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Diffstat (limited to 'ffmpeg/libavresample/utils.c')
| -rw-r--r-- | ffmpeg/libavresample/utils.c | 635 |
1 files changed, 635 insertions, 0 deletions
diff --git a/ffmpeg/libavresample/utils.c b/ffmpeg/libavresample/utils.c new file mode 100644 index 0000000..c159e33 --- /dev/null +++ b/ffmpeg/libavresample/utils.c @@ -0,0 +1,635 @@ +/* + * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> + * + * 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 + */ + +#include "libavutil/common.h" +#include "libavutil/dict.h" +// #include "libavutil/error.h" +#include "libavutil/log.h" +#include "libavutil/mem.h" +#include "libavutil/opt.h" + +#include "avresample.h" +#include "internal.h" +#include "audio_data.h" +#include "audio_convert.h" +#include "audio_mix.h" +#include "resample.h" + +int avresample_open(AVAudioResampleContext *avr) +{ + int ret; + + /* set channel mixing parameters */ + avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); + if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n", + avr->in_channel_layout); + return AVERROR(EINVAL); + } + avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); + if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n", + avr->out_channel_layout); + return AVERROR(EINVAL); + } + avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels); + avr->downmix_needed = avr->in_channels > avr->out_channels; + avr->upmix_needed = avr->out_channels > avr->in_channels || + (!avr->downmix_needed && (avr->mix_matrix || + avr->in_channel_layout != avr->out_channel_layout)); + avr->mixing_needed = avr->downmix_needed || avr->upmix_needed; + + /* set resampling parameters */ + avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate || + avr->force_resampling; + + /* select internal sample format if not specified by the user */ + if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE && + (avr->mixing_needed || avr->resample_needed)) { + enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt); + enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); + int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt), + av_get_bytes_per_sample(out_fmt)); + if (max_bps <= 2) { + avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P; + } else if (avr->mixing_needed) { + avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; + } else { + if (max_bps <= 4) { + if (in_fmt == AV_SAMPLE_FMT_S32P || + out_fmt == AV_SAMPLE_FMT_S32P) { + if (in_fmt == AV_SAMPLE_FMT_FLTP || + out_fmt == AV_SAMPLE_FMT_FLTP) { + /* if one is s32 and the other is flt, use dbl */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP; + } else { + /* if one is s32 and the other is s32, s16, or u8, use s32 */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P; + } + } else { + /* if one is flt and the other is flt, s16 or u8, use flt */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; + } + } else { + /* if either is dbl, use dbl */ + avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP; + } + } + av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n", + av_get_sample_fmt_name(avr->internal_sample_fmt)); + } + + /* treat all mono as planar for easier comparison */ + if (avr->in_channels == 1) + avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt); + if (avr->out_channels == 1) + avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); + + /* we may need to add an extra conversion in order to remap channels if + the output format is not planar */ + if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed && + !av_sample_fmt_is_planar(avr->out_sample_fmt)) { + avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); + } + + /* set sample format conversion parameters */ + if (avr->resample_needed || avr->mixing_needed) + avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt; + else + avr->in_convert_needed = avr->use_channel_map && + !av_sample_fmt_is_planar(avr->out_sample_fmt); + + if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed) + avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt; + else + avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt; + + avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed || + (avr->use_channel_map && avr->resample_needed)); + + if (avr->use_channel_map) { + if (avr->in_copy_needed) { + avr->remap_point = REMAP_IN_COPY; + av_dlog(avr, "remap channels during in_copy\n"); + } else if (avr->in_convert_needed) { + avr->remap_point = REMAP_IN_CONVERT; + av_dlog(avr, "remap channels during in_convert\n"); + } else if (avr->out_convert_needed) { + avr->remap_point = REMAP_OUT_CONVERT; + av_dlog(avr, "remap channels during out_convert\n"); + } else { + avr->remap_point = REMAP_OUT_COPY; + av_dlog(avr, "remap channels during out_copy\n"); + } + +#ifdef DEBUG + { + int ch; + av_dlog(avr, "output map: "); + if (avr->ch_map_info.do_remap) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_map[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "copy map: "); + if (avr->ch_map_info.do_copy) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_copy[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "zero map: "); + if (avr->ch_map_info.do_zero) + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.channel_zero[ch]); + else + av_dlog(avr, "n/a"); + av_dlog(avr, "\n"); + av_dlog(avr, "input map: "); + for (ch = 0; ch < avr->in_channels; ch++) + av_dlog(avr, " % 2d", avr->ch_map_info.input_map[ch]); + av_dlog(avr, "\n"); + } +#endif + } else + avr->remap_point = REMAP_NONE; + + /* allocate buffers */ + if (avr->in_copy_needed || avr->in_convert_needed) { + avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels), + 0, avr->internal_sample_fmt, + "in_buffer"); + if (!avr->in_buffer) { + ret = AVERROR(EINVAL); + goto error; + } + } + if (avr->resample_needed) { + avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels, + 0, avr->internal_sample_fmt, + "resample_out_buffer"); + if (!avr->resample_out_buffer) { + ret = AVERROR(EINVAL); + goto error; + } + } + if (avr->out_convert_needed) { + avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0, + avr->out_sample_fmt, "out_buffer"); + if (!avr->out_buffer) { + ret = AVERROR(EINVAL); + goto error; + } + } + avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels, + 1024); + if (!avr->out_fifo) { + ret = AVERROR(ENOMEM); + goto error; + } + + /* setup contexts */ + if (avr->in_convert_needed) { + avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt, + avr->in_sample_fmt, avr->in_channels, + avr->in_sample_rate, + avr->remap_point == REMAP_IN_CONVERT); + if (!avr->ac_in) { + ret = AVERROR(ENOMEM); + goto error; + } + } + if (avr->out_convert_needed) { + enum AVSampleFormat src_fmt; + if (avr->in_convert_needed) + src_fmt = avr->internal_sample_fmt; + else + src_fmt = avr->in_sample_fmt; + avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt, + avr->out_channels, + avr->out_sample_rate, + avr->remap_point == REMAP_OUT_CONVERT); + if (!avr->ac_out) { + ret = AVERROR(ENOMEM); + goto error; + } + } + if (avr->resample_needed) { + avr->resample = ff_audio_resample_init(avr); + if (!avr->resample) { + ret = AVERROR(ENOMEM); + goto error; + } + } + if (avr->mixing_needed) { + avr->am = ff_audio_mix_alloc(avr); + if (!avr->am) { + ret = AVERROR(ENOMEM); + goto error; + } + } + + return 0; + +error: + avresample_close(avr); + return ret; +} + +void avresample_close(AVAudioResampleContext *avr) +{ + ff_audio_data_free(&avr->in_buffer); + ff_audio_data_free(&avr->resample_out_buffer); + ff_audio_data_free(&avr->out_buffer); + av_audio_fifo_free(avr->out_fifo); + avr->out_fifo = NULL; + ff_audio_convert_free(&avr->ac_in); + ff_audio_convert_free(&avr->ac_out); + ff_audio_resample_free(&avr->resample); + ff_audio_mix_free(&avr->am); + av_freep(&avr->mix_matrix); + + avr->use_channel_map = 0; +} + +void avresample_free(AVAudioResampleContext **avr) +{ + if (!*avr) + return; + avresample_close(*avr); + av_opt_free(*avr); + av_freep(avr); +} + +static int handle_buffered_output(AVAudioResampleContext *avr, + AudioData *output, AudioData *converted) +{ + int ret; + + if (!output || av_audio_fifo_size(avr->out_fifo) > 0 || + (converted && output->allocated_samples < converted->nb_samples)) { + if (converted) { + /* if there are any samples in the output FIFO or if the + user-supplied output buffer is not large enough for all samples, + we add to the output FIFO */ + av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name); + ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0, + converted->nb_samples); + if (ret < 0) + return ret; + } + + /* if the user specified an output buffer, read samples from the output + FIFO to the user output */ + if (output && output->allocated_samples > 0) { + av_dlog(avr, "[FIFO] read from out_fifo to output\n"); + av_dlog(avr, "[end conversion]\n"); + return ff_audio_data_read_from_fifo(avr->out_fifo, output, + output->allocated_samples); + } + } else if (converted) { + /* copy directly to output if it is large enough or there is not any + data in the output FIFO */ + av_dlog(avr, "[copy] %s to output\n", converted->name); + output->nb_samples = 0; + ret = ff_audio_data_copy(output, converted, + avr->remap_point == REMAP_OUT_COPY ? + &avr->ch_map_info : NULL); + if (ret < 0) + return ret; + av_dlog(avr, "[end conversion]\n"); + return output->nb_samples; + } + av_dlog(avr, "[end conversion]\n"); + return 0; +} + +int attribute_align_arg avresample_convert(AVAudioResampleContext *avr, + uint8_t **output, int out_plane_size, + int out_samples, uint8_t **input, + int in_plane_size, int in_samples) +{ + AudioData input_buffer; + AudioData output_buffer; + AudioData *current_buffer; + int ret, direct_output; + + /* reset internal buffers */ + if (avr->in_buffer) { + avr->in_buffer->nb_samples = 0; + ff_audio_data_set_channels(avr->in_buffer, + avr->in_buffer->allocated_channels); + } + if (avr->resample_out_buffer) { + avr->resample_out_buffer->nb_samples = 0; + ff_audio_data_set_channels(avr->resample_out_buffer, + avr->resample_out_buffer->allocated_channels); + } + if (avr->out_buffer) { + avr->out_buffer->nb_samples = 0; + ff_audio_data_set_channels(avr->out_buffer, + avr->out_buffer->allocated_channels); + } + + av_dlog(avr, "[start conversion]\n"); + + /* initialize output_buffer with output data */ + direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0; + if (output) { + ret = ff_audio_data_init(&output_buffer, output, out_plane_size, + avr->out_channels, out_samples, + avr->out_sample_fmt, 0, "output"); + if (ret < 0) + return ret; + output_buffer.nb_samples = 0; + } + + if (input) { + /* initialize input_buffer with input data */ + ret = ff_audio_data_init(&input_buffer, input, in_plane_size, + avr->in_channels, in_samples, + avr->in_sample_fmt, 1, "input"); + if (ret < 0) + return ret; + current_buffer = &input_buffer; + + if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed && + !avr->out_convert_needed && direct_output && out_samples >= in_samples) { + /* in some rare cases we can copy input to output and upmix + directly in the output buffer */ + av_dlog(avr, "[copy] %s to output\n", current_buffer->name); + ret = ff_audio_data_copy(&output_buffer, current_buffer, + avr->remap_point == REMAP_OUT_COPY ? + &avr->ch_map_info : NULL); + if (ret < 0) + return ret; + current_buffer = &output_buffer; + } else if (avr->remap_point == REMAP_OUT_COPY && + (!direct_output || out_samples < in_samples)) { + /* if remapping channels during output copy, we may need to + * use an intermediate buffer in order to remap before adding + * samples to the output fifo */ + av_dlog(avr, "[copy] %s to out_buffer\n", current_buffer->name); + ret = ff_audio_data_copy(avr->out_buffer, current_buffer, + &avr->ch_map_info); + if (ret < 0) + return ret; + current_buffer = avr->out_buffer; + } else if (avr->in_copy_needed || avr->in_convert_needed) { + /* if needed, copy or convert input to in_buffer, and downmix if + applicable */ + if (avr->in_convert_needed) { + ret = ff_audio_data_realloc(avr->in_buffer, + current_buffer->nb_samples); + if (ret < 0) + return ret; + av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name); + ret = ff_audio_convert(avr->ac_in, avr->in_buffer, + current_buffer); + if (ret < 0) + return ret; + } else { + av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name); + ret = ff_audio_data_copy(avr->in_buffer, current_buffer, + avr->remap_point == REMAP_IN_COPY ? + &avr->ch_map_info : NULL); + if (ret < 0) + return ret; + } + ff_audio_data_set_channels(avr->in_buffer, avr->in_channels); + if (avr->downmix_needed) { + av_dlog(avr, "[downmix] in_buffer\n"); + ret = ff_audio_mix(avr->am, avr->in_buffer); + if (ret < 0) + return ret; + } + current_buffer = avr->in_buffer; + } + } else { + /* flush resampling buffer and/or output FIFO if input is NULL */ + if (!avr->resample_needed) + return handle_buffered_output(avr, output ? &output_buffer : NULL, + NULL); + current_buffer = NULL; + } + + if (avr->resample_needed) { + AudioData *resample_out; + + if (!avr->out_convert_needed && direct_output && out_samples > 0) + resample_out = &output_buffer; + else + resample_out = avr->resample_out_buffer; + av_dlog(avr, "[resample] %s to %s\n", current_buffer->name, + resample_out->name); + ret = ff_audio_resample(avr->resample, resample_out, + current_buffer); + if (ret < 0) + return ret; + + /* if resampling did not produce any samples, just return 0 */ + if (resample_out->nb_samples == 0) { + av_dlog(avr, "[end conversion]\n"); + return 0; + } + + current_buffer = resample_out; + } + + if (avr->upmix_needed) { + av_dlog(avr, "[upmix] %s\n", current_buffer->name); + ret = ff_audio_mix(avr->am, current_buffer); + if (ret < 0) + return ret; + } + + /* if we resampled or upmixed directly to output, return here */ + if (current_buffer == &output_buffer) { + av_dlog(avr, "[end conversion]\n"); + return current_buffer->nb_samples; + } + + if (avr->out_convert_needed) { + if (direct_output && out_samples >= current_buffer->nb_samples) { + /* convert directly to output */ + av_dlog(avr, "[convert] %s to output\n", current_buffer->name); + ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer); + if (ret < 0) + return ret; + + av_dlog(avr, "[end conversion]\n"); + return output_buffer.nb_samples; + } else { + ret = ff_audio_data_realloc(avr->out_buffer, + current_buffer->nb_samples); + if (ret < 0) + return ret; + av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name); + ret = ff_audio_convert(avr->ac_out, avr->out_buffer, + current_buffer); + if (ret < 0) + return ret; + current_buffer = avr->out_buffer; + } + } + + return handle_buffered_output(avr, output ? &output_buffer : NULL, + current_buffer); +} + +int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, + int stride) +{ + int in_channels, out_channels, i, o; + + if (avr->am) + return ff_audio_mix_get_matrix(avr->am, matrix, stride); + + in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); + out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); + + if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS || + out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n"); + return AVERROR(EINVAL); + } + + if (!avr->mix_matrix) { + av_log(avr, AV_LOG_ERROR, "matrix is not set\n"); + return AVERROR(EINVAL); + } + + for (o = 0; o < out_channels; o++) + for (i = 0; i < in_channels; i++) + matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i]; + + return 0; +} + +int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, + int stride) +{ + int in_channels, out_channels, i, o; + + if (avr->am) + return ff_audio_mix_set_matrix(avr->am, matrix, stride); + + in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); + out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); + + if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS || + out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n"); + return AVERROR(EINVAL); + } + + if (avr->mix_matrix) + av_freep(&avr->mix_matrix); + avr->mix_matrix = av_malloc(in_channels * out_channels * + sizeof(*avr->mix_matrix)); + if (!avr->mix_matrix) + return AVERROR(ENOMEM); + + for (o = 0; o < out_channels; o++) + for (i = 0; i < in_channels; i++) + avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i]; + + return 0; +} + +int avresample_set_channel_mapping(AVAudioResampleContext *avr, + const int *channel_map) +{ + ChannelMapInfo *info = &avr->ch_map_info; + int in_channels, ch, i; + + in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); + if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) { + av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n"); + return AVERROR(EINVAL); + } + + memset(info, 0, sizeof(*info)); + memset(info->input_map, -1, sizeof(info->input_map)); + + for (ch = 0; ch < in_channels; ch++) { + if (channel_map[ch] >= in_channels) { + av_log(avr, AV_LOG_ERROR, "Invalid channel map\n"); + return AVERROR(EINVAL); + } + if (channel_map[ch] < 0) { + info->channel_zero[ch] = 1; + info->channel_map[ch] = -1; + info->do_zero = 1; + } else if (info->input_map[channel_map[ch]] >= 0) { + info->channel_copy[ch] = info->input_map[channel_map[ch]]; + info->channel_map[ch] = -1; + info->do_copy = 1; + } else { + info->channel_map[ch] = channel_map[ch]; + info->input_map[channel_map[ch]] = ch; + info->do_remap = 1; + } + } + /* Fill-in unmapped input channels with unmapped output channels. + This is used when remapping during conversion from interleaved to + planar format. */ + for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) { + while (ch < in_channels && info->input_map[ch] >= 0) + ch++; + while (i < in_channels && info->channel_map[i] >= 0) + i++; + if (ch >= in_channels || i >= in_channels) + break; + info->input_map[ch] = i; + } + + avr->use_channel_map = 1; + return 0; +} + +int avresample_available(AVAudioResampleContext *avr) +{ + return av_audio_fifo_size(avr->out_fifo); +} + +int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples) +{ + if (!output) + return av_audio_fifo_drain(avr->out_fifo, nb_samples); + return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples); +} + +unsigned avresample_version(void) +{ + return LIBAVRESAMPLE_VERSION_INT; +} + +const char *avresample_license(void) +{ +#define LICENSE_PREFIX "libavresample license: " + return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; +} + +const char *avresample_configuration(void) +{ + return FFMPEG_CONFIGURATION; +} |