize + +@section hqdn3d + +High precision/quality 3d denoise filter. This filter aims to reduce +image noise producing smooth images and making still images really +still. It should enhance compressibility. + +It accepts the following optional parameters: +@var{luma_spatial}:@var{chroma_spatial}:@var{luma_tmp}:@var{chroma_tmp} + +@table @option +@item luma_spatial +a non-negative float number which specifies spatial luma strength, +defaults to 4.0 + +@item chroma_spatial +a non-negative float number which specifies spatial chroma strength, +defaults to 3.0*@var{luma_spatial}/4.0 + +@item luma_tmp +a float number which specifies luma temporal strength, defaults to +6.0*@var{luma_spatial}/4.0 + +@item chroma_tmp +a float number which specifies chroma temporal strength, defaults to +@var{luma_tmp}*@var{chroma_spatial}/@var{luma_spatial} +@end table + +@section hue + +Modify the hue and/or the saturation of the input. + +This filter accepts the following optional named options: + +@table @option +@item h +Specify the hue angle as a number of degrees. It accepts a float +number or an expression, and defaults to 0.0. + +@item H +Specify the hue angle as a number of radians. It accepts a float +number or an expression, and defaults to 0.0. + +@item s +Specify the saturation in the [-10,10] range. It accepts a float number and +defaults to 1.0. +@end table + +The @var{h}, @var{H} and @var{s} parameters are expressions containing the +following constants: + +@table @option +@item n +frame count of the input frame starting from 0 + +@item pts +presentation timestamp of the input frame expressed in time base units + +@item r +frame rate of the input video, NAN if the input frame rate is unknown + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@item tb +time base of the input video +@end table + +The options can also be set using the syntax: @var{hue}:@var{saturation} + +In this case @var{hue} is expressed in degrees. + +@subsection Examples + +@itemize +@item +Set the hue to 90 degrees and the saturation to 1.0: +@example +hue=h=90:s=1 +@end example + +@item +Same command but expressing the hue in radians: +@example +hue=H=PI/2:s=1 +@end example + +@item +Same command without named options, hue must be expressed in degrees: +@example +hue=90:1 +@end example + +@item +Note that "h:s" syntax does not support expressions for the values of +h and s, so the following example will issue an error: +@example +hue=PI/2:1 +@end example + +@item +Rotate hue and make the saturation swing between 0 +and 2 over a period of 1 second: +@example +hue="H=2*PI*t: s=sin(2*PI*t)+1" +@end example + +@item +Apply a 3 seconds saturation fade-in effect starting at 0: +@example +hue="s=min(t/3\,1)" +@end example + +The general fade-in expression can be written as: +@example +hue="s=min(0\, max((t-START)/DURATION\, 1))" +@end example + +@item +Apply a 3 seconds saturation fade-out effect starting at 5 seconds: +@example +hue="s=max(0\, min(1\, (8-t)/3))" +@end example + +The general fade-out expression can be written as: +@example +hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" +@end example + +@end itemize + +@subsection Commands + +This filter supports the following command: +@table @option +@item reinit +Modify the hue and/or the saturation of the input video. +The command accepts the same named options and syntax than when calling the +filter from the command-line. + +If a parameter is omitted, it is kept at its current value. +@end table + +@section idet + +Detect video interlacing type. + +This filter tries to detect if the input is interlaced or progressive, +top or bottom field first. + +@section il + +Deinterleave or interleave fields. + +This filter allows to process interlaced images fields without +deinterlacing them. Deinterleaving splits the input frame into 2 +fields (so called half pictures). Odd lines are moved to the top +half of the output image, even lines to the bottom half. +You can process (filter) them independently and then re-interleave them. + +It accepts a list of options in the form of @var{key}=@var{value} pairs +separated by ":". A description of the accepted options follows. + +@table @option +@item luma_mode, l +@item chroma_mode, s +@item alpha_mode, a +Available values for @var{luma_mode}, @var{chroma_mode} and +@var{alpha_mode} are: + +@table @samp +@item none +Do nothing. + +@item deinterleave, d +Deinterleave fields, placing one above the other. + +@item interleave, i +Interleave fields. Reverse the effect of deinterleaving. +@end table +Default value is @code{none}. + +@item luma_swap, ls +@item chroma_swap, cs +@item alpha_swap, as +Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is @code{0}. +@end table + +@section kerndeint + +Deinterlace input video by applying Donald Graft's adaptive kernel +deinterling. Work on interlaced parts of a video to produce +progressive frames. + +This filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to the following syntax: +@var{thresh}:@var{map}:@var{order}:@var{sharp}:@var{twoway}. + +The description of the accepted parameters follows. + +@table @option +@item thresh +Set the threshold which affects the filter's tolerance when +determining if a pixel line must be processed. It must be an integer +in the range [0,255] and defaults to 10. A value of 0 will result in +applying the process on every pixels. + +@item map +Paint pixels exceeding the threshold value to white if set to 1. +Default is 0. + +@item order +Set the fields order. Swap fields if set to 1, leave fields alone if +0. Default is 0. + +@item sharp +Enable additional sharpening if set to 1. Default is 0. + +@item twoway +Enable twoway sharpening if set to 1. Default is 0. +@end table + +@subsection Examples + +@itemize +@item +Apply default values: +@example +kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0 +@end example + +@item +Enable additional sharpening: +@example +kerndeint=sharp=1 +@end example + +@item +Paint processed pixels in white: +@example +kerndeint=map=1 +@end example +@end itemize + +@section lut, lutrgb, lutyuv + +Compute a look-up table for binding each pixel component input value +to an output value, and apply it to input video. + +@var{lutyuv} applies a lookup table to a YUV input video, @var{lutrgb} +to an RGB input video. + +These filters accept in input a ":"-separated list of options, which +specify the expressions used for computing the lookup table for the +corresponding pixel component values. + +The @var{lut} filter requires either YUV or RGB pixel formats in +input, and accepts the options: +@table @option +@item c0 +set first pixel component expression +@item c1 +set second pixel component expression +@item c2 +set third pixel component expression +@item c3 +set fourth pixel component expression, corresponds to the alpha component +@end table + +The exact component associated to each option depends on the format in +input. + +The @var{lutrgb} filter requires RGB pixel formats in input, and +accepts the options: +@table @option +@item r +set red component expression +@item g +set green component expression +@item b +set blue component expression +@item a +alpha component expression +@end table + +The @var{lutyuv} filter requires YUV pixel formats in input, and +accepts the options: +@table @option +@item y +set Y/luminance component expression +@item u +set U/Cb component expression +@item v +set V/Cr component expression +@item a +set alpha component expression +@end table + +The expressions can contain the following constants and functions: + +@table @option +@item w, h +the input width and height + +@item val +input value for the pixel component + +@item clipval +the input value clipped in the @var{minval}-@var{maxval} range + +@item maxval +maximum value for the pixel component + +@item minval +minimum value for the pixel component + +@item negval +the negated value for the pixel component value clipped in the +@var{minval}-@var{maxval} range , it corresponds to the expression +"maxval-clipval+minval" + +@item clip(val) +the computed value in @var{val} clipped in the +@var{minval}-@var{maxval} range + +@item gammaval(gamma) +the computed gamma correction value of the pixel component value +clipped in the @var{minval}-@var{maxval} range, corresponds to the +expression +"pow((clipval-minval)/(maxval-minval)\,@var{gamma})*(maxval-minval)+minval" + +@end table + +All expressions default to "val". + +@subsection Examples + +@itemize +@item +Negate input video: +@example +lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" +lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" +@end example + +The above is the same as: +@example +lutrgb="r=negval:g=negval:b=negval" +lutyuv="y=negval:u=negval:v=negval" +@end example + +@item +Negate luminance: +@example +lutyuv=y=negval +@end example + +@item +Remove chroma components, turns the video into a graytone image: +@example +lutyuv="u=128:v=128" +@end example + +@item +Apply a luma burning effect: +@example +lutyuv="y=2*val" +@end example + +@item +Remove green and blue components: +@example +lutrgb="g=0:b=0" +@end example + +@item +Set a constant alpha channel value on input: +@example +format=rgba,lutrgb=a="maxval-minval/2" +@end example + +@item +Correct luminance gamma by a 0.5 factor: +@example +lutyuv=y=gammaval(0.5) +@end example + +@item +Discard least significant bits of luma: +@example +lutyuv=y='bitand(val, 128+64+32)' +@end example +@end itemize + +@section mp + +Apply an MPlayer filter to the input video. + +This filter provides a wrapper around most of the filters of +MPlayer/MEncoder. + +This wrapper is considered experimental. Some of the wrapped filters +may not work properly and we may drop support for them, as they will +be implemented natively into FFmpeg. Thus you should avoid +depending on them when writing portable scripts. + +The filters accepts the parameters: +@var{filter_name}[:=]@var{filter_params} + +@var{filter_name} is the name of a supported MPlayer filter, +@var{filter_params} is a string containing the parameters accepted by +the named filter. + +The list of the currently supported filters follows: +@table @var +@item detc +@item dint +@item divtc +@item down3dright +@item eq2 +@item eq +@item fil +@item fspp +@item ilpack +@item ivtc +@item mcdeint +@item ow +@item perspective +@item phase +@item pp7 +@item pullup +@item qp +@item sab +@item softpulldown +@item spp +@item telecine +@item tinterlace +@item uspp +@end table + +The parameter syntax and behavior for the listed filters are the same +of the corresponding MPlayer filters. For detailed instructions check +the "VIDEO FILTERS" section in the MPlayer manual. + +@subsection Examples + +@itemize +@item +Adjust gamma, brightness, contrast: +@example +mp=eq2=1.0:2:0.5 +@end example +@end itemize + +See also mplayer(1), @url{http://www.mplayerhq.hu/}. + +@section negate + +Negate input video. + +This filter accepts an integer in input, if non-zero it negates the +alpha component (if available). The default value in input is 0. + +@section noformat + +Force libavfilter not to use any of the specified pixel formats for the +input to the next filter. + +The filter accepts a list of pixel format names, separated by ":", +for example "yuv420p:monow:rgb24". + +@subsection Examples + +@itemize +@item +Force libavfilter to use a format different from @var{yuv420p} for the +input to the vflip filter: +@example +noformat=yuv420p,vflip +@end example + +@item +Convert the input video to any of the formats not contained in the list: +@example +noformat=yuv420p:yuv444p:yuv410p +@end example +@end itemize + +@section noise + +Add noise on video input frame. + +This filter accepts a list of options in the form of @var{key}=@var{value} +pairs separated by ":". A description of the accepted options follows. + +@table @option +@item all_seed +@item c0_seed +@item c1_seed +@item c2_seed +@item c3_seed +Set noise seed for specific pixel component or all pixel components in case +of @var{all_seed}. Default value is @code{123457}. + +@item all_strength, alls +@item c0_strength, c0s +@item c1_strength, c1s +@item c2_strength, c2s +@item c3_strength, c3s +Set noise strength for specific pixel component or all pixel components in case +@var{all_strength}. Default value is @code{0}. Allowed range is [0, 100]. + +@item all_flags, allf +@item c0_flags, c0f +@item c1_flags, c1f +@item c2_flags, c2f +@item c3_flags, c3f +Set pixel component flags or set flags for all components if @var{all_flags}. +Available values for component flags are: +@table @samp +@item a +averaged temporal noise (smoother) +@item p +mix random noise with a (semi)regular pattern +@item q +higher quality (slightly better looking, slightly slower) +@item t +temporal noise (noise pattern changes between frames) +@item u +uniform noise (gaussian otherwise) +@end table +@end table + +@subsection Examples + +Add temporal and uniform noise to input video: +@example +noise=alls=20:allf=t+u +@end example + +@section null + +Pass the video source unchanged to the output. + +@section ocv + +Apply video transform using libopencv. + +To enable this filter install libopencv library and headers and +configure FFmpeg with @code{--enable-libopencv}. + +The filter takes the parameters: @var{filter_name}@{:=@}@var{filter_params}. + +@var{filter_name} is the name of the libopencv filter to apply. + +@var{filter_params} specifies the parameters to pass to the libopencv +filter. If not specified the default values are assumed. + +Refer to the official libopencv documentation for more precise +information: +@url{http://opencv.willowgarage.com/documentation/c/image_filtering.html} + +Follows the list of supported libopencv filters. + +@anchor{dilate} +@subsection dilate + +Dilate an image by using a specific structuring element. +This filter corresponds to the libopencv function @code{cvDilate}. + +It accepts the parameters: @var{struct_el}:@var{nb_iterations}. + +@var{struct_el} represents a structuring element, and has the syntax: +@var{cols}x@var{rows}+@var{anchor_x}x@var{anchor_y}/@var{shape} + +@var{cols} and @var{rows} represent the number of columns and rows of +the structuring element, @var{anchor_x} and @var{anchor_y} the anchor +point, and @var{shape} the shape for the structuring element, and +can be one of the values "rect", "cross", "ellipse", "custom". + +If the value for @var{shape} is "custom", it must be followed by a +string of the form "=@var{filename}". The file with name +@var{filename} is assumed to represent a binary image, with each +printable character corresponding to a bright pixel. When a custom +@var{shape} is used, @var{cols} and @var{rows} are ignored, the number +or columns and rows of the read file are assumed instead. + +The default value for @var{struct_el} is "3x3+0x0/rect". + +@var{nb_iterations} specifies the number of times the transform is +applied to the image, and defaults to 1. + +Follow some example: +@example +# use the default values +ocv=dilate + +# dilate using a structuring element with a 5x5 cross, iterate two times +ocv=dilate=5x5+2x2/cross:2 + +# read the shape from the file diamond.shape, iterate two times +# the file diamond.shape may contain a pattern of characters like this: +# * +# *** +# ***** +# *** +# * +# the specified cols and rows are ignored (but not the anchor point coordinates) +ocv=0x0+2x2/custom=diamond.shape:2 +@end example + +@subsection erode + +Erode an image by using a specific structuring element. +This filter corresponds to the libopencv function @code{cvErode}. + +The filter accepts the parameters: @var{struct_el}:@var{nb_iterations}, +with the same syntax and semantics as the @ref{dilate} filter. + +@subsection smooth + +Smooth the input video. + +The filter takes the following parameters: +@var{type}:@var{param1}:@var{param2}:@var{param3}:@var{param4}. + +@var{type} is the type of smooth filter to apply, and can be one of +the following values: "blur", "blur_no_scale", "median", "gaussian", +"bilateral". The default value is "gaussian". + +@var{param1}, @var{param2}, @var{param3}, and @var{param4} are +parameters whose meanings depend on smooth type. @var{param1} and +@var{param2} accept integer positive values or 0, @var{param3} and +@var{param4} accept float values. + +The default value for @var{param1} is 3, the default value for the +other parameters is 0. + +These parameters correspond to the parameters assigned to the +libopencv function @code{cvSmooth}. + +@anchor{overlay} +@section overlay + +Overlay one video on top of another. + +It takes two inputs and one output, the first input is the "main" +video on which the second input is overlayed. + +This filter accepts a list of @var{key}=@var{value} pairs as argument, +separated by ":". If the key of the first options is omitted, the +arguments are interpreted according to the syntax @var{x}:@var{y}. + +A description of the accepted options follows. + +@table @option +@item x, y +Set the expression for the x and y coordinates of the overlayed video +on the main video. Default value is 0. + +The @var{x} and @var{y} expressions can contain the following +parameters: +@table @option +@item main_w, main_h +main input width and height + +@item W, H +same as @var{main_w} and @var{main_h} + +@item overlay_w, overlay_h +overlay input width and height + +@item w, h +same as @var{overlay_w} and @var{overlay_h} +@end table + +@item format +Set the format for the output video. + +It accepts the following values: +@table @samp +@item yuv420 +force YUV420 output + +@item yuv444 +force YUV444 output + +@item rgb +force RGB output +@end table + +Default value is @samp{yuv420}. + +@item rgb @emph{(deprecated)} +If set to 1, force the filter to accept inputs in the RGB +color space. Default value is 0. This option is deprecated, use +@option{format} instead. + +@item shortest +If set to 1, force the output to terminate when the shortest input +terminates. Default value is 0. +@end table + +Be aware that frames are taken from each input video in timestamp +order, hence, if their initial timestamps differ, it is a a good idea +to pass the two inputs through a @var{setpts=PTS-STARTPTS} filter to +have them begin in the same zero timestamp, as it does the example for +the @var{movie} filter. + +You can chain together more overlays but you should test the +efficiency of such approach. + +@subsection Examples + +@itemize +@item +Draw the overlay at 10 pixels from the bottom right corner of the main +video: +@example +overlay=main_w-overlay_w-10:main_h-overlay_h-10 +@end example + +Using named options the example above becomes: +@example +overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10 +@end example + +@item +Insert a transparent PNG logo in the bottom left corner of the input, +using the @command{ffmpeg} tool with the @code{-filter_complex} option: +@example +ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output +@end example + +@item +Insert 2 different transparent PNG logos (second logo on bottom +right corner) using the @command{ffmpeg} tool: +@example +ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=10:H-h-10,overlay=W-w-10:H-h-10' output +@end example + +@item +Add a transparent color layer on top of the main video, WxH specifies +the size of the main input to the overlay filter: +@example +color=red@@.3:WxH [over]; [in][over] overlay [out] +@end example + +@item +Play an original video and a filtered version (here with the deshake +filter) side by side using the @command{ffplay} tool: +@example +ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' +@end example + +The above command is the same as: +@example +ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' +@end example + +@item +Compose output by putting two input videos side to side: +@example +ffmpeg -i left.avi -i right.avi -filter_complex " +nullsrc=size=200x100 [background]; +[0:v] setpts=PTS-STARTPTS, scale=100x100 [left]; +[1:v] setpts=PTS-STARTPTS, scale=100x100 [right]; +[background][left] overlay=shortest=1 [background+left]; +[background+left][right] overlay=shortest=1:x=100 [left+right] +" +@end example + +@item +Chain several overlays in cascade: +@example +nullsrc=s=200x200 [bg]; +testsrc=s=100x100, split=4 [in0][in1][in2][in3]; +[in0] lutrgb=r=0, [bg] overlay=0:0 [mid0]; +[in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1]; +[in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2]; +[in3] null, [mid2] overlay=100:100 [out0] +@end example + +@end itemize + +@section pad + +Add paddings to the input image, and place the original input at the +given coordinates @var{x}, @var{y}. + +The filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +If the key of the first options is omitted, the arguments are +interpreted according to the syntax +@var{width}:@var{height}:@var{x}:@var{y}:@var{color}. + +A description of the accepted options follows. + +@table @option +@item width, w +@item height, h +Specify an expression for the size of the output image with the +paddings added. If the value for @var{width} or @var{height} is 0, the +corresponding input size is used for the output. + +The @var{width} expression can reference the value set by the +@var{height} expression, and vice versa. + +The default value of @var{width} and @var{height} is 0. + +@item x +@item y +Specify an expression for the offsets where to place the input image +in the padded area with respect to the top/left border of the output +image. + +The @var{x} expression can reference the value set by the @var{y} +expression, and vice versa. + +The default value of @var{x} and @var{y} is 0. + +@item color +Specify the color of the padded area, it can be the name of a color +(case insensitive match) or a 0xRRGGBB[AA] sequence. + +The default value of @var{color} is "black". +@end table + +The value for the @var{width}, @var{height}, @var{x}, and @var{y} +options are expressions containing the following constants: + +@table @option +@item in_w, in_h +the input video width and height + +@item iw, ih +same as @var{in_w} and @var{in_h} + +@item out_w, out_h +the output width and height, that is the size of the padded area as +specified by the @var{width} and @var{height} expressions + +@item ow, oh +same as @var{out_w} and @var{out_h} + +@item x, y +x and y offsets as specified by the @var{x} and @var{y} +expressions, or NAN if not yet specified + +@item a +same as @var{iw} / @var{ih} + +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} + +@item hsub, vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@end table + +@subsection Examples + +@itemize +@item +Add paddings with color "violet" to the input video. Output video +size is 640x480, the top-left corner of the input video is placed at +column 0, row 40: +@example +pad=640:480:0:40:violet +@end example + +The example above is equivalent to the following command: +@example +pad=width=640:height=480:x=0:y=40:color=violet +@end example + +@item +Pad the input to get an output with dimensions increased by 3/2, +and put the input video at the center of the padded area: +@example +pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" +@end example + +@item +Pad the input to get a squared output with size equal to the maximum +value between the input width and height, and put the input video at +the center of the padded area: +@example +pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" +@end example + +@item +Pad the input to get a final w/h ratio of 16:9: +@example +pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" +@end example + +@item +In case of anamorphic video, in order to set the output display aspect +correctly, it is necessary to use @var{sar} in the expression, +according to the relation: +@example +(ih * X / ih) * sar = output_dar +X = output_dar / sar +@end example + +Thus the previous example needs to be modified to: +@example +pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" +@end example + +@item +Double output size and put the input video in the bottom-right +corner of the output padded area: +@example +pad="2*iw:2*ih:ow-iw:oh-ih" +@end example +@end itemize + +@section pixdesctest + +Pixel format descriptor test filter, mainly useful for internal +testing. The output video should be equal to the input video. + +For example: +@example +format=monow, pixdesctest +@end example + +can be used to test the monowhite pixel format descriptor definition. + +@section pp + +Enable the specified chain of postprocessing subfilters using libpostproc. This +library should be automatically selected with a GPL build (@code{--enable-gpl}). +Subfilters must be separated by '/' and can be disabled by prepending a '-'. +Each subfilter and some options have a short and a long name that can be used +interchangeably, i.e. dr/dering are the same. + +All subfilters share common options to determine their scope: + +@table @option +@item a/autoq +Honor the quality commands for this subfilter. + +@item c/chrom +Do chrominance filtering, too (default). + +@item y/nochrom +Do luminance filtering only (no chrominance). + +@item n/noluma +Do chrominance filtering only (no luminance). +@end table + +These options can be appended after the subfilter name, separated by a ':'. + +Available subfilters are: + +@table @option +@item hb/hdeblock[:difference[:flatness]] +Horizontal deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table + +@item vb/vdeblock[:difference[:flatness]] +Vertical deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table + +@item ha/hadeblock[:difference[:flatness]] +Accurate horizontal deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table + +@item va/vadeblock[:difference[:flatness]] +Accurate vertical deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table +@end table + +The horizontal and vertical deblocking filters share the difference and +flatness values so you cannot set different horizontal and vertical +thresholds. + +@table @option +@item h1/x1hdeblock +Experimental horizontal deblocking filter + +@item v1/x1vdeblock +Experimental vertical deblocking filter + +@item dr/dering +Deringing filter + +@item tn/tmpnoise[:threshold1[:threshold2[:threshold3]]], temporal noise reducer +@table @option +@item threshold1 +larger -> stronger filtering +@item threshold2 +larger -> stronger filtering +@item threshold3 +larger -> stronger filtering +@end table + +@item al/autolevels[:f/fullyrange], automatic brightness / contrast correction +@table @option +@item f/fullyrange +Stretch luminance to @code{0-255}. +@end table + +@item lb/linblenddeint +Linear blend deinterlacing filter that deinterlaces the given block by +filtering all lines with a @code{(1 2 1)} filter. + +@item li/linipoldeint +Linear interpolating deinterlacing filter that deinterlaces the given block by +linearly interpolating every second line. + +@item ci/cubicipoldeint +Cubic interpolating deinterlacing filter deinterlaces the given block by +cubically interpolating every second line. + +@item md/mediandeint +Median deinterlacing filter that deinterlaces the given block by applying a +median filter to every second line. + +@item fd/ffmpegdeint +FFmpeg deinterlacing filter that deinterlaces the given block by filtering every +second line with a @code{(-1 4 2 4 -1)} filter. + +@item l5/lowpass5 +Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given +block by filtering all lines with a @code{(-1 2 6 2 -1)} filter. + +@item fq/forceQuant[:quantizer] +Overrides the quantizer table from the input with the constant quantizer you +specify. +@table @option +@item quantizer +Quantizer to use +@end table + +@item de/default +Default pp filter combination (@code{hb:a,vb:a,dr:a}) + +@item fa/fast +Fast pp filter combination (@code{h1:a,v1:a,dr:a}) + +@item ac +High quality pp filter combination (@code{ha:a:128:7,va:a,dr:a}) +@end table + +@subsection Examples + +@itemize +@item +Apply horizontal and vertical deblocking, deringing and automatic +brightness/contrast: +@example +pp=hb/vb/dr/al +@end example + +@item +Apply default filters without brightness/contrast correction: +@example +pp=de/-al +@end example + +@item +Apply default filters and temporal denoiser: +@example +pp=default/tmpnoise:1:2:3 +@end example + +@item +Apply deblocking on luminance only, and switch vertical deblocking on or off +automatically depending on available CPU time: +@example +pp=hb:y/vb:a +@end example +@end itemize + +@section removelogo + +Suppress a TV station logo, using an image file to determine which +pixels comprise the logo. It works by filling in the pixels that +comprise the logo with neighboring pixels. + +This filter requires one argument which specifies the filter bitmap +file, which can be any image format supported by libavformat. The +width and height of the image file must match those of the video +stream being processed. + +Pixels in the provided bitmap image with a value of zero are not +considered part of the logo, non-zero pixels are considered part of +the logo. If you use white (255) for the logo and black (0) for the +rest, you will be safe. For making the filter bitmap, it is +recommended to take a screen capture of a black frame with the logo +visible, and then using a threshold filter followed by the erode +filter once or twice. + +If needed, little splotches can be fixed manually. Remember that if +logo pixels are not covered, the filter quality will be much +reduced. Marking too many pixels as part of the logo does not hurt as +much, but it will increase the amount of blurring needed to cover over +the image and will destroy more information than necessary, and extra +pixels will slow things down on a large logo. + +@section scale + +Scale (resize) the input video, using the libswscale library. + +The scale filter forces the output display aspect ratio to be the same +of the input, by changing the output sample aspect ratio. + +This filter accepts a list of named options in the form of +@var{key}=@var{value} pairs separated by ":". If the key for the first +two options is not specified, the assumed keys for the first two +values are @code{w} and @code{h}. If the first option has no key and +can be interpreted like a video size specification, it will be used +to set the video size. + +A description of the accepted options follows. + +@table @option +@item width, w +Set the video width expression, default value is @code{iw}. See below +for the list of accepted constants. + +@item height, h +Set the video heiht expression, default value is @code{ih}. +See below for the list of accepted constants. + +@item interl +Set the interlacing. It accepts the following values: + +@table @option +@item 1 +force interlaced aware scaling + +@item 0 +do not apply interlaced scaling + +@item -1 +select interlaced aware scaling depending on whether the source frames +are flagged as interlaced or not +@end table + +Default value is @code{0}. + +@item flags +Set libswscale scaling flags. If not explictly specified the filter +applies a bilinear scaling algorithm. + +@item size, s +Set the video size, the value must be a valid abbreviation or in the +form @var{width}x@var{height}. +@end table + +The values of the @var{w} and @var{h} options are expressions +containing the following constants: + +@table @option +@item in_w, in_h +the input width and height + +@item iw, ih +same as @var{in_w} and @var{in_h} + +@item out_w, out_h +the output (cropped) width and height + +@item ow, oh +same as @var{out_w} and @var{out_h} + +@item a +same as @var{iw} / @var{ih} + +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} + +@item hsub, vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@end table + +If the input image format is different from the format requested by +the next filter, the scale filter will convert the input to the +requested format. + +If the value for @var{width} or @var{height} is 0, the respective input +size is used for the output. + +If the value for @var{width} or @var{height} is -1, the scale filter will +use, for the respective output size, a value that maintains the aspect +ratio of the input image. + +@subsection Examples + +@itemize +@item +Scale the input video to a size of 200x100: +@example +scale=200:100 +@end example + +This is equivalent to: +@example +scale=w=200:h=100 +@end example + +or: +@example +scale=200x100 +@end example + +@item +Specify a size abbreviation for the output size: +@example +scale=qcif +@end example + +which can also be written as: +@example +scale=size=qcif +@end example + +@item +Scale the input to 2x: +@example +scale=2*iw:2*ih +@end example + +@item +The above is the same as: +@example +scale=2*in_w:2*in_h +@end example + +@item +Scale the input to 2x with forced interlaced scaling: +@example +scale=2*iw:2*ih:interl=1 +@end example + +@item +Scale the input to half size: +@example +scale=iw/2:ih/2 +@end example + +@item +Increase the width, and set the height to the same size: +@example +scale=3/2*iw:ow +@end example + +@item +Seek for Greek harmony: +@example +scale=iw:1/PHI*iw +scale=ih*PHI:ih +@end example + +@item +Increase the height, and set the width to 3/2 of the height: +@example +scale=3/2*oh:3/5*ih +@end example + +@item +Increase the size, but make the size a multiple of the chroma: +@example +scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" +@end example + +@item +Increase the width to a maximum of 500 pixels, keep the same input +aspect ratio: +@example +scale='min(500\, iw*3/2):-1' +@end example +@end itemize + +@section setdar, setsar + +The @code{setdar} filter sets the Display Aspect Ratio for the filter +output video. + +This is done by changing the specified Sample (aka Pixel) Aspect +Ratio, according to the following equation: +@example +@var{DAR} = @var{HORIZONTAL_RESOLUTION} / @var{VERTICAL_RESOLUTION} * @var{SAR} +@end example + +Keep in mind that the @code{setdar} filter does not modify the pixel +dimensions of the video frame. Also the display aspect ratio set by +this filter may be changed by later filters in the filterchain, +e.g. in case of scaling or if another "setdar" or a "setsar" filter is +applied. + +The @code{setsar} filter sets the Sample (aka Pixel) Aspect Ratio for +the filter output video. + +Note that as a consequence of the application of this filter, the +output display aspect ratio will change according to the equation +above. + +Keep in mind that the sample aspect ratio set by the @code{setsar} +filter may be changed by later filters in the filterchain, e.g. if +another "setsar" or a "setdar" filter is applied. + +The @code{setdar} and @code{setsar} filters accept a string in the +form @var{num}:@var{den} expressing an aspect ratio, or the following +named options, expressed as a sequence of @var{key}=@var{value} pairs, +separated by ":". + +@table @option +@item max +Set the maximum integer value to use for expressing numerator and +denominator when reducing the expressed aspect ratio to a rational. +Default value is @code{100}. + +@item r, ratio: +Set the aspect ratio used by the filter. + +The parameter can be a floating point number string, an expression, or +a string of the form @var{num}:@var{den}, where @var{num} and +@var{den} are the numerator and denominator of the aspect ratio. If +the parameter is not specified, it is assumed the value "0". +In case the form "@var{num}:@var{den}" the @code{:} character should +be escaped. +@end table + +If the keys are omitted in the named options list, the specifed values +are assumed to be @var{ratio} and @var{max} in that order. + +For example to change the display aspect ratio to 16:9, specify: +@example +setdar='16:9' +@end example + +The example above is equivalent to: +@example +setdar=1.77777 +@end example + +To change the sample aspect ratio to 10:11, specify: +@example +setsar='10:11' +@end example + +To set a display aspect ratio of 16:9, and specify a maximum integer value of +1000 in the aspect ratio reduction, use the command: +@example +setdar=ratio='16:9':max=1000 +@end example + +@section setfield + +Force field for the output video frame. + +The @code{setfield} filter marks the interlace type field for the +output frames. It does not change the input frame, but only sets the +corresponding property, which affects how the frame is treated by +following filters (e.g. @code{fieldorder} or @code{yadif}). + +This filter accepts a single option @option{mode}, which can be +specified either by setting @code{mode=VALUE} or setting the value +alone. Available values are: + +@table @samp +@item auto +Keep the same field property. + +@item bff +Mark the frame as bottom-field-first. + +@item tff +Mark the frame as top-field-first. + +@item prog +Mark the frame as progressive. +@end table + +@section showinfo + +Show a line containing various information for each input video frame. +The input video is not modified. + +The shown line contains a sequence of key/value pairs of the form +@var{key}:@var{value}. + +A description of each shown parameter follows: + +@table @option +@item n +sequential number of the input frame, starting from 0 + +@item pts +Presentation TimeStamp of the input frame, expressed as a number of +time base units. The time base unit depends on the filter input pad. + +@item pts_time +Presentation TimeStamp of the input frame, expressed as a number of +seconds + +@item pos +position of the frame in the input stream, -1 if this information in +unavailable and/or meaningless (for example in case of synthetic video) + +@item fmt +pixel format name + +@item sar +sample aspect ratio of the input frame, expressed in the form +@var{num}/@var{den} + +@item s +size of the input frame, expressed in the form +@var{width}x@var{height} + +@item i +interlaced mode ("P" for "progressive", "T" for top field first, "B" +for bottom field first) + +@item iskey +1 if the frame is a key frame, 0 otherwise + +@item type +picture type of the input frame ("I" for an I-frame, "P" for a +P-frame, "B" for a B-frame, "?" for unknown type). +Check also the documentation of the @code{AVPictureType} enum and of +the @code{av_get_picture_type_char} function defined in +@file{libavutil/avutil.h}. + +@item checksum +Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame + +@item plane_checksum +Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, +expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3}]" +@end table + +@section smartblur + +Blur the input video without impacting the outlines. + +This filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +If the key of the first options is omitted, the arguments are +interpreted according to the syntax: +@var{luma_radius}:@var{luma_strength}:@var{luma_threshold}[:@var{chroma_radius}:@var{chroma_strength}:@var{chroma_threshold}] + +A description of the accepted options follows. + +@table @option +@item luma_radius, lr +@item chroma_radius, cr +Set the luma/chroma radius. The option value must be a float number in +the range [0.1,5.0] that specifies the variance of the gaussian filter +used to blur the image (slower if larger). Default value is 1.0. + +@item luma_strength, ls +@item chroma_strength, cs +Set the luma/chroma strength. The option value must be a float number +in the range [-1.0,1.0] that configures the blurring. A value included +in [0.0,1.0] will blur the image whereas a value included in +[-1.0,0.0] will sharpen the image. Default value is 1.0. + +@item luma_threshold, lt +@item chroma_threshold, ct +Set the luma/chroma threshold used as a coefficient to determine +whether a pixel should be blurred or not. The option value must be an +integer in the range [-30,30]. A value of 0 will filter all the image, +a value included in [0,30] will filter flat areas and a value included +in [-30,0] will filter edges. Default value is 0. +@end table + +If a chroma option is not explicitly set, the corresponding luma value +is set. + +@section stereo3d + +Convert between different stereoscopic image formats. + +This filter accepts the following named options, expressed as a +sequence of @var{key}=@var{value} pairs, separated by ":". + +@table @option +@item in +Set stereoscopic image format of input. + +Available values for input image formats are: +@table @samp +@item sbsl +side by side parallel (left eye left, right eye right) + +@item sbsr +side by side crosseye (right eye left, left eye right) + +@item sbs2l +side by side parallel with half width resolution +(left eye left, right eye right) + +@item sbs2r +side by side crosseye with half width resolution +(right eye left, left eye right) + +@item abl +above-below (left eye above, right eye below) + +@item abr +above-below (right eye above, left eye below) + +@item ab2l +above-below with half height resolution +(left eye above, right eye below) + +@item ab2r +above-below with half height resolution +(right eye above, left eye below) + +Default value is @samp{sbsl}. +@end table + +@item out +Set stereoscopic image format of output. + +Available values for output image formats are all the input formats as well as: +@table @samp +@item arbg +anaglyph red/blue gray +(red filter on left eye, blue filter on right eye) + +@item argg +anaglyph red/green gray +(red filter on left eye, green filter on right eye) + +@item arcg +anaglyph red/cyan gray +(red filter on left eye, cyan filter on right eye) + +@item arch +anaglyph red/cyan half colored +(red filter on left eye, cyan filter on right eye) + +@item arcc +anaglyph red/cyan color +(red filter on left eye, cyan filter on right eye) + +@item arcd +anaglyph red/cyan color optimized with the least squares projection of dubois +(red filter on left eye, cyan filter on right eye) + +@item agmg +anaglyph green/magenta gray +(green filter on left eye, magenta filter on right eye) + +@item agmh +anaglyph green/magenta half colored +(green filter on left eye, magenta filter on right eye) + +@item agmc +anaglyph green/magenta colored +(green filter on left eye, magenta filter on right eye) + +@item agmd +anaglyph green/magenta color optimized with the least squares projection of dubois +(green filter on left eye, magenta filter on right eye) + +@item aybg +anaglyph yellow/blue gray +(yellow filter on left eye, blue filter on right eye) + +@item aybh +anaglyph yellow/blue half colored +(yellow filter on left eye, blue filter on right eye) + +@item aybc +anaglyph yellow/blue colored +(yellow filter on left eye, blue filter on right eye) + +@item aybd +anaglyph yellow/blue color optimized with the least squares projection of dubois +(yellow filter on left eye, blue filter on right eye) + +@item irl +interleaved rows (left eye has top row, right eye starts on next row) + +@item irr +interleaved rows (right eye has top row, left eye starts on next row) + +@item ml +mono output (left eye only) + +@item mr +mono output (right eye only) +@end table + +Default value is @samp{arcd}. +@end table + +@anchor{subtitles} +@section subtitles + +Draw subtitles on top of input video using the libass library. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libass}. This filter also requires a build with libavcodec and +libavformat to convert the passed subtitles file to ASS (Advanced Substation +Alpha) subtitles format. + +This filter accepts the following named options, expressed as a +sequence of @var{key}=@var{value} pairs, separated by ":". + +@table @option +@item filename, f +Set the filename of the subtitle file to read. It must be specified. + +@item original_size +Specify the size of the original video, the video for which the ASS file +was composed. Due to a misdesign in ASS aspect ratio arithmetic, this is +necessary to correctly scale the fonts if the aspect ratio has been changed. + +@item charenc +Set subtitles input character encoding. @code{subtitles} filter only. Only +useful if not UTF-8. +@end table + +If the first key is not specified, it is assumed that the first value +specifies the @option{filename}. + +For example, to render the file @file{sub.srt} on top of the input +video, use the command: +@example +subtitles=sub.srt +@end example + +which is equivalent to: +@example +subtitles=filename=sub.srt +@end example + +@section split + +Split input video into several identical outputs. + +The filter accepts a single parameter which specifies the number of outputs. If +unspecified, it defaults to 2. + +For example +@example +ffmpeg -i INPUT -filter_complex split=5 OUTPUT +@end example +will create 5 copies of the input video. + +For example: +@example +[in] split [splitout1][splitout2]; +[splitout1] crop=100:100:0:0 [cropout]; +[splitout2] pad=200:200:100:100 [padout]; +@end example + +will create two separate outputs from the same input, one cropped and +one padded. + +@section super2xsai + +Scale the input by 2x and smooth using the Super2xSaI (Scale and +Interpolate) pixel art scaling algorithm. + +Useful for enlarging pixel art images without reducing sharpness. + +@section swapuv +Swap U & V plane. + +@section thumbnail +Select the most representative frame in a given sequence of consecutive frames. + +It accepts as argument the frames batch size to analyze (default @var{N}=100); +in a set of @var{N} frames, the filter will pick one of them, and then handle +the next batch of @var{N} frames until the end. + +Since the filter keeps track of the whole frames sequence, a bigger @var{N} +value will result in a higher memory usage, so a high value is not recommended. + +The following example extract one picture each 50 frames: +@example +thumbnail=50 +@end example + +Complete example of a thumbnail creation with @command{ffmpeg}: +@example +ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png +@end example + +@section tile + +Tile several successive frames together. + +It accepts a list of options in the form of @var{key}=@var{value} pairs +separated by ":". A description of the accepted options follows. + +@table @option + +@item layout +Set the grid size (i.e. the number of lines and columns) in the form +"@var{w}x@var{h}". + +@item margin +Set the outer border margin in pixels. + +@item padding +Set the inner border thickness (i.e. the number of pixels between frames). For +more advanced padding options (such as having different values for the edges), +refer to the pad video filter. + +@item nb_frames +Set the maximum number of frames to render in the given area. It must be less +than or equal to @var{w}x@var{h}. The default value is @code{0}, meaning all +the area will be used. + +@end table + +Alternatively, the options can be specified as a flat string: + +@var{layout}[:@var{nb_frames}[:@var{margin}[:@var{padding}]]] + +For example, produce 8x8 PNG tiles of all keyframes (@option{-skip_frame +nokey}) in a movie: +@example +ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png +@end example +The @option{-vsync 0} is necessary to prevent @command{ffmpeg} from +duplicating each output frame to accomodate the originally detected frame +rate. + +Another example to display @code{5} pictures in an area of @code{3x2} frames, +with @code{7} pixels between them, and @code{2} pixels of initial margin, using +mixed flat and named options: +@example +tile=3x2:nb_frames=5:padding=7:margin=2 +@end example + +@section tinterlace + +Perform various types of temporal field interlacing. + +Frames are counted starting from 1, so the first input frame is +considered odd. + +This filter accepts options in the form of @var{key}=@var{value} pairs +separated by ":". +Alternatively, the @var{mode} option can be specified as a value alone, +optionally followed by a ":" and further ":" separated @var{key}=@var{value} +pairs. + +A description of the accepted options follows. + +@table @option + +@item mode +Specify the mode of the interlacing. This option can also be specified +as a value alone. See below for a list of values for this option. + +Available values are: + +@table @samp +@item merge, 0 +Move odd frames into the upper field, even into the lower field, +generating a double height frame at half framerate. + +@item drop_odd, 1 +Only output even frames, odd frames are dropped, generating a frame with +unchanged height at half framerate. + +@item drop_even, 2 +Only output odd frames, even frames are dropped, generating a frame with +unchanged height at half framerate. + +@item pad, 3 +Expand each frame to full height, but pad alternate lines with black, +generating a frame with double height at the same input framerate. + +@item interleave_top, 4 +Interleave the upper field from odd frames with the lower field from +even frames, generating a frame with unchanged height at half framerate. + +@item interleave_bottom, 5 +Interleave the lower field from odd frames with the upper field from +even frames, generating a frame with unchanged height at half framerate. + +@item interlacex2, 6 +Double frame rate with unchanged height. Frames are inserted each +containing the second temporal field from the previous input frame and +the first temporal field from the next input frame. This mode relies on +the top_field_first flag. Useful for interlaced video displays with no +field synchronisation. +@end table + +Numeric values are deprecated but are accepted for backward +compatibility reasons. + +Default mode is @code{merge}. + +@item flags +Specify flags influencing the filter process. + +Available value for @var{flags} is: + +@table @option +@item low_pass_filter, vlfp +Enable vertical low-pass filtering in the filter. +Vertical low-pass filtering is required when creating an interlaced +destination from a progressive source which contains high-frequency +vertical detail. Filtering will reduce interlace 'twitter' and Moire +patterning. + +Vertical low-pass filtering can only be enabled for @option{mode} +@var{interleave_top} and @var{interleave_bottom}. + +@end table +@end table + +@section transpose + +Transpose rows with columns in the input video and optionally flip it. + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ':'. If the key of the first options is omitted, +the arguments are interpreted according to the syntax +@var{dir}:@var{passthrough}. + +@table @option +@item dir +Specify the transposition direction. Can assume the following values: + +@table @samp +@item 0, 4 +Rotate by 90 degrees counterclockwise and vertically flip (default), that is: +@example +L.R L.l +. . -> . . +l.r R.r +@end example + +@item 1, 5 +Rotate by 90 degrees clockwise, that is: +@example +L.R l.L +. . -> . . +l.r r.R +@end example + +@item 2, 6 +Rotate by 90 degrees counterclockwise, that is: +@example +L.R R.r +. . -> . . +l.r L.l +@end example + +@item 3, 7 +Rotate by 90 degrees clockwise and vertically flip, that is: +@example +L.R r.R +. . -> . . +l.r l.L +@end example +@end table + +For values between 4-7, the transposition is only done if the input +video geometry is portrait and not landscape. These values are +deprecated, the @code{passthrough} option should be used instead. + +@item passthrough +Do not apply the transposition if the input geometry matches the one +specified by the specified value. It accepts the following values: +@table @samp +@item none +Always apply transposition. +@item portrait +Preserve portrait geometry (when @var{height} >= @var{width}). +@item landscape +Preserve landscape geometry (when @var{width} >= @var{height}). +@end table + +Default value is @code{none}. +@end table + +For example to rotate by 90 degrees clockwise and preserve portrait +layout: +@example +transpose=dir=1:passthrough=portrait +@end example + +The command above can also be specified as: +@example +transpose=1:portrait +@end example + +@section unsharp + +Sharpen or blur the input video. + +This filter accepts parameters as a list of @var{key}=@var{value} pairs, +separated by ":". + +If the key of the first options is omitted, the arguments are +interpreted according to the syntax: +@var{luma_msize_x}:@var{luma_msize_y}:@var{luma_amount}:@var{chroma_msize_x}:@var{chroma_msize_y}:@var{chroma_amount} + +A description of the accepted options follows. + +@table @option +@item luma_msize_x, lx +@item chroma_msize_x, cx +Set the luma/chroma matrix horizontal size. It must be an odd integer +between 3 and 63, default value is 5. + +@item luma_msize_y, ly +@item chroma_msize_y, cy +Set the luma/chroma matrix vertical size. It must be an odd integer +between 3 and 63, default value is 5. + +@item luma_amount, la +@item chroma_amount, ca +Set the luma/chroma effect strength. It can be a float number, +reasonable values lay between -1.5 and 1.5. + +Negative values will blur the input video, while positive values will +sharpen it, a value of zero will disable the effect. + +Default value is 1.0 for @option{luma_amount}, 0.0 for +@option{chroma_amount}. +@end table + +@subsection Examples + +@itemize +@item +Apply strong luma sharpen effect: +@example +unsharp=7:7:2.5 +@end example + +@item +Apply strong blur of both luma and chroma parameters: +@example +unsharp=7:7:-2:7:7:-2 +@end example +@end itemize + +@section vflip + +Flip the input video vertically. + +@example +ffmpeg -i in.avi -vf "vflip" out.avi +@end example + +@section yadif + +Deinterlace the input video ("yadif" means "yet another deinterlacing +filter"). + +The filter accepts parameters as a list of @var{key}=@var{value} +pairs, separated by ":". If the key of the first options is omitted, +the arguments are interpreted according to syntax +@var{mode}:@var{parity}:@var{deint}. + +The description of the accepted parameters follows. + +@table @option +@item mode +Specify the interlacing mode to adopt. Accept one of the following +values: + +@table @option +@item 0, send_frame +output 1 frame for each frame +@item 1, send_field +output 1 frame for each field +@item 2, send_frame_nospatial +like @code{send_frame} but skip spatial interlacing check +@item 3, send_field_nospatial +like @code{send_field} but skip spatial interlacing check +@end table + +Default value is @code{send_frame}. + +@item parity +Specify the picture field parity assumed for the input interlaced +video. Accept one of the following values: + +@table @option +@item 0, tff +assume top field first +@item 1, bff +assume bottom field first +@item -1, auto +enable automatic detection +@end table + +Default value is @code{auto}. +If interlacing is unknown or decoder does not export this information, +top field first will be assumed. + +@item deint +Specify which frames to deinterlace. Accept one of the following +values: + +@table @option +@item 0, all +deinterlace all frames +@item 1, interlaced +only deinterlace frames marked as interlaced +@end table + +Default value is @code{all}. +@end table + +@c man end VIDEO FILTERS + +@chapter Video Sources +@c man begin VIDEO SOURCES + +Below is a description of the currently available video sources. + +@section buffer + +Buffer video frames, and make them available to the filter chain. + +This source is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/vsrc_buffer.h}. + +It accepts a list of options in the form of @var{key}=@var{value} pairs +separated by ":". A description of the accepted options follows. + +@table @option + +@item video_size +Specify the size (width and height) of the buffered video frames. + +@item pix_fmt +A string representing the pixel format of the buffered video frames. +It may be a number corresponding to a pixel format, or a pixel format +name. + +@item time_base +Specify the timebase assumed by the timestamps of the buffered frames. + +@item time_base +Specify the frame rate expected for the video stream. + +@item pixel_aspect +Specify the sample aspect ratio assumed by the video frames. + +@item sws_param +Specify the optional parameters to be used for the scale filter which +is automatically inserted when an input change is detected in the +input size or format. +@end table + +For example: +@example +buffer=size=320x240:pix_fmt=yuv410p:time_base=1/24:pixel_aspect=1/1 +@end example + +will instruct the source to accept video frames with size 320x240 and +with format "yuv410p", assuming 1/24 as the timestamps timebase and +square pixels (1:1 sample aspect ratio). +Since the pixel format with name "yuv410p" corresponds to the number 6 +(check the enum AVPixelFormat definition in @file{libavutil/pixfmt.h}), +this example corresponds to: +@example +buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1 +@end example + +Alternatively, the options can be specified as a flat string, but this +syntax is deprecated: + +@var{width}:@var{height}:@var{pix_fmt}:@var{time_base.num}:@var{time_base.den}:@var{pixel_aspect.num}:@var{pixel_aspect.den}[:@var{sws_param}] + +@section cellauto + +Create a pattern generated by an elementary cellular automaton. + +The initial state of the cellular automaton can be defined through the +@option{filename}, and @option{pattern} options. If such options are +not specified an initial state is created randomly. + +At each new frame a new row in the video is filled with the result of +the cellular automaton next generation. The behavior when the whole +frame is filled is defined by the @option{scroll} option. + +This source accepts a list of options in the form of +@var{key}=@var{value} pairs separated by ":". A description of the +accepted options follows. + +@table @option +@item filename, f +Read the initial cellular automaton state, i.e. the starting row, from +the specified file. +In the file, each non-whitespace character is considered an alive +cell, a newline will terminate the row, and further characters in the +file will be ignored. + +@item pattern, p +Read the initial cellular automaton state, i.e. the starting row, from +the specified string. + +Each non-whitespace character in the string is considered an alive +cell, a newline will terminate the row, and further characters in the +string will be ignored. + +@item rate, r +Set the video rate, that is the number of frames generated per second. +Default is 25. + +@item random_fill_ratio, ratio +Set the random fill ratio for the initial cellular automaton row. It +is a floating point number value ranging from 0 to 1, defaults to +1/PHI. + +This option is ignored when a file or a pattern is specified. + +@item random_seed, seed +Set the seed for filling randomly the initial row, must be an integer +included between 0 and UINT32_MAX. If not specified, or if explicitly +set to -1, the filter will try to use a good random seed on a best +effort basis. + +@item rule +Set the cellular automaton rule, it is a number ranging from 0 to 255. +Default value is 110. + +@item size, s +Set the size of the output video. + +If @option{filename} or @option{pattern} is specified, the size is set +by default to the width of the specified initial state row, and the +height is set to @var{width} * PHI. + +If @option{size} is set, it must contain the width of the specified +pattern string, and the specified pattern will be centered in the +larger row. + +If a filename or a pattern string is not specified, the size value +defaults to "320x518" (used for a randomly generated initial state). + +@item scroll +If set to 1, scroll the output upward when all the rows in the output +have been already filled. If set to 0, the new generated row will be +written over the top row just after the bottom row is filled. +Defaults to 1. + +@item start_full, full +If set to 1, completely fill the output with generated rows before +outputting the first frame. +This is the default behavior, for disabling set the value to 0. + +@item stitch +If set to 1, stitch the left and right row edges together. +This is the default behavior, for disabling set the value to 0. +@end table + +@subsection Examples + +@itemize +@item +Read the initial state from @file{pattern}, and specify an output of +size 200x400. +@example +cellauto=f=pattern:s=200x400 +@end example + +@item +Generate a random initial row with a width of 200 cells, with a fill +ratio of 2/3: +@example +cellauto=ratio=2/3:s=200x200 +@end example + +@item +Create a pattern generated by rule 18 starting by a single alive cell +centered on an initial row with width 100: +@example +cellauto=p=@@:s=100x400:full=0:rule=18 +@end example + +@item +Specify a more elaborated initial pattern: +@example +cellauto=p='@@@@ @@ @@@@':s=100x400:full=0:rule=18 +@end example + +@end itemize + +@section mandelbrot + +Generate a Mandelbrot set fractal, and progressively zoom towards the +point specified with @var{start_x} and @var{start_y}. + +This source accepts a list of options in the form of +@var{key}=@var{value} pairs separated by ":". A description of the +accepted options follows. + +@table @option + +@item end_pts +Set the terminal pts value. Default value is 400. + +@item end_scale +Set the terminal scale value. +Must be a floating point value. Default value is 0.3. + +@item inner +Set the inner coloring mode, that is the algorithm used to draw the +Mandelbrot fractal internal region. + +It shall assume one of the following values: +@table @option +@item black +Set black mode. +@item convergence +Show time until convergence. +@item mincol +Set color based on point closest to the origin of the iterations. +@item period +Set period mode. +@end table + +Default value is @var{mincol}. + +@item bailout +Set the bailout value. Default value is 10.0. + +@item maxiter +Set the maximum of iterations performed by the rendering +algorithm. Default value is 7189. + +@item outer +Set outer coloring mode. +It shall assume one of following values: +@table @option +@item iteration_count +Set iteration cound mode. +@item normalized_iteration_count +set normalized iteration count mode. +@end table +Default value is @var{normalized_iteration_count}. + +@item rate, r +Set frame rate, expressed as number of frames per second. Default +value is "25". + +@item size, s +Set frame size. Default value is "640x480". + +@item start_scale +Set the initial scale value. Default value is 3.0. + +@item start_x +Set the initial x position. Must be a floating point value between +-100 and 100. Default value is -0.743643887037158704752191506114774. + +@item start_y +Set the initial y position. Must be a floating point value between +-100 and 100. Default value is -0.131825904205311970493132056385139. +@end table + +@section mptestsrc + +Generate various test patterns, as generated by the MPlayer test filter. + +The size of the generated video is fixed, and is 256x256. +This source is useful in particular for testing encoding features. + +This source accepts an optional sequence of @var{key}=@var{value} pairs, +separated by ":". The description of the accepted options follows. + +@table @option + +@item rate, r +Specify the frame rate of the sourced video, as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float +number or a valid video frame rate abbreviation. The default value is +"25". + +@item duration, d +Set the video duration of the sourced video. The accepted syntax is: +@example +[-]HH:MM:SS[.m...] +[-]S+[.m...] +@end example +See also the function @code{av_parse_time()}. + +If not specified, or the expressed duration is negative, the video is +supposed to be generated forever. + +@item test, t + +Set the number or the name of the test to perform. Supported tests are: +@table @option +@item dc_luma +@item dc_chroma +@item freq_luma +@item freq_chroma +@item amp_luma +@item amp_chroma +@item cbp +@item mv +@item ring1 +@item ring2 +@item all +@end table + +Default value is "all", which will cycle through the list of all tests. +@end table + +For example the following: +@example +testsrc=t=dc_luma +@end example + +will generate a "dc_luma" test pattern. + +@section frei0r_src + +Provide a frei0r source. + +To enable compilation of this filter you need to install the frei0r +header and configure FFmpeg with @code{--enable-frei0r}. + +The source supports the syntax: +@example +@var{size}:@var{rate}:@var{src_name}[@{=|:@}@var{param1}:@var{param2}:...:@var{paramN}] +@end example + +@var{size} is the size of the video to generate, may be a string of the +form @var{width}x@var{height} or a frame size abbreviation. +@var{rate} is the rate of the video to generate, may be a string of +the form @var{num}/@var{den} or a frame rate abbreviation. +@var{src_name} is the name to the frei0r source to load. For more +information regarding frei0r and how to set the parameters read the +section @ref{frei0r} in the description of the video filters. + +For example, to generate a frei0r partik0l source with size 200x200 +and frame rate 10 which is overlayed on the overlay filter main input: +@example +frei0r_src=200x200:10:partik0l=1234 [overlay]; [in][overlay] overlay +@end example + +@section life + +Generate a life pattern. + +This source is based on a generalization of John Conway's life game. + +The sourced input represents a life grid, each pixel represents a cell +which can be in one of two possible states, alive or dead. Every cell +interacts with its eight neighbours, which are the cells that are +horizontally, vertically, or diagonally adjacent. + +At each interaction the grid evolves according to the adopted rule, +which specifies the number of neighbor alive cells which will make a +cell stay alive or born. The @option{rule} option allows to specify +the rule to adopt. + +This source accepts a list of options in the form of +@var{key}=@var{value} pairs separated by ":". A description of the +accepted options follows. + +@table @option +@item filename, f +Set the file from which to read the initial grid state. In the file, +each non-whitespace character is considered an alive cell, and newline +is used to delimit the end of each row. + +If this option is not specified, the initial grid is generated +randomly. + +@item rate, r +Set the video rate, that is the number of frames generated per second. +Default is 25. + +@item random_fill_ratio, ratio +Set the random fill ratio for the initial random grid. It is a +floating point number value ranging from 0 to 1, defaults to 1/PHI. +It is ignored when a file is specified. + +@item random_seed, seed +Set the seed for filling the initial random grid, must be an integer +included between 0 and UINT32_MAX. If not specified, or if explicitly +set to -1, the filter will try to use a good random seed on a best +effort basis. + +@item rule +Set the life rule. + +A rule can be specified with a code of the kind "S@var{NS}/B@var{NB}", +where @var{NS} and @var{NB} are sequences of numbers in the range 0-8, +@var{NS} specifies the number of alive neighbor cells which make a +live cell stay alive, and @var{NB} the number of alive neighbor cells +which make a dead cell to become alive (i.e. to "born"). +"s" and "b" can be used in place of "S" and "B", respectively. + +Alternatively a rule can be specified by an 18-bits integer. The 9 +high order bits are used to encode the next cell state if it is alive +for each number of neighbor alive cells, the low order bits specify +the rule for "borning" new cells. Higher order bits encode for an +higher number of neighbor cells. +For example the number 6153 = @code{(12<<9)+9} specifies a stay alive +rule of 12 and a born rule of 9, which corresponds to "S23/B03". + +Default value is "S23/B3", which is the original Conway's game of life +rule, and will keep a cell alive if it has 2 or 3 neighbor alive +cells, and will born a new cell if there are three alive cells around +a dead cell. + +@item size, s +Set the size of the output video. + +If @option{filename} is specified, the size is set by default to the +same size of the input file. If @option{size} is set, it must contain +the size specified in the input file, and the initial grid defined in +that file is centered in the larger resulting area. + +If a filename is not specified, the size value defaults to "320x240" +(used for a randomly generated initial grid). + +@item stitch +If set to 1, stitch the left and right grid edges together, and the +top and bottom edges also. Defaults to 1. + +@item mold +Set cell mold speed. If set, a dead cell will go from @option{death_color} to +@option{mold_color} with a step of @option{mold}. @option{mold} can have a +value from 0 to 255. + +@item life_color +Set the color of living (or new born) cells. + +@item death_color +Set the color of dead cells. If @option{mold} is set, this is the first color +used to represent a dead cell. + +@item mold_color +Set mold color, for definitely dead and moldy cells. +@end table + +@subsection Examples + +@itemize +@item +Read a grid from @file{pattern}, and center it on a grid of size +300x300 pixels: +@example +life=f=pattern:s=300x300 +@end example + +@item +Generate a random grid of size 200x200, with a fill ratio of 2/3: +@example +life=ratio=2/3:s=200x200 +@end example + +@item +Specify a custom rule for evolving a randomly generated grid: +@example +life=rule=S14/B34 +@end example + +@item +Full example with slow death effect (mold) using @command{ffplay}: +@example +ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 +@end example +@end itemize + +@section color, nullsrc, rgbtestsrc, smptebars, testsrc + +The @code{color} source provides an uniformly colored input. + +The @code{nullsrc} source returns unprocessed video frames. It is +mainly useful to be employed in analysis / debugging tools, or as the +source for filters which ignore the input data. + +The @code{rgbtestsrc} source generates an RGB test pattern useful for +detecting RGB vs BGR issues. You should see a red, green and blue +stripe from top to bottom. + +The @code{smptebars} source generates a color bars pattern, based on +the SMPTE Engineering Guideline EG 1-1990. + +The @code{testsrc} source generates a test video pattern, showing a +color pattern, a scrolling gradient and a timestamp. This is mainly +intended for testing purposes. + +These sources accept an optional sequence of @var{key}=@var{value} pairs, +separated by ":". The description of the accepted options follows. + +@table @option + +@item color, c +Specify the color of the source, only used in the @code{color} +source. It can be the name of a color (case insensitive match) or a +0xRRGGBB[AA] sequence, possibly followed by an alpha specifier. The +default value is "black". + +@item size, s +Specify the size of the sourced video, it may be a string of the form +@var{width}x@var{height}, or the name of a size abbreviation. The +default value is "320x240". + +@item rate, r +Specify the frame rate of the sourced video, as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float +number or a valid video frame rate abbreviation. The default value is +"25". + +@item sar +Set the sample aspect ratio of the sourced video. + +@item duration, d +Set the video duration of the sourced video. The accepted syntax is: +@example +[-]HH[:MM[:SS[.m...]]] +[-]S+[.m...] +@end example +See also the function @code{av_parse_time()}. + +If not specified, or the expressed duration is negative, the video is +supposed to be generated forever. + +@item decimals, n +Set the number of decimals to show in the timestamp, only used in the +@code{testsrc} source. + +The displayed timestamp value will correspond to the original +timestamp value multiplied by the power of 10 of the specified +value. Default value is 0. +@end table + +For example the following: +@example +testsrc=duration=5.3:size=qcif:rate=10 +@end example + +will generate a video with a duration of 5.3 seconds, with size +176x144 and a frame rate of 10 frames per second. + +The following graph description will generate a red source +with an opacity of 0.2, with size "qcif" and a frame rate of 10 +frames per second. +@example +color=c=red@@0.2:s=qcif:r=10 +@end example + +If the input content is to be ignored, @code{nullsrc} can be used. The +following command generates noise in the luminance plane by employing +the @code{geq} filter: +@example +nullsrc=s=256x256, geq=random(1)*255:128:128 +@end example + +@c man end VIDEO SOURCES + +@chapter Video Sinks +@c man begin VIDEO SINKS + +Below is a description of the currently available video sinks. + +@section buffersink + +Buffer video frames, and make them available to the end of the filter +graph. + +This sink is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/buffersink.h}. + +It does not require a string parameter in input, but you need to +specify a pointer to a list of supported pixel formats terminated by +-1 in the opaque parameter provided to @code{avfilter_init_filter} +when initializing this sink. + +@section nullsink + +Null video sink, do absolutely nothing with the input video. It is +mainly useful as a template and to be employed in analysis / debugging +tools. + +@c man end VIDEO SINKS + +@chapter Multimedia Filters +@c man begin MULTIMEDIA FILTERS + +Below is a description of the currently available multimedia filters. + +@section aperms, perms + +Set read/write permissions for the output frames. + +These filters are mainly aimed at developers to test direct path in the +following filter in the filtergraph. + +The filters accept parameters as a list of @var{key}=@var{value} pairs, +separated by ":". If the key of the first options is omitted, the argument is +assumed to be the @var{mode}. + +A description of the accepted parameters follows. + +@table @option +@item mode +Select the permissions mode. + +It accepts the following values: +@table @samp +@item none +Do nothing. This is the default. +@item ro +Set all the output frames read-only. +@item rw +Set all the output frames directly writable. +@item toggle +Make the frame read-only if writable, and writable if read-only. +@item random +Set each output frame read-only or writable randomly. +@end table +@end table + +Note: in case of auto-inserted filter between the permission filter and the +following one, the permission might not be received as expected in that +following filter. Inserting a @ref{format} or @ref{aformat} filter before the +perms/aperms filter can avoid this problem. + +@section aselect, select +Select frames to pass in output. + +These filters accept a single option @option{expr} or @option{e} +specifying the select expression, which can be specified either by +specyfing @code{expr=VALUE} or specifying the expression +alone. + +The select expression is evaluated for each input frame. If the +evaluation result is a non-zero value, the frame is selected and +passed to the output, otherwise it is discarded. + +The expression can contain the following constants: + +@table @option +@item n +the sequential number of the filtered frame, starting from 0 + +@item selected_n +the sequential number of the selected frame, starting from 0 + +@item prev_selected_n +the sequential number of the last selected frame, NAN if undefined + +@item TB +timebase of the input timestamps + +@item pts +the PTS (Presentation TimeStamp) of the filtered video frame, +expressed in @var{TB} units, NAN if undefined + +@item t +the PTS (Presentation TimeStamp) of the filtered video frame, +expressed in seconds, NAN if undefined + +@item prev_pts +the PTS of the previously filtered video frame, NAN if undefined + +@item prev_selected_pts +the PTS of the last previously filtered video frame, NAN if undefined + +@item prev_selected_t +the PTS of the last previously selected video frame, NAN if undefined + +@item start_pts +the PTS of the first video frame in the video, NAN if undefined + +@item start_t +the time of the first video frame in the video, NAN if undefined + +@item pict_type @emph{(video only)} +the type of the filtered frame, can assume one of the following +values: +@table @option +@item I +@item P +@item B +@item S +@item SI +@item SP +@item BI +@end table + +@item interlace_type @emph{(video only)} +the frame interlace type, can assume one of the following values: +@table @option +@item PROGRESSIVE +the frame is progressive (not interlaced) +@item TOPFIRST +the frame is top-field-first +@item BOTTOMFIRST +the frame is bottom-field-first +@end table + +@item consumed_sample_n @emph{(audio only)} +the number of selected samples before the current frame + +@item samples_n @emph{(audio only)} +the number of samples in the current frame + +@item sample_rate @emph{(audio only)} +the input sample rate + +@item key +1 if the filtered frame is a key-frame, 0 otherwise + +@item pos +the position in the file of the filtered frame, -1 if the information +is not available (e.g. for synthetic video) + +@item scene @emph{(video only)} +value between 0 and 1 to indicate a new scene; a low value reflects a low +probability for the current frame to introduce a new scene, while a higher +value means the current frame is more likely to be one (see the example below) + +@end table + +The default value of the select expression is "1". + +@subsection Examples + +@itemize +@item +Select all frames in input: +@example +select +@end example + +The example above is the same as: +@example +select=1 +@end example + +@item +Skip all frames: +@example +select=0 +@end example + +@item +Select only I-frames: +@example +select='eq(pict_type\,I)' +@end example + +@item +Select one frame every 100: +@example +select='not(mod(n\,100))' +@end example + +@item +Select only frames contained in the 10-20 time interval: +@example +select='gte(t\,10)*lte(t\,20)' +@end example + +@item +Select only I frames contained in the 10-20 time interval: +@example +select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)' +@end example + +@item +Select frames with a minimum distance of 10 seconds: +@example +select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' +@end example + +@item +Use aselect to select only audio frames with samples number > 100: +@example +aselect='gt(samples_n\,100)' +@end example + +@item +Create a mosaic of the first scenes: +@example +ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png +@end example + +Comparing @var{scene} against a value between 0.3 and 0.5 is generally a sane +choice. +@end itemize + +@section asendcmd, sendcmd + +Send commands to filters in the filtergraph. + +These filters read commands to be sent to other filters in the +filtergraph. + +@code{asendcmd} must be inserted between two audio filters, +@code{sendcmd} must be inserted between two video filters, but apart +from that they act the same way. + +The specification of commands can be provided in the filter arguments +with the @var{commands} option, or in a file specified by the +@var{filename} option. + +These filters accept the following options: +@table @option +@item commands, c +Set the commands to be read and sent to the other filters. +@item filename, f +Set the filename of the commands to be read and sent to the other +filters. +@end table + +@subsection Commands syntax + +A commands description consists of a sequence of interval +specifications, comprising a list of commands to be executed when a +particular event related to that interval occurs. The occurring event +is typically the current frame time entering or leaving a given time +interval. + +An interval is specified by the following syntax: +@example +@var{START}[-@var{END}] @var{COMMANDS}; +@end example + +The time interval is specified by the @var{START} and @var{END} times. +@var{END} is optional and defaults to the maximum time. + +The current frame time is considered within the specified interval if +it is included in the interval [@var{START}, @var{END}), that is when +the time is greater or equal to @var{START} and is lesser than +@var{END}. + +@var{COMMANDS} consists of a sequence of one or more command +specifications, separated by ",", relating to that interval. The +syntax of a command specification is given by: +@example +[@var{FLAGS}] @var{TARGET} @var{COMMAND} @var{ARG} +@end example + +@var{FLAGS} is optional and specifies the type of events relating to +the time interval which enable sending the specified command, and must +be a non-null sequence of identifier flags separated by "+" or "|" and +enclosed between "[" and "]". + +The following flags are recognized: +@table @option +@item enter +The command is sent when the current frame timestamp enters the +specified interval. In other words, the command is sent when the +previous frame timestamp was not in the given interval, and the +current is. + +@item leave +The command is sent when the current frame timestamp leaves the +specified interval. In other words, the command is sent when the +previous frame timestamp was in the given interval, and the +current is not. +@end table + +If @var{FLAGS} is not specified, a default value of @code{[enter]} is +assumed. + +@var{TARGET} specifies the target of the command, usually the name of +the filter class or a specific filter instance name. + +@var{COMMAND} specifies the name of the command for the target filter. + +@var{ARG} is optional and specifies the optional list of argument for +the given @var{COMMAND}. + +Between one interval specification and another, whitespaces, or +sequences of characters starting with @code{#} until the end of line, +are ignored and can be used to annotate comments. + +A simplified BNF description of the commands specification syntax +follows: +@example +@var{COMMAND_FLAG} ::= "enter" | "leave" +@var{COMMAND_FLAGS} ::= @var{COMMAND_FLAG} [(+|"|")@var{COMMAND_FLAG}] +@var{COMMAND} ::= ["[" @var{COMMAND_FLAGS} "]"] @var{TARGET} @var{COMMAND} [@var{ARG}] +@var{COMMANDS} ::= @var{COMMAND} [,@var{COMMANDS}] +@var{INTERVAL} ::= @var{START}[-@var{END}] @var{COMMANDS} +@var{INTERVALS} ::= @var{INTERVAL}[;@var{INTERVALS}] +@end example + +@subsection Examples + +@itemize +@item +Specify audio tempo change at second 4: +@example +asendcmd=c='4.0 atempo tempo 1.5',atempo +@end example + +@item +Specify a list of drawtext and hue commands in a file. +@example +# show text in the interval 5-10 +5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world', + [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text='; + +# desaturate the image in the interval 15-20 +15.0-20.0 [enter] hue reinit s=0, + [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor', + [leave] hue reinit s=1, + [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color'; + +# apply an exponential saturation fade-out effect, starting from time 25 +25 [enter] hue s=exp(t-25) +@end example + +A filtergraph allowing to read and process the above command list +stored in a file @file{test.cmd}, can be specified with: +@example +sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue +@end example +@end itemize + +@anchor{setpts} +@section asetpts, setpts + +Change the PTS (presentation timestamp) of the input frames. + +@code{asetpts} works on audio frames, @code{setpts} on video frames. + +Accept in input an expression evaluated through the eval API, which +can contain the following constants: + +@table @option +@item FRAME_RATE +frame rate, only defined for constant frame-rate video + +@item PTS +the presentation timestamp in input + +@item N +the count of the input frame, starting from 0. + +@item NB_CONSUMED_SAMPLES +the number of consumed samples, not including the current frame (only +audio) + +@item NB_SAMPLES +the number of samples in the current frame (only audio) + +@item SAMPLE_RATE +audio sample rate + +@item STARTPTS +the PTS of the first frame + +@item STARTT +the time in seconds of the first frame + +@item INTERLACED +tell if the current frame is interlaced + +@item T +the time in seconds of the current frame + +@item TB +the time base + +@item POS +original position in the file of the frame, or undefined if undefined +for the current frame + +@item PREV_INPTS +previous input PTS + +@item PREV_INT +previous input time in seconds + +@item PREV_OUTPTS +previous output PTS + +@item PREV_OUTT +previous output time in seconds + +@item RTCTIME +wallclock (RTC) time in microseconds. This is deprecated, use time(0) +instead. + +@item RTCSTART +wallclock (RTC) time at the start of the movie in microseconds +@end table + +@subsection Examples + +@itemize +@item +Start counting PTS from zero +@example +setpts=PTS-STARTPTS +@end example + +@item +Apply fast motion effect: +@example +setpts=0.5*PTS +@end example + +@item +Apply slow motion effect: +@example +setpts=2.0*PTS +@end example + +@item +Set fixed rate of 25 frames per second: +@example +setpts=N/(25*TB) +@end example + +@item +Set fixed rate 25 fps with some jitter: +@example +setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' +@end example + +@item +Apply an offset of 10 seconds to the input PTS: +@example +setpts=PTS+10/TB +@end example + +@item +Generate timestamps from a "live source" and rebase onto the current timebase: +@example +setpts='(RTCTIME - RTCSTART) / (TB * 1000000)' +@end example +@end itemize + +@section ebur128 + +EBU R128 scanner filter. This filter takes an audio stream as input and outputs +it unchanged. By default, it logs a message at a frequency of 10Hz with the +Momentary loudness (identified by @code{M}), Short-term loudness (@code{S}), +Integrated loudness (@code{I}) and Loudness Range (@code{LRA}). + +The filter also has a video output (see the @var{video} option) with a real +time graph to observe the loudness evolution. The graphic contains the logged +message mentioned above, so it is not printed anymore when this option is set, +unless the verbose logging is set. The main graphing area contains the +short-term loudness (3 seconds of analysis), and the gauge on the right is for +the momentary loudness (400 milliseconds). + +More information about the Loudness Recommendation EBU R128 on +@url{http://tech.ebu.ch/loudness}. + +The filter accepts the following named parameters: + +@table @option + +@item video +Activate the video output. The audio stream is passed unchanged whether this +option is set or no. The video stream will be the first output stream if +activated. Default is @code{0}. + +@item size +Set the video size. This option is for video only. Default and minimum +resolution is @code{640x480}. + +@item meter +Set the EBU scale meter. Default is @code{9}. Common values are @code{9} and +@code{18}, respectively for EBU scale meter +9 and EBU scale meter +18. Any +other integer value between this range is allowed. + +@item metadata +Set metadata injection. If set to @code{1}, the audio input will be segmented +into 100ms output frames, each of them containing various loudness information +in metadata. All the metadata keys are prefixed with @code{lavfi.r128.}. + +Default is @code{0}. + +@item framelog +Force the frame logging level. + +Available values are: +@table @samp +@item info +information logging level +@item verbose +verbose logging level +@end table + +By default, the logging level is set to @var{info}. If the @option{video} or +the @option{metadata} options are set, it switches to @var{verbose}. +@end table + +@subsection Examples + +@itemize +@item +Real-time graph using @command{ffplay}, with a EBU scale meter +18: +@example +ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" +@end example + +@item +Run an analysis with @command{ffmpeg}: +@example +ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null - +@end example +@end itemize + +@section settb, asettb + +Set the timebase to use for the output frames timestamps. +It is mainly useful for testing timebase configuration. + +This filter accepts a single option @option{tb}, which can be +specified either by setting @option{tb}=@var{VALUE} or setting the +value alone. + +The value for @option{tb} is an arithmetic expression representing a +rational. The expression can contain the constants "AVTB" (the default +timebase), "intb" (the input timebase) and "sr" (the sample rate, +audio only). Default value is "intb". + +@subsection Examples + +@itemize +@item +Set the timebase to 1/25: +@example +settb=1/25 +@end example + +@item +Set the timebase to 1/10: +@example +settb=0.1 +@end example + +@item +Set the timebase to 1001/1000: +@example +settb=1+0.001 +@end example + +@item +Set the timebase to 2*intb: +@example +settb=2*intb +@end example + +@item +Set the default timebase value: +@example +settb=AVTB +@end example +@end itemize + +@section concat + +Concatenate audio and video streams, joining them together one after the +other. + +The filter works on segments of synchronized video and audio streams. All +segments must have the same number of streams of each type, and that will +also be the number of streams at output. + +The filter accepts the following named parameters: +@table @option + +@item n +Set the number of segments. Default is 2. + +@item v +Set the number of output video streams, that is also the number of video +streams in each segment. Default is 1. + +@item a +Set the number of output audio streams, that is also the number of video +streams in each segment. Default is 0. + +@item unsafe +Activate unsafe mode: do not fail if segments have a different format. + +@end table + +The filter has @var{v}+@var{a} outputs: first @var{v} video outputs, then +@var{a} audio outputs. + +There are @var{n}x(@var{v}+@var{a}) inputs: first the inputs for the first +segment, in the same order as the outputs, then the inputs for the second +segment, etc. + +Related streams do not always have exactly the same duration, for various +reasons including codec frame size or sloppy authoring. For that reason, +related synchronized streams (e.g. a video and its audio track) should be +concatenated at once. The concat filter will use the duration of the longest +stream in each segment (except the last one), and if necessary pad shorter +audio streams with silence. + +For this filter to work correctly, all segments must start at timestamp 0. + +All corresponding streams must have the same parameters in all segments; the +filtering system will automatically select a common pixel format for video +streams, and a common sample format, sample rate and channel layout for +audio streams, but other settings, such as resolution, must be converted +explicitly by the user. + +Different frame rates are acceptable but will result in variable frame rate +at output; be sure to configure the output file to handle it. + +@subsection Examples + +@itemize +@item +Concatenate an opening, an episode and an ending, all in bilingual version +(video in stream 0, audio in streams 1 and 2): +@example +ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \ + '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] + concat=n=3:v=1:a=2 [v] [a1] [a2]' \ + -map '[v]' -map '[a1]' -map '[a2]' output.mkv +@end example + +@item +Concatenate two parts, handling audio and video separately, using the +(a)movie sources, and adjusting the resolution: +@example +movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ; +movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ; +[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa] +@end example +Note that a desync will happen at the stitch if the audio and video streams +do not have exactly the same duration in the first file. + +@end itemize + +@section showspectrum + +Convert input audio to a video output, representing the audio frequency +spectrum. + +The filter accepts the following named parameters: +@table @option +@item size, s +Specify the video size for the output. Default value is @code{640x512}. + +@item slide +Specify if the spectrum should slide along the window. Default value is +@code{0}. + +@item mode +Specify display mode. + +It accepts the following values: +@table @samp +@item combined +all channels are displayed in the same row +@item separate +all channels are displayed in separate rows +@end table + +Default value is @samp{combined}. + +@item color +Specify display color mode. + +It accepts the following values: +@table @samp +@item channel +each channel is displayed in a separate color +@item intensity +each channel is is displayed using the same color scheme +@end table + +Default value is @samp{channel}. + +@item scale +Specify scale used for calculating intensity color values. + +It accepts the following values: +@table @samp +@item lin +linear +@item sqrt +square root, default +@item cbrt +cubic root +@item log +logarithmic +@end table + +Default value is @samp{sqrt}. + +@item saturation +Set saturation modifier for displayed colors. Negative values provide +alternative color scheme. @code{0} is no saturation at all. +Saturation must be in [-10.0, 10.0] range. +Default value is @code{1}. +@end table + +The usage is very similar to the showwaves filter; see the examples in that +section. + +@subsection Examples + +@itemize +@item +Large window with logarithmic color scaling: +@example +showspectrum=s=1280x480:scale=log +@end example + +@item +Complete example for a colored and sliding spectrum per channel using @command{ffplay}: +@example +ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1]; + [a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]' +@end example +@end itemize + +@section showwaves + +Convert input audio to a video output, representing the samples waves. + +The filter accepts the following named parameters: +@table @option +@item mode +Set display mode. + +Available values are: +@table @samp +@item point +Draw a point for each sample. + +@item line +Draw a vertical line for each sample. +@end table + +Default value is @code{point}. + +@item n +Set the number of samples which are printed on the same column. A +larger value will decrease the frame rate. Must be a positive +integer. This option can be set only if the value for @var{rate} +is not explicitly specified. + +@item rate, r +Set the (approximate) output frame rate. This is done by setting the +option @var{n}. Default value is "25". + +@item size, s +Specify the video size for the output. Default value is "600x240". +@end table + +@subsection Examples + +@itemize +@item +Output the input file audio and the corresponding video representation +at the same time: +@example +amovie=a.mp3,asplit[out0],showwaves[out1] +@end example + +@item +Create a synthetic signal and show it with showwaves, forcing a +framerate of 30 frames per second: +@example +aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1] +@end example +@end itemize + +@c man end MULTIMEDIA FILTERS + +@chapter Multimedia Sources +@c man begin MULTIMEDIA SOURCES + +Below is a description of the currently available multimedia sources. + +@section amovie + +This is the same as @ref{movie} source, except it selects an audio +stream by default. + +@anchor{movie} +@section movie + +Read audio and/or video stream(s) from a movie container. + +It accepts the syntax: @var{movie_name}[:@var{options}] where +@var{movie_name} is the name of the resource to read (not necessarily +a file but also a device or a stream accessed through some protocol), +and @var{options} is an optional sequence of @var{key}=@var{value} +pairs, separated by ":". + +The description of the accepted options follows. + +@table @option + +@item format_name, f +Specifies the format assumed for the movie to read, and can be either +the name of a container or an input device. If not specified the +format is guessed from @var{movie_name} or by probing. + +@item seek_point, sp +Specifies the seek point in seconds, the frames will be output +starting from this seek point, the parameter is evaluated with +@code{av_strtod} so the numerical value may be suffixed by an IS +postfix. Default value is "0". + +@item streams, s +Specifies the streams to read. Several streams can be specified, +separated by "+". The source will then have as many outputs, in the +same order. The syntax is explained in the ``Stream specifiers'' +section in the ffmpeg manual. Two special names, "dv" and "da" specify +respectively the default (best suited) video and audio stream. Default +is "dv", or "da" if the filter is called as "amovie". + +@item stream_index, si +Specifies the index of the video stream to read. If the value is -1, +the best suited video stream will be automatically selected. Default +value is "-1". Deprecated. If the filter is called "amovie", it will select +audio instead of video. + +@item loop +Specifies how many times to read the stream in sequence. +If the value is less than 1, the stream will be read again and again. +Default value is "1". + +Note that when the movie is looped the source timestamps are not +changed, so it will generate non monotonically increasing timestamps. +@end table + +This filter allows to overlay a second video on top of main input of +a filtergraph as shown in this graph: +@example +input -----------> deltapts0 --> overlay --> output + ^ + | +movie --> scale--> deltapts1 -------+ +@end example + +@subsection Examples + +@itemize +@item +Skip 3.2 seconds from the start of the avi file in.avi, and overlay it +on top of the input labelled as "in": +@example +movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; +[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] +@end example + +@item +Read from a video4linux2 device, and overlay it on top of the input +labelled as "in": +@example +movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; +[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] +@end example + +@item +Read the first video stream and the audio stream with id 0x81 from +dvd.vob; the video is connected to the pad named "video" and the audio is +connected to the pad named "audio": +@example +movie=dvd.vob:s=v:0+#0x81 [video] [audio] +@end example +@end itemize + +@c man end MULTIMEDIA SOURCES diff --git a/ffmpeg1/doc/general.texi b/ffmpeg1/doc/general.texi new file mode 100644 index 0000000..39b9360 --- /dev/null +++ b/ffmpeg1/doc/general.texi @@ -0,0 +1,1016 @@ +\input texinfo @c -*- texinfo -*- + +@settitle General Documentation +@titlepage +@center @titlefont{General Documentation} +@end titlepage + +@top + +@contents + +@chapter External libraries + +FFmpeg can be hooked up with a number of external libraries to add support +for more formats. None of them are used by default, their use has to be +explicitly requested by passing the appropriate flags to +@command{./configure}. + +@section OpenJPEG + +FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to +@url{http://www.openjpeg.org/} to get the libraries and follow the installation +instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to +@file{./configure}. + + +@section OpenCORE and VisualOn libraries + +Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer +libraries provide encoders for a number of audio codecs. + +@float NOTE +OpenCORE and VisualOn libraries are under the Apache License 2.0 +(see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is +incompatible with the LGPL version 2.1 and GPL version 2. You have to +upgrade FFmpeg's license to LGPL version 3 (or if you have enabled +GPL components, GPL version 3) to use it. +@end float + +@subsection OpenCORE AMR + +FFmpeg can make use of the OpenCORE libraries for AMR-NB +decoding/encoding and AMR-WB decoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the +instructions for installing the libraries. +Then pass @code{--enable-libopencore-amrnb} and/or +@code{--enable-libopencore-amrwb} to configure to enable them. + +@subsection VisualOn AAC encoder library + +FFmpeg can make use of the VisualOn AACenc library for AAC encoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the +instructions for installing the library. +Then pass @code{--enable-libvo-aacenc} to configure to enable it. + +@subsection VisualOn AMR-WB encoder library + +FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the +instructions for installing the library. +Then pass @code{--enable-libvo-amrwbenc} to configure to enable it. + +@subsection Fraunhofer AAC library + +FFmpeg can make use of the Fraunhofer AAC library for AAC encoding. + +Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the +instructions for installing the library. +Then pass @code{--enable-libfdk-aac} to configure to enable it. + +@section LAME + +FFmpeg can make use of the LAME library for MP3 encoding. + +Go to @url{http://lame.sourceforge.net/} and follow the +instructions for installing the library. +Then pass @code{--enable-libmp3lame} to configure to enable it. + +@section TwoLAME + +FFmpeg can make use of the TwoLAME library for MP2 encoding. + +Go to @url{http://www.twolame.org/} and follow the +instructions for installing the library. +Then pass @code{--enable-libtwolame} to configure to enable it. + +@section libvpx + +FFmpeg can make use of the libvpx library for VP8 encoding. + +Go to @url{http://www.webmproject.org/} and follow the instructions for +installing the library. Then pass @code{--enable-libvpx} to configure to +enable it. + +@section x264 + +FFmpeg can make use of the x264 library for H.264 encoding. + +Go to @url{http://www.videolan.org/developers/x264.html} and follow the +instructions for installing the library. Then pass @code{--enable-libx264} to +configure to enable it. + +@float NOTE +x264 is under the GNU Public License Version 2 or later +(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for +details), you must upgrade FFmpeg's license to GPL in order to use it. +@end float + +@section libilbc + +iLBC is a narrowband speech codec that has been made freely available +by Google as part of the WebRTC project. libilbc is a packaging friendly +copy of the iLBC codec. FFmpeg can make use of the libilbc library for +iLBC encoding and decoding. + +Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for +installing the library. Then pass @code{--enable-libilbc} to configure to +enable it. + + + +@chapter Supported File Formats, Codecs or Features + +You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list. + +@section File Formats + +FFmpeg supports the following file formats through the @code{libavformat} +library: + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item 4xm @tab @tab X + @tab 4X Technologies format, used in some games. +@item 8088flex TMV @tab @tab X +@item ACT Voice @tab @tab X + @tab contains G.729 audio +@item Adobe Filmstrip @tab X @tab X +@item Audio IFF (AIFF) @tab X @tab X +@item American Laser Games MM @tab @tab X + @tab Multimedia format used in games like Mad Dog McCree. +@item 3GPP AMR @tab X @tab X +@item Amazing Studio Packed Animation File @tab @tab X + @tab Multimedia format used in game Heart Of Darkness. +@item Apple HTTP Live Streaming @tab @tab X +@item Artworx Data Format @tab @tab X +@item AFC @tab @tab X + @tab Audio format used on the Nintendo Gamecube. +@item ASF @tab X @tab X +@item AST @tab X @tab X + @tab Audio format used on the Nintendo Wii. +@item AVI @tab X @tab X +@item AVISynth @tab @tab X +@item AVR @tab @tab X + @tab Audio format used on Mac. +@item AVS @tab @tab X + @tab Multimedia format used by the Creature Shock game. +@item Beam Software SIFF @tab @tab X + @tab Audio and video format used in some games by Beam Software. +@item Bethesda Softworks VID @tab @tab X + @tab Used in some games from Bethesda Softworks. +@item Binary text @tab @tab X +@item Bink @tab @tab X + @tab Multimedia format used by many games. +@item Bitmap Brothers JV @tab @tab X + @tab Used in Z and Z95 games. +@item Brute Force & Ignorance @tab @tab X + @tab Used in the game Flash Traffic: City of Angels. +@item BRSTM @tab @tab X + @tab Audio format used on the Nintendo Wii. +@item BWF @tab X @tab X +@item CRI ADX @tab X @tab X + @tab Audio-only format used in console video games. +@item Discworld II BMV @tab @tab X +@item Interplay C93 @tab @tab X + @tab Used in the game Cyberia from Interplay. +@item Delphine Software International CIN @tab @tab X + @tab Multimedia format used by Delphine Software games. +@item CD+G @tab @tab X + @tab Video format used by CD+G karaoke disks +@item Commodore CDXL @tab @tab X + @tab Amiga CD video format +@item Core Audio Format @tab X @tab X + @tab Apple Core Audio Format +@item CRC testing format @tab X @tab +@item Creative Voice @tab X @tab X + @tab Created for the Sound Blaster Pro. +@item CRYO APC @tab @tab X + @tab Audio format used in some games by CRYO Interactive Entertainment. +@item D-Cinema audio @tab X @tab X +@item Deluxe Paint Animation @tab @tab X +@item DFA @tab @tab X + @tab This format is used in Chronomaster game +@item DV video @tab X @tab X +@item DXA @tab @tab X + @tab This format is used in the non-Windows version of the Feeble Files + game and different game cutscenes repacked for use with ScummVM. +@item Electronic Arts cdata @tab @tab X +@item Electronic Arts Multimedia @tab @tab X + @tab Used in various EA games; files have extensions like WVE and UV2. +@item Ensoniq Paris Audio File @tab @tab X +@item FFM (FFserver live feed) @tab X @tab X +@item Flash (SWF) @tab X @tab X +@item Flash 9 (AVM2) @tab X @tab X + @tab Only embedded audio is decoded. +@item FLI/FLC/FLX animation @tab @tab X + @tab .fli/.flc files +@item Flash Video (FLV) @tab X @tab X + @tab Macromedia Flash video files +@item framecrc testing format @tab X @tab +@item FunCom ISS @tab @tab X + @tab Audio format used in various games from FunCom like The Longest Journey. +@item G.723.1 @tab X @tab X +@item G.729 BIT @tab X @tab X +@item G.729 raw @tab @tab X +@item GIF Animation @tab X @tab X +@item GXF @tab X @tab X + @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley + playout servers. +@item iCEDraw File @tab @tab X +@item ICO @tab X @tab X + @tab Microsoft Windows ICO +@item id Quake II CIN video @tab @tab X +@item id RoQ @tab X @tab X + @tab Used in Quake III, Jedi Knight 2 and other computer games. +@item IEC61937 encapsulation @tab X @tab X +@item IFF @tab @tab X + @tab Interchange File Format +@item iLBC @tab X @tab X +@item Interplay MVE @tab @tab X + @tab Format used in various Interplay computer games. +@item IV8 @tab @tab X + @tab A format generated by IndigoVision 8000 video server. +@item IVF (On2) @tab X @tab X + @tab A format used by libvpx +@item IRCAM @tab X @tab X +@item LATM @tab X @tab X +@item LMLM4 @tab @tab X + @tab Used by Linux Media Labs MPEG-4 PCI boards +@item LOAS @tab @tab X + @tab contains LATM multiplexed AAC audio +@item LVF @tab @tab X +@item LXF @tab @tab X + @tab VR native stream format, used by Leitch/Harris' video servers. +@item Matroska @tab X @tab X +@item Matroska audio @tab X @tab +@item FFmpeg metadata @tab X @tab X + @tab Metadata in text format. +@item MAXIS XA @tab @tab X + @tab Used in Sim City 3000; file extension .xa. +@item MD Studio @tab @tab X +@item Metal Gear Solid: The Twin Snakes @tab @tab X +@item Megalux Frame @tab @tab X + @tab Used by Megalux Ultimate Paint +@item Mobotix .mxg @tab @tab X +@item Monkey's Audio @tab @tab X +@item Motion Pixels MVI @tab @tab X +@item MOV/QuickTime/MP4 @tab X @tab X + @tab 3GP, 3GP2, PSP, iPod variants supported +@item MP2 @tab X @tab X +@item MP3 @tab X @tab X +@item MPEG-1 System @tab X @tab X + @tab muxed audio and video, VCD format supported +@item MPEG-PS (program stream) @tab X @tab X + @tab also known as @code{VOB} file, SVCD and DVD format supported +@item MPEG-TS (transport stream) @tab X @tab X + @tab also known as DVB Transport Stream +@item MPEG-4 @tab X @tab X + @tab MPEG-4 is a variant of QuickTime. +@item MIME multipart JPEG @tab X @tab +@item MSN TCP webcam @tab @tab X + @tab Used by MSN Messenger webcam streams. +@item MTV @tab @tab X +@item Musepack @tab @tab X +@item Musepack SV8 @tab @tab X +@item Material eXchange Format (MXF) @tab X @tab X + @tab SMPTE 377M, used by D-Cinema, broadcast industry. +@item Material eXchange Format (MXF), D-10 Mapping @tab X @tab X + @tab SMPTE 386M, D-10/IMX Mapping. +@item NC camera feed @tab @tab X + @tab NC (AVIP NC4600) camera streams +@item NIST SPeech HEader REsources @tab @tab X +@item NTT TwinVQ (VQF) @tab @tab X + @tab Nippon Telegraph and Telephone Corporation TwinVQ. +@item Nullsoft Streaming Video @tab @tab X +@item NuppelVideo @tab @tab X +@item NUT @tab X @tab X + @tab NUT Open Container Format +@item Ogg @tab X @tab X +@item Playstation Portable PMP @tab @tab X +@item Portable Voice Format @tab @tab X +@item TechnoTrend PVA @tab @tab X + @tab Used by TechnoTrend DVB PCI boards. +@item QCP @tab @tab X +@item raw ADTS (AAC) @tab X @tab X +@item raw AC-3 @tab X @tab X +@item raw Chinese AVS video @tab X @tab X +@item raw CRI ADX @tab X @tab X +@item raw Dirac @tab X @tab X +@item raw DNxHD @tab X @tab X +@item raw DTS @tab X @tab X +@item raw DTS-HD @tab @tab X +@item raw E-AC-3 @tab X @tab X +@item raw FLAC @tab X @tab X +@item raw GSM @tab @tab X +@item raw H.261 @tab X @tab X +@item raw H.263 @tab X @tab X +@item raw H.264 @tab X @tab X +@item raw Ingenient MJPEG @tab @tab X +@item raw MJPEG @tab X @tab X +@item raw MLP @tab @tab X +@item raw MPEG @tab @tab X +@item raw MPEG-1 @tab @tab X +@item raw MPEG-2 @tab @tab X +@item raw MPEG-4 @tab X @tab X +@item raw NULL @tab X @tab +@item raw video @tab X @tab X +@item raw id RoQ @tab X @tab +@item raw Shorten @tab @tab X +@item raw TAK @tab @tab X +@item raw TrueHD @tab X @tab X +@item raw VC-1 @tab @tab X +@item raw PCM A-law @tab X @tab X +@item raw PCM mu-law @tab X @tab X +@item raw PCM signed 8 bit @tab X @tab X +@item raw PCM signed 16 bit big-endian @tab X @tab X +@item raw PCM signed 16 bit little-endian @tab X @tab X +@item raw PCM signed 24 bit big-endian @tab X @tab X +@item raw PCM signed 24 bit little-endian @tab X @tab X +@item raw PCM signed 32 bit big-endian @tab X @tab X +@item raw PCM signed 32 bit little-endian @tab X @tab X +@item raw PCM unsigned 8 bit @tab X @tab X +@item raw PCM unsigned 16 bit big-endian @tab X @tab X +@item raw PCM unsigned 16 bit little-endian @tab X @tab X +@item raw PCM unsigned 24 bit big-endian @tab X @tab X +@item raw PCM unsigned 24 bit little-endian @tab X @tab X +@item raw PCM unsigned 32 bit big-endian @tab X @tab X +@item raw PCM unsigned 32 bit little-endian @tab X @tab X +@item raw PCM floating-point 32 bit big-endian @tab X @tab X +@item raw PCM floating-point 32 bit little-endian @tab X @tab X +@item raw PCM floating-point 64 bit big-endian @tab X @tab X +@item raw PCM floating-point 64 bit little-endian @tab X @tab X +@item RDT @tab @tab X +@item REDCODE R3D @tab @tab X + @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio. +@item RealMedia @tab X @tab X +@item Redirector @tab @tab X +@item Renderware TeXture Dictionary @tab @tab X +@item RL2 @tab @tab X + @tab Audio and video format used in some games by Entertainment Software Partners. +@item RPL/ARMovie @tab @tab X +@item Lego Mindstorms RSO @tab X @tab X +@item RTMP @tab X @tab X + @tab Output is performed by publishing stream to RTMP server +@item RTP @tab X @tab X +@item RTSP @tab X @tab X +@item SAP @tab X @tab X +@item SBG @tab @tab X +@item SDP @tab @tab X +@item Sega FILM/CPK @tab @tab X + @tab Used in many Sega Saturn console games. +@item Silicon Graphics Movie @tab @tab X +@item Sierra SOL @tab @tab X + @tab .sol files used in Sierra Online games. +@item Sierra VMD @tab @tab X + @tab Used in Sierra CD-ROM games. +@item Smacker @tab @tab X + @tab Multimedia format used by many games. +@item SMJPEG @tab X @tab X + @tab Used in certain Loki game ports. +@item Smush @tab @tab X + @tab Multimedia format used in some LucasArts games. +@item Sony OpenMG (OMA) @tab X @tab X + @tab Audio format used in Sony Sonic Stage and Sony Vegas. +@item Sony PlayStation STR @tab @tab X +@item Sony Wave64 (W64) @tab X @tab X +@item SoX native format @tab X @tab X +@item SUN AU format @tab X @tab X +@item Text files @tab @tab X +@item THP @tab @tab X + @tab Used on the Nintendo GameCube. +@item Tiertex Limited SEQ @tab @tab X + @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback. +@item True Audio @tab @tab X +@item VC-1 test bitstream @tab X @tab X +@item Vivo @tab @tab X +@item WAV @tab X @tab X +@item WavPack @tab X @tab X +@item WebM @tab X @tab X +@item Windows Televison (WTV) @tab X @tab X +@item Wing Commander III movie @tab @tab X + @tab Multimedia format used in Origin's Wing Commander III computer game. +@item Westwood Studios audio @tab @tab X + @tab Multimedia format used in Westwood Studios games. +@item Westwood Studios VQA @tab @tab X + @tab Multimedia format used in Westwood Studios games. +@item XMV @tab @tab X + @tab Microsoft video container used in Xbox games. +@item xWMA @tab @tab X + @tab Microsoft audio container used by XAudio 2. +@item eXtended BINary text (XBIN) @tab @tab X +@item YUV4MPEG pipe @tab X @tab X +@item Psygnosis YOP @tab @tab X +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@section Image Formats + +FFmpeg can read and write images for each frame of a video sequence. The +following image formats are supported: + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item .Y.U.V @tab X @tab X + @tab one raw file per component +@item animated GIF @tab X @tab X + @tab Only uncompressed GIFs are generated. +@item BMP @tab X @tab X + @tab Microsoft BMP image +@item PIX @tab @tab X + @tab PIX is an image format used in the Argonaut BRender engine. +@item DPX @tab X @tab X + @tab Digital Picture Exchange +@item EXR @tab @tab X + @tab OpenEXR +@item JPEG @tab X @tab X + @tab Progressive JPEG is not supported. +@item JPEG 2000 @tab X @tab X +@item JPEG-LS @tab X @tab X +@item LJPEG @tab X @tab + @tab Lossless JPEG +@item PAM @tab X @tab X + @tab PAM is a PNM extension with alpha support. +@item PBM @tab X @tab X + @tab Portable BitMap image +@item PCX @tab X @tab X + @tab PC Paintbrush +@item PGM @tab X @tab X + @tab Portable GrayMap image +@item PGMYUV @tab X @tab X + @tab PGM with U and V components in YUV 4:2:0 +@item PIC @tab @tab X + @tab Pictor/PC Paint +@item PNG @tab X @tab X +@item PPM @tab X @tab X + @tab Portable PixelMap image +@item PTX @tab @tab X + @tab V.Flash PTX format +@item SGI @tab X @tab X + @tab SGI RGB image format +@item Sun Rasterfile @tab X @tab X + @tab Sun RAS image format +@item TIFF @tab X @tab X + @tab YUV, JPEG and some extension is not supported yet. +@item Truevision Targa @tab X @tab X + @tab Targa (.TGA) image format +@item XBM @tab X @tab X + @tab X BitMap image format +@item XFace @tab X @tab X + @tab X-Face image format +@item XWD @tab X @tab X + @tab X Window Dump image format +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@section Video Codecs + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item 4X Movie @tab @tab X + @tab Used in certain computer games. +@item 8088flex TMV @tab @tab X +@item A64 multicolor @tab X @tab + @tab Creates video suitable to be played on a commodore 64 (multicolor mode). +@item Amazing Studio PAF Video @tab @tab X +@item American Laser Games MM @tab @tab X + @tab Used in games like Mad Dog McCree. +@item AMV Video @tab X @tab X + @tab Used in Chinese MP3 players. +@item ANSI/ASCII art @tab @tab X +@item Apple MJPEG-B @tab @tab X +@item Apple ProRes @tab X @tab X +@item Apple QuickDraw @tab @tab X + @tab fourcc: qdrw +@item Asus v1 @tab X @tab X + @tab fourcc: ASV1 +@item Asus v2 @tab X @tab X + @tab fourcc: ASV2 +@item ATI VCR1 @tab @tab X + @tab fourcc: VCR1 +@item ATI VCR2 @tab @tab X + @tab fourcc: VCR2 +@item Auravision Aura @tab @tab X +@item Auravision Aura 2 @tab @tab X +@item Autodesk Animator Flic video @tab @tab X +@item Autodesk RLE @tab @tab X + @tab fourcc: AASC +@item Avid 1:1 10-bit RGB Packer @tab X @tab X + @tab fourcc: AVrp +@item AVS (Audio Video Standard) video @tab @tab X + @tab Video encoding used by the Creature Shock game. +@item AYUV @tab X @tab X + @tab Microsoft uncompressed packed 4:4:4:4 +@item Beam Software VB @tab @tab X +@item Bethesda VID video @tab @tab X + @tab Used in some games from Bethesda Softworks. +@item Bink Video @tab @tab X +@item Bitmap Brothers JV video @tab @tab X +@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X +@item Brute Force & Ignorance @tab @tab X + @tab Used in the game Flash Traffic: City of Angels. +@item C93 video @tab @tab X + @tab Codec used in Cyberia game. +@item CamStudio @tab @tab X + @tab fourcc: CSCD +@item CD+G @tab @tab X + @tab Video codec for CD+G karaoke disks +@item CDXL @tab @tab X + @tab Amiga CD video codec +@item Chinese AVS video @tab E @tab X + @tab AVS1-P2, JiZhun profile, encoding through external library libxavs +@item Delphine Software International CIN video @tab @tab X + @tab Codec used in Delphine Software International games. +@item Discworld II BMV Video @tab @tab X +@item Canopus Lossless Codec @tab @tab X +@item Cinepak @tab @tab X +@item Cirrus Logic AccuPak @tab X @tab X + @tab fourcc: CLJR +@item CPiA Video Format @tab @tab X +@item Creative YUV (CYUV) @tab @tab X +@item DFA @tab @tab X + @tab Codec used in Chronomaster game. +@item Dirac @tab E @tab X + @tab supported through external library libschroedinger +@item Deluxe Paint Animation @tab @tab X +@item DNxHD @tab X @tab X + @tab aka SMPTE VC3 +@item Duck TrueMotion 1.0 @tab @tab X + @tab fourcc: DUCK +@item Duck TrueMotion 2.0 @tab @tab X + @tab fourcc: TM20 +@item DV (Digital Video) @tab X @tab X +@item Dxtory capture format @tab @tab X +@item Feeble Files/ScummVM DXA @tab @tab X + @tab Codec originally used in Feeble Files game. +@item Electronic Arts CMV video @tab @tab X + @tab Used in NHL 95 game. +@item Electronic Arts Madcow video @tab @tab X +@item Electronic Arts TGV video @tab @tab X +@item Electronic Arts TGQ video @tab @tab X +@item Electronic Arts TQI video @tab @tab X +@item Escape 124 @tab @tab X +@item Escape 130 @tab @tab X +@item FFmpeg video codec #1 @tab X @tab X + @tab lossless codec (fourcc: FFV1) +@item Flash Screen Video v1 @tab X @tab X + @tab fourcc: FSV1 +@item Flash Screen Video v2 @tab X @tab X +@item Flash Video (FLV) @tab X @tab X + @tab Sorenson H.263 used in Flash +@item Forward Uncompressed @tab @tab X +@item Fraps @tab @tab X +@item H.261 @tab X @tab X +@item H.263 / H.263-1996 @tab X @tab X +@item H.263+ / H.263-1998 / H.263 version 2 @tab X @tab X +@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 @tab E @tab X + @tab encoding supported through external library libx264 +@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration) @tab E @tab X +@item HuffYUV @tab X @tab X +@item HuffYUV FFmpeg variant @tab X @tab X +@item IBM Ultimotion @tab @tab X + @tab fourcc: ULTI +@item id Cinematic video @tab @tab X + @tab Used in Quake II. +@item id RoQ video @tab X @tab X + @tab Used in Quake III, Jedi Knight 2, other computer games. +@item IFF ILBM @tab @tab X + @tab IFF interleaved bitmap +@item IFF ByteRun1 @tab @tab X + @tab IFF run length encoded bitmap +@item Intel H.263 @tab @tab X +@item Intel Indeo 2 @tab @tab X +@item Intel Indeo 3 @tab @tab X +@item Intel Indeo 4 @tab @tab X +@item Intel Indeo 5 @tab @tab X +@item Interplay C93 @tab @tab X + @tab Used in the game Cyberia from Interplay. +@item Interplay MVE video @tab @tab X + @tab Used in Interplay .MVE files. +@item J2K @tab X @tab X +@item Karl Morton's video codec @tab @tab X + @tab Codec used in Worms games. +@item Kega Game Video (KGV1) @tab @tab X + @tab Kega emulator screen capture codec. +@item Lagarith @tab @tab X +@item LCL (LossLess Codec Library) MSZH @tab @tab X +@item LCL (LossLess Codec Library) ZLIB @tab E @tab E +@item LOCO @tab @tab X +@item LucasArts Smush @tab @tab X + @tab Used in LucasArts games. +@item lossless MJPEG @tab X @tab X +@item Microsoft ATC Screen @tab @tab X + @tab Also known as Microsoft Screen 3. +@item Microsoft Expression Encoder Screen @tab @tab X + @tab Also known as Microsoft Titanium Screen 2. +@item Microsoft RLE @tab @tab X +@item Microsoft Screen 1 @tab @tab X + @tab Also known as Windows Media Video V7 Screen. +@item Microsoft Screen 2 @tab @tab X + @tab Also known as Windows Media Video V9 Screen. +@item Microsoft Video 1 @tab @tab X +@item Mimic @tab @tab X + @tab Used in MSN Messenger Webcam streams. +@item Miro VideoXL @tab @tab X + @tab fourcc: VIXL +@item MJPEG (Motion JPEG) @tab X @tab X +@item Mobotix MxPEG video @tab @tab X +@item Motion Pixels video @tab @tab X +@item MPEG-1 video @tab X @tab X +@item MPEG-1/2 video XvMC (X-Video Motion Compensation) @tab @tab X +@item MPEG-1/2 video (VDPAU acceleration) @tab @tab X +@item MPEG-2 video @tab X @tab X +@item MPEG-4 part 2 @tab X @tab X + @tab libxvidcore can be used alternatively for encoding. +@item MPEG-4 part 2 Microsoft variant version 1 @tab @tab X +@item MPEG-4 part 2 Microsoft variant version 2 @tab X @tab X +@item MPEG-4 part 2 Microsoft variant version 3 @tab X @tab X +@item Nintendo Gamecube THP video @tab @tab X +@item NuppelVideo/RTjpeg @tab @tab X + @tab Video encoding used in NuppelVideo files. +@item On2 VP3 @tab @tab X + @tab still experimental +@item On2 VP5 @tab @tab X + @tab fourcc: VP50 +@item On2 VP6 @tab @tab X + @tab fourcc: VP60,VP61,VP62 +@item VP8 @tab E @tab X + @tab fourcc: VP80, encoding supported through external library libvpx +@item Pinnacle TARGA CineWave YUV16 @tab @tab X + @tab fourcc: Y216 +@item Prores @tab @tab X + @tab fourcc: apch,apcn,apcs,apco +@item Q-team QPEG @tab @tab X + @tab fourccs: QPEG, Q1.0, Q1.1 +@item QuickTime 8BPS video @tab @tab X +@item QuickTime Animation (RLE) video @tab X @tab X + @tab fourcc: 'rle ' +@item QuickTime Graphics (SMC) @tab @tab X + @tab fourcc: 'smc ' +@item QuickTime video (RPZA) @tab @tab X + @tab fourcc: rpza +@item R10K AJA Kona 10-bit RGB Codec @tab X @tab X +@item R210 Quicktime Uncompressed RGB 10-bit @tab X @tab X +@item Raw Video @tab X @tab X +@item RealVideo 1.0 @tab X @tab X +@item RealVideo 2.0 @tab X @tab X +@item RealVideo 3.0 @tab @tab X + @tab still far from ideal +@item RealVideo 4.0 @tab @tab X +@item Renderware TXD (TeXture Dictionary) @tab @tab X + @tab Texture dictionaries used by the Renderware Engine. +@item RL2 video @tab @tab X + @tab used in some games by Entertainment Software Partners +@item SGI RLE 8-bit @tab @tab X +@item Sierra VMD video @tab @tab X + @tab Used in Sierra VMD files. +@item Silicon Graphics Motion Video Compressor 1 (MVC1) @tab @tab X +@item Silicon Graphics Motion Video Compressor 2 (MVC2) @tab @tab X +@item Smacker video @tab @tab X + @tab Video encoding used in Smacker. +@item SMPTE VC-1 @tab @tab X +@item Snow @tab X @tab X + @tab experimental wavelet codec (fourcc: SNOW) +@item Sony PlayStation MDEC (Motion DECoder) @tab @tab X +@item Sorenson Vector Quantizer 1 @tab X @tab X + @tab fourcc: SVQ1 +@item Sorenson Vector Quantizer 3 @tab @tab X + @tab fourcc: SVQ3 +@item Sunplus JPEG (SP5X) @tab @tab X + @tab fourcc: SP5X +@item TechSmith Screen Capture Codec @tab @tab X + @tab fourcc: TSCC +@item TechSmith Screen Capture Codec 2 @tab @tab X + @tab fourcc: TSC2 +@item Theora @tab E @tab X + @tab encoding supported through external library libtheora +@item Tiertex Limited SEQ video @tab @tab X + @tab Codec used in DOS CD-ROM FlashBack game. +@item Ut Video @tab X @tab X +@item v210 QuickTime uncompressed 4:2:2 10-bit @tab X @tab X +@item v308 QuickTime uncompressed 4:4:4 @tab X @tab X +@item v408 QuickTime uncompressed 4:4:4:4 @tab X @tab X +@item v410 QuickTime uncompressed 4:4:4 10-bit @tab X @tab X +@item VBLE Lossless Codec @tab @tab X +@item VMware Screen Codec / VMware Video @tab @tab X + @tab Codec used in videos captured by VMware. +@item Westwood Studios VQA (Vector Quantized Animation) video @tab @tab X +@item Windows Media Image @tab @tab X +@item Windows Media Video 7 @tab X @tab X +@item Windows Media Video 8 @tab X @tab X +@item Windows Media Video 9 @tab @tab X + @tab not completely working +@item Wing Commander III / Xan @tab @tab X + @tab Used in Wing Commander III .MVE files. +@item Wing Commander IV / Xan @tab @tab X + @tab Used in Wing Commander IV. +@item Winnov WNV1 @tab @tab X +@item WMV7 @tab X @tab X +@item YAMAHA SMAF @tab X @tab X +@item Psygnosis YOP Video @tab @tab X +@item yuv4 @tab X @tab X + @tab libquicktime uncompressed packed 4:2:0 +@item ZeroCodec Lossless Video @tab @tab X +@item ZLIB @tab X @tab X + @tab part of LCL, encoder experimental +@item Zip Motion Blocks Video @tab X @tab X + @tab Encoder works only in PAL8. +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@section Audio Codecs + +@multitable @columnfractions .4 .1 .1 .4 +@item Name @tab Encoding @tab Decoding @tab Comments +@item 8SVX exponential @tab @tab X +@item 8SVX fibonacci @tab @tab X +@item AAC+ @tab E @tab X + @tab encoding supported through external library libaacplus +@item AAC @tab E @tab X + @tab encoding supported through external library libfaac and libvo-aacenc +@item AC-3 @tab IX @tab X +@item ADPCM 4X Movie @tab @tab X +@item ADPCM CDROM XA @tab @tab X +@item ADPCM Creative Technology @tab @tab X + @tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2 +@item ADPCM Electronic Arts @tab @tab X + @tab Used in various EA titles. +@item ADPCM Electronic Arts Maxis CDROM XS @tab @tab X + @tab Used in Sim City 3000. +@item ADPCM Electronic Arts R1 @tab @tab X +@item ADPCM Electronic Arts R2 @tab @tab X +@item ADPCM Electronic Arts R3 @tab @tab X +@item ADPCM Electronic Arts XAS @tab @tab X +@item ADPCM G.722 @tab X @tab X +@item ADPCM G.726 @tab X @tab X +@item ADPCM IMA AMV @tab @tab X + @tab Used in AMV files +@item ADPCM IMA Electronic Arts EACS @tab @tab X +@item ADPCM IMA Electronic Arts SEAD @tab @tab X +@item ADPCM IMA Funcom @tab @tab X +@item ADPCM IMA QuickTime @tab X @tab X +@item ADPCM IMA Loki SDL MJPEG @tab @tab X +@item ADPCM IMA WAV @tab X @tab X +@item ADPCM IMA Westwood @tab @tab X +@item ADPCM ISS IMA @tab @tab X + @tab Used in FunCom games. +@item ADPCM IMA Dialogic @tab @tab X +@item ADPCM IMA Duck DK3 @tab @tab X + @tab Used in some Sega Saturn console games. +@item ADPCM IMA Duck DK4 @tab @tab X + @tab Used in some Sega Saturn console games. +@item ADPCM Microsoft @tab X @tab X +@item ADPCM MS IMA @tab X @tab X +@item ADPCM Nintendo Gamecube AFC @tab @tab X +@item ADPCM Nintendo Gamecube THP @tab @tab X +@item ADPCM QT IMA @tab X @tab X +@item ADPCM SEGA CRI ADX @tab X @tab X + @tab Used in Sega Dreamcast games. +@item ADPCM Shockwave Flash @tab X @tab X +@item ADPCM Sound Blaster Pro 2-bit @tab @tab X +@item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X +@item ADPCM Sound Blaster Pro 4-bit @tab @tab X +@item ADPCM Westwood Studios IMA @tab @tab X + @tab Used in Westwood Studios games like Command and Conquer. +@item ADPCM Yamaha @tab X @tab X +@item AMR-NB @tab E @tab X + @tab encoding supported through external library libopencore-amrnb +@item AMR-WB @tab E @tab X + @tab encoding supported through external library libvo-amrwbenc +@item Amazing Studio PAF Audio @tab @tab X +@item Apple lossless audio @tab X @tab X + @tab QuickTime fourcc 'alac' +@item Atrac 1 @tab @tab X +@item Atrac 3 @tab @tab X +@item Bink Audio @tab @tab X + @tab Used in Bink and Smacker files in many games. +@item CELT @tab @tab E + @tab decoding supported through external library libcelt +@item Delphine Software International CIN audio @tab @tab X + @tab Codec used in Delphine Software International games. +@item Discworld II BMV Audio @tab @tab X +@item COOK @tab @tab X + @tab All versions except 5.1 are supported. +@item DCA (DTS Coherent Acoustics) @tab X @tab X +@item DPCM id RoQ @tab X @tab X + @tab Used in Quake III, Jedi Knight 2 and other computer games. +@item DPCM Interplay @tab @tab X + @tab Used in various Interplay computer games. +@item DPCM Sierra Online @tab @tab X + @tab Used in Sierra Online game audio files. +@item DPCM Sol @tab @tab X +@item DPCM Xan @tab @tab X + @tab Used in Origin's Wing Commander IV AVI files. +@item DSP Group TrueSpeech @tab @tab X +@item DV audio @tab @tab X +@item Enhanced AC-3 @tab X @tab X +@item EVRC (Enhanced Variable Rate Codec) @tab @tab X +@item FLAC (Free Lossless Audio Codec) @tab X @tab IX +@item G.723.1 @tab X @tab X +@item G.729 @tab @tab X +@item GSM @tab E @tab X + @tab encoding supported through external library libgsm +@item GSM Microsoft variant @tab E @tab X + @tab encoding supported through external library libgsm +@item IAC (Indeo Audio Coder) @tab @tab X +@item iLBC (Internet Low Bitrate Codec) @tab E @tab E + @tab encoding and decoding supported through external library libilbc +@item IMC (Intel Music Coder) @tab @tab X +@item MACE (Macintosh Audio Compression/Expansion) 3:1 @tab @tab X +@item MACE (Macintosh Audio Compression/Expansion) 6:1 @tab @tab X +@item MLP (Meridian Lossless Packing) @tab @tab X + @tab Used in DVD-Audio discs. +@item Monkey's Audio @tab @tab X + @tab Only versions 3.97-3.99 are supported. +@item MP1 (MPEG audio layer 1) @tab @tab IX +@item MP2 (MPEG audio layer 2) @tab IX @tab IX + @tab libtwolame can be used alternatively for encoding. +@item MP3 (MPEG audio layer 3) @tab E @tab IX + @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported +@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X +@item Musepack SV7 @tab @tab X +@item Musepack SV8 @tab @tab X +@item Nellymoser Asao @tab X @tab X +@item Opus @tab E @tab E + @tab supported through external library libopus +@item PCM A-law @tab X @tab X +@item PCM mu-law @tab X @tab X +@item PCM signed 8-bit planar @tab X @tab X +@item PCM signed 16-bit big-endian planar @tab X @tab X +@item PCM signed 16-bit little-endian planar @tab X @tab X +@item PCM signed 24-bit little-endian planar @tab X @tab X +@item PCM signed 32-bit little-endian planar @tab X @tab X +@item PCM 32-bit floating point big-endian @tab X @tab X +@item PCM 32-bit floating point little-endian @tab X @tab X +@item PCM 64-bit floating point big-endian @tab X @tab X +@item PCM 64-bit floating point little-endian @tab X @tab X +@item PCM D-Cinema audio signed 24-bit @tab X @tab X +@item PCM signed 8-bit @tab X @tab X +@item PCM signed 16-bit big-endian @tab X @tab X +@item PCM signed 16-bit little-endian @tab X @tab X +@item PCM signed 24-bit big-endian @tab X @tab X +@item PCM signed 24-bit little-endian @tab X @tab X +@item PCM signed 32-bit big-endian @tab X @tab X +@item PCM signed 32-bit little-endian @tab X @tab X +@item PCM signed 16/20/24-bit big-endian in MPEG-TS @tab @tab X +@item PCM unsigned 8-bit @tab X @tab X +@item PCM unsigned 16-bit big-endian @tab X @tab X +@item PCM unsigned 16-bit little-endian @tab X @tab X +@item PCM unsigned 24-bit big-endian @tab X @tab X +@item PCM unsigned 24-bit little-endian @tab X @tab X +@item PCM unsigned 32-bit big-endian @tab X @tab X +@item PCM unsigned 32-bit little-endian @tab X @tab X +@item PCM Zork @tab @tab X +@item QCELP / PureVoice @tab @tab X +@item QDesign Music Codec 2 @tab @tab X + @tab There are still some distortions. +@item RealAudio 1.0 (14.4K) @tab X @tab X + @tab Real 14400 bit/s codec +@item RealAudio 2.0 (28.8K) @tab @tab X + @tab Real 28800 bit/s codec +@item RealAudio 3.0 (dnet) @tab IX @tab X + @tab Real low bitrate AC-3 codec +@item RealAudio Lossless @tab @tab X +@item RealAudio SIPR / ACELP.NET @tab @tab X +@item Shorten @tab @tab X +@item Sierra VMD audio @tab @tab X + @tab Used in Sierra VMD files. +@item Smacker audio @tab @tab X +@item SMPTE 302M AES3 audio @tab @tab X +@item Sonic @tab X @tab X + @tab experimental codec +@item Sonic lossless @tab X @tab X + @tab experimental codec +@item Speex @tab E @tab E + @tab supported through external library libspeex +@item TAK (Tom's lossless Audio Kompressor) @tab @tab X +@item True Audio (TTA) @tab @tab X +@item TrueHD @tab @tab X + @tab Used in HD-DVD and Blu-Ray discs. +@item TwinVQ (VQF flavor) @tab @tab X +@item VIMA @tab @tab X + @tab Used in LucasArts SMUSH animations. +@item Vorbis @tab E @tab X + @tab A native but very primitive encoder exists. +@item WavPack @tab @tab X +@item Westwood Audio (SND1) @tab @tab X +@item Windows Media Audio 1 @tab X @tab X +@item Windows Media Audio 2 @tab X @tab X +@item Windows Media Audio Lossless @tab @tab X +@item Windows Media Audio Pro @tab @tab X +@item Windows Media Audio Voice @tab @tab X +@end multitable + +@code{X} means that encoding (resp. decoding) is supported. + +@code{E} means that support is provided through an external library. + +@code{I} means that an integer-only version is available, too (ensures high +performance on systems without hardware floating point support). + +@section Subtitle Formats + +@multitable @columnfractions .4 .1 .1 .1 .1 +@item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding +@item 3GPP Timed Text @tab @tab @tab X @tab X +@item AQTitle @tab @tab X @tab @tab X +@item DVB @tab X @tab X @tab X @tab X +@item DVD @tab X @tab X @tab X @tab X +@item JACOsub @tab X @tab X @tab @tab X +@item MicroDVD @tab X @tab X @tab @tab X +@item MPL2 @tab @tab X @tab @tab X +@item MPsub (MPlayer) @tab @tab X @tab @tab X +@item PGS @tab @tab @tab @tab X +@item PJS (Phoenix) @tab @tab X @tab @tab X +@item RealText @tab @tab X @tab @tab X +@item SAMI @tab @tab X @tab @tab X +@item SSA/ASS @tab X @tab X @tab X @tab X +@item SubRip (SRT) @tab X @tab X @tab X @tab X +@item SubViewer v1 @tab @tab X @tab @tab X +@item SubViewer @tab @tab X @tab @tab X +@item TED Talks captions @tab @tab X @tab @tab X +@item VobSub (IDX+SUB) @tab @tab X @tab @tab X +@item VPlayer @tab @tab X @tab @tab X +@item WebVTT @tab @tab X @tab @tab X +@item XSUB @tab @tab @tab X @tab X +@end multitable + +@code{X} means that the feature is supported. + +@section Network Protocols + +@multitable @columnfractions .4 .1 +@item Name @tab Support +@item file @tab X +@item Gopher @tab X +@item HLS @tab X +@item HTTP @tab X +@item HTTPS @tab X +@item MMSH @tab X +@item MMST @tab X +@item pipe @tab X +@item RTMP @tab X +@item RTMPE @tab X +@item RTMPS @tab X +@item RTMPT @tab X +@item RTMPTE @tab X +@item RTMPTS @tab X +@item RTP @tab X +@item SCTP @tab X +@item TCP @tab X +@item TLS @tab X +@item UDP @tab X +@end multitable + +@code{X} means that the protocol is supported. + +@code{E} means that support is provided through an external library. + + +@section Input/Output Devices + +@multitable @columnfractions .4 .1 .1 +@item Name @tab Input @tab Output +@item ALSA @tab X @tab X +@item BKTR @tab X @tab +@item caca @tab @tab X +@item DV1394 @tab X @tab +@item Lavfi virtual device @tab X @tab +@item Linux framebuffer @tab X @tab +@item JACK @tab X @tab +@item LIBCDIO @tab X +@item LIBDC1394 @tab X @tab +@item OpenAL @tab X +@item OSS @tab X @tab X +@item Pulseaudio @tab X @tab +@item SDL @tab @tab X +@item Video4Linux2 @tab X @tab +@item VfW capture @tab X @tab +@item X11 grabbing @tab X @tab +@end multitable + +@code{X} means that input/output is supported. + +@section Timecode + +@multitable @columnfractions .4 .1 .1 +@item Codec/format @tab Read @tab Write +@item AVI @tab X @tab X +@item DV @tab X @tab X +@item GXF @tab X @tab X +@item MOV @tab X @tab X +@item MPEG1/2 @tab X @tab X +@item MXF @tab X @tab X +@end multitable + +@bye diff --git a/ffmpeg1/doc/git-howto.texi b/ffmpeg1/doc/git-howto.texi new file mode 100644 index 0000000..44e1cc6 --- /dev/null +++ b/ffmpeg1/doc/git-howto.texi @@ -0,0 +1,415 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Using git to develop FFmpeg + +@titlepage +@center @titlefont{Using git to develop FFmpeg} +@end titlepage + +@top + +@contents + +@chapter Introduction + +This document aims in giving some quick references on a set of useful git +commands. You should always use the extensive and detailed documentation +provided directly by git: + +@example +git --help +man git +@end example + +shows you the available subcommands, + +@example +git <command> --help +man git-<command> +@end example + +shows information about the subcommand <command>. + +Additional information could be found on the +@url{http://gitref.org, Git Reference} website + +For more information about the Git project, visit the + +@url{http://git-scm.com/, Git website} + +Consult these resources whenever you have problems, they are quite exhaustive. + +What follows now is a basic introduction to Git and some FFmpeg-specific +guidelines to ease the contribution to the project + +@chapter Basics Usage + +@section Get GIT + +You can get git from @url{http://git-scm.com/} +Most distribution and operating system provide a package for it. + + +@section Cloning the source tree + +@example +git clone git://source.ffmpeg.org/ffmpeg <target> +@end example + +This will put the FFmpeg sources into the directory @var{<target>}. + +@example +git clone git@@source.ffmpeg.org:ffmpeg <target> +@end example + +This will put the FFmpeg sources into the directory @var{<target>} and let +you push back your changes to the remote repository. + +Make sure that you do not have Windows line endings in your checkouts, +otherwise you may experience spurious compilation failures. One way to +achieve this is to run + +@example +git config --global core.autocrlf false +@end example + + +@section Updating the source tree to the latest revision + +@example +git pull (--rebase) +@end example + +pulls in the latest changes from the tracked branch. The tracked branch +can be remote. By default the master branch tracks the branch master in +the remote origin. + +@float IMPORTANT +@command{--rebase} (see below) is recommended. +@end float + +@section Rebasing your local branches + +@example +git pull --rebase +@end example + +fetches the changes from the main repository and replays your local commits +over it. This is required to keep all your local changes at the top of +FFmpeg's master tree. The master tree will reject pushes with merge commits. + + +@section Adding/removing files/directories + +@example +git add [-A] <filename/dirname> +git rm [-r] <filename/dirname> +@end example + +GIT needs to get notified of all changes you make to your working +directory that makes files appear or disappear. +Line moves across files are automatically tracked. + + +@section Showing modifications + +@example +git diff <filename(s)> +@end example + +will show all local modifications in your working directory as unified diff. + + +@section Inspecting the changelog + +@example +git log <filename(s)> +@end example + +You may also use the graphical tools like gitview or gitk or the web +interface available at http://source.ffmpeg.org/ + +@section Checking source tree status + +@example +git status +@end example + +detects all the changes you made and lists what actions will be taken in case +of a commit (additions, modifications, deletions, etc.). + + +@section Committing + +@example +git diff --check +@end example + +to double check your changes before committing them to avoid trouble later +on. All experienced developers do this on each and every commit, no matter +how small. +Every one of them has been saved from looking like a fool by this many times. +It's very easy for stray debug output or cosmetic modifications to slip in, +please avoid problems through this extra level of scrutiny. + +For cosmetics-only commits you should get (almost) empty output from + +@example +git diff -w -b <filename(s)> +@end example + +Also check the output of + +@example +git status +@end example + +to make sure you don't have untracked files or deletions. + +@example +git add [-i|-p|-A] <filenames/dirnames> +@end example + +Make sure you have told git your name and email address + +@example +git config --global user.name "My Name" +git config --global user.email my@@email.invalid +@end example + +Use @var{--global} to set the global configuration for all your git checkouts. + +Git will select the changes to the files for commit. Optionally you can use +the interactive or the patch mode to select hunk by hunk what should be +added to the commit. + + +@example +git commit +@end example + +Git will commit the selected changes to your current local branch. + +You will be prompted for a log message in an editor, which is either +set in your personal configuration file through + +@example +git config --global core.editor +@end example + +or set by one of the following environment variables: +@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}. + +Log messages should be concise but descriptive. Explain why you made a change, +what you did will be obvious from the changes themselves most of the time. +Saying just "bug fix" or "10l" is bad. Remember that people of varying skill +levels look at and educate themselves while reading through your code. Don't +include filenames in log messages, Git provides that information. + +Possibly make the commit message have a terse, descriptive first line, an +empty line and then a full description. The first line will be used to name +the patch by git format-patch. + +@section Preparing a patchset + +@example +git format-patch <commit> [-o directory] +@end example + +will generate a set of patches for each commit between @var{<commit>} and +current @var{HEAD}. E.g. + +@example +git format-patch origin/master +@end example + +will generate patches for all commits on current branch which are not +present in upstream. +A useful shortcut is also + +@example +git format-patch -n +@end example + +which will generate patches from last @var{n} commits. +By default the patches are created in the current directory. + +@section Sending patches for review + +@example +git send-email <commit list|directory> +@end example + +will send the patches created by @command{git format-patch} or directly +generates them. All the email fields can be configured in the global/local +configuration or overridden by command line. +Note that this tool must often be installed separately (e.g. @var{git-email} +package on Debian-based distros). + + +@section Renaming/moving/copying files or contents of files + +Git automatically tracks such changes, making those normal commits. + +@example +mv/cp path/file otherpath/otherfile +git add [-A] . +git commit +@end example + + +@chapter Git configuration + +In order to simplify a few workflows, it is advisable to configure both +your personal Git installation and your local FFmpeg repository. + +@section Personal Git installation + +Add the following to your @file{~/.gitconfig} to help @command{git send-email} +and @command{git format-patch} detect renames: + +@example +[diff] + renames = copy +@end example + +@section Repository configuration + +In order to have @command{git send-email} automatically send patches +to the ffmpeg-devel mailing list, add the following stanza +to @file{/path/to/ffmpeg/repository/.git/config}: + +@example +[sendemail] + to = ffmpeg-devel@@ffmpeg.org +@end example + +@chapter FFmpeg specific + +@section Reverting broken commits + +@example +git reset <commit> +@end example + +@command{git reset} will uncommit the changes till @var{<commit>} rewriting +the current branch history. + +@example +git commit --amend +@end example + +allows to amend the last commit details quickly. + +@example +git rebase -i origin/master +@end example + +will replay local commits over the main repository allowing to edit, merge +or remove some of them in the process. + +@float NOTE +@command{git reset}, @command{git commit --amend} and @command{git rebase} +rewrite history, so you should use them ONLY on your local or topic branches. +The main repository will reject those changes. +@end float + +@example +git revert <commit> +@end example + +@command{git revert} will generate a revert commit. This will not make the +faulty commit disappear from the history. + +@section Pushing changes to remote trees + +@example +git push +@end example + +Will push the changes to the default remote (@var{origin}). +Git will prevent you from pushing changes if the local and remote trees are +out of sync. Refer to and to sync the local tree. + +@example +git remote add <name> <url> +@end example + +Will add additional remote with a name reference, it is useful if you want +to push your local branch for review on a remote host. + +@example +git push <remote> <refspec> +@end example + +Will push the changes to the @var{<remote>} repository. +Omitting @var{<refspec>} makes @command{git push} update all the remote +branches matching the local ones. + +@section Finding a specific svn revision + +Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits +based on a regular expression. see man gitrevisions + +@example +git show :/'as revision 23456' +@end example + +will show the svn changeset @var{r23456}. With older git versions searching in +the @command{git log} output is the easiest option (especially if a pager with +search capabilities is used). +This commit can be checked out with + +@example +git checkout -b svn_23456 :/'as revision 23456' +@end example + +or for git < 1.7.1 with + +@example +git checkout -b svn_23456 $SHA1 +@end example + +where @var{$SHA1} is the commit hash from the @command{git log} output. + + +@chapter pre-push checklist + +Once you have a set of commits that you feel are ready for pushing, +work through the following checklist to doublecheck everything is in +proper order. This list tries to be exhaustive. In case you are just +pushing a typo in a comment, some of the steps may be unnecessary. +Apply your common sense, but if in doubt, err on the side of caution. + +First, make sure that the commits and branches you are going to push +match what you want pushed and that nothing is missing, extraneous or +wrong. You can see what will be pushed by running the git push command +with --dry-run first. And then inspecting the commits listed with +@command{git log -p 1234567..987654}. The @command{git status} command +may help in finding local changes that have been forgotten to be added. + +Next let the code pass through a full run of our testsuite. + +@itemize +@item @command{make distclean} +@item @command{/path/to/ffmpeg/configure} +@item @command{make check} +@item if fate fails due to missing samples run @command{make fate-rsync} and retry +@end itemize + +Make sure all your changes have been checked before pushing them, the +testsuite only checks against regressions and that only to some extend. It does +obviously not check newly added features/code to be working unless you have +added a test for that (which is recommended). + +Also note that every single commit should pass the test suite, not just +the result of a series of patches. + +Once everything passed, push the changes to your public ffmpeg clone and post a +merge request to ffmpeg-devel. You can also push them directly but this is not +recommended. + +@chapter Server Issues + +Contact the project admins @email{root@@ffmpeg.org} if you have technical +problems with the GIT server. diff --git a/ffmpeg1/doc/git-howto.txt b/ffmpeg1/doc/git-howto.txt new file mode 100644 index 0000000..5ba72ee --- /dev/null +++ b/ffmpeg1/doc/git-howto.txt @@ -0,0 +1,273 @@ + +About Git write access: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +Before everything else, you should know how to use GIT properly. +Luckily Git comes with excellent documentation. + + git --help + man git + +shows you the available subcommands, + + git <command> --help + man git-<command> + +shows information about the subcommand <command>. + +The most comprehensive manual is the website Git Reference + +http://gitref.org/ + +For more information about the Git project, visit + +http://git-scm.com/ + +Consult these resources whenever you have problems, they are quite exhaustive. + +You do not need a special username or password. +All you need is to provide a ssh public key to the Git server admin. + +What follows now is a basic introduction to Git and some FFmpeg-specific +guidelines. Read it at least once, if you are granted commit privileges to the +FFmpeg project you are expected to be familiar with these rules. + + + +I. BASICS: +========== + +0. Get GIT: + + Most distributions have a git package, if not + You can get git from http://git-scm.com/ + + +1. Cloning the source tree: + + git clone git://source.ffmpeg.org/ffmpeg <target> + + This will put the FFmpeg sources into the directory <target>. + + git clone git@source.ffmpeg.org:ffmpeg <target> + + This will put the FFmpeg sources into the directory <target> and let + you push back your changes to the remote repository. + + +2. Updating the source tree to the latest revision: + + git pull (--ff-only) + + pulls in the latest changes from the tracked branch. The tracked branch + can be remote. By default the master branch tracks the branch master in + the remote origin. + Caveat: Since merge commits are forbidden at least for the initial + months of git --ff-only or --rebase (see below) are recommended. + --ff-only will fail and not create merge commits if your branch + has diverged (has a different history) from the tracked branch. + +2.a Rebasing your local branches: + + git pull --rebase + + fetches the changes from the main repository and replays your local commits + over it. This is required to keep all your local changes at the top of + FFmpeg's master tree. The master tree will reject pushes with merge commits. + + +3. Adding/removing files/directories: + + git add [-A] <filename/dirname> + git rm [-r] <filename/dirname> + + GIT needs to get notified of all changes you make to your working + directory that makes files appear or disappear. + Line moves across files are automatically tracked. + + +4. Showing modifications: + + git diff <filename(s)> + + will show all local modifications in your working directory as unified diff. + + +5. Inspecting the changelog: + + git log <filename(s)> + + You may also use the graphical tools like gitview or gitk or the web + interface available at http://source.ffmpeg.org + +6. Checking source tree status: + + git status + + detects all the changes you made and lists what actions will be taken in case + of a commit (additions, modifications, deletions, etc.). + + +7. Committing: + + git diff --check + + to double check your changes before committing them to avoid trouble later + on. All experienced developers do this on each and every commit, no matter + how small. + Every one of them has been saved from looking like a fool by this many times. + It's very easy for stray debug output or cosmetic modifications to slip in, + please avoid problems through this extra level of scrutiny. + + For cosmetics-only commits you should get (almost) empty output from + + git diff -w -b <filename(s)> + + Also check the output of + + git status + + to make sure you don't have untracked files or deletions. + + git add [-i|-p|-A] <filenames/dirnames> + + Make sure you have told git your name and email address, e.g. by running + git config --global user.name "My Name" + git config --global user.email my@email.invalid + (--global to set the global configuration for all your git checkouts). + + Git will select the changes to the files for commit. Optionally you can use + the interactive or the patch mode to select hunk by hunk what should be + added to the commit. + + git commit + + Git will commit the selected changes to your current local branch. + + You will be prompted for a log message in an editor, which is either + set in your personal configuration file through + + git config core.editor + + or set by one of the following environment variables: + GIT_EDITOR, VISUAL or EDITOR. + + Log messages should be concise but descriptive. Explain why you made a change, + what you did will be obvious from the changes themselves most of the time. + Saying just "bug fix" or "10l" is bad. Remember that people of varying skill + levels look at and educate themselves while reading through your code. Don't + include filenames in log messages, Git provides that information. + + Possibly make the commit message have a terse, descriptive first line, an + empty line and then a full description. The first line will be used to name + the patch by git format-patch. + + +8. Renaming/moving/copying files or contents of files: + + Git automatically tracks such changes, making those normal commits. + + mv/cp path/file otherpath/otherfile + + git add [-A] . + + git commit + + Do not move, rename or copy files of which you are not the maintainer without + discussing it on the mailing list first! + +9. Reverting broken commits + + git revert <commit> + + git revert will generate a revert commit. This will not make the faulty + commit disappear from the history. + + git reset <commit> + + git reset will uncommit the changes till <commit> rewriting the current + branch history. + + git commit --amend + + allows to amend the last commit details quickly. + + git rebase -i origin/master + + will replay local commits over the main repository allowing to edit, + merge or remove some of them in the process. + + Note that the reset, commit --amend and rebase rewrite history, so you + should use them ONLY on your local or topic branches. + + The main repository will reject those changes. + +10. Preparing a patchset. + + git format-patch <commit> [-o directory] + + will generate a set of patches for each commit between <commit> and + current HEAD. E.g. + + git format-patch origin/master + + will generate patches for all commits on current branch which are not + present in upstream. + A useful shortcut is also + + git format-patch -n + + which will generate patches from last n commits. + By default the patches are created in the current directory. + +11. Sending patches for review + + git send-email <commit list|directory> + + will send the patches created by git format-patch or directly generates + them. All the email fields can be configured in the global/local + configuration or overridden by command line. + Note that this tool must often be installed separately (e.g. git-email + package on Debian-based distros). + +12. Pushing changes to remote trees + + git push + + Will push the changes to the default remote (origin). + Git will prevent you from pushing changes if the local and remote trees are + out of sync. Refer to 2 and 2.a to sync the local tree. + + git remote add <name> <url> + + Will add additional remote with a name reference, it is useful if you want + to push your local branch for review on a remote host. + + git push <remote> <refspec> + + Will push the changes to the remote repository. Omitting refspec makes git + push update all the remote branches matching the local ones. + +13. Finding a specific svn revision + + Since version 1.7.1 git supports ':/foo' syntax for specifying commits + based on a regular expression. see man gitrevisions + + git show :/'as revision 23456' + + will show the svn changeset r23456. With older git versions searching in + the git log output is the easiest option (especially if a pager with + search capabilities is used). + This commit can be checked out with + + git checkout -b svn_23456 :/'as revision 23456' + + or for git < 1.7.1 with + + git checkout -b svn_23456 $SHA1 + + where $SHA1 is the commit SHA1 from the 'git log' output. + + +Contact the project admins <root at ffmpeg dot org> if you have technical +problems with the GIT server. diff --git a/ffmpeg1/doc/indevs.texi b/ffmpeg1/doc/indevs.texi new file mode 100644 index 0000000..cc5d666 --- /dev/null +++ b/ffmpeg1/doc/indevs.texi @@ -0,0 +1,797 @@ +@chapter Input Devices +@c man begin INPUT DEVICES + +Input devices are configured elements in FFmpeg which allow to access +the data coming from a multimedia device attached to your system. + +When you configure your FFmpeg build, all the supported input devices +are enabled by default. You can list all available ones using the +configure option "--list-indevs". + +You can disable all the input devices using the configure option +"--disable-indevs", and selectively enable an input device using the +option "--enable-indev=@var{INDEV}", or you can disable a particular +input device using the option "--disable-indev=@var{INDEV}". + +The option "-formats" of the ff* tools will display the list of +supported input devices (amongst the demuxers). + +A description of the currently available input devices follows. + +@section alsa + +ALSA (Advanced Linux Sound Architecture) input device. + +To enable this input device during configuration you need libasound +installed on your system. + +This device allows capturing from an ALSA device. The name of the +device to capture has to be an ALSA card identifier. + +An ALSA identifier has the syntax: +@example +hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]] +@end example + +where the @var{DEV} and @var{SUBDEV} components are optional. + +The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV}) +specify card number or identifier, device number and subdevice number +(-1 means any). + +To see the list of cards currently recognized by your system check the +files @file{/proc/asound/cards} and @file{/proc/asound/devices}. + +For example to capture with @command{ffmpeg} from an ALSA device with +card id 0, you may run the command: +@example +ffmpeg -f alsa -i hw:0 alsaout.wav +@end example + +For more information see: +@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html} + +@section bktr + +BSD video input device. + +@section dshow + +Windows DirectShow input device. + +DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. +Currently only audio and video devices are supported. + +Multiple devices may be opened as separate inputs, but they may also be +opened on the same input, which should improve synchronism between them. + +The input name should be in the format: + +@example +@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}] +@end example + +where @var{TYPE} can be either @var{audio} or @var{video}, +and @var{NAME} is the device's name. + +@subsection Options + +If no options are specified, the device's defaults are used. +If the device does not support the requested options, it will +fail to open. + +@table @option + +@item video_size +Set the video size in the captured video. + +@item framerate +Set the framerate in the captured video. + +@item sample_rate +Set the sample rate (in Hz) of the captured audio. + +@item sample_size +Set the sample size (in bits) of the captured audio. + +@item channels +Set the number of channels in the captured audio. + +@item list_devices +If set to @option{true}, print a list of devices and exit. + +@item list_options +If set to @option{true}, print a list of selected device's options +and exit. + +@item video_device_number +Set video device number for devices with same name (starts at 0, +defaults to 0). + +@item audio_device_number +Set audio device number for devices with same name (starts at 0, +defaults to 0). + +@item pixel_format +Select pixel format to be used by DirectShow. This may only be set when +the video codec is not set or set to rawvideo. + +@item audio_buffer_size +Set audio device buffer size in milliseconds (which can directly +impact latency, depending on the device). +Defaults to using the audio device's +default buffer size (typically some multiple of 500ms). +Setting this value too low can degrade performance. +See also +@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx} + +@end table + +@subsection Examples + +@itemize + +@item +Print the list of DirectShow supported devices and exit: +@example +$ ffmpeg -list_devices true -f dshow -i dummy +@end example + +@item +Open video device @var{Camera}: +@example +$ ffmpeg -f dshow -i video="Camera" +@end example + +@item +Open second video device with name @var{Camera}: +@example +$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" +@end example + +@item +Open video device @var{Camera} and audio device @var{Microphone}: +@example +$ ffmpeg -f dshow -i video="Camera":audio="Microphone" +@end example + +@item +Print the list of supported options in selected device and exit: +@example +$ ffmpeg -list_options true -f dshow -i video="Camera" +@end example + +@end itemize + +@section dv1394 + +Linux DV 1394 input device. + +@section fbdev + +Linux framebuffer input device. + +The Linux framebuffer is a graphic hardware-independent abstraction +layer to show graphics on a computer monitor, typically on the +console. It is accessed through a file device node, usually +@file{/dev/fb0}. + +For more detailed information read the file +Documentation/fb/framebuffer.txt included in the Linux source tree. + +To record from the framebuffer device @file{/dev/fb0} with +@command{ffmpeg}: +@example +ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi +@end example + +You can take a single screenshot image with the command: +@example +ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg +@end example + +See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). + +@section iec61883 + +FireWire DV/HDV input device using libiec61883. + +To enable this input device, you need libiec61883, libraw1394 and +libavc1394 installed on your system. Use the configure option +@code{--enable-libiec61883} to compile with the device enabled. + +The iec61883 capture device supports capturing from a video device +connected via IEEE1394 (FireWire), using libiec61883 and the new Linux +FireWire stack (juju). This is the default DV/HDV input method in Linux +Kernel 2.6.37 and later, since the old FireWire stack was removed. + +Specify the FireWire port to be used as input file, or "auto" +to choose the first port connected. + +@subsection Options + +@table @option + +@item dvtype +Override autodetection of DV/HDV. This should only be used if auto +detection does not work, or if usage of a different device type +should be prohibited. Treating a DV device as HDV (or vice versa) will +not work and result in undefined behavior. +The values @option{auto}, @option{dv} and @option{hdv} are supported. + +@item dvbuffer +Set maxiumum size of buffer for incoming data, in frames. For DV, this +is an exact value. For HDV, it is not frame exact, since HDV does +not have a fixed frame size. + +@item dvguid +Select the capture device by specifying it's GUID. Capturing will only +be performed from the specified device and fails if no device with the +given GUID is found. This is useful to select the input if multiple +devices are connected at the same time. +Look at /sys/bus/firewire/devices to find out the GUIDs. + +@end table + +@subsection Examples + +@itemize + +@item +Grab and show the input of a FireWire DV/HDV device. +@example +ffplay -f iec61883 -i auto +@end example + +@item +Grab and record the input of a FireWire DV/HDV device, +using a packet buffer of 100000 packets if the source is HDV. +@example +ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg +@end example + +@end itemize + +@section jack + +JACK input device. + +To enable this input device during configuration you need libjack +installed on your system. + +A JACK input device creates one or more JACK writable clients, one for +each audio channel, with name @var{client_name}:input_@var{N}, where +@var{client_name} is the name provided by the application, and @var{N} +is a number which identifies the channel. +Each writable client will send the acquired data to the FFmpeg input +device. + +Once you have created one or more JACK readable clients, you need to +connect them to one or more JACK writable clients. + +To connect or disconnect JACK clients you can use the @command{jack_connect} +and @command{jack_disconnect} programs, or do it through a graphical interface, +for example with @command{qjackctl}. + +To list the JACK clients and their properties you can invoke the command +@command{jack_lsp}. + +Follows an example which shows how to capture a JACK readable client +with @command{ffmpeg}. +@example +# Create a JACK writable client with name "ffmpeg". +$ ffmpeg -f jack -i ffmpeg -y out.wav + +# Start the sample jack_metro readable client. +$ jack_metro -b 120 -d 0.2 -f 4000 + +# List the current JACK clients. +$ jack_lsp -c +system:capture_1 +system:capture_2 +system:playback_1 +system:playback_2 +ffmpeg:input_1 +metro:120_bpm + +# Connect metro to the ffmpeg writable client. +$ jack_connect metro:120_bpm ffmpeg:input_1 +@end example + +For more information read: +@url{http://jackaudio.org/} + +@section lavfi + +Libavfilter input virtual device. + +This input device reads data from the open output pads of a libavfilter +filtergraph. + +For each filtergraph open output, the input device will create a +corresponding stream which is mapped to the generated output. Currently +only video data is supported. The filtergraph is specified through the +option @option{graph}. + +@subsection Options + +@table @option + +@item graph +Specify the filtergraph to use as input. Each video open output must be +labelled by a unique string of the form "out@var{N}", where @var{N} is a +number starting from 0 corresponding to the mapped input stream +generated by the device. +The first unlabelled output is automatically assigned to the "out0" +label, but all the others need to be specified explicitly. + +If not specified defaults to the filename specified for the input +device. + +@item graph_file +Set the filename of the filtergraph to be read and sent to the other +filters. Syntax of the filtergraph is the same as the one specified by +the option @var{graph}. + +@end table + +@subsection Examples + +@itemize +@item +Create a color video stream and play it back with @command{ffplay}: +@example +ffplay -f lavfi -graph "color=c=pink [out0]" dummy +@end example + +@item +As the previous example, but use filename for specifying the graph +description, and omit the "out0" label: +@example +ffplay -f lavfi color=c=pink +@end example + +@item +Create three different video test filtered sources and play them: +@example +ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 +@end example + +@item +Read an audio stream from a file using the amovie source and play it +back with @command{ffplay}: +@example +ffplay -f lavfi "amovie=test.wav" +@end example + +@item +Read an audio stream and a video stream and play it back with +@command{ffplay}: +@example +ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" +@end example + +@end itemize + +@section libdc1394 + +IIDC1394 input device, based on libdc1394 and libraw1394. + +@section openal + +The OpenAL input device provides audio capture on all systems with a +working OpenAL 1.1 implementation. + +To enable this input device during configuration, you need OpenAL +headers and libraries installed on your system, and need to configure +FFmpeg with @code{--enable-openal}. + +OpenAL headers and libraries should be provided as part of your OpenAL +implementation, or as an additional download (an SDK). Depending on your +installation you may need to specify additional flags via the +@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build +system to locate the OpenAL headers and libraries. + +An incomplete list of OpenAL implementations follows: + +@table @strong +@item Creative +The official Windows implementation, providing hardware acceleration +with supported devices and software fallback. +See @url{http://openal.org/}. +@item OpenAL Soft +Portable, open source (LGPL) software implementation. Includes +backends for the most common sound APIs on the Windows, Linux, +Solaris, and BSD operating systems. +See @url{http://kcat.strangesoft.net/openal.html}. +@item Apple +OpenAL is part of Core Audio, the official Mac OS X Audio interface. +See @url{http://developer.apple.com/technologies/mac/audio-and-video.html} +@end table + +This device allows to capture from an audio input device handled +through OpenAL. + +You need to specify the name of the device to capture in the provided +filename. If the empty string is provided, the device will +automatically select the default device. You can get the list of the +supported devices by using the option @var{list_devices}. + +@subsection Options + +@table @option + +@item channels +Set the number of channels in the captured audio. Only the values +@option{1} (monaural) and @option{2} (stereo) are currently supported. +Defaults to @option{2}. + +@item sample_size +Set the sample size (in bits) of the captured audio. Only the values +@option{8} and @option{16} are currently supported. Defaults to +@option{16}. + +@item sample_rate +Set the sample rate (in Hz) of the captured audio. +Defaults to @option{44.1k}. + +@item list_devices +If set to @option{true}, print a list of devices and exit. +Defaults to @option{false}. + +@end table + +@subsection Examples + +Print the list of OpenAL supported devices and exit: +@example +$ ffmpeg -list_devices true -f openal -i dummy out.ogg +@end example + +Capture from the OpenAL device @file{DR-BT101 via PulseAudio}: +@example +$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg +@end example + +Capture from the default device (note the empty string '' as filename): +@example +$ ffmpeg -f openal -i '' out.ogg +@end example + +Capture from two devices simultaneously, writing to two different files, +within the same @command{ffmpeg} command: +@example +$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg +@end example +Note: not all OpenAL implementations support multiple simultaneous capture - +try the latest OpenAL Soft if the above does not work. + +@section oss + +Open Sound System input device. + +The filename to provide to the input device is the device node +representing the OSS input device, and is usually set to +@file{/dev/dsp}. + +For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the +command: +@example +ffmpeg -f oss -i /dev/dsp /tmp/oss.wav +@end example + +For more information about OSS see: +@url{http://manuals.opensound.com/usersguide/dsp.html} + +@section pulse + +pulseaudio input device. + +To enable this input device during configuration you need libpulse-simple +installed in your system. + +The filename to provide to the input device is a source device or the +string "default" + +To list the pulse source devices and their properties you can invoke +the command @command{pactl list sources}. + +@example +ffmpeg -f pulse -i default /tmp/pulse.wav +@end example + +@subsection @var{server} AVOption + +The syntax is: +@example +-server @var{server name} +@end example + +Connects to a specific server. + +@subsection @var{name} AVOption + +The syntax is: +@example +-name @var{application name} +@end example + +Specify the application name pulse will use when showing active clients, +by default it is the LIBAVFORMAT_IDENT string + +@subsection @var{stream_name} AVOption + +The syntax is: +@example +-stream_name @var{stream name} +@end example + +Specify the stream name pulse will use when showing active streams, +by default it is "record" + +@subsection @var{sample_rate} AVOption + +The syntax is: +@example +-sample_rate @var{samplerate} +@end example + +Specify the samplerate in Hz, by default 48kHz is used. + +@subsection @var{channels} AVOption + +The syntax is: +@example +-channels @var{N} +@end example + +Specify the channels in use, by default 2 (stereo) is set. + +@subsection @var{frame_size} AVOption + +The syntax is: +@example +-frame_size @var{bytes} +@end example + +Specify the number of byte per frame, by default it is set to 1024. + +@subsection @var{fragment_size} AVOption + +The syntax is: +@example +-fragment_size @var{bytes} +@end example + +Specify the minimal buffering fragment in pulseaudio, it will affect the +audio latency. By default it is unset. + +@section sndio + +sndio input device. + +To enable this input device during configuration you need libsndio +installed on your system. + +The filename to provide to the input device is the device node +representing the sndio input device, and is usually set to +@file{/dev/audio0}. + +For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the +command: +@example +ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav +@end example + +@section video4linux2, v4l2 + +Video4Linux2 input video device. + +"v4l2" can be used as alias for "video4linux2". + +If FFmpeg is built with v4l-utils support (by using the +@code{--enable-libv4l2} configure option), the device will always rely +on libv4l2. + +The name of the device to grab is a file device node, usually Linux +systems tend to automatically create such nodes when the device +(e.g. an USB webcam) is plugged into the system, and has a name of the +kind @file{/dev/video@var{N}}, where @var{N} is a number associated to +the device. + +Video4Linux2 devices usually support a limited set of +@var{width}x@var{height} sizes and framerates. You can check which are +supported using @command{-list_formats all} for Video4Linux2 devices. +Some devices, like TV cards, support one or more standards. It is possible +to list all the supported standards using @command{-list_standards all}. + +The time base for the timestamps is 1 microsecond. Depending on the kernel +version and configuration, the timestamps may be derived from the real time +clock (origin at the Unix Epoch) or the monotonic clock (origin usually at +boot time, unaffected by NTP or manual changes to the clock). The +@option{-timestamps abs} or @option{-ts abs} option can be used to force +conversion into the real time clock. + +Some usage examples of the video4linux2 device with @command{ffmpeg} +and @command{ffplay}: +@itemize +@item +Grab and show the input of a video4linux2 device: +@example +ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 +@end example + +@item +Grab and record the input of a video4linux2 device, leave the +framerate and size as previously set: +@example +ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg +@end example +@end itemize + +For more information about Video4Linux, check @url{http://linuxtv.org/}. + +@subsection Options + +@table @option +@item standard +Set the standard. Must be the name of a supported standard. To get a +list of the supported standards, use the @option{list_standards} +option. + +@item channel +Set the input channel number. Default to 0. + +@item video_size +Set the video frame size. The argument must be a string in the form +@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation. + +@item pixel_format +Select the pixel format (only valid for raw video input). + +@item input_format +Set the preferred pixel format (for raw video) or a codec name. +This option allows to select the input format, when several are +available. + +@item framerate +Set the preferred video framerate. + +@item list_formats +List available formats (supported pixel formats, codecs, and frame +sizes) and exit. + +Available values are: +@table @samp +@item all +Show all available (compressed and non-compressed) formats. + +@item raw +Show only raw video (non-compressed) formats. + +@item compressed +Show only compressed formats. +@end table + +@item list_standards +List supported standards and exit. + +Available values are: +@table @samp +@item all +Show all supported standards. +@end table + +@item timestamps, ts +Set type of timestamps for grabbed frames. + +Available values are: +@table @samp +@item default +Use timestamps from the kernel. + +@item abs +Use absolute timestamps (wall clock). + +@item mono2abs +Force conversion from monotonic to absolute timestamps. +@end table + +Default value is @code{default}. +@end table + +@section vfwcap + +VfW (Video for Windows) capture input device. + +The filename passed as input is the capture driver number, ranging from +0 to 9. You may use "list" as filename to print a list of drivers. Any +other filename will be interpreted as device number 0. + +@section x11grab + +X11 video input device. + +This device allows to capture a region of an X11 display. + +The filename passed as input has the syntax: +@example +[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}] +@end example + +@var{hostname}:@var{display_number}.@var{screen_number} specifies the +X11 display name of the screen to grab from. @var{hostname} can be +omitted, and defaults to "localhost". The environment variable +@env{DISPLAY} contains the default display name. + +@var{x_offset} and @var{y_offset} specify the offsets of the grabbed +area with respect to the top-left border of the X11 screen. They +default to 0. + +Check the X11 documentation (e.g. man X) for more detailed information. + +Use the @command{dpyinfo} program for getting basic information about the +properties of your X11 display (e.g. grep for "name" or "dimensions"). + +For example to grab from @file{:0.0} using @command{ffmpeg}: +@example +ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg +@end example + +Grab at position @code{10,20}: +@example +ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg +@end example + +@subsection Options + +@table @option +@item draw_mouse +Specify whether to draw the mouse pointer. A value of @code{0} specify +not to draw the pointer. Default value is @code{1}. + +@item follow_mouse +Make the grabbed area follow the mouse. The argument can be +@code{centered} or a number of pixels @var{PIXELS}. + +When it is specified with "centered", the grabbing region follows the mouse +pointer and keeps the pointer at the center of region; otherwise, the region +follows only when the mouse pointer reaches within @var{PIXELS} (greater than +zero) to the edge of region. + +For example: +@example +ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg +@end example + +To follow only when the mouse pointer reaches within 100 pixels to edge: +@example +ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg +@end example + +@item framerate +Set the grabbing frame rate. Default value is @code{ntsc}, +corresponding to a framerate of @code{30000/1001}. + +@item show_region +Show grabbed region on screen. + +If @var{show_region} is specified with @code{1}, then the grabbing +region will be indicated on screen. With this option, it is easy to +know what is being grabbed if only a portion of the screen is grabbed. + +For example: +@example +ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg +@end example + +With @var{follow_mouse}: +@example +ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg +@end example + +@item video_size +Set the video frame size. Default value is @code{vga}. +@end table + +@c man end INPUT DEVICES diff --git a/ffmpeg1/doc/issue_tracker.txt b/ffmpeg1/doc/issue_tracker.txt new file mode 100644 index 0000000..d487f66 --- /dev/null +++ b/ffmpeg1/doc/issue_tracker.txt @@ -0,0 +1,213 @@ +FFmpeg's bug/patch/feature request tracker manual +================================================= + +NOTE: This is a draft. + +Overview: +--------- + +FFmpeg uses Trac for tracking issues, new issues and changes to +existing issues can be done through a web interface. + +Issues can be different kinds of things we want to keep track of +but that do not belong into the source tree itself. This includes +bug reports, patches, feature requests and license violations. We +might add more items to this list in the future, so feel free to +propose a new `type of issue' on the ffmpeg-devel mailing list if +you feel it is worth tracking. + +It is possible to subscribe to individual issues by adding yourself to the +Cc list or to subscribe to the ffmpeg-trac mailing list which receives +a mail for every change to every issue. +(the above does all work already after light testing) + +The subscription URL for the ffmpeg-trac list is: +http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac +The URL of the webinterface of the tracker is: +http(s)://ffmpeg.org/trac/ffmpeg + +Type: +----- +bug / defect + An error, flaw, mistake, failure, or fault in FFmpeg or libav* that + prevents it from behaving as intended. + +feature request / enhancement + Request of support for encoding or decoding of a new codec, container + or variant. + Request of support for more, less or plain different output or behavior + where the current implementation cannot be considered wrong. + +license violation + ticket to keep track of (L)GPL violations of ffmpeg by others + +patch + A patch as generated by diff which conforms to the patch submission and + development policy. + + +Priority: +--------- +critical + Bugs and patches which deal with data loss and security issues. + No feature request can be critical. + +important + Bugs which make FFmpeg unusable for a significant number of users, and + patches fixing them. + Examples here might be completely broken MPEG-4 decoding or a build issue + on Linux. + While broken 4xm decoding or a broken OS/2 build would not be important, + the separation to normal is somewhat fuzzy. + For feature requests this priority would be used for things many people + want. + Regressions also should be marked as important, regressions are bugs that + don't exist in a past revision or another branch. + +normal + + +minor + Bugs and patches about things like spelling errors, "mp2" instead of + "mp3" being shown and such. + Feature requests about things few people want or which do not make a big + difference. + +wish + Something that is desirable to have but that there is no urgency at + all to implement, e.g. something completely cosmetic like a website + restyle or a personalized doxy template or the FFmpeg logo. + This priority is not valid for bugs. + + +Status: +------- +new + initial state + +open + intermediate states + +closed + final state + + +Analyzed flag: +-------------- +Bugs which have been analyzed and where it is understood what causes them +and which exact chain of events triggers them. This analysis should be +available as a message in the bug report. +Note, do not change the status to analyzed without also providing a clear +and understandable analysis. +This state implicates that the bug either has been reproduced or that +reproduction is not needed as the bug is already understood. + + +Type/Status/Substatus: +---------- +*/new/new + Initial state of new bugs, patches and feature requests submitted by + users. + +*/open/open + Issues which have been briefly looked at and which did not look outright + invalid. + This implicates that no real more detailed state applies yet. Conversely, + the more detailed states below implicate that the issue has been briefly + looked at. + +*/closed/duplicate + Bugs, patches or feature requests which are duplicates. + Note that patches dealing with the same thing in a different way are not + duplicates. + Note, if you mark something as duplicate, do not forget setting the + superseder so bug reports are properly linked. + +*/closed/invalid + Bugs caused by user errors, random ineligible or otherwise nonsense stuff. + +*/closed/needs_more_info + Issues for which some information has been requested by the developers, + but which has not been provided by anyone within reasonable time. + + +bug/closed/fixed + Bugs which have to the best of our knowledge been fixed. + +bug/closed/wont_fix + Bugs which we will not fix. Possible reasons include legality, high + complexity for the sake of supporting obscure corner cases, speed loss + for similarly esoteric purposes, et cetera. + This also means that we would reject a patch. + If we are just too lazy to fix a bug then the correct state is open + and unassigned. Closed means that the case is closed which is not + the case if we are just waiting for a patch. + +bug/closed/works_for_me + Bugs for which sufficient information was provided to reproduce but + reproduction failed - that is the code seems to work correctly to the + best of our knowledge. + +patch/open/approved + Patches which have been reviewed and approved by a developer. + Such patches can be applied anytime by any other developer after some + reasonable testing (compile + regression tests + does the patch do + what the author claimed). + +patch/open/needs_changes + Patches which have been reviewed and need changes to be accepted. + +patch/closed/applied + Patches which have been applied. + +patch/closed/rejected + Patches which have been rejected. + +feature_request/closed/implemented + Feature requests which have been implemented. + +feature_request/closed/wont_implement + Feature requests which will not be implemented. The reasons here could + be legal, philosophical or others. + +Note, please do not use type-status-substatus combinations other than the +above without asking on ffmpeg-dev first! + +Note2, if you provide the requested info do not forget to remove the +needs_more_info substatus. + +Component: +---------- + +avcodec + issues in libavcodec/* + +avformat + issues in libavformat/* + +avutil + issues in libavutil/* + +regression test + issues in tests/* + +ffmpeg + issues in or related to ffmpeg.c + +ffplay + issues in or related to ffplay.c + +ffprobe + issues in or related to ffprobe.c + +ffserver + issues in or related to ffserver.c + +build system + issues in or related to configure/Makefile + +regression + bugs which were not present in a past revision + +trac + issues related to our issue tracker diff --git a/ffmpeg1/doc/libavcodec.texi b/ffmpeg1/doc/libavcodec.texi new file mode 100644 index 0000000..618f9f6 --- /dev/null +++ b/ffmpeg1/doc/libavcodec.texi @@ -0,0 +1,48 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavcodec Documentation +@titlepage +@center @titlefont{Libavcodec Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavcodec library provides a generic encoding/decoding framework +and contains multiple decoders and encoders for audio, video and +subtitle streams, and several bitstream filters. + +The shared architecture provides various services ranging from bit +stream I/O to DSP optimizations, and makes it suitable for +implementing robust and fast codecs as well as for experimentation. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavcodec +@settitle media streams decoding and encoding library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libavdevice.texi b/ffmpeg1/doc/libavdevice.texi new file mode 100644 index 0000000..d5f790b --- /dev/null +++ b/ffmpeg1/doc/libavdevice.texi @@ -0,0 +1,45 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavdevice Documentation +@titlepage +@center @titlefont{Libavdevice Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavdevice library provides a generic framework for grabbing from +and rendering to many common multimedia input/output devices, and +supports several input and output devices, including Video4Linux2, +VfW, DShow, and ALSA. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-devices(1), +libavutil(3), libavcodec(3), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavdevice +@settitle multimedia device handling library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libavfilter.texi b/ffmpeg1/doc/libavfilter.texi new file mode 100644 index 0000000..4f82944 --- /dev/null +++ b/ffmpeg1/doc/libavfilter.texi @@ -0,0 +1,44 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavfilter Documentation +@titlepage +@center @titlefont{Libavfilter Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavfilter library provides a generic audio/video filtering +framework containing several filters, sources and sinks. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-filters.html,ffmpeg-filters}, +@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample}, +@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-filters(1), +libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavfilter +@settitle multimedia filtering library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libavformat.texi b/ffmpeg1/doc/libavformat.texi new file mode 100644 index 0000000..85e49cb --- /dev/null +++ b/ffmpeg1/doc/libavformat.texi @@ -0,0 +1,48 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavformat Documentation +@titlepage +@center @titlefont{Libavformat Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavformat library provides a generic framework for multiplexing +and demultiplexing (muxing and demuxing) audio, video and subtitle +streams. It encompasses multiple muxers and demuxers for multimedia +container formats. + +It also supports several input and output protocols to access a media +resource. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-formats(1), ffmpeg-protocols(1), +libavutil(3), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavformat +@settitle multimedia muxing and demuxing library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libavutil.texi b/ffmpeg1/doc/libavutil.texi new file mode 100644 index 0000000..50b0d0e --- /dev/null +++ b/ffmpeg1/doc/libavutil.texi @@ -0,0 +1,44 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libavutil Documentation +@titlepage +@center @titlefont{Libavutil Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavutil library is a utility library to aid portable +multimedia programming. It contains safe portable string functions, +random number generators, data structures, additional mathematics +functions, cryptography and multimedia related functionality (like +enumerations for pixel and sample formats). + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavutil +@settitle multimedia-biased utility library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libswresample.texi b/ffmpeg1/doc/libswresample.texi new file mode 100644 index 0000000..1a5b01f --- /dev/null +++ b/ffmpeg1/doc/libswresample.texi @@ -0,0 +1,70 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libswresample Documentation +@titlepage +@center @titlefont{Libswresample Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libswresample library performs highly optimized audio resampling, +rematrixing and sample format conversion operations. + +Specifically, this library performs the following conversions: + +@itemize +@item +@emph{Resampling}: is the process of changing the audio rate, for +example from an high sample rate of 44100Hz to 8000Hz. Audio +conversion from high to low sample rate is a lossy process. Several +resampling options and algorithms are available. + +@item +@emph{Format conversion}: is the process of converting the type of +samples, for example from 16-bit signed samples to unsigned 8-bit or +float samples. It also handles packing conversion, when passing from +packed layout (all samples belonging to distinct channels interleaved +in the same buffer), to planar layout (all samples belonging to the +same channel stored in a dedicated buffer or "plane"). + +@item +@emph{Rematrixing}: is the process of changing the channel layout, for +example from stereo to mono. When the input channels cannot be mapped +to the output streams, the process is lossy, since it involves +different gain factors and mixing. +@end itemize + +Various other audio conversions (e.g. stretching and padding) are +enabled through dedicated options. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-resampler(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libswresample +@settitle audio resampling library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/libswscale.texi b/ffmpeg1/doc/libswscale.texi new file mode 100644 index 0000000..818e988 --- /dev/null +++ b/ffmpeg1/doc/libswscale.texi @@ -0,0 +1,63 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Libswscale Documentation +@titlepage +@center @titlefont{Libswscale Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libswscale library performs highly optimized image scaling and +colorspace and pixel format conversion operations. + +Specifically, this library performs the following conversions: + +@itemize +@item +@emph{Rescaling}: is the process of changing the video size. Several +rescaling options and algorithms are available. This is usually a +lossy process. + +@item +@emph{Pixel format conversion}: is the process of converting the image +format and colorspace of the image, for example from planar YUV420P to +RGB24 packed. It also handles packing conversion, that is converts +from packed layout (all pixels belonging to distinct planes +interleaved in the same buffer), to planar layout (all samples +belonging to the same plane stored in a dedicated buffer or "plane"). + +This is usually a lossy process in case the source and destination +colorspaces differ. +@end itemize + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-scaler(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libswscale +@settitle video scaling and pixel format conversion library + +@end ignore + +@bye diff --git a/ffmpeg1/doc/metadata.texi b/ffmpeg1/doc/metadata.texi new file mode 100644 index 0000000..2a28575 --- /dev/null +++ b/ffmpeg1/doc/metadata.texi @@ -0,0 +1,68 @@ +@chapter Metadata +@c man begin METADATA + +FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded +INI-like text file and then load it back using the metadata muxer/demuxer. + +The file format is as follows: +@enumerate + +@item +A file consists of a header and a number of metadata tags divided into sections, +each on its own line. + +@item +The header is a ';FFMETADATA' string, followed by a version number (now 1). + +@item +Metadata tags are of the form 'key=value' + +@item +Immediately after header follows global metadata + +@item +After global metadata there may be sections with per-stream/per-chapter +metadata. + +@item +A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in +brackets ('[', ']') and ends with next section or end of file. + +@item +At the beginning of a chapter section there may be an optional timebase to be +used for start/end values. It must be in form 'TIMEBASE=num/den', where num and +den are integers. If the timebase is missing then start/end times are assumed to +be in milliseconds. +Next a chapter section must contain chapter start and end times in form +'START=num', 'END=num', where num is a positive integer. + +@item +Empty lines and lines starting with ';' or '#' are ignored. + +@item +Metadata keys or values containing special characters ('=', ';', '#', '\' and a +newline) must be escaped with a backslash '\'. + +@item +Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of +the tag (in the example above key is 'foo ', value is ' bar'). +@end enumerate + +A ffmetadata file might look like this: +@example +;FFMETADATA1 +title=bike\\shed +;this is a comment +artist=FFmpeg troll team + +[CHAPTER] +TIMEBASE=1/1000 +START=0 +#chapter ends at 0:01:00 +END=60000 +title=chapter \#1 +[STREAM] +title=multi\ +line +@end example +@c man end METADATA diff --git a/ffmpeg1/doc/mips.txt b/ffmpeg1/doc/mips.txt new file mode 100644 index 0000000..959b32c --- /dev/null +++ b/ffmpeg1/doc/mips.txt @@ -0,0 +1,69 @@ +MIPS optimizations info +=============================================== + +MIPS optimizations of codecs are targeting MIPS 74k family of +CPUs. Some of these optimizations are relying more on properties of +this architecture and some are relying less (and can be used on most +MIPS architectures without degradation in performance). + +Along with FFMPEG copyright notice, there is MIPS copyright notice in +all the files that are created by people from MIPS Technologies. + +Example of copyright notice: +=============================================== +/* + * Copyright (c) 2012 + * MIPS Technologies, Inc., California. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its + * contributors may be used to endorse or promote products derived from + * this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND + * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE + * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL + * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS + * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT + * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * Author: Author Name (author_name@@mips.com) + */ + +Files that have MIPS copyright notice in them: +=============================================== +* libavutil/mips/ + float_dsp_mips.c + libm_mips.h +* libavcodec/mips/ + aaccoder_mips.c + ac3dsp_mips.c + acelp_filters_mips.c + acelp_vectors_mips.c + amrwbdec_mips.c + amrwbdec_mips.h + celp_filters_mips.c + celp_math_mips.c + compute_antialias_fixed.h + compute_antialias_float.h + lsp_mips.h + dsputil_mips.c + fft_mips.c + fft_table.h + fft_init_table.c + fmtconvert_mips.c + iirfilter_mips.c + mpegaudiodsp_mips_fixed.c + mpegaudiodsp_mips_float.c diff --git a/ffmpeg1/doc/multithreading.txt b/ffmpeg1/doc/multithreading.txt new file mode 100644 index 0000000..2b992fc --- /dev/null +++ b/ffmpeg1/doc/multithreading.txt @@ -0,0 +1,70 @@ +FFmpeg multithreading methods +============================================== + +FFmpeg provides two methods for multithreading codecs. + +Slice threading decodes multiple parts of a frame at the same time, using +AVCodecContext execute() and execute2(). + +Frame threading decodes multiple frames at the same time. +It accepts N future frames and delays decoded pictures by N-1 frames. +The later frames are decoded in separate threads while the user is +displaying the current one. + +Restrictions on clients +============================================== + +Slice threading - +* The client's draw_horiz_band() must be thread-safe according to the comment + in avcodec.h. + +Frame threading - +* Restrictions with slice threading also apply. +* For best performance, the client should set thread_safe_callbacks if it + provides a thread-safe get_buffer() callback. +* There is one frame of delay added for every thread beyond the first one. + Clients must be able to handle this; the pkt_dts and pkt_pts fields in + AVFrame will work as usual. + +Restrictions on codec implementations +============================================== + +Slice threading - + None except that there must be something worth executing in parallel. + +Frame threading - +* Codecs can only accept entire pictures per packet. +* Codecs similar to ffv1, whose streams don't reset across frames, + will not work because their bitstreams cannot be decoded in parallel. + +* The contents of buffers must not be read before ff_thread_await_progress() + has been called on them. reget_buffer() and buffer age optimizations no longer work. +* The contents of buffers must not be written to after ff_thread_report_progress() + has been called on them. This includes draw_edges(). + +Porting codecs to frame threading +============================================== + +Find all context variables that are needed by the next frame. Move all +code changing them, as well as code calling get_buffer(), up to before +the decode process starts. Call ff_thread_finish_setup() afterwards. If +some code can't be moved, have update_thread_context() run it in the next +thread. + +If the codec allocates writable tables in its init(), add an init_thread_copy() +which re-allocates them for other threads. + +Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little +speed gain at this point but it should work. + +If there are inter-frame dependencies, so the codec calls +ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The +frames must then be freed with ff_thread_release_buffer(). +Otherwise leave it at zero and decode directly into the user-supplied frames. + +Call ff_thread_report_progress() after some part of the current picture has decoded. +A good place to put this is where draw_horiz_band() is called - add this if it isn't +called anywhere, as it's useful too and the implementation is trivial when you're +doing this. Note that draw_edges() needs to be called before reporting progress. + +Before accessing a reference frame or its MVs, call ff_thread_await_progress(). diff --git a/ffmpeg1/doc/muxers.texi b/ffmpeg1/doc/muxers.texi new file mode 100644 index 0000000..9d119c3 --- /dev/null +++ b/ffmpeg1/doc/muxers.texi @@ -0,0 +1,794 @@ +@chapter Muxers +@c man begin MUXERS + +Muxers are configured elements in FFmpeg which allow writing +multimedia streams to a particular type of file. + +When you configure your FFmpeg build, all the supported muxers +are enabled by default. You can list all available muxers using the +configure option @code{--list-muxers}. + +You can disable all the muxers with the configure option +@code{--disable-muxers} and selectively enable / disable single muxers +with the options @code{--enable-muxer=@var{MUXER}} / +@code{--disable-muxer=@var{MUXER}}. + +The option @code{-formats} of the ff* tools will display the list of +enabled muxers. + +A description of some of the currently available muxers follows. + +@anchor{crc} +@section crc + +CRC (Cyclic Redundancy Check) testing format. + +This muxer computes and prints the Adler-32 CRC of all the input audio +and video frames. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +CRC. + +The output of the muxer consists of a single line of the form: +CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to +8 digits containing the CRC for all the decoded input frames. + +For example to compute the CRC of the input, and store it in the file +@file{out.crc}: +@example +ffmpeg -i INPUT -f crc out.crc +@end example + +You can print the CRC to stdout with the command: +@example +ffmpeg -i INPUT -f crc - +@end example + +You can select the output format of each frame with @command{ffmpeg} by +specifying the audio and video codec and format. For example to +compute the CRC of the input audio converted to PCM unsigned 8-bit +and the input video converted to MPEG-2 video, use the command: +@example +ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - +@end example + +See also the @ref{framecrc} muxer. + +@anchor{framecrc} +@section framecrc + +Per-packet CRC (Cyclic Redundancy Check) testing format. + +This muxer computes and prints the Adler-32 CRC for each audio +and video packet. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +CRC. + +The output of the muxer consists of a line for each audio and video +packet of the form: +@example +@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC} +@end example + +@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the +CRC of the packet. + +For example to compute the CRC of the audio and video frames in +@file{INPUT}, converted to raw audio and video packets, and store it +in the file @file{out.crc}: +@example +ffmpeg -i INPUT -f framecrc out.crc +@end example + +To print the information to stdout, use the command: +@example +ffmpeg -i INPUT -f framecrc - +@end example + +With @command{ffmpeg}, you can select the output format to which the +audio and video frames are encoded before computing the CRC for each +packet by specifying the audio and video codec. For example, to +compute the CRC of each decoded input audio frame converted to PCM +unsigned 8-bit and of each decoded input video frame converted to +MPEG-2 video, use the command: +@example +ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - +@end example + +See also the @ref{crc} muxer. + +@anchor{framemd5} +@section framemd5 + +Per-packet MD5 testing format. + +This muxer computes and prints the MD5 hash for each audio +and video packet. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +hash. + +The output of the muxer consists of a line for each audio and video +packet of the form: +@example +@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5} +@end example + +@var{MD5} is a hexadecimal number representing the computed MD5 hash +for the packet. + +For example to compute the MD5 of the audio and video frames in +@file{INPUT}, converted to raw audio and video packets, and store it +in the file @file{out.md5}: +@example +ffmpeg -i INPUT -f framemd5 out.md5 +@end example + +To print the information to stdout, use the command: +@example +ffmpeg -i INPUT -f framemd5 - +@end example + +See also the @ref{md5} muxer. + +@anchor{hls} +@section hls + +Apple HTTP Live Streaming muxer that segments MPEG-TS according to +the HTTP Live Streaming specification. + +It creates a playlist file and numbered segment files. The output +filename specifies the playlist filename; the segment filenames +receive the same basename as the playlist, a sequential number and +a .ts extension. + +@example +ffmpeg -i in.nut out.m3u8 +@end example + +@table @option +@item -hls_time @var{seconds} +Set the segment length in seconds. +@item -hls_list_size @var{size} +Set the maximum number of playlist entries. +@item -hls_wrap @var{wrap} +Set the number after which index wraps. +@item -start_number @var{number} +Start the sequence from @var{number}. +@end table + +@anchor{ico} +@section ico + +ICO file muxer. + +Microsoft's icon file format (ICO) has some strict limitations that should be noted: + +@itemize +@item +Size cannot exceed 256 pixels in any dimension + +@item +Only BMP and PNG images can be stored + +@item +If a BMP image is used, it must be one of the following pixel formats: +@example +BMP Bit Depth FFmpeg Pixel Format +1bit pal8 +4bit pal8 +8bit pal8 +16bit rgb555le +24bit bgr24 +32bit bgra +@end example + +@item +If a BMP image is used, it must use the BITMAPINFOHEADER DIB header + +@item +If a PNG image is used, it must use the rgba pixel format +@end itemize + +@anchor{image2} +@section image2 + +Image file muxer. + +The image file muxer writes video frames to image files. + +The output filenames are specified by a pattern, which can be used to +produce sequentially numbered series of files. +The pattern may contain the string "%d" or "%0@var{N}d", this string +specifies the position of the characters representing a numbering in +the filenames. If the form "%0@var{N}d" is used, the string +representing the number in each filename is 0-padded to @var{N} +digits. The literal character '%' can be specified in the pattern with +the string "%%". + +If the pattern contains "%d" or "%0@var{N}d", the first filename of +the file list specified will contain the number 1, all the following +numbers will be sequential. + +The pattern may contain a suffix which is used to automatically +determine the format of the image files to write. + +For example the pattern "img-%03d.bmp" will specify a sequence of +filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., +@file{img-010.bmp}, etc. +The pattern "img%%-%d.jpg" will specify a sequence of filenames of the +form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg}, +etc. + +The following example shows how to use @command{ffmpeg} for creating a +sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ..., +taking one image every second from the input video: +@example +ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' +@end example + +Note that with @command{ffmpeg}, if the format is not specified with the +@code{-f} option and the output filename specifies an image file +format, the image2 muxer is automatically selected, so the previous +command can be written as: +@example +ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' +@end example + +Note also that the pattern must not necessarily contain "%d" or +"%0@var{N}d", for example to create a single image file +@file{img.jpeg} from the input video you can employ the command: +@example +ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg +@end example + +@table @option +@item start_number @var{number} +Start the sequence from @var{number}. Default value is 1. Must be a +positive number. + +@item updatefirst 1|0 +If set to 1, update the first written image file again and +again. Default value is 0. +@end table + +The image muxer supports the .Y.U.V image file format. This format is +special in that that each image frame consists of three files, for +each of the YUV420P components. To read or write this image file format, +specify the name of the '.Y' file. The muxer will automatically open the +'.U' and '.V' files as required. + +@anchor{md5} +@section md5 + +MD5 testing format. + +This muxer computes and prints the MD5 hash of all the input audio +and video frames. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +hash. + +The output of the muxer consists of a single line of the form: +MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing +the computed MD5 hash. + +For example to compute the MD5 hash of the input converted to raw +audio and video, and store it in the file @file{out.md5}: +@example +ffmpeg -i INPUT -f md5 out.md5 +@end example + +You can print the MD5 to stdout with the command: +@example +ffmpeg -i INPUT -f md5 - +@end example + +See also the @ref{framemd5} muxer. + +@section MOV/MP4/ISMV + +The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 +file has all the metadata about all packets stored in one location +(written at the end of the file, it can be moved to the start for +better playback by adding @var{faststart} to the @var{movflags}, or +using the @command{qt-faststart} tool). A fragmented +file consists of a number of fragments, where packets and metadata +about these packets are stored together. Writing a fragmented +file has the advantage that the file is decodable even if the +writing is interrupted (while a normal MOV/MP4 is undecodable if +it is not properly finished), and it requires less memory when writing +very long files (since writing normal MOV/MP4 files stores info about +every single packet in memory until the file is closed). The downside +is that it is less compatible with other applications. + +Fragmentation is enabled by setting one of the AVOptions that define +how to cut the file into fragments: + +@table @option +@item -moov_size @var{bytes} +Reserves space for the moov atom at the beginning of the file instead of placing the +moov atom at the end. If the space reserved is insufficient, muxing will fail. +@item -movflags frag_keyframe +Start a new fragment at each video keyframe. +@item -frag_duration @var{duration} +Create fragments that are @var{duration} microseconds long. +@item -frag_size @var{size} +Create fragments that contain up to @var{size} bytes of payload data. +@item -movflags frag_custom +Allow the caller to manually choose when to cut fragments, by +calling @code{av_write_frame(ctx, NULL)} to write a fragment with +the packets written so far. (This is only useful with other +applications integrating libavformat, not from @command{ffmpeg}.) +@item -min_frag_duration @var{duration} +Don't create fragments that are shorter than @var{duration} microseconds long. +@end table + +If more than one condition is specified, fragments are cut when +one of the specified conditions is fulfilled. The exception to this is +@code{-min_frag_duration}, which has to be fulfilled for any of the other +conditions to apply. + +Additionally, the way the output file is written can be adjusted +through a few other options: + +@table @option +@item -movflags empty_moov +Write an initial moov atom directly at the start of the file, without +describing any samples in it. Generally, an mdat/moov pair is written +at the start of the file, as a normal MOV/MP4 file, containing only +a short portion of the file. With this option set, there is no initial +mdat atom, and the moov atom only describes the tracks but has +a zero duration. + +Files written with this option set do not work in QuickTime. +This option is implicitly set when writing ismv (Smooth Streaming) files. +@item -movflags separate_moof +Write a separate moof (movie fragment) atom for each track. Normally, +packets for all tracks are written in a moof atom (which is slightly +more efficient), but with this option set, the muxer writes one moof/mdat +pair for each track, making it easier to separate tracks. + +This option is implicitly set when writing ismv (Smooth Streaming) files. +@item -movflags faststart +Run a second pass moving the moov atom on top of the file. This +operation can take a while, and will not work in various situations such +as fragmented output, thus it is not enabled by default. +@item -movflags rtphint +Add RTP hinting tracks to the output file. +@end table + +Smooth Streaming content can be pushed in real time to a publishing +point on IIS with this muxer. Example: +@example +ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) +@end example + +@section mpegts + +MPEG transport stream muxer. + +This muxer implements ISO 13818-1 and part of ETSI EN 300 468. + +The muxer options are: + +@table @option +@item -mpegts_original_network_id @var{number} +Set the original_network_id (default 0x0001). This is unique identifier +of a network in DVB. Its main use is in the unique identification of a +service through the path Original_Network_ID, Transport_Stream_ID. +@item -mpegts_transport_stream_id @var{number} +Set the transport_stream_id (default 0x0001). This identifies a +transponder in DVB. +@item -mpegts_service_id @var{number} +Set the service_id (default 0x0001) also known as program in DVB. +@item -mpegts_pmt_start_pid @var{number} +Set the first PID for PMT (default 0x1000, max 0x1f00). +@item -mpegts_start_pid @var{number} +Set the first PID for data packets (default 0x0100, max 0x0f00). +@end table + +The recognized metadata settings in mpegts muxer are @code{service_provider} +and @code{service_name}. If they are not set the default for +@code{service_provider} is "FFmpeg" and the default for +@code{service_name} is "Service01". + +@example +ffmpeg -i file.mpg -c copy \ + -mpegts_original_network_id 0x1122 \ + -mpegts_transport_stream_id 0x3344 \ + -mpegts_service_id 0x5566 \ + -mpegts_pmt_start_pid 0x1500 \ + -mpegts_start_pid 0x150 \ + -metadata service_provider="Some provider" \ + -metadata service_name="Some Channel" \ + -y out.ts +@end example + +@section null + +Null muxer. + +This muxer does not generate any output file, it is mainly useful for +testing or benchmarking purposes. + +For example to benchmark decoding with @command{ffmpeg} you can use the +command: +@example +ffmpeg -benchmark -i INPUT -f null out.null +@end example + +Note that the above command does not read or write the @file{out.null} +file, but specifying the output file is required by the @command{ffmpeg} +syntax. + +Alternatively you can write the command as: +@example +ffmpeg -benchmark -i INPUT -f null - +@end example + +@section matroska + +Matroska container muxer. + +This muxer implements the matroska and webm container specs. + +The recognized metadata settings in this muxer are: + +@table @option + +@item title=@var{title name} +Name provided to a single track +@end table + +@table @option + +@item language=@var{language name} +Specifies the language of the track in the Matroska languages form +@end table + +@table @option + +@item stereo_mode=@var{mode} +Stereo 3D video layout of two views in a single video track +@table @option +@item mono +video is not stereo +@item left_right +Both views are arranged side by side, Left-eye view is on the left +@item bottom_top +Both views are arranged in top-bottom orientation, Left-eye view is at bottom +@item top_bottom +Both views are arranged in top-bottom orientation, Left-eye view is on top +@item checkerboard_rl +Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first +@item checkerboard_lr +Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first +@item row_interleaved_rl +Each view is constituted by a row based interleaving, Right-eye view is first row +@item row_interleaved_lr +Each view is constituted by a row based interleaving, Left-eye view is first row +@item col_interleaved_rl +Both views are arranged in a column based interleaving manner, Right-eye view is first column +@item col_interleaved_lr +Both views are arranged in a column based interleaving manner, Left-eye view is first column +@item anaglyph_cyan_red +All frames are in anaglyph format viewable through red-cyan filters +@item right_left +Both views are arranged side by side, Right-eye view is on the left +@item anaglyph_green_magenta +All frames are in anaglyph format viewable through green-magenta filters +@item block_lr +Both eyes laced in one Block, Left-eye view is first +@item block_rl +Both eyes laced in one Block, Right-eye view is first +@end table +@end table + +For example a 3D WebM clip can be created using the following command line: +@example +ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm +@end example + +@section segment, stream_segment, ssegment + +Basic stream segmenter. + +The segmenter muxer outputs streams to a number of separate files of nearly +fixed duration. Output filename pattern can be set in a fashion similar to +@ref{image2}. + +@code{stream_segment} is a variant of the muxer used to write to +streaming output formats, i.e. which do not require global headers, +and is recommended for outputting e.g. to MPEG transport stream segments. +@code{ssegment} is a shorter alias for @code{stream_segment}. + +Every segment starts with a keyframe of the selected reference stream, +which is set through the @option{reference_stream} option. + +Note that if you want accurate splitting for a video file, you need to +make the input key frames correspond to the exact splitting times +expected by the segmenter, or the segment muxer will start the new +segment with the key frame found next after the specified start +time. + +The segment muxer works best with a single constant frame rate video. + +Optionally it can generate a list of the created segments, by setting +the option @var{segment_list}. The list type is specified by the +@var{segment_list_type} option. + +The segment muxer supports the following options: + +@table @option +@item reference_stream @var{specifier} +Set the reference stream, as specified by the string @var{specifier}. +If @var{specifier} is set to @code{auto}, the reference is choosen +automatically. Otherwise it must be a stream specifier (see the ``Stream +specifiers'' chapter in the ffmpeg manual) which specifies the +reference stream. The default value is ``auto''. + +@item segment_format @var{format} +Override the inner container format, by default it is guessed by the filename +extension. + +@item segment_list @var{name} +Generate also a listfile named @var{name}. If not specified no +listfile is generated. + +@item segment_list_flags @var{flags} +Set flags affecting the segment list generation. + +It currently supports the following flags: +@table @var +@item cache +Allow caching (only affects M3U8 list files). + +@item live +Allow live-friendly file generation. +@end table + +Default value is @code{cache}. + +@item segment_list_size @var{size} +Update the list file so that it contains at most the last @var{size} +segments. If 0 the list file will contain all the segments. Default +value is 0. + +@item segment_list type @var{type} +Specify the format for the segment list file. + +The following values are recognized: +@table @option +@item flat +Generate a flat list for the created segments, one segment per line. + +@item csv, ext +Generate a list for the created segments, one segment per line, +each line matching the format (comma-separated values): +@example +@var{segment_filename},@var{segment_start_time},@var{segment_end_time} +@end example + +@var{segment_filename} is the name of the output file generated by the +muxer according to the provided pattern. CSV escaping (according to +RFC4180) is applied if required. + +@var{segment_start_time} and @var{segment_end_time} specify +the segment start and end time expressed in seconds. + +A list file with the suffix @code{".csv"} or @code{".ext"} will +auto-select this format. + +@code{ext} is deprecated in favor or @code{csv}. + +@item ffconcat +Generate an ffconcat file for the created segments. The resulting file +can be read using the FFmpeg @ref{concat} demuxer. + +A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will +auto-select this format. + +@item m3u8 +Generate an extended M3U8 file, version 3, compliant with +@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}. + +A list file with the suffix @code{".m3u8"} will auto-select this format. +@end table + +If not specified the type is guessed from the list file name suffix. + +@item segment_time @var{time} +Set segment duration to @var{time}, the value must be a duration +specification. Default value is "2". See also the +@option{segment_times} option. + +Note that splitting may not be accurate, unless you force the +reference stream key-frames at the given time. See the introductory +notice and the examples below. + +@item segment_time_delta @var{delta} +Specify the accuracy time when selecting the start time for a +segment, expressed as a duration specification. Default value is "0". + +When delta is specified a key-frame will start a new segment if its +PTS satisfies the relation: +@example +PTS >= start_time - time_delta +@end example + +This option is useful when splitting video content, which is always +split at GOP boundaries, in case a key frame is found just before the +specified split time. + +In particular may be used in combination with the @file{ffmpeg} option +@var{force_key_frames}. The key frame times specified by +@var{force_key_frames} may not be set accurately because of rounding +issues, with the consequence that a key frame time may result set just +before the specified time. For constant frame rate videos a value of +1/2*@var{frame_rate} should address the worst case mismatch between +the specified time and the time set by @var{force_key_frames}. + +@item segment_times @var{times} +Specify a list of split points. @var{times} contains a list of comma +separated duration specifications, in increasing order. See also +the @option{segment_time} option. + +@item segment_frames @var{frames} +Specify a list of split video frame numbers. @var{frames} contains a +list of comma separated integer numbers, in increasing order. + +This option specifies to start a new segment whenever a reference +stream key frame is found and the sequential number (starting from 0) +of the frame is greater or equal to the next value in the list. + +@item segment_wrap @var{limit} +Wrap around segment index once it reaches @var{limit}. + +@item segment_start_number @var{number} +Set the sequence number of the first segment. Defaults to @code{0}. + +@item reset_timestamps @var{1|0} +Reset timestamps at the begin of each segment, so that each segment +will start with near-zero timestamps. It is meant to ease the playback +of the generated segments. May not work with some combinations of +muxers/codecs. It is set to @code{0} by default. +@end table + +@subsection Examples + +@itemize +@item +To remux the content of file @file{in.mkv} to a list of segments +@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of +generated segments to @file{out.list}: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut +@end example + +@item +As the example above, but segment the input file according to the split +points specified by the @var{segment_times} option: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut +@end example + +@item +As the example above, but use the @code{ffmpeg} @var{force_key_frames} +option to force key frames in the input at the specified location, together +with the segment option @var{segment_time_delta} to account for +possible roundings operated when setting key frame times. +@example +ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \ +-f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut +@end example +In order to force key frames on the input file, transcoding is +required. + +@item +Segment the input file by splitting the input file according to the +frame numbers sequence specified with the @var{segment_frames} option: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut +@end example + +@item +To convert the @file{in.mkv} to TS segments using the @code{libx264} +and @code{libfaac} encoders: +@example +ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts +@end example + +@item +Segment the input file, and create an M3U8 live playlist (can be used +as live HLS source): +@example +ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ +-segment_list_flags +live -segment_time 10 out%03d.mkv +@end example +@end itemize + +@section mp3 + +The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and +optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the +@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is +not written by default, but may be enabled with the @code{write_id3v1} option. + +For seekable output the muxer also writes a Xing frame at the beginning, which +contains the number of frames in the file. It is useful for computing duration +of VBR files. + +The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures +are supplied to the muxer in form of a video stream with a single packet. There +can be any number of those streams, each will correspond to a single APIC frame. +The stream metadata tags @var{title} and @var{comment} map to APIC +@var{description} and @var{picture type} respectively. See +@url{http://id3.org/id3v2.4.0-frames} for allowed picture types. + +Note that the APIC frames must be written at the beginning, so the muxer will +buffer the audio frames until it gets all the pictures. It is therefore advised +to provide the pictures as soon as possible to avoid excessive buffering. + +Examples: + +Write an mp3 with an ID3v2.3 header and an ID3v1 footer: +@example +ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 +@end example + +To attach a picture to an mp3 file select both the audio and the picture stream +with @code{map}: +@example +ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 +-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 +@end example + +@section ogg + +Ogg container muxer. + +@table @option +@item -page_duration @var{duration} +Preferred page duration, in microseconds. The muxer will attempt to create +pages that are approximately @var{duration} microseconds long. This allows the +user to compromise between seek granularity and container overhead. The default +is 1 second. A value of 0 will fill all segments, making pages as large as +possible. A value of 1 will effectively use 1 packet-per-page in most +situations, giving a small seek granularity at the cost of additional container +overhead. +@end table + +@section tee + +The tee muxer can be used to write the same data to several files or any +other kind of muxer. It can be used, for example, to both stream a video to +the network and save it to disk at the same time. + +It is different from specifying several outputs to the @command{ffmpeg} +command-line tool because the audio and video data will be encoded only once +with the tee muxer; encoding can be a very expensive process. It is not +useful when using the libavformat API directly because it is then possible +to feed the same packets to several muxers directly. + +The slave outputs are specified in the file name given to the muxer, +separated by '|'. If any of the slave name contains the '|' separator, +leading or trailing spaces or any special character, it must be +escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils +manual). + +Options can be specified for each slave by prepending them as a list of +@var{key}=@var{value} pairs separated by ':', between square brackets. If +the options values contain a special character or the ':' separator, they +must be escaped; note that this is a second level escaping. + +Example: encode something and both archive it in a WebM file and stream it +as MPEG-TS over UDP (the streams need to be explicitly mapped): + +@example +ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a + "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/" +@end example + +Note: some codecs may need different options depending on the output format; +the auto-detection of this can not work with the tee muxer. The main example +is the @option{global_header} flag. + +@c man end MUXERS diff --git a/ffmpeg1/doc/nut.texi b/ffmpeg1/doc/nut.texi new file mode 100644 index 0000000..0026a12 --- /dev/null +++ b/ffmpeg1/doc/nut.texi @@ -0,0 +1,138 @@ +\input texinfo @c -*- texinfo -*- + +@settitle NUT + +@titlepage +@center @titlefont{NUT} +@end titlepage + +@top + +@contents + +@chapter Description +NUT is a low overhead generic container format. It stores audio, video, +subtitle and user-defined streams in a simple, yet efficient, way. + +It was created by a group of FFmpeg and MPlayer developers in 2003 +and was finalized in 2008. + +The official nut specification is at svn://svn.mplayerhq.hu/nut +In case of any differences between this text and the official specification, +the official specification shall prevail. + +@chapter Container-specific codec tags + +@section Generic raw YUVA formats + +Since many exotic planar YUVA pixel formats are not considered by +the AVI/QuickTime FourCC lists, the following scheme is adopted for +representing them. + +The first two bytes can contain the values: +Y1 = only Y +Y2 = Y+A +Y3 = YUV +Y4 = YUVA + +The third byte represents the width and height chroma subsampling +values for the UV planes, that is the amount to shift the luma +width/height right to find the chroma width/height. + +The fourth byte is the number of bits used (8, 16, ...). + +If the order of bytes is inverted, that means that each component has +to be read big-endian. + +@section Raw Audio + +@multitable @columnfractions .4 .4 +@item ALAW @tab A-LAW +@item ULAW @tab MU-LAW +@item P<type><interleaving><bits> @tab little-endian PCM +@item <bits><interleaving><type>P @tab big-endian PCM +@end multitable + +<type> is S for signed integer, U for unsigned integer, F for IEEE float +<interleaving> is D for default, P is for planar. +<bits> is 8/16/24/32 + +@example +PFD[32] would for example be signed 32 bit little-endian IEEE float +@end example + +@section Subtitles + +@multitable @columnfractions .4 .4 +@item UTF8 @tab Raw UTF-8 +@item SSA[0] @tab SubStation Alpha +@item DVDS @tab DVD subtitles +@item DVBS @tab DVB subtitles +@end multitable + +@section Raw Data + +@multitable @columnfractions .4 .4 +@item UTF8 @tab Raw UTF-8 +@end multitable + +@section Codecs + +@multitable @columnfractions .4 .4 +@item 3IV1 @tab non-compliant MPEG-4 generated by old 3ivx +@item ASV1 @tab Asus Video +@item ASV2 @tab Asus Video 2 +@item CVID @tab Cinepak +@item CYUV @tab Creative YUV +@item DIVX @tab non-compliant MPEG-4 generated by old DivX +@item DUCK @tab Truemotion 1 +@item FFV1 @tab FFmpeg video 1 +@item FFVH @tab FFmpeg Huffyuv +@item H261 @tab ITU H.261 +@item H262 @tab ITU H.262 +@item H263 @tab ITU H.263 +@item H264 @tab ITU H.264 +@item HFYU @tab Huffyuv +@item I263 @tab Intel H.263 +@item IV31 @tab Indeo 3.1 +@item IV32 @tab Indeo 3.2 +@item IV50 @tab Indeo 5.0 +@item LJPG @tab ITU JPEG (lossless) +@item MJLS @tab ITU JPEG-LS +@item MJPG @tab ITU JPEG +@item MPG4 @tab MS MPEG-4v1 (not ISO MPEG-4) +@item MP42 @tab MS MPEG-4v2 +@item MP43 @tab MS MPEG-4v3 +@item MP4V @tab ISO MPEG-4 Part 2 Video (from old encoders) +@item mpg1 @tab ISO MPEG-1 Video +@item mpg2 @tab ISO MPEG-2 Video +@item MRLE @tab MS RLE +@item MSVC @tab MS Video 1 +@item RT21 @tab Indeo 2.1 +@item RV10 @tab RealVideo 1.0 +@item RV20 @tab RealVideo 2.0 +@item RV30 @tab RealVideo 3.0 +@item RV40 @tab RealVideo 4.0 +@item SNOW @tab FFmpeg Snow +@item SVQ1 @tab Sorenson Video 1 +@item SVQ3 @tab Sorenson Video 3 +@item theo @tab Xiph Theora +@item TM20 @tab Truemotion 2.0 +@item UMP4 @tab non-compliant MPEG-4 generated by UB Video MPEG-4 +@item VCR1 @tab ATI VCR1 +@item VP30 @tab VP 3.0 +@item VP31 @tab VP 3.1 +@item VP50 @tab VP 5.0 +@item VP60 @tab VP 6.0 +@item VP61 @tab VP 6.1 +@item VP62 @tab VP 6.2 +@item VP70 @tab VP 7.0 +@item WMV1 @tab MS WMV7 +@item WMV2 @tab MS WMV8 +@item WMV3 @tab MS WMV9 +@item WV1F @tab non-compliant MPEG-4 generated by ? +@item WVC1 @tab VC-1 +@item XVID @tab non-compliant MPEG-4 generated by old Xvid +@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug +@end multitable + diff --git a/ffmpeg1/doc/optimization.txt b/ffmpeg1/doc/optimization.txt new file mode 100644 index 0000000..5a66d6b --- /dev/null +++ b/ffmpeg1/doc/optimization.txt @@ -0,0 +1,288 @@ +optimization Tips (for libavcodec): +=================================== + +What to optimize: +----------------- +If you plan to do non-x86 architecture specific optimizations (SIMD normally), +then take a look in the x86/ directory, as most important functions are +already optimized for MMX. + +If you want to do x86 optimizations then you can either try to finetune the +stuff in the x86 directory or find some other functions in the C source to +optimize, but there aren't many left. + + +Understanding these overoptimized functions: +-------------------------------------------- +As many functions tend to be a bit difficult to understand because +of optimizations, it can be hard to optimize them further, or write +architecture-specific versions. It is recommended to look at older +revisions of the interesting files (web frontends for the various FFmpeg +branches are listed at http://ffmpeg.org/download.html). +Alternatively, look into the other architecture-specific versions in +the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly +comprehend the instructions, it could help understanding the functions +and how they can be optimized. + +NOTE: If you still don't understand some function, ask at our mailing list!!! +(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel) + + +When is an optimization justified? +---------------------------------- +Normally, clean and simple optimizations for widely used codecs are +justified even if they only achieve an overall speedup of 0.1%. These +speedups accumulate and can make a big difference after awhile. Also, if +none of the following factors get worse due to an optimization -- speed, +binary code size, source size, source readability -- and at least one +factor improves, then an optimization is always a good idea even if the +overall gain is less than 0.1%. For obscure codecs that are not often +used, the goal is more toward keeping the code clean, small, and +readable instead of making it 1% faster. + + +WTF is that function good for ....: +----------------------------------- +The primary purpose of this list is to avoid wasting time optimizing functions +which are rarely used. + +put(_no_rnd)_pixels{,_x2,_y2,_xy2} + Used in motion compensation (en/decoding). + +avg_pixels{,_x2,_y2,_xy2} + Used in motion compensation of B-frames. + These are less important than the put*pixels functions. + +avg_no_rnd_pixels* + unused + +pix_abs16x16{,_x2,_y2,_xy2} + Used in motion estimation (encoding) with SAD. + +pix_abs8x8{,_x2,_y2,_xy2} + Used in motion estimation (encoding) with SAD of MPEG-4 4MV only. + These are less important than the pix_abs16x16* functions. + +put_mspel8_mc* / wmv2_mspel8* + Used only in WMV2. + it is not recommended that you waste your time with these, as WMV2 + is an ugly and relatively useless codec. + +mpeg4_qpel* / *qpel_mc* + Used in MPEG-4 qpel motion compensation (encoding & decoding). + The qpel8 functions are used only for 4mv, + the avg_* functions are used only for B-frames. + Optimizing them should have a significant impact on qpel + encoding & decoding. + +qpel{8,16}_mc??_old_c / *pixels{8,16}_l4 + Just used to work around a bug in an old libavcodec encoder version. + Don't optimize them. + +tpel_mc_func {put,avg}_tpel_pixels_tab + Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding. + +add_bytes/diff_bytes + For huffyuv only, optimize if you want a faster ffhuffyuv codec. + +get_pixels / diff_pixels + Used for encoding, easy. + +clear_blocks + easiest to optimize + +gmc + Used for MPEG-4 gmc. + Optimizing this should have a significant effect on the gmc decoding + speed. + +gmc1 + Used for chroma blocks in MPEG-4 gmc with 1 warp point + (there are 4 luma & 2 chroma blocks per macroblock, so + only 1/3 of the gmc blocks use this, the other 2/3 + use the normal put_pixel* code, but only if there is + just 1 warp point). + Note: DivX5 gmc always uses just 1 warp point. + +pix_sum + Used for encoding. + +hadamard8_diff / sse / sad == pix_norm1 / dct_sad / quant_psnr / rd / bit + Specific compare functions used in encoding, it depends upon the + command line switches which of these are used. + Don't waste your time with dct_sad & quant_psnr, they aren't + really useful. + +put_pixels_clamped / add_pixels_clamped + Used for en/decoding in the IDCT, easy. + Note, some optimized IDCTs have the add/put clamped code included and + then put_pixels_clamped / add_pixels_clamped will be unused. + +idct/fdct + idct (encoding & decoding) + fdct (encoding) + difficult to optimize + +dct_quantize_trellis + Used for encoding with trellis quantization. + difficult to optimize + +dct_quantize + Used for encoding. + +dct_unquantize_mpeg1 + Used in MPEG-1 en/decoding. + +dct_unquantize_mpeg2 + Used in MPEG-2 en/decoding. + +dct_unquantize_h263 + Used in MPEG-4/H.263 en/decoding. + +FIXME remaining functions? +BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h. + + + +Alignment: +Some instructions on some architectures have strict alignment restrictions, +for example most SSE/SSE2 instructions on x86. +The minimum guaranteed alignment is written in the .h files, for example: + void (*put_pixels_clamped)(const int16_t *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size); + + +General Tips: +------------- +Use asm loops like: +__asm__( + "1: .... + ... + "jump_instruction .... +Do not use C loops: +do{ + __asm__( + ... +}while() + +For x86, mark registers that are clobbered in your asm. This means both +general x86 registers (e.g. eax) as well as XMM registers. This last one is +particularly important on Win64, where xmm6-15 are callee-save, and not +restoring their contents leads to undefined results. In external asm (e.g. +yasm), you do this by using: +cglobal functon_name, num_args, num_regs, num_xmm_regs +In inline asm, you specify clobbered registers at the end of your asm: +__asm__(".." ::: "%eax"). +If gcc is not set to support sse (-msse) it will not accept xmm registers +in the clobber list. For that we use two macros to declare the clobbers. +XMM_CLOBBERS should be used when there are other clobbers, for example: +__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax"); +and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers: +__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0")); + +Do not expect a compiler to maintain values in your registers between separate +(inline) asm code blocks. It is not required to. For example, this is bad: +__asm__("movdqa %0, %%xmm7" : src); +/* do something */ +__asm__("movdqa %%xmm7, %1" : dst); +- first of all, you're assuming that the compiler will not use xmm7 in + between the two asm blocks. It probably won't when you test it, but it's + a poor assumption that will break at some point for some --cpu compiler flag +- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would + have restored the original value of xmm7 after the first asm block, thus + rendering the combination of the two blocks of code invalid +Code that depends on data in registries being untouched, should be written as +a single __asm__() statement. Ideally, a single function contains only one +__asm__() block. + +Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics. +The latter requires a good optimizing compiler which gcc is not. + +Inline asm vs. external asm +--------------------------- +Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc) +and external asm (.s or .asm files, handled by an assembler such as yasm/nasm) +are accepted in FFmpeg. Which one to use differs per specific case. + +- if your code is intended to be inlined in a C function, inline asm is always + better, because external asm cannot be inlined +- if your code calls external functions, yasm is always better +- if your code takes huge and complex structs as function arguments (e.g. + MpegEncContext; note that this is not ideal and is discouraged if there + are alternatives), then inline asm is always better, because predicting + member offsets in complex structs is almost impossible. It's safest to let + the compiler take care of that +- in many cases, both can be used and it just depends on the preference of the + person writing the asm. For new asm, the choice is up to you. For existing + asm, you'll likely want to maintain whatever form it is currently in unless + there is a good reason to change it. +- if, for some reason, you believe that a particular chunk of existing external + asm could be improved upon further if written in inline asm (or the other + way around), then please make the move from external asm <-> inline asm a + separate patch before your patches that actually improve the asm. + + +Links: +====== +http://www.aggregate.org/MAGIC/ + +x86-specific: +------------- +http://developer.intel.com/design/pentium4/manuals/248966.htm + +The IA-32 Intel Architecture Software Developer's Manual, Volume 2: +Instruction Set Reference +http://developer.intel.com/design/pentium4/manuals/245471.htm + +http://www.agner.org/assem/ + +AMD Athlon Processor x86 Code Optimization Guide: +http://www.amd.com/us-en/assets/content_type/white_papers_and_tech_docs/22007.pdf + + +ARM-specific: +------------- +ARM Architecture Reference Manual (up to ARMv5TE): +http://www.arm.com/community/university/eulaarmarm.html + +Procedure Call Standard for the ARM Architecture: +http://www.arm.com/pdfs/aapcs.pdf + +Optimization guide for ARM9E (used in Nokia 770 Internet Tablet): +http://infocenter.arm.com/help/topic/com.arm.doc.ddi0240b/DDI0240A.pdf +Optimization guide for ARM11 (used in Nokia N800 Internet Tablet): +http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf +Optimization guide for Intel XScale (used in Sharp Zaurus PDA): +http://download.intel.com/design/intelxscale/27347302.pdf +Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual +http://download.intel.com/design/intelxscale/31451001.pdf + +PowerPC-specific: +----------------- +PowerPC32/AltiVec PIM: +www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPEM.pdf + +PowerPC32/AltiVec PEM: +www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPIM.pdf + +CELL/SPU: +http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf +http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf + +SPARC-specific: +--------------- +SPARC Joint Programming Specification (JPS1): Commonality +http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf + +UltraSPARC III Processor User's Manual (contains instruction timings) +http://www.sun.com/processors/manuals/USIIIv2.pdf + +VIS Whitepaper (contains optimization guidelines) +http://www.sun.com/processors/vis/download/vis/vis_whitepaper.pdf + +GCC asm links: +-------------- +official doc but quite ugly +http://gcc.gnu.org/onlinedocs/gcc/Extended-Asm.html + +a bit old (note "+" is valid for input-output, even though the next disagrees) +http://www.cs.virginia.edu/~clc5q/gcc-inline-asm.pdf diff --git a/ffmpeg1/doc/outdevs.texi b/ffmpeg1/doc/outdevs.texi new file mode 100644 index 0000000..371d63a --- /dev/null +++ b/ffmpeg1/doc/outdevs.texi @@ -0,0 +1,156 @@ +@chapter Output Devices +@c man begin OUTPUT DEVICES + +Output devices are configured elements in FFmpeg which allow to write +multimedia data to an output device attached to your system. + +When you configure your FFmpeg build, all the supported output devices +are enabled by default. You can list all available ones using the +configure option "--list-outdevs". + +You can disable all the output devices using the configure option +"--disable-outdevs", and selectively enable an output device using the +option "--enable-outdev=@var{OUTDEV}", or you can disable a particular +input device using the option "--disable-outdev=@var{OUTDEV}". + +The option "-formats" of the ff* tools will display the list of +enabled output devices (amongst the muxers). + +A description of the currently available output devices follows. + +@section alsa + +ALSA (Advanced Linux Sound Architecture) output device. + +@section caca + +CACA output device. + +This output devices allows to show a video stream in CACA window. +Only one CACA window is allowed per application, so you can +have only one instance of this output device in an application. + +To enable this output device you need to configure FFmpeg with +@code{--enable-libcaca}. +libcaca is a graphics library that outputs text instead of pixels. + +For more information about libcaca, check: +@url{http://caca.zoy.org/wiki/libcaca} + +@subsection Options + +@table @option + +@item window_title +Set the CACA window title, if not specified default to the filename +specified for the output device. + +@item window_size +Set the CACA window size, can be a string of the form +@var{width}x@var{height} or a video size abbreviation. +If not specified it defaults to the size of the input video. + +@item driver +Set display driver. + +@item algorithm +Set dithering algorithm. Dithering is necessary +because the picture being rendered has usually far more colours than +the available palette. +The accepted values are listed with @code{-list_dither algorithms}. + +@item antialias +Set antialias method. Antialiasing smoothens the rendered +image and avoids the commonly seen staircase effect. +The accepted values are listed with @code{-list_dither antialiases}. + +@item charset +Set which characters are going to be used when rendering text. +The accepted values are listed with @code{-list_dither charsets}. + +@item color +Set color to be used when rendering text. +The accepted values are listed with @code{-list_dither colors}. + +@item list_drivers +If set to @option{true}, print a list of available drivers and exit. + +@item list_dither +List available dither options related to the argument. +The argument must be one of @code{algorithms}, @code{antialiases}, +@code{charsets}, @code{colors}. +@end table + +@subsection Examples + +@itemize +@item +The following command shows the @command{ffmpeg} output is an +CACA window, forcing its size to 80x25: +@example +ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - +@end example + +@item +Show the list of available drivers and exit: +@example +ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - +@end example + +@item +Show the list of available dither colors and exit: +@example +ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - +@end example +@end itemize + +@section oss + +OSS (Open Sound System) output device. + +@section sdl + +SDL (Simple DirectMedia Layer) output device. + +This output devices allows to show a video stream in an SDL +window. Only one SDL window is allowed per application, so you can +have only one instance of this output device in an application. + +To enable this output device you need libsdl installed on your system +when configuring your build. + +For more information about SDL, check: +@url{http://www.libsdl.org/} + +@subsection Options + +@table @option + +@item window_title +Set the SDL window title, if not specified default to the filename +specified for the output device. + +@item icon_title +Set the name of the iconified SDL window, if not specified it is set +to the same value of @var{window_title}. + +@item window_size +Set the SDL window size, can be a string of the form +@var{width}x@var{height} or a video size abbreviation. +If not specified it defaults to the size of the input video, +downscaled according to the aspect ratio. +@end table + +@subsection Examples + +The following command shows the @command{ffmpeg} output is an +SDL window, forcing its size to the qcif format: +@example +ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" +@end example + +@section sndio + +sndio audio output device. + +@c man end OUTPUT DEVICES diff --git a/ffmpeg1/doc/platform.texi b/ffmpeg1/doc/platform.texi new file mode 100644 index 0000000..bb8e6ca --- /dev/null +++ b/ffmpeg1/doc/platform.texi @@ -0,0 +1,369 @@ +\input texinfo @c -*- texinfo -*- + +@settitle Platform Specific Information +@titlepage +@center @titlefont{Platform Specific Information} +@end titlepage + +@top + +@contents + +@chapter Unix-like + +Some parts of FFmpeg cannot be built with version 2.15 of the GNU +assembler which is still provided by a few AMD64 distributions. To +make sure your compiler really uses the required version of gas +after a binutils upgrade, run: + +@example +$(gcc -print-prog-name=as) --version +@end example + +If not, then you should install a different compiler that has no +hard-coded path to gas. In the worst case pass @code{--disable-asm} +to configure. + +@section BSD + +BSD make will not build FFmpeg, you need to install and use GNU Make +(@command{gmake}). + +@section (Open)Solaris + +GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}), +standard Solaris Make will not work. When building with a non-c99 front-end +(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o} +or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options +since the libc is not c99-compliant by default. The probes performed by +configure may raise an exception leading to the death of configure itself +due to a bug in the system shell. Simply invoke a different shell such as +bash directly to work around this: + +@example +bash ./configure +@end example + +@anchor{Darwin} +@section Darwin (Mac OS X, iPhone) + +The toolchain provided with Xcode is sufficient to build the basic +unacelerated code. + +Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from +@url{http://github.com/yuvi/gas-preprocessor} to build the optimized +assembler functions. Just download the Perl script and put it somewhere +in your PATH, FFmpeg's configure will pick it up automatically. + +Mac OS X on amd64 and x86 requires @command{yasm} to build most of the +optimized assembler functions. @uref{http://www.finkproject.org/, Fink}, +@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix}, +@uref{http://mxcl.github.com/homebrew/, Homebrew} +or @uref{http://www.macports.org, MacPorts} can easily provide it. + + +@chapter DOS + +Using a cross-compiler is preferred for various reasons. +@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html} + + +@chapter OS/2 + +For information about compiling FFmpeg on OS/2 see +@url{http://www.edm2.com/index.php/FFmpeg}. + + +@chapter Windows + +To get help and instructions for building FFmpeg under Windows, check out +the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}. + +@section Native Windows compilation using MinGW or MinGW-w64 + +FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64 +toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from +@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}. +You can find detailed installation instructions in the download section and +the FAQ. + +Notes: + +@itemize + +@item Building natively using MSYS can be sped up by disabling implicit rules +in the Makefile by calling @code{make -r} instead of plain @code{make}. This +speed up is close to non-existent for normal one-off builds and is only +noticeable when running make for a second time (for example during +@code{make install}). + +@item In order to compile FFplay, you must have the MinGW development library +of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed. + +@item By using @code{./configure --enable-shared} when configuring FFmpeg, +you can build the FFmpeg libraries (e.g. libavutil, libavcodec, +libavformat) as DLLs. + +@end itemize + +@section Microsoft Visual C++ + +FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and +wrapper. + +You will need the following prerequisites: + +@itemize +@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper} +@item @uref{http://code.google.com/p/msinttypes/, msinttypes} +@item @uref{http://www.mingw.org/, MSYS} +@item @uref{http://yasm.tortall.net/, YASM} +@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if +you want to run @uref{fate.html, FATE}. +@end itemize + +To set up a proper MSVC environment in MSYS, you simply need to run +@code{msys.bat} from the Visual Studio command prompt. + +Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe} +somewhere in your @code{PATH}. + +Next, make sure @code{inttypes.h} and any other headers and libs you want to use +are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and +@code{INCLUDE} environment variables to include the @strong{Windows} paths to +these directories. Alternatively, you can try and use the +@code{--extra-cflags}/@code{--extra-ldflags} configure options. + +Finally, run: + +@example +./configure --toolchain=msvc +make +make install +@end example + +If you wish to compile shared libraries, add @code{--enable-shared} to your +configure options. Note that due to the way MSVC handles DLL imports and +exports, you cannot compile static and shared libraries at the same time, and +enabling shared libraries will automatically disable the static ones. + +Notes: + +@itemize + +@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker. +You can find out by running @code{which link} to see which @code{link.exe} you +are using. If it is located at @code{/bin/link.exe}, then you have the wrong one +in your @code{PATH}. Either move or remove that copy, or make sure MSVC's +@code{link.exe} takes precedence in your @code{PATH} over coreutils'. + +@item If you wish to build with zlib support, you will have to grab a compatible +zlib binary from somewhere, with an MSVC import lib, or if you wish to link +statically, you can follow the instructions below to build a compatible +@code{zlib.lib} with MSVC. Regardless of which method you use, you must still +follow step 3, or compilation will fail. +@enumerate +@item Grab the @uref{http://zlib.net/, zlib sources}. +@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since +this is how FFmpeg is built as well. +@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets +erroneously included when building FFmpeg. +@item Run @code{nmake -f win32/Makefile.msc}. +@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC +can see. +@end enumerate + +@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express. +Anything else is not officially supported. + +@end itemize + +@subsection Linking to FFmpeg with Microsoft Visual C++ + +If you plan to link with MSVC-built static libraries, you will need +to make sure you have @code{Runtime Library} set to +@code{Multi-threaded (/MT)} in your project's settings. + +FFmpeg headers do not declare global data for Windows DLLs through the usual +dllexport/dllimport interface. Such data will be exported properly while +building, but to use them in your MSVC code you will have to edit the +appropriate headers and mark the data as dllimport. For example, in +libavutil/pixdesc.h you should have: +@example +extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[]; +@end example + +You will also need to define @code{inline} to something MSVC understands: +@example +#define inline __inline +@end example + +Also note, that as stated in @strong{Microsoft Visual C++}, you will need +an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}. + +If you plan on using import libraries created by dlltool, you must +set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization +settings, otherwise the resulting binaries will fail during runtime. +This is not required when using import libraries generated by @code{lib.exe}. +This issue is reported upstream at +@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}. + +To create import libraries that work with the @code{/OPT:REF} option +(which is enabled by default in Release mode), follow these steps: + +@enumerate + +@item Open the @emph{Visual Studio Command Prompt}. + +Alternatively, in a normal command line prompt, call @file{vcvars32.bat} +which sets up the environment variables for the Visual C++ tools +(the standard location for this file is something like +@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}). + +@item Enter the @file{bin} directory where the created LIB and DLL files +are stored. + +@item Generate new import libraries with @command{lib.exe}: + +@example +lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib +@end example + +Replace @code{foo-version} and @code{foo} with the respective library names. + +@end enumerate + +@anchor{Cross compilation for Windows with Linux} +@section Cross compilation for Windows with Linux + +You must use the MinGW cross compilation tools available at +@url{http://www.mingw.org/}. + +Then configure FFmpeg with the following options: +@example +./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc- +@end example +(you can change the cross-prefix according to the prefix chosen for the +MinGW tools). + +Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}. + +@section Compilation under Cygwin + +Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack +llrint() in its C library. + +Install your Cygwin with all the "Base" packages, plus the +following "Devel" ones: +@example +binutils, gcc4-core, make, git, mingw-runtime, texi2html +@end example + +In order to run FATE you will also need the following "Utils" packages: +@example +bc, diffutils +@end example + +If you want to build FFmpeg with additional libraries, download Cygwin +"Devel" packages for Ogg and Vorbis from any Cygwin packages repository: +@example +libogg-devel, libvorbis-devel +@end example + +These library packages are only available from +@uref{http://sourceware.org/cygwinports/, Cygwin Ports}: + +@example +yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel, +libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel +@end example + +The recommendation for x264 is to build it from source, as it evolves too +quickly for Cygwin Ports to be up to date. + +@section Crosscompilation for Windows under Cygwin + +With Cygwin you can create Windows binaries that do not need the cygwin1.dll. + +Just install your Cygwin as explained before, plus these additional +"Devel" packages: +@example +gcc-mingw-core, mingw-runtime, mingw-zlib +@end example + +and add some special flags to your configure invocation. + +For a static build run +@example +./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin +@end example + +and for a build with shared libraries +@example +./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin +@end example + +@chapter Plan 9 + +The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler +does not implement all the C99 features needed by FFmpeg so the gcc +port must be used. Furthermore, a few items missing from the C +library and shell environment need to be fixed. + +@itemize + +@item GNU awk, grep, make, and sed + +Working packages of these tools can be found at +@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}. +They can be installed with @uref{http://9front.org/, 9front's} @code{pkg} +utility by setting @code{pkgpath} to +@code{http://ports2plan9.googlecode.com/files/}. + +@item Missing/broken @code{head} and @code{printf} commands + +Replacements adequate for building FFmpeg can be found in the +@code{compat/plan9} directory. Place these somewhere they will be +found by the shell. These are not full implementations of the +commands and are @emph{not} suitable for general use. + +@item Missing C99 @code{stdint.h} and @code{inttypes.h} + +Replacement headers are available from +@url{http://code.google.com/p/plan9front/issues/detail?id=152}. + +@item Missing or non-standard library functions + +Some functions in the C library are missing or incomplete. The +@code{@uref{http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz, +gcc-apelibs-1207}} package from +@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9} +includes an updated C library, but installing the full package gives +unusable executables. Instead, keep the files from @code{gccbin.tgz} +under @code{/386/lib/gnu}. From the @code{libc.a} archive in the +@code{gcc-apelibs-1207} package, extract the following object files and +turn them into a library: + +@itemize +@item @code{strerror.o} +@item @code{strtoll.o} +@item @code{snprintf.o} +@item @code{vsnprintf.o} +@item @code{vfprintf.o} +@item @code{_IO_getc.o} +@item @code{_IO_putc.o} +@end itemize + +Use the @code{--extra-libs} option of @code{configure} to inform the +build system of this library. + +@item FPU exceptions enabled by default + +Unlike most other systems, Plan 9 enables FPU exceptions by default. +These must be disabled before calling any FFmpeg functions. While the +included tools will do this automatically, other users of the +libraries must do it themselves. + +@end itemize + +@bye diff --git a/ffmpeg1/doc/print_options.c b/ffmpeg1/doc/print_options.c new file mode 100644 index 0000000..c369cfd --- /dev/null +++ b/ffmpeg1/doc/print_options.c @@ -0,0 +1,128 @@ +/* + * Copyright (c) 2012 Anton Khirnov + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/* + * generate texinfo manpages for avoptions + */ + +#include <stddef.h> +#include <string.h> +#include <float.h> + +#include "libavformat/avformat.h" +#include "libavcodec/avcodec.h" +#include "libavutil/opt.h" + +static void print_usage(void) +{ + fprintf(stderr, "Usage: enum_options type\n" + "type: format codec\n"); + exit(1); +} + +static void print_option(const AVOption *opts, const AVOption *o, int per_stream) +{ + if (!(o->flags & (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM))) + return; + + printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : ""); + switch (o->type) { + case AV_OPT_TYPE_BINARY: printf("hexadecimal string"); break; + case AV_OPT_TYPE_STRING: printf("string"); break; + case AV_OPT_TYPE_INT: + case AV_OPT_TYPE_INT64: printf("integer"); break; + case AV_OPT_TYPE_FLOAT: + case AV_OPT_TYPE_DOUBLE: printf("float"); break; + case AV_OPT_TYPE_RATIONAL: printf("rational number"); break; + case AV_OPT_TYPE_FLAGS: printf("flags"); break; + default: printf("value"); break; + } + printf("} (@emph{"); + + if (o->flags & AV_OPT_FLAG_DECODING_PARAM) { + printf("input"); + if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) + printf("/"); + } + if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output"); + if (o->flags & AV_OPT_FLAG_AUDIO_PARAM) printf(",audio"); + if (o->flags & AV_OPT_FLAG_VIDEO_PARAM) printf(",video"); + if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles"); + + printf("})\n"); + if (o->help) + printf("%s\n", o->help); + + if (o->unit) { + const AVOption *u; + printf("\nPossible values:\n@table @samp\n"); + + for (u = opts; u->name; u++) { + if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit)) + printf("@item %s\n%s\n", u->name, u->help ? u->help : ""); + } + printf("@end table\n"); + } +} + +static void show_opts(const AVOption *opts, int per_stream) +{ + const AVOption *o; + + printf("@table @option\n"); + for (o = opts; o->name; o++) { + if (o->type != AV_OPT_TYPE_CONST) + print_option(opts, o, per_stream); + } + printf("@end table\n"); +} + +static void show_format_opts(void) +{ +#include "libavformat/options_table.h" + + printf("@section Format AVOptions\n"); + show_opts(options, 0); +} + +static void show_codec_opts(void) +{ +#include "libavcodec/options_table.h" + + printf("@section Codec AVOptions\n"); + show_opts(options, 1); +} + +int main(int argc, char **argv) +{ + if (argc < 2) + print_usage(); + + printf("@c DO NOT EDIT THIS FILE!\n" + "@c It was generated by print_options.\n\n"); + if (!strcmp(argv[1], "format")) + show_format_opts(); + else if (!strcmp(argv[1], "codec")) + show_codec_opts(); + else + print_usage(); + + return 0; +} diff --git a/ffmpeg1/doc/protocols.texi b/ffmpeg1/doc/protocols.texi new file mode 100644 index 0000000..9940b67 --- /dev/null +++ b/ffmpeg1/doc/protocols.texi @@ -0,0 +1,790 @@ +@chapter Protocols +@c man begin PROTOCOLS + +Protocols are configured elements in FFmpeg which allow to access +resources which require the use of a particular protocol. + +When you configure your FFmpeg build, all the supported protocols are +enabled by default. You can list all available ones using the +configure option "--list-protocols". + +You can disable all the protocols using the configure option +"--disable-protocols", and selectively enable a protocol using the +option "--enable-protocol=@var{PROTOCOL}", or you can disable a +particular protocol using the option +"--disable-protocol=@var{PROTOCOL}". + +The option "-protocols" of the ff* tools will display the list of +supported protocols. + +A description of the currently available protocols follows. + +@section bluray + +Read BluRay playlist. + +The accepted options are: +@table @option + +@item angle +BluRay angle + +@item chapter +Start chapter (1...N) + +@item playlist +Playlist to read (BDMV/PLAYLIST/?????.mpls) + +@end table + +Examples: + +Read longest playlist from BluRay mounted to /mnt/bluray: +@example +bluray:/mnt/bluray +@end example + +Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: +@example +-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray +@end example + +@section concat + +Physical concatenation protocol. + +Allow to read and seek from many resource in sequence as if they were +a unique resource. + +A URL accepted by this protocol has the syntax: +@example +concat:@var{URL1}|@var{URL2}|...|@var{URLN} +@end example + +where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the +resource to be concatenated, each one possibly specifying a distinct +protocol. + +For example to read a sequence of files @file{split1.mpeg}, +@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the +command: +@example +ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg +@end example + +Note that you may need to escape the character "|" which is special for +many shells. + +@section data + +Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}. + +For example, to convert a GIF file given inline with @command{ffmpeg}: +@example +ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png +@end example + +@section file + +File access protocol. + +Allow to read from or read to a file. + +For example to read from a file @file{input.mpeg} with @command{ffmpeg} +use the command: +@example +ffmpeg -i file:input.mpeg output.mpeg +@end example + +The ff* tools default to the file protocol, that is a resource +specified with the name "FILE.mpeg" is interpreted as the URL +"file:FILE.mpeg". + +@section gopher + +Gopher protocol. + +@section hls + +Read Apple HTTP Live Streaming compliant segmented stream as +a uniform one. The M3U8 playlists describing the segments can be +remote HTTP resources or local files, accessed using the standard +file protocol. +The nested protocol is declared by specifying +"+@var{proto}" after the hls URI scheme name, where @var{proto} +is either "file" or "http". + +@example +hls+http://host/path/to/remote/resource.m3u8 +hls+file://path/to/local/resource.m3u8 +@end example + +Using this protocol is discouraged - the hls demuxer should work +just as well (if not, please report the issues) and is more complete. +To use the hls demuxer instead, simply use the direct URLs to the +m3u8 files. + +@section http + +HTTP (Hyper Text Transfer Protocol). + +This protocol accepts the following options. + +@table @option +@item seekable +Control seekability of connection. If set to 1 the resource is +supposed to be seekable, if set to 0 it is assumed not to be seekable, +if set to -1 it will try to autodetect if it is seekable. Default +value is -1. + +@item chunked_post +If set to 1 use chunked transfer-encoding for posts, default is 1. + +@item headers +Set custom HTTP headers, can override built in default headers. The +value must be a string encoding the headers. + +@item content_type +Force a content type. + +@item user-agent +Override User-Agent header. If not specified the protocol will use a +string describing the libavformat build. + +@item multiple_requests +Use persistent connections if set to 1. By default it is 0. + +@item post_data +Set custom HTTP post data. + +@item timeout +Set timeout of socket I/O operations used by the underlying low level +operation. By default it is set to -1, which means that the timeout is +not specified. + +@item mime_type +Set MIME type. + +@item cookies +Set the cookies to be sent in future requests. The format of each cookie is the +same as the value of a Set-Cookie HTTP response field. Multiple cookies can be +delimited by a newline character. +@end table + +@subsection HTTP Cookies + +Some HTTP requests will be denied unless cookie values are passed in with the +request. The @option{cookies} option allows these cookies to be specified. At +the very least, each cookie must specify a value along with a path and domain. +HTTP requests that match both the domain and path will automatically include the +cookie value in the HTTP Cookie header field. Multiple cookies can be delimited +by a newline. + +The required syntax to play a stream specifying a cookie is: +@example +ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8 +@end example + +@section mmst + +MMS (Microsoft Media Server) protocol over TCP. + +@section mmsh + +MMS (Microsoft Media Server) protocol over HTTP. + +The required syntax is: +@example +mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] +@end example + +@section md5 + +MD5 output protocol. + +Computes the MD5 hash of the data to be written, and on close writes +this to the designated output or stdout if none is specified. It can +be used to test muxers without writing an actual file. + +Some examples follow. +@example +# Write the MD5 hash of the encoded AVI file to the file output.avi.md5. +ffmpeg -i input.flv -f avi -y md5:output.avi.md5 + +# Write the MD5 hash of the encoded AVI file to stdout. +ffmpeg -i input.flv -f avi -y md5: +@end example + +Note that some formats (typically MOV) require the output protocol to +be seekable, so they will fail with the MD5 output protocol. + +@section pipe + +UNIX pipe access protocol. + +Allow to read and write from UNIX pipes. + +The accepted syntax is: +@example +pipe:[@var{number}] +@end example + +@var{number} is the number corresponding to the file descriptor of the +pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number} +is not specified, by default the stdout file descriptor will be used +for writing, stdin for reading. + +For example to read from stdin with @command{ffmpeg}: +@example +cat test.wav | ffmpeg -i pipe:0 +# ...this is the same as... +cat test.wav | ffmpeg -i pipe: +@end example + +For writing to stdout with @command{ffmpeg}: +@example +ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi +# ...this is the same as... +ffmpeg -i test.wav -f avi pipe: | cat > test.avi +@end example + +Note that some formats (typically MOV), require the output protocol to +be seekable, so they will fail with the pipe output protocol. + +@section rtmp + +Real-Time Messaging Protocol. + +The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia +content across a TCP/IP network. + +The required syntax is: +@example +rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}] +@end example + +The accepted parameters are: +@table @option + +@item server +The address of the RTMP server. + +@item port +The number of the TCP port to use (by default is 1935). + +@item app +It is the name of the application to access. It usually corresponds to +the path where the application is installed on the RTMP server +(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override +the value parsed from the URI through the @code{rtmp_app} option, too. + +@item playpath +It is the path or name of the resource to play with reference to the +application specified in @var{app}, may be prefixed by "mp4:". You +can override the value parsed from the URI through the @code{rtmp_playpath} +option, too. + +@item listen +Act as a server, listening for an incoming connection. + +@item timeout +Maximum time to wait for the incoming connection. Implies listen. +@end table + +Additionally, the following parameters can be set via command line options +(or in code via @code{AVOption}s): +@table @option + +@item rtmp_app +Name of application to connect on the RTMP server. This option +overrides the parameter specified in the URI. + +@item rtmp_buffer +Set the client buffer time in milliseconds. The default is 3000. + +@item rtmp_conn +Extra arbitrary AMF connection parameters, parsed from a string, +e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}. +Each value is prefixed by a single character denoting the type, +B for Boolean, N for number, S for string, O for object, or Z for null, +followed by a colon. For Booleans the data must be either 0 or 1 for +FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or +1 to end or begin an object, respectively. Data items in subobjects may +be named, by prefixing the type with 'N' and specifying the name before +the value (i.e. @code{NB:myFlag:1}). This option may be used multiple +times to construct arbitrary AMF sequences. + +@item rtmp_flashver +Version of the Flash plugin used to run the SWF player. The default +is LNX 9,0,124,2. + +@item rtmp_flush_interval +Number of packets flushed in the same request (RTMPT only). The default +is 10. + +@item rtmp_live +Specify that the media is a live stream. No resuming or seeking in +live streams is possible. The default value is @code{any}, which means the +subscriber first tries to play the live stream specified in the +playpath. If a live stream of that name is not found, it plays the +recorded stream. The other possible values are @code{live} and +@code{recorded}. + +@item rtmp_pageurl +URL of the web page in which the media was embedded. By default no +value will be sent. + +@item rtmp_playpath +Stream identifier to play or to publish. This option overrides the +parameter specified in the URI. + +@item rtmp_subscribe +Name of live stream to subscribe to. By default no value will be sent. +It is only sent if the option is specified or if rtmp_live +is set to live. + +@item rtmp_swfhash +SHA256 hash of the decompressed SWF file (32 bytes). + +@item rtmp_swfsize +Size of the decompressed SWF file, required for SWFVerification. + +@item rtmp_swfurl +URL of the SWF player for the media. By default no value will be sent. + +@item rtmp_swfverify +URL to player swf file, compute hash/size automatically. + +@item rtmp_tcurl +URL of the target stream. Defaults to proto://host[:port]/app. + +@end table + +For example to read with @command{ffplay} a multimedia resource named +"sample" from the application "vod" from an RTMP server "myserver": +@example +ffplay rtmp://myserver/vod/sample +@end example + +@section rtmpe + +Encrypted Real-Time Messaging Protocol. + +The Encrypted Real-Time Messaging Protocol (RTMPE) is used for +streaming multimedia content within standard cryptographic primitives, +consisting of Diffie-Hellman key exchange and HMACSHA256, generating +a pair of RC4 keys. + +@section rtmps + +Real-Time Messaging Protocol over a secure SSL connection. + +The Real-Time Messaging Protocol (RTMPS) is used for streaming +multimedia content across an encrypted connection. + +@section rtmpt + +Real-Time Messaging Protocol tunneled through HTTP. + +The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used +for streaming multimedia content within HTTP requests to traverse +firewalls. + +@section rtmpte + +Encrypted Real-Time Messaging Protocol tunneled through HTTP. + +The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) +is used for streaming multimedia content within HTTP requests to traverse +firewalls. + +@section rtmpts + +Real-Time Messaging Protocol tunneled through HTTPS. + +The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used +for streaming multimedia content within HTTPS requests to traverse +firewalls. + +@section rtmp, rtmpe, rtmps, rtmpt, rtmpte + +Real-Time Messaging Protocol and its variants supported through +librtmp. + +Requires the presence of the librtmp headers and library during +configuration. You need to explicitly configure the build with +"--enable-librtmp". If enabled this will replace the native RTMP +protocol. + +This protocol provides most client functions and a few server +functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), +encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled +variants of these encrypted types (RTMPTE, RTMPTS). + +The required syntax is: +@example +@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options} +@end example + +where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe", +"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and +@var{server}, @var{port}, @var{app} and @var{playpath} have the same +meaning as specified for the RTMP native protocol. +@var{options} contains a list of space-separated options of the form +@var{key}=@var{val}. + +See the librtmp manual page (man 3 librtmp) for more information. + +For example, to stream a file in real-time to an RTMP server using +@command{ffmpeg}: +@example +ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream +@end example + +To play the same stream using @command{ffplay}: +@example +ffplay "rtmp://myserver/live/mystream live=1" +@end example + +@section rtp + +Real-Time Protocol. + +@section rtsp + +RTSP is not technically a protocol handler in libavformat, it is a demuxer +and muxer. The demuxer supports both normal RTSP (with data transferred +over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with +data transferred over RDT). + +The muxer can be used to send a stream using RTSP ANNOUNCE to a server +supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's +@uref{http://github.com/revmischa/rtsp-server, RTSP server}). + +The required syntax for a RTSP url is: +@example +rtsp://@var{hostname}[:@var{port}]/@var{path} +@end example + +The following options (set on the @command{ffmpeg}/@command{ffplay} command +line, or set in code via @code{AVOption}s or in @code{avformat_open_input}), +are supported: + +Flags for @code{rtsp_transport}: + +@table @option + +@item udp +Use UDP as lower transport protocol. + +@item tcp +Use TCP (interleaving within the RTSP control channel) as lower +transport protocol. + +@item udp_multicast +Use UDP multicast as lower transport protocol. + +@item http +Use HTTP tunneling as lower transport protocol, which is useful for +passing proxies. +@end table + +Multiple lower transport protocols may be specified, in that case they are +tried one at a time (if the setup of one fails, the next one is tried). +For the muxer, only the @code{tcp} and @code{udp} options are supported. + +Flags for @code{rtsp_flags}: + +@table @option +@item filter_src +Accept packets only from negotiated peer address and port. +@item listen +Act as a server, listening for an incoming connection. +@end table + +When receiving data over UDP, the demuxer tries to reorder received packets +(since they may arrive out of order, or packets may get lost totally). This +can be disabled by setting the maximum demuxing delay to zero (via +the @code{max_delay} field of AVFormatContext). + +When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the +streams to display can be chosen with @code{-vst} @var{n} and +@code{-ast} @var{n} for video and audio respectively, and can be switched +on the fly by pressing @code{v} and @code{a}. + +Example command lines: + +To watch a stream over UDP, with a max reordering delay of 0.5 seconds: + +@example +ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 +@end example + +To watch a stream tunneled over HTTP: + +@example +ffplay -rtsp_transport http rtsp://server/video.mp4 +@end example + +To send a stream in realtime to a RTSP server, for others to watch: + +@example +ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp +@end example + +To receive a stream in realtime: + +@example +ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} +@end example + +@section sap + +Session Announcement Protocol (RFC 2974). This is not technically a +protocol handler in libavformat, it is a muxer and demuxer. +It is used for signalling of RTP streams, by announcing the SDP for the +streams regularly on a separate port. + +@subsection Muxer + +The syntax for a SAP url given to the muxer is: +@example +sap://@var{destination}[:@var{port}][?@var{options}] +@end example + +The RTP packets are sent to @var{destination} on port @var{port}, +or to port 5004 if no port is specified. +@var{options} is a @code{&}-separated list. The following options +are supported: + +@table @option + +@item announce_addr=@var{address} +Specify the destination IP address for sending the announcements to. +If omitted, the announcements are sent to the commonly used SAP +announcement multicast address 224.2.127.254 (sap.mcast.net), or +ff0e::2:7ffe if @var{destination} is an IPv6 address. + +@item announce_port=@var{port} +Specify the port to send the announcements on, defaults to +9875 if not specified. + +@item ttl=@var{ttl} +Specify the time to live value for the announcements and RTP packets, +defaults to 255. + +@item same_port=@var{0|1} +If set to 1, send all RTP streams on the same port pair. If zero (the +default), all streams are sent on unique ports, with each stream on a +port 2 numbers higher than the previous. +VLC/Live555 requires this to be set to 1, to be able to receive the stream. +The RTP stack in libavformat for receiving requires all streams to be sent +on unique ports. +@end table + +Example command lines follow. + +To broadcast a stream on the local subnet, for watching in VLC: + +@example +ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 +@end example + +Similarly, for watching in @command{ffplay}: + +@example +ffmpeg -re -i @var{input} -f sap sap://224.0.0.255 +@end example + +And for watching in @command{ffplay}, over IPv6: + +@example +ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] +@end example + +@subsection Demuxer + +The syntax for a SAP url given to the demuxer is: +@example +sap://[@var{address}][:@var{port}] +@end example + +@var{address} is the multicast address to listen for announcements on, +if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port} +is the port that is listened on, 9875 if omitted. + +The demuxers listens for announcements on the given address and port. +Once an announcement is received, it tries to receive that particular stream. + +Example command lines follow. + +To play back the first stream announced on the normal SAP multicast address: + +@example +ffplay sap:// +@end example + +To play back the first stream announced on one the default IPv6 SAP multicast address: + +@example +ffplay sap://[ff0e::2:7ffe] +@end example + +@section tcp + +Trasmission Control Protocol. + +The required syntax for a TCP url is: +@example +tcp://@var{hostname}:@var{port}[?@var{options}] +@end example + +@table @option + +@item listen +Listen for an incoming connection + +@item timeout=@var{microseconds} +In read mode: if no data arrived in more than this time interval, raise error. +In write mode: if socket cannot be written in more than this time interval, raise error. +This also sets timeout on TCP connection establishing. + +@example +ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen +ffplay tcp://@var{hostname}:@var{port} +@end example + +@end table + +@section tls + +Transport Layer Security/Secure Sockets Layer + +The required syntax for a TLS/SSL url is: +@example +tls://@var{hostname}:@var{port}[?@var{options}] +@end example + +@table @option + +@item listen +Act as a server, listening for an incoming connection. + +@item cafile=@var{filename} +Certificate authority file. The file must be in OpenSSL PEM format. + +@item cert=@var{filename} +Certificate file. The file must be in OpenSSL PEM format. + +@item key=@var{filename} +Private key file. + +@item verify=@var{0|1} +Verify the peer's certificate. + +@end table + +Example command lines: + +To create a TLS/SSL server that serves an input stream. + +@example +ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key} +@end example + +To play back a stream from the TLS/SSL server using @command{ffplay}: + +@example +ffplay tls://@var{hostname}:@var{port} +@end example + +@section udp + +User Datagram Protocol. + +The required syntax for a UDP url is: +@example +udp://@var{hostname}:@var{port}[?@var{options}] +@end example + +@var{options} contains a list of &-separated options of the form @var{key}=@var{val}. + +In case threading is enabled on the system, a circular buffer is used +to store the incoming data, which allows to reduce loss of data due to +UDP socket buffer overruns. The @var{fifo_size} and +@var{overrun_nonfatal} options are related to this buffer. + +The list of supported options follows. + +@table @option + +@item buffer_size=@var{size} +Set the UDP socket buffer size in bytes. This is used both for the +receiving and the sending buffer size. + +@item localport=@var{port} +Override the local UDP port to bind with. + +@item localaddr=@var{addr} +Choose the local IP address. This is useful e.g. if sending multicast +and the host has multiple interfaces, where the user can choose +which interface to send on by specifying the IP address of that interface. + +@item pkt_size=@var{size} +Set the size in bytes of UDP packets. + +@item reuse=@var{1|0} +Explicitly allow or disallow reusing UDP sockets. + +@item ttl=@var{ttl} +Set the time to live value (for multicast only). + +@item connect=@var{1|0} +Initialize the UDP socket with @code{connect()}. In this case, the +destination address can't be changed with ff_udp_set_remote_url later. +If the destination address isn't known at the start, this option can +be specified in ff_udp_set_remote_url, too. +This allows finding out the source address for the packets with getsockname, +and makes writes return with AVERROR(ECONNREFUSED) if "destination +unreachable" is received. +For receiving, this gives the benefit of only receiving packets from +the specified peer address/port. + +@item sources=@var{address}[,@var{address}] +Only receive packets sent to the multicast group from one of the +specified sender IP addresses. + +@item block=@var{address}[,@var{address}] +Ignore packets sent to the multicast group from the specified +sender IP addresses. + +@item fifo_size=@var{units} +Set the UDP receiving circular buffer size, expressed as a number of +packets with size of 188 bytes. If not specified defaults to 7*4096. + +@item overrun_nonfatal=@var{1|0} +Survive in case of UDP receiving circular buffer overrun. Default +value is 0. + +@item timeout=@var{microseconds} +In read mode: if no data arrived in more than this time interval, raise error. +@end table + +Some usage examples of the UDP protocol with @command{ffmpeg} follow. + +To stream over UDP to a remote endpoint: +@example +ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} +@end example + +To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer: +@example +ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 +@end example + +To receive over UDP from a remote endpoint: +@example +ffmpeg -i udp://[@var{multicast-address}]:@var{port} +@end example + +@c man end PROTOCOLS diff --git a/ffmpeg1/doc/rate_distortion.txt b/ffmpeg1/doc/rate_distortion.txt new file mode 100644 index 0000000..e9711c2 --- /dev/null +++ b/ffmpeg1/doc/rate_distortion.txt @@ -0,0 +1,61 @@ +A Quick Description Of Rate Distortion Theory. + +We want to encode a video, picture or piece of music optimally. What does +"optimally" really mean? It means that we want to get the best quality at a +given filesize OR we want to get the smallest filesize at a given quality +(in practice, these 2 goals are usually the same). + +Solving this directly is not practical; trying all byte sequences 1 +megabyte in length and selecting the "best looking" sequence will yield +256^1000000 cases to try. + +But first, a word about quality, which is also called distortion. +Distortion can be quantified by almost any quality measurement one chooses. +Commonly, the sum of squared differences is used but more complex methods +that consider psychovisual effects can be used as well. It makes no +difference in this discussion. + + +First step: that rate distortion factor called lambda... +Let's consider the problem of minimizing: + + distortion + lambda*rate + +rate is the filesize +distortion is the quality +lambda is a fixed value chosen as a tradeoff between quality and filesize +Is this equivalent to finding the best quality for a given max +filesize? The answer is yes. For each filesize limit there is some lambda +factor for which minimizing above will get you the best quality (using your +chosen quality measurement) at the desired (or lower) filesize. + + +Second step: splitting the problem. +Directly splitting the problem of finding the best quality at a given +filesize is hard because we do not know how many bits from the total +filesize should be allocated to each of the subproblems. But the formula +from above: + + distortion + lambda*rate + +can be trivially split. Consider: + + (distortion0 + distortion1) + lambda*(rate0 + rate1) + +This creates a problem made of 2 independent subproblems. The subproblems +might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize: + + (distortion0 + distortion1) + lambda*(rate0 + rate1) + +we just have to minimize: + + distortion0 + lambda*rate0 + +and + + distortion1 + lambda*rate1 + +I.e, the 2 problems can be solved independently. + +Author: Michael Niedermayer +Copyright: LGPL diff --git a/ffmpeg1/doc/snow.txt b/ffmpeg1/doc/snow.txt new file mode 100644 index 0000000..f991339 --- /dev/null +++ b/ffmpeg1/doc/snow.txt @@ -0,0 +1,630 @@ +============================================= +Snow Video Codec Specification Draft 20080110 +============================================= + +Introduction: +============= +This specification describes the Snow bitstream syntax and semantics as +well as the formal Snow decoding process. + +The decoding process is described precisely and any compliant decoder +MUST produce the exact same output for a spec-conformant Snow stream. +For encoding, though, any process which generates a stream compliant to +the syntactical and semantic requirements and which is decodable by +the process described in this spec shall be considered a conformant +Snow encoder. + +Definitions: +============ + +MUST the specific part must be done to conform to this standard +SHOULD it is recommended to be done that way, but not strictly required + +ilog2(x) is the rounded down logarithm of x with basis 2 +ilog2(0) = 0 + +Type definitions: +================= + +b 1-bit range coded +u unsigned scalar value range coded +s signed scalar value range coded + + +Bitstream syntax: +================= + +frame: + header + prediction + residual + +header: + keyframe b MID_STATE + if(keyframe || always_reset) + reset_contexts + if(keyframe){ + version u header_state + always_reset b header_state + temporal_decomposition_type u header_state + temporal_decomposition_count u header_state + spatial_decomposition_count u header_state + colorspace_type u header_state + chroma_h_shift u header_state + chroma_v_shift u header_state + spatial_scalability b header_state + max_ref_frames-1 u header_state + qlogs + } + if(!keyframe){ + update_mc b header_state + if(update_mc){ + for(plane=0; plane<2; plane++){ + diag_mc b header_state + htaps/2-1 u header_state + for(i= p->htaps/2; i; i--) + |hcoeff[i]| u header_state + } + } + update_qlogs b header_state + if(update_qlogs){ + spatial_decomposition_count u header_state + qlogs + } + } + + spatial_decomposition_type s header_state + qlog s header_state + mv_scale s header_state + qbias s header_state + block_max_depth s header_state + +qlogs: + for(plane=0; plane<2; plane++){ + quant_table[plane][0][0] s header_state + for(level=0; level < spatial_decomposition_count; level++){ + quant_table[plane][level][1]s header_state + quant_table[plane][level][3]s header_state + } + } + +reset_contexts + *_state[*]= MID_STATE + +prediction: + for(y=0; y<block_count_vertical; y++) + for(x=0; x<block_count_horizontal; x++) + block(0) + +block(level): + mvx_diff=mvy_diff=y_diff=cb_diff=cr_diff=0 + if(keyframe){ + intra=1 + }else{ + if(level!=max_block_depth){ + s_context= 2*left->level + 2*top->level + topleft->level + topright->level + leaf b block_state[4 + s_context] + } + if(level==max_block_depth || leaf){ + intra b block_state[1 + left->intra + top->intra] + if(intra){ + y_diff s block_state[32] + cb_diff s block_state[64] + cr_diff s block_state[96] + }else{ + ref_context= ilog2(2*left->ref) + ilog2(2*top->ref) + if(ref_frames > 1) + ref u block_state[128 + 1024 + 32*ref_context] + mx_context= ilog2(2*abs(left->mx - top->mx)) + my_context= ilog2(2*abs(left->my - top->my)) + mvx_diff s block_state[128 + 32*(mx_context + 16*!!ref)] + mvy_diff s block_state[128 + 32*(my_context + 16*!!ref)] + } + }else{ + block(level+1) + block(level+1) + block(level+1) + block(level+1) + } + } + + +residual: + residual2(luma) + residual2(chroma_cr) + residual2(chroma_cb) + +residual2: + for(level=0; level<spatial_decomposition_count; level++){ + if(level==0) + subband(LL, 0) + subband(HL, level) + subband(LH, level) + subband(HH, level) + } + +subband: + FIXME + + + +Tag description: +---------------- + +version + 0 + this MUST NOT change within a bitstream + +always_reset + if 1 then the range coder contexts will be reset after each frame + +temporal_decomposition_type + 0 + +temporal_decomposition_count + 0 + +spatial_decomposition_count + FIXME + +colorspace_type + 0 + this MUST NOT change within a bitstream + +chroma_h_shift + log2(luma.width / chroma.width) + this MUST NOT change within a bitstream + +chroma_v_shift + log2(luma.height / chroma.height) + this MUST NOT change within a bitstream + +spatial_scalability + 0 + +max_ref_frames + maximum number of reference frames + this MUST NOT change within a bitstream + +update_mc + indicates that motion compensation filter parameters are stored in the + header + +diag_mc + flag to enable faster diagonal interpolation + this SHOULD be 1 unless it turns out to be covered by a valid patent + +htaps + number of half pel interpolation filter taps, MUST be even, >0 and <10 + +hcoeff + half pel interpolation filter coefficients, hcoeff[0] are the 2 middle + coefficients [1] are the next outer ones and so on, resulting in a filter + like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... + the sign of the coefficients is not explicitly stored but alternates + after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... + hcoeff[0] is not explicitly stored but found by subtracting the sum + of all stored coefficients with signs from 32 + hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ... + a good choice for hcoeff and htaps is + htaps= 6 + hcoeff={40,-10,2} + an alternative which requires more computations at both encoder and + decoder side and may or may not be better is + htaps= 8 + hcoeff={42,-14,6,-2} + + +ref_frames + minimum of the number of available reference frames and max_ref_frames + for example the first frame after a key frame always has ref_frames=1 + +spatial_decomposition_type + wavelet type + 0 is a 9/7 symmetric compact integer wavelet + 1 is a 5/3 symmetric compact integer wavelet + others are reserved + stored as delta from last, last is reset to 0 if always_reset || keyframe + +qlog + quality (logarthmic quantizer scale) + stored as delta from last, last is reset to 0 if always_reset || keyframe + +mv_scale + stored as delta from last, last is reset to 0 if always_reset || keyframe + FIXME check that everything works fine if this changes between frames + +qbias + dequantization bias + stored as delta from last, last is reset to 0 if always_reset || keyframe + +block_max_depth + maximum depth of the block tree + stored as delta from last, last is reset to 0 if always_reset || keyframe + +quant_table + quantiztation table + + +Highlevel bitstream structure: +============================= + -------------------------------------------- +| Header | + -------------------------------------------- +| ------------------------------------ | +| | Block0 | | +| | split? | | +| | yes no | | +| | ......... intra? | | +| | : Block01 : yes no | | +| | : Block02 : ....... .......... | | +| | : Block03 : : y DC : : ref index: | | +| | : Block04 : : cb DC : : motion x : | | +| | ......... : cr DC : : motion y : | | +| | ....... .......... | | +| ------------------------------------ | +| ------------------------------------ | +| | Block1 | | +| ... | + -------------------------------------------- +| ------------ ------------ ------------ | +|| Y subbands | | Cb subbands| | Cr subbands|| +|| --- --- | | --- --- | | --- --- || +|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| || +|| ... | | ... | | ... || +| ------------ ------------ ------------ | + -------------------------------------------- + +Decoding process: +================= + + ------------ + | | + | Subbands | + ------------ | | + | | ------------ + | Intra DC | | + | | LL0 subband prediction + ------------ | + \ Dequantizaton + ------------------- \ | +| Reference frames | \ IDWT +| ------- ------- | Motion \ | +||Frame 0| |Frame 1|| Compensation . OBMC v ------- +| ------- ------- | --------------. \------> + --->|Frame n|-->output +| ------- ------- | ------- +||Frame 2| |Frame 3||<----------------------------------/ +| ... | + ------------------- + + +Range Coder: +============ + +Binary Range Coder: +------------------- +The implemented range coder is an adapted version based upon "Range encoding: +an algorithm for removing redundancy from a digitised message." by G. N. N. +Martin. +The symbols encoded by the Snow range coder are bits (0|1). The +associated probabilities are not fix but change depending on the symbol mix +seen so far. + + +bit seen | new state +---------+----------------------------------------------- + 0 | 256 - state_transition_table[256 - old_state]; + 1 | state_transition_table[ old_state]; + +state_transition_table = { + 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, + 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, + 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, + 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, + 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, + 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, +104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, +119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, +134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, +150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, +165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, +180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, +195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, +210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, +226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, +241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0}; + +FIXME + + +Range Coding of integers: +------------------------- +FIXME + + +Neighboring Blocks: +=================== +left and top are set to the respective blocks unless they are outside of +the image in which case they are set to the Null block + +top-left is set to the top left block unless it is outside of the image in +which case it is set to the left block + +if this block has no larger parent block or it is at the left side of its +parent block and the top right block is not outside of the image then the +top right block is used for top-right else the top-left block is used + +Null block +y,cb,cr are 128 +level, ref, mx and my are 0 + + +Motion Vector Prediction: +========================= +1. the motion vectors of all the neighboring blocks are scaled to +compensate for the difference of reference frames + +scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8 + +2. the median of the scaled left, top and top-right vectors is used as +motion vector prediction + +3. the used motion vector is the sum of the predictor and + (mvx_diff, mvy_diff)*mv_scale + + +Intra DC Predicton: +====================== +the luma and chroma values of the left block are used as predictors + +the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff +to reverse this in the decoder apply the following: +block[y][x].dc[0] = block[y][x-1].dc[0] + y_diff; +block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff; +block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff; +block[*][-1].dc[*]= 128; + + +Motion Compensation: +==================== + +Halfpel interpolation: +---------------------- +halfpel interpolation is done by convolution with the halfpel filter stored +in the header: + +horizontal halfpel samples are found by +H1[y][x] = hcoeff[0]*(F[y][x ] + F[y][x+1]) + + hcoeff[1]*(F[y][x-1] + F[y][x+2]) + + hcoeff[2]*(F[y][x-2] + F[y][x+3]) + + ... +h1[y][x] = (H1[y][x] + 32)>>6; + +vertical halfpel samples are found by +H2[y][x] = hcoeff[0]*(F[y ][x] + F[y+1][x]) + + hcoeff[1]*(F[y-1][x] + F[y+2][x]) + + ... +h2[y][x] = (H2[y][x] + 32)>>6; + +vertical+horizontal halfpel samples are found by +H3[y][x] = hcoeff[0]*(H2[y][x ] + H2[y][x+1]) + + hcoeff[1]*(H2[y][x-1] + H2[y][x+2]) + + ... +H3[y][x] = hcoeff[0]*(H1[y ][x] + H1[y+1][x]) + + hcoeff[1]*(H1[y+1][x] + H1[y+2][x]) + + ... +h3[y][x] = (H3[y][x] + 2048)>>12; + + + F H1 F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F-------F-------F-> H1<-F-------F-------F + v v v + H2 H3 H2 + ^ ^ ^ + F-------F-------F-> H1<-F-------F-------F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F H1 F + + +unavailable fullpel samples (outside the picture for example) shall be equal +to the closest available fullpel sample + + +Smaller pel interpolation: +-------------------------- +if diag_mc is set then points which lie on a line between 2 vertically, +horiziontally or diagonally adjacent halfpel points shall be interpolated +linearls with rounding to nearest and halfway values rounded up. +points which lie on 2 diagonals at the same time should only use the one +diagonal not containing the fullpel point + + + + F-->O---q---O<--h1->O---q---O<--F + v \ / v \ / v + O O O O O O O + | / | \ | + q q q q q + | / | \ | + O O O O O O O + ^ / \ ^ / \ ^ + h2-->O---q---O<--h3->O---q---O<--h2 + v \ / v \ / v + O O O O O O O + | \ | / | + q q q q q + | \ | / | + O O O O O O O + ^ / \ ^ / \ ^ + F-->O---q---O<--h1->O---q---O<--F + + + +the remaining points shall be bilinearly interpolated from the +up to 4 surrounding halfpel and fullpel points, again rounding should be to +nearest and halfway values rounded up + +compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma +interpolation at least + + +Overlapped block motion compensation: +------------------------------------- +FIXME + +LL band prediction: +=================== +Each sample in the LL0 subband is predicted by the median of the left, top and +left+top-topleft samples, samples outside the subband shall be considered to +be 0. To reverse this prediction in the decoder apply the following. +for(y=0; y<height; y++){ + for(x=0; x<width; x++){ + sample[y][x] += median(sample[y-1][x], + sample[y][x-1], + sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]); + } +} +sample[-1][*]=sample[*][-1]= 0; +width,height here are the width and height of the LL0 subband not of the final +video + + +Dequantizaton: +============== +FIXME + +Wavelet Transform: +================== + +Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integer +transform and a integer approximation of the symmetric biorthogonal 9/7 +daubechies wavelet. + +2D IDWT (inverse discrete wavelet transform) +-------------------------------------------- +The 2D IDWT applies a 2D filter recursively, each time combining the +4 lowest frequency subbands into a single subband until only 1 subband +remains. +The 2D filter is done by first applying a 1D filter in the vertical direction +and then applying it in the horizontal one. + --------------- --------------- --------------- --------------- +|LL0|HL0| | | | | | | | | | | | +|---+---| HL1 | | L0|H0 | HL1 | | LL1 | HL1 | | | | +|LH0|HH0| | | | | | | | | | | | +|-------+-------|->|-------+-------|->|-------+-------|->| L1 | H1 |->... +| | | | | | | | | | | | +| LH1 | HH1 | | LH1 | HH1 | | LH1 | HH1 | | | | +| | | | | | | | | | | | + --------------- --------------- --------------- --------------- + + +1D Filter: +---------- +1. interleave the samples of the low and high frequency subbands like +s={L0, H0, L1, H1, L2, H2, L3, H3, ... } +note, this can end with a L or a H, the number of elements shall be w +s[-1] shall be considered equivalent to s[1 ] +s[w ] shall be considered equivalent to s[w-2] + +2. perform the lifting steps in order as described below + +5/3 Integer filter: +1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w +2. s[i] += (s[i-1] + s[i+1] )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -1/4 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +1/2 + + +Snow's 9/7 Integer filter: +1. s[i] -= (3*(s[i-1] + s[i+1]) + 4)>>3; for all even i < w +2. s[i] -= s[i-1] + s[i+1] ; for all odd i < w +3. s[i] += ( s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w +4. s[i] += (3*(s[i-1] + s[i+1]) )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -3/8 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + (| + (| + (| + (| + -1 +\ + /|\ + /|\ + /|\ + /|\ +1/4 + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | +1/16 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +3/2 + +optimization tips: +following are exactly identical +(3a)>>1 == a + (a>>1) +(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2 + +16bit implementation note: +The IDWT can be implemented with 16bits, but this requires some care to +prevent overflows, the following list, lists the minimum number of bits needed +for some terms +1. lifting step +A= s[i-1] + s[i+1] 16bit +3*A + 4 18bit +A + (A>>1) + 2 17bit + +3. lifting step +s[i-1] + s[i+1] 17bit + +4. lifiting step +3*(s[i-1] + s[i+1]) 17bit + + +TODO: +===== +Important: +finetune initial contexts +flip wavelet? +try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients +try the MV length as context for coding the residual coefficients +use extradata for stuff which is in the keyframes now? +the MV median predictor is patented IIRC +implement per picture halfpel interpolation +try different range coder state transition tables for different contexts + +Not Important: +compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality) +spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later) + + +Credits: +======== +Michael Niedermayer +Loren Merritt + + +Copyright: +========== +GPL + GFDL + whatever is needed to make this a RFC diff --git a/ffmpeg1/doc/soc.txt b/ffmpeg1/doc/soc.txt new file mode 100644 index 0000000..2504dba --- /dev/null +++ b/ffmpeg1/doc/soc.txt @@ -0,0 +1,24 @@ +Google Summer of Code and similar project guidelines + +Summer of Code is a project by Google in which students are paid to implement +some nice new features for various participating open source projects ... + +This text is a collection of things to take care of for the next soc as +it's a little late for this year's soc (2006). + +The Goal: +Our goal in respect to soc is and must be of course exactly one thing and +that is to improve FFmpeg, to reach this goal, code must +* conform to the development policy and patch submission guidelines +* must improve FFmpeg somehow (faster, smaller, "better", + more codecs supported, fewer bugs, cleaner, ...) + +for mentors and other developers to help students to reach that goal it is +essential that changes to their codebase are publicly visible, clean and +easy reviewable that again leads us to: +* use of a revision control system like git +* separation of cosmetic from non-cosmetic changes (this is almost entirely + ignored by mentors and students in soc 2006 which might lead to a surprise + when the code will be reviewed at the end before a possible inclusion in + FFmpeg, individual changes were generally not reviewable due to cosmetics). +* frequent commits, so that comments can be provided early diff --git a/ffmpeg1/doc/swresample.txt b/ffmpeg1/doc/swresample.txt new file mode 100644 index 0000000..2d192a3 --- /dev/null +++ b/ffmpeg1/doc/swresample.txt @@ -0,0 +1,46 @@ + The official guide to swresample for confused developers. + ========================================================= + +Current (simplified) Architecture: +--------------------------------- + Input + v + __________________/|\___________ + / | \ + / input sample format convert v + / | ___________/ + | |/ + | v + | ___________/|\___________ _____________ + | / | \ | | + | Rematrix | resample <---->| Buffers | + | \___________ | ___________/ |_____________| + v \|/ +Special Converter v + v ___________/|\___________ _____________ + | / | \ | | + | Rematrix | resample <---->| Buffers | + | \___________ | ___________/ |_____________| + | \|/ + | v + | |\___________ + \ | \ + \ output sample format convert v + \_________________ | ___________/ + \|/ + v + Output + +Planar/Packed conversion is done when needed during sample format conversion. +Every step can be skipped without memcpy when it is not needed. +Either Resampling and Rematrixing can be performed first depending on which +way it is faster. +The Buffers are needed for resampling due to resamplng being a process that +requires future and past data, it thus also introduces inevitably a delay when +used. +Internally 32bit float and 16bit int is supported currently, other formats can +easily be added. +Externally all sample formats in packed and planar configuration are supported +It's also trivial to add special converters for common cases. +If only sample format and/or packed/planar conversion is needed, it +is performed from input to output directly in a single pass with no intermediates. diff --git a/ffmpeg1/doc/swscale.txt b/ffmpeg1/doc/swscale.txt new file mode 100644 index 0000000..2066009 --- /dev/null +++ b/ffmpeg1/doc/swscale.txt @@ -0,0 +1,98 @@ + The official guide to swscale for confused developers. + ======================================================== + +Current (simplified) Architecture: +--------------------------------- + Input + v + _______OR_________ + / \ + / \ + special converter [Input to YUV converter] + | | + | (8bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 ) + | | + | v + | Horizontal scaler + | | + | (15bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 ) + | | + | v + | Vertical scaler and output converter + | | + v v + output + + +Swscale has 2 scaler paths. Each side must be capable of handling +slices, that is, consecutive non-overlapping rectangles of dimension +(0,slice_top) - (picture_width, slice_bottom). + +special converter + These generally are unscaled converters of common + formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also + in principle contain scalers optimized for specific common cases. + +Main path + The main path is used when no special converter can be used. The code + is designed as a destination line pull architecture. That is, for each + output line the vertical scaler pulls lines from a ring buffer. When + the ring buffer does not contain the wanted line, then it is pulled from + the input slice through the input converter and horizontal scaler. + The result is also stored in the ring buffer to serve future vertical + scaler requests. + When no more output can be generated because lines from a future slice + would be needed, then all remaining lines in the current slice are + converted, horizontally scaled and put in the ring buffer. + [This is done for luma and chroma, each with possibly different numbers + of lines per picture.] + +Input to YUV Converter + When the input to the main path is not planar 8 bits per component YUV or + 8-bit gray, it is converted to planar 8-bit YUV. Two sets of converters + exist for this currently: One performs horizontal downscaling by 2 + before the conversion, the other leaves the full chroma resolution, + but is slightly slower. The scaler will try to preserve full chroma + when the output uses it. It is possible to force full chroma with + SWS_FULL_CHR_H_INP even for cases where the scaler thinks it is useless. + +Horizontal scaler + There are several horizontal scalers. A special case worth mentioning is + the fast bilinear scaler that is made of runtime-generated MMXEXT code + using specially tuned pshufw instructions. + The remaining scalers are specially-tuned for various filter lengths. + They scale 8-bit unsigned planar data to 16-bit signed planar data. + Future >8 bits per component inputs will need to add a new horizontal + scaler that preserves the input precision. + +Vertical scaler and output converter + There is a large number of combined vertical scalers + output converters. + Some are: + * unscaled output converters + * unscaled output converters that average 2 chroma lines + * bilinear converters (C, MMX and accurate MMX) + * arbitrary filter length converters (C, MMX and accurate MMX) + And + * Plain C 8-bit 4:2:2 YUV -> RGB converters using LUTs + * Plain C 17-bit 4:4:4 YUV -> RGB converters using multiplies + * MMX 11-bit 4:2:2 YUV -> RGB converters + * Plain C 16-bit Y -> 16-bit gray + ... + + RGB with less than 8 bits per component uses dither to improve the + subjective quality and low-frequency accuracy. + + +Filter coefficients: +-------------------- +There are several different scalers (bilinear, bicubic, lanczos, area, +sinc, ...). Their coefficients are calculated in initFilter(). +Horizontal filter coefficients have a 1.0 point at 1 << 14, vertical ones at +1 << 12. The 1.0 points have been chosen to maximize precision while leaving +a little headroom for convolutional filters like sharpening filters and +minimizing SIMD instructions needed to apply them. +It would be trivial to use a different 1.0 point if some specific scaler +would benefit from it. +Also, as already hinted at, initFilter() accepts an optional convolutional +filter as input that can be used for contrast, saturation, blur, sharpening +shift, chroma vs. luma shift, ... diff --git a/ffmpeg1/doc/syntax.texi b/ffmpeg1/doc/syntax.texi new file mode 100644 index 0000000..af22d6c --- /dev/null +++ b/ffmpeg1/doc/syntax.texi @@ -0,0 +1,258 @@ +@chapter Syntax +@c man begin SYNTAX + +This section documents the syntax and formats employed by the FFmpeg +libraries and tools. + +@anchor{quoting_and_escaping} +@section Quoting and escaping + +FFmpeg adopts the following quoting and escaping mechanism, unless +explicitly specified. The following rules are applied: + +@itemize +@item +@code{'} and @code{\} are special characters (respectively used for +quoting and escaping). In addition to them, there might be other +special characters depending on the specific syntax where the escaping +and quoting are employed. + +@item +A special character is escaped by prefixing it with a '\'. + +@item +All characters enclosed between '' are included literally in the +parsed string. The quote character @code{'} itself cannot be quoted, +so you may need to close the quote and escape it. + +@item +Leading and trailing whitespaces, unless escaped or quoted, are +removed from the parsed string. +@end itemize + +Note that you may need to add a second level of escaping when using +the command line or a script, which depends on the syntax of the +adopted shell language. + +The function @code{av_get_token} defined in +@file{libavutil/avstring.h} can be used to parse a token quoted or +escaped according to the rules defined above. + +The tool @file{tools/ffescape} in the FFmpeg source tree can be used +to automatically quote or escape a string in a script. + +@subsection Examples + +@itemize +@item +Escape the string @code{Crime d'Amour} containing the @code{'} special +character: +@example +Crime d\'Amour +@end example + +@item +The string above contains a quote, so the @code{'} needs to be escaped +when quoting it: +@example +'Crime d'\''Amour' +@end example + +@item +Include leading or trailing whitespaces using quoting: +@example +' this string starts and ends with whitespaces ' +@end example + +@item +Escaping and quoting can be mixed together: +@example +' The string '\'string\'' is a string ' +@end example + +@item +To include a literal @code{\} you can use either escaping or quoting: +@example +'c:\foo' can be written as c:\\foo +@end example +@end itemize + +@anchor{date syntax} +@section Date + +The accepted syntax is: +@example +[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z] +now +@end example + +If the value is "now" it takes the current time. + +Time is local time unless Z is appended, in which case it is +interpreted as UTC. +If the year-month-day part is not specified it takes the current +year-month-day. + +@anchor{time duration syntax} +@section Time duration + +The accepted syntax is: +@example +[-][HH:]MM:SS[.m...] +[-]S+[.m...] +@end example + +@var{HH} expresses the number of hours, @var{MM} the number a of minutes +and @var{SS} the number of seconds. + +@anchor{video size syntax} +@section Video size +Specify the size of the sourced video, it may be a string of the form +@var{width}x@var{height}, or the name of a size abbreviation. + +The following abbreviations are recognized: +@table @samp +@item ntsc +720x480 +@item pal +720x576 +@item qntsc +352x240 +@item qpal +352x288 +@item sntsc +640x480 +@item spal +768x576 +@item film +352x240 +@item ntsc-film +352x240 +@item sqcif +128x96 +@item qcif +176x144 +@item cif +352x288 +@item 4cif +704x576 +@item 16cif +1408x1152 +@item qqvga +160x120 +@item qvga +320x240 +@item vga +640x480 +@item svga +800x600 +@item xga +1024x768 +@item uxga +1600x1200 +@item qxga +2048x1536 +@item sxga +1280x1024 +@item qsxga +2560x2048 +@item hsxga +5120x4096 +@item wvga +852x480 +@item wxga +1366x768 +@item wsxga +1600x1024 +@item wuxga +1920x1200 +@item woxga +2560x1600 +@item wqsxga +3200x2048 +@item wquxga +3840x2400 +@item whsxga +6400x4096 +@item whuxga +7680x4800 +@item cga +320x200 +@item ega +640x350 +@item hd480 +852x480 +@item hd720 +1280x720 +@item hd1080 +1920x1080 +@item 2k +2048x1080 +@item 2kflat +1998x1080 +@item 2kscope +2048x858 +@item 4k +4096x2160 +@item 4kflat +3996x2160 +@item 4kscope +4096x1716 +@end table + +@anchor{video rate syntax} +@section Video rate + +Specify the frame rate of a video, expressed as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float +number or a valid video frame rate abbreviation. + +The following abbreviations are recognized: +@table @samp +@item ntsc +30000/1001 +@item pal +25/1 +@item qntsc +30000/1001 +@item qpal +25/1 +@item sntsc +30000/1001 +@item spal +25/1 +@item film +24/1 +@item ntsc-film +24000/1001 +@end table + +@anchor{ratio syntax} +@section Ratio + +A ratio can be expressed as an expression, or in the form +@var{numerator}:@var{denominator}. + +Note that a ratio with infinite (1/0) or negative value is +considered valid, so you should check on the returned value if you +want to exclude those values. + +The undefined value can be expressed using the "0:0" string. + +@anchor{color syntax} +@section Color + +It can be the name of a color (case insensitive match) or a +[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string +representing the alpha component. + +The alpha component may be a string composed by "0x" followed by an +hexadecimal number or a decimal number between 0.0 and 1.0, which +represents the opacity value (0x00/0.0 means completely transparent, +0xff/1.0 completely opaque). +If the alpha component is not specified then 0xff is assumed. + +The string "random" will result in a random color. + +@c man end SYNTAX diff --git a/ffmpeg1/doc/t2h.init b/ffmpeg1/doc/t2h.init new file mode 100644 index 0000000..2aab488 --- /dev/null +++ b/ffmpeg1/doc/t2h.init @@ -0,0 +1,116 @@ +# no horiz rules between sections +$end_section = \&FFmpeg_end_section; +sub FFmpeg_end_section($$) +{ +} + +$EXTRA_HEAD = +'<link rel="icon" href="favicon.png" type="image/png" /> +'; + +$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT; +<link rel="stylesheet" type="text/css" href="default.css" /> +EOT + +my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT; +<link rel="icon" href="favicon.png" type="image/png" /> +</head> +<body> +<div id="container"> +EOT + +$PRE_BODY_CLOSE = '</div></div>'; + +$SMALL_RULE = ''; +$BODYTEXT = ''; + +$print_page_foot = \&FFmpeg_print_page_foot; +sub FFmpeg_print_page_foot($$) +{ + my $fh = shift; + my $program_string = defined &T2H_DEFAULT_program_string ? + T2H_DEFAULT_program_string() : program_string(); + print $fh '<footer class="footer pagination-right">' . "\n"; + print $fh '<span class="label label-info">' . $program_string; + print $fh "</span></footer></div>\n"; +} + +$float = \&FFmpeg_float; + +sub FFmpeg_float($$$$) +{ + my $text = shift; + my $float = shift; + my $caption = shift; + my $shortcaption = shift; + + my $label = ''; + if (exists($float->{'id'})) + { + $label = &$anchor($float->{'id'}); + } + my $class = ''; + my $subject = ''; + + if ($caption =~ /NOTE/) + { + $class = "alert alert-info"; + } + elsif ($caption =~ /IMPORTANT/) + { + $class = "alert alert-warning"; + } + + return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>'; +} + +$print_page_head = \&FFmpeg_print_page_head; +sub FFmpeg_print_page_head($$) +{ + my $fh = shift; + my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}"; + $longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'}; + my $description = $DOCUMENT_DESCRIPTION; + $description = $longtitle if (!defined($description)); + $description = "<meta name=\"description\" content=\"$description\">" if + ($description ne ''); + $description = $Texi2HTML::THISDOC{'documentdescription'} if (defined($Texi2HTML::THISDOC{'documentdescription'})); + my $encoding = ''; + $encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne '')); + $longtitle =~ s/Documentation.*//g; + $longtitle = "FFmpeg documentation : " . $longtitle; + + print $fh <<EOT; +<!DOCTYPE html> +<html> +$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} --> +<!-- +$Texi2HTML::THISDOC{program_authors} +--> +<head> +<title>$longtitle</title> + +$description +<meta name="keywords" content="$longtitle"> +<meta name="resource-type" content="document"> +<meta name="distribution" content="global"> +<meta name="Generator" content="$Texi2HTML::THISDOC{program}"> +$encoding +$CSS_LINES +$TEMPLATE_HEADER +EOT +} + +# declare encoding in header +$IN_ENCODING = $ENCODING = "utf-8"; + +# no navigation elements +$SECTION_NAVIGATION = 0; +# the same for texi2html 5.0 +$HEADERS = 0; + +# TOC and Chapter headings link +$TOC_LINKS = 1; + +# print the TOC where @contents is used +$INLINE_CONTENTS = 1; diff --git a/ffmpeg1/doc/tablegen.txt b/ffmpeg1/doc/tablegen.txt new file mode 100644 index 0000000..4c4f036 --- /dev/null +++ b/ffmpeg1/doc/tablegen.txt @@ -0,0 +1,70 @@ +Writing a table generator + +This documentation is preliminary. +Parts of the API are not good and should be changed. + +Basic concepts + +A table generator consists of two files, *_tablegen.c and *_tablegen.h. +The .h file will provide the variable declarations and initialization +code for the tables, the .c calls the initialization code and then prints +the tables as a header file using the tableprint.h helpers. +Both of these files will be compiled for the host system, so to avoid +breakage with cross-compilation neither of them may include, directly +or indirectly, config.h or avconfig.h. +This means that e.g. libavutil/mathematics.h is ok but libavutil/libm.h is not. +Due to this, the .c file or Makefile may have to provide additional defines +or stubs, though if possible this should be avoided. +In particular, CONFIG_HARDCODED_TABLES should always be defined to 0. + +The .c file + +This file should include the *_tablegen.h and tableprint.h files and +anything else it needs as long as it does not depend on config.h or +avconfig.h. +In addition to that it must contain a main() function which initializes +all tables by calling the init functions from the .h file and then prints +them. +The printing code typically looks like this: + write_fileheader(); + printf("static const uint8_t my_array[100] = {\n"); + write_uint8_t_array(my_array, 100); + printf("};\n"); + +This is the more generic form, in case you need to do something special. +Usually you should instead use the short form: + write_fileheader(); + WRITE_ARRAY("static const", uint8_t, my_array); + +write_fileheader() adds some minor things like a "this is a generated file" +comment and some standard includes. +tablegen.h defines some write functions for one- and two-dimensional arrays +for standard types - they print only the "core" parts so they are easier +to reuse for multi-dimensional arrays so the outermost {} must be printed +separately. +If there's no standard function for printing the type you need, the +WRITE_1D_FUNC_ARGV macro is a very quick way to create one. +See libavcodec/dv_tablegen.c for an example. + + +The .h file + +This file should contain: + - one or more initialization functions + - the table variable declarations +If CONFIG_HARDCODED_TABLES is set, the initialization functions should +not do anything, and instead of the variable declarations the +generated *_tables.h file should be included. +Since that will be generated in the build directory, the path must be +included, i.e. +#include "libavcodec/example_tables.h" +not +#include "example_tables.h" + +Makefile changes + +To make the automatic table creation work, you must manually declare the +new dependency. +For this add a line similar to this: +$(SUBDIR)example.o: $(SUBDIR)example_tables.h +under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile. diff --git a/ffmpeg1/doc/texi2pod.pl b/ffmpeg1/doc/texi2pod.pl new file mode 100755 index 0000000..697576c --- /dev/null +++ b/ffmpeg1/doc/texi2pod.pl @@ -0,0 +1,453 @@ +#! /usr/bin/perl + +# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc. + +# This file is part of GNU CC. + +# GNU CC is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2, or (at your option) +# any later version. + +# GNU CC is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. + +# You should have received a copy of the GNU General Public License +# along with GNU CC; see the file COPYING. If not, write to +# the Free Software Foundation, 51 Franklin Street, Fifth Floor, +# Boston, MA 02110-1301 USA + +# This does trivial (and I mean _trivial_) conversion of Texinfo +# markup to Perl POD format. It's intended to be used to extract +# something suitable for a manpage from a Texinfo document. + +use warnings; + +$output = 0; +$skipping = 0; +%chapters = (); +@chapters_sequence = (); +$chapter = ""; +@icstack = (); +@endwstack = (); +@skstack = (); +@instack = (); +$shift = ""; +%defs = (); +$fnno = 1; +$inf = ""; +@ibase = (); + +while ($_ = shift) { + if (/^-D(.*)$/) { + if ($1 ne "") { + $flag = $1; + } else { + $flag = shift; + } + $value = ""; + ($flag, $value) = ($flag =~ /^([^=]+)(?:=(.+))?/); + die "no flag specified for -D\n" + unless $flag ne ""; + die "flags may only contain letters, digits, hyphens, dashes and underscores\n" + unless $flag =~ /^[a-zA-Z0-9_-]+$/; + $defs{$flag} = $value; + } elsif (/^-I(.*)$/) { + push @ibase, $1 ne "" ? $1 : shift; + } elsif (/^-/) { + usage(); + } else { + $in = $_, next unless defined $in; + $out = $_, next unless defined $out; + usage(); + } +} + +push @ibase, "."; + +if (defined $in) { + $inf = gensym(); + open($inf, "<$in") or die "opening \"$in\": $!\n"; + push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|; +} else { + $inf = \*STDIN; +} + +if (defined $out) { + open(STDOUT, ">$out") or die "opening \"$out\": $!\n"; +} + +while(defined $inf) { +INF: while(<$inf>) { + # Certain commands are discarded without further processing. + /^\@(?: + [a-z]+index # @*index: useful only in complete manual + |need # @need: useful only in printed manual + |(?:end\s+)?group # @group .. @end group: ditto + |page # @page: ditto + |node # @node: useful only in .info file + |(?:end\s+)?ifnottex # @ifnottex .. @end ifnottex: use contents + )\b/x and next; + + chomp; + + # Look for filename and title markers. + /^\@setfilename\s+([^.]+)/ and $fn = $1, next; + /^\@settitle\s+([^.]+)/ and $tl = postprocess($1), next; + + # Identify a man title but keep only the one we are interested in. + /^\@c\s+man\s+title\s+([A-Za-z0-9-]+)\s+(.+)/ and do { + if (exists $defs{$1}) { + $fn = $1; + $tl = postprocess($2); + } + next; + }; + + /^\@include\s+(.+)$/ and do { + push @instack, $inf; + $inf = gensym(); + + for (@ibase) { + open($inf, "<" . $_ . "/" . $1) and next INF; + } + die "cannot open $1: $!\n"; + }; + + /^\@chapter\s+([A-Za-z ]+)/ and do { + # close old chapter + $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); + + # start new chapter + $chapter_name = $1, push (@chapters_sequence, $chapter_name); + $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name}; + $chapter = ""; + $output = 1; + next; + }; + + /^\@bye/ and do { + # close old chapter + $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); + last INF; + }; + + # handle variables + /^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do { + $defs{$1} = $2; + next; + }; + /^\@clear\s+([a-zA-Z0-9_-]+)/ and do { + delete $defs{$1}; + next; + }; + + next unless $output; + + # Discard comments. (Can't do it above, because then we'd never see + # @c man lines.) + /^\@c\b/ and next; + + # End-block handler goes up here because it needs to operate even + # if we are skipping. + /^\@end\s+([a-z]+)/ and do { + # Ignore @end foo, where foo is not an operation which may + # cause us to skip, if we are presently skipping. + my $ended = $1; + next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/; + + die "\@end $ended without \@$ended at line $.\n" unless defined $endw; + die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw; + + $endw = pop @endwstack; + + if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) { + $skipping = pop @skstack; + next; + } elsif ($ended =~ /^(?:example|smallexample|display)$/) { + $shift = ""; + $_ = ""; # need a paragraph break + } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) { + $_ = "\n=back\n"; + $ic = pop @icstack; + } else { + die "unknown command \@end $ended at line $.\n"; + } + }; + + # We must handle commands which can cause skipping even while we + # are skipping, otherwise we will not process nested conditionals + # correctly. + /^\@ifset\s+([a-zA-Z0-9_-]+)/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = "ifset"; + $skipping = 1 unless exists $defs{$1}; + next; + }; + + /^\@ifclear\s+([a-zA-Z0-9_-]+)/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = "ifclear"; + $skipping = 1 if exists $defs{$1}; + next; + }; + + /^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do { + push @endwstack, $endw; + push @skstack, $skipping; + $endw = $1; + $skipping = $endw !~ /ifnothtml/; + next; + }; + + next if $skipping; + + # Character entities. First the ones that can be replaced by raw text + # or discarded outright: + s/\@copyright\{\}/(c)/g; + s/\@dots\{\}/.../g; + s/\@enddots\{\}/..../g; + s/\@([.!? ])/$1/g; + s/\@[:-]//g; + s/\@bullet(?:\{\})?/*/g; + s/\@TeX\{\}/TeX/g; + s/\@pounds\{\}/\#/g; + s/\@minus(?:\{\})?/-/g; + + # Now the ones that have to be replaced by special escapes + # (which will be turned back into text by unmunge()) + s/&/&/g; + s/\@\{/{/g; + s/\@\}/}/g; + s/\@\@/&at;/g; + + # Inside a verbatim block, handle @var specially. + if ($shift ne "") { + s/\@var\{([^\}]*)\}/<$1>/g; + } + + # POD doesn't interpret E<> inside a verbatim block. + if ($shift eq "") { + s/</</g; + s/>/>/g; + } else { + s/</</g; + s/>/>/g; + } + + # Single line command handlers. + + /^\@(?:section|unnumbered|unnumberedsec|center|heading)\s+(.+)$/ + and $_ = "\n=head2 $1\n"; + /^\@(?:subsection|subheading)\s+(.+)$/ + and $_ = "\n=head3 $1\n"; + /^\@(?:subsubsection|subsubheading)\s+(.+)$/ + and $_ = "\n=head4 $1\n"; + + # Block command handlers: + /^\@itemize\s*(\@[a-z]+|\*|-)?/ and do { + push @endwstack, $endw; + push @icstack, $ic; + $ic = $1 ? $1 : "*"; + $_ = "\n=over 4\n"; + $endw = "itemize"; + }; + + /^\@enumerate(?:\s+([a-zA-Z0-9]+))?/ and do { + push @endwstack, $endw; + push @icstack, $ic; + if (defined $1) { + $ic = $1 . "."; + } else { + $ic = "1."; + } + $_ = "\n=over 4\n"; + $endw = "enumerate"; + }; + + /^\@((?:multi|[fv])?table)\s+(\@[a-z]+)/ and do { + push @endwstack, $endw; + push @icstack, $ic; + $endw = $1; + $ic = $2; + $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env|command)/B/; + $ic =~ s/\@(?:code|kbd)/C/; + $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/; + $ic =~ s/\@(?:file)/F/; + $ic =~ s/\@(?:columnfractions)//; + $_ = "\n=over 4\n"; + }; + + /^\@((?:small)?example|display)/ and do { + push @endwstack, $endw; + $endw = $1; + $shift = "\t"; + $_ = ""; # need a paragraph break + }; + + /^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do { + my $columns = $1; + $columns =~ s/\@tab/ : /; + + $_ = "\n=item B<". $columns .">\n"; + }; + + /^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do { + my $columns = $1; + $columns =~ s/\@tab/ : /; + + $_ = " : ". $columns; + $chapter =~ s/\n+\s+$//; + }; + + /^\@itemx?\s*(.+)?$/ and do { + if (defined $1) { + # Entity escapes prevent munging by the <> processing below. + $_ = "\n=item $ic\<$1\>\n"; + } else { + $_ = "\n=item $ic\n"; + $ic =~ y/A-Ya-y/B-Zb-z/; + $ic =~ s/(\d+)/$1 + 1/eg; + } + }; + + $chapter .= $shift.$_."\n"; +} +# End of current file. +close($inf); +$inf = pop @instack; +} + +die "No filename or title\n" unless defined $fn && defined $tl; + +$chapters{NAME} = "$fn \- $tl\n"; +$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES}; + +unshift @chapters_sequence, "NAME"; +for $chapter (@chapters_sequence) { + if (exists $chapters{$chapter}) { + $head = uc($chapter); + print "=head1 $head\n\n"; + print scalar unmunge ($chapters{$chapter}); + print "\n"; + } +} + +sub usage +{ + die "usage: $0 [-D toggle...] [infile [outfile]]\n"; +} + +sub postprocess +{ + local $_ = $_[0]; + + # @value{foo} is replaced by whatever 'foo' is defined as. + while (m/(\@value\{([a-zA-Z0-9_-]+)\})/g) { + if (! exists $defs{$2}) { + print STDERR "Option $2 not defined\n"; + s/\Q$1\E//; + } else { + $value = $defs{$2}; + s/\Q$1\E/$value/; + } + } + + # Formatting commands. + # Temporary escape for @r. + s/\@r\{([^\}]*)\}/R<$1>/g; + s/\@(?:dfn|var|emph|cite|i)\{([^\}]*)\}/I<$1>/g; + s/\@(?:code|kbd)\{([^\}]*)\}/C<$1>/g; + s/\@(?:gccoptlist|samp|strong|key|option|env|command|b)\{([^\}]*)\}/B<$1>/g; + s/\@sc\{([^\}]*)\}/\U$1/g; + s/\@file\{([^\}]*)\}/F<$1>/g; + s/\@w\{([^\}]*)\}/S<$1>/g; + s/\@(?:dmn|math)\{([^\}]*)\}/$1/g; + + # Cross references are thrown away, as are @noindent and @refill. + # (@noindent is impossible in .pod, and @refill is unnecessary.) + # @* is also impossible in .pod; we discard it and any newline that + # follows it. Similarly, our macro @gol must be discarded. + + s/\@anchor{(?:[^\}]*)\}//g; + s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g; + s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g; + s/;\s+\@pxref\{(?:[^\}]*)\}//g; + s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g; + s/\@ref\{([^\}]*)\}/$1/g; + s/\@noindent\s*//g; + s/\@refill//g; + s/\@gol//g; + s/\@\*\s*\n?//g; + + # @uref can take one, two, or three arguments, with different + # semantics each time. @url and @email are just like @uref with + # one argument, for our purposes. + s/\@(?:uref|url|email)\{([^\},]*),?[^\}]*\}/<B<$1>>/g; + s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g; + s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g; + + # Turn B<blah I<blah> blah> into B<blah> I<blah> B<blah> to + # match Texinfo semantics of @emph inside @samp. Also handle @r + # inside bold. + s/</</g; + s/>/>/g; + 1 while s/B<((?:[^<>]|I<[^<>]*>)*)R<([^>]*)>/B<$1>${2}B</g; + 1 while (s/B<([^<>]*)I<([^>]+)>/B<$1>I<$2>B</g); + 1 while (s/I<([^<>]*)B<([^>]+)>/I<$1>B<$2>I</g); + s/[BI]<>//g; + s/([BI])<(\s+)([^>]+)>/$2$1<$3>/g; + s/([BI])<([^>]+?)(\s+)>/$1<$2>$3/g; + + # Extract footnotes. This has to be done after all other + # processing because otherwise the regexp will choke on formatting + # inside @footnote. + while (/\@footnote/g) { + s/\@footnote\{([^\}]+)\}/[$fnno]/; + add_footnote($1, $fnno); + $fnno++; + } + + return $_; +} + +sub unmunge +{ + # Replace escaped symbols with their equivalents. + local $_ = $_[0]; + + s/</E<lt>/g; + s/>/E<gt>/g; + s/{/\{/g; + s/}/\}/g; + s/&at;/\@/g; + s/&/&/g; + return $_; +} + +sub add_footnote +{ + unless (exists $chapters{FOOTNOTES}) { + $chapters{FOOTNOTES} = "\n=over 4\n\n"; + } + + $chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; + $chapters{FOOTNOTES} .= $_[0]; + $chapters{FOOTNOTES} .= "\n\n"; +} + +# stolen from Symbol.pm +{ + my $genseq = 0; + sub gensym + { + my $name = "GEN" . $genseq++; + my $ref = \*{$name}; + delete $::{$name}; + return $ref; + } +} diff --git a/ffmpeg1/doc/viterbi.txt b/ffmpeg1/doc/viterbi.txt new file mode 100644 index 0000000..9782546 --- /dev/null +++ b/ffmpeg1/doc/viterbi.txt @@ -0,0 +1,109 @@ +This is a quick description of the viterbi aka dynamic programing +algorthm. + +Its reason for existence is that wikipedia has become very poor on +describing algorithms in a way that makes it useable for understanding +them or anything else actually. It tends now to describe the very same +algorithm under 50 different names and pages with few understandable +by even people who fully understand the algorithm and the theory behind. + +Problem description: (that is what it can solve) +assume we have a 2d table, or you could call it a graph or matrix if you +prefer + + O O O O O O O + + O O O O O O O + + O O O O O O O + + O O O O O O O + + +That table has edges connecting points from each column to the next column +and each edge has a score like: (only some edge and scores shown to keep it +readable) + + + O--5--O-----O-----O-----O-----O + 2 / 7 / \ / \ / \ / + \ / \ / \ / \ / \ / + O7-/--O--/--O--/--O--/--O--/--O + \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/ + /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\ + O3-/--O--/--O--/--O--/--O--/--O + / \ / \ / \ / \ / \ + 1 \ 9 \ / \ / \ / \ + O--2--O--1--O--5--O--3--O--8--O + + + +Our goal is to find a path from left to right through it which +minimizes the sum of the score of all edges. +(and of course left/right is just a convention here it could be top down too) +Similarly the minimum could be the maximum by just fliping the sign, +Example of a path with scores: + + O O O O O O O + +>---O. O O .O-2-O O O + 5. .7 . + O O-1-O O O 8 O O + . + O O O O O O-1-O---> (sum here is 24) + + +The viterbi algorthm now solves this simply column by column +For the previous column each point has a best path and a associated +score: + + O-----5 O + \ + \ + O \ 1 O + \/ + /\ + O / 2 O + / + / + O-----2 O + + +To move one column forward we just need to find the best path and associated +scores for the next column +here are some edges we could choose from: + + + O-----5--3--O + \ \8 + \ \ + O \ 1--9--O + \/ \3 + /\ \ + O / 2--1--O + / \2 + / \ + O-----2--4--O + +Finding the new best paths and scores for each point of our new column is +trivial given we know the previous column best paths and scores: + + O-----0-----8 + \ + \ + O \ 0----10 + \/ + /\ + O / 0-----3 + / \ + / \ + O 0 4 + + +the viterbi algorthm continues exactly like this column for column until the +end and then just picks the path with the best score (above that would be the +one with score 3) + + +Author: Michael niedermayer +Copyright LGPL |