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Diffstat (limited to 'ffmpeg/libavformat/audiointerleave.c')
| -rw-r--r-- | ffmpeg/libavformat/audiointerleave.c | 148 |
1 files changed, 148 insertions, 0 deletions
diff --git a/ffmpeg/libavformat/audiointerleave.c b/ffmpeg/libavformat/audiointerleave.c new file mode 100644 index 0000000..35dd8d5 --- /dev/null +++ b/ffmpeg/libavformat/audiointerleave.c @@ -0,0 +1,148 @@ +/* + * Audio Interleaving functions + * + * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/fifo.h" +#include "libavutil/mathematics.h" +#include "avformat.h" +#include "audiointerleave.h" +#include "internal.h" + +void ff_audio_interleave_close(AVFormatContext *s) +{ + int i; + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + AudioInterleaveContext *aic = st->priv_data; + + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) + av_fifo_free(aic->fifo); + } +} + +int ff_audio_interleave_init(AVFormatContext *s, + const int *samples_per_frame, + AVRational time_base) +{ + int i; + + if (!samples_per_frame) + return -1; + + if (!time_base.num) { + av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); + return -1; + } + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + AudioInterleaveContext *aic = st->priv_data; + + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + aic->sample_size = (st->codec->channels * + av_get_bits_per_sample(st->codec->codec_id)) / 8; + if (!aic->sample_size) { + av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); + return -1; + } + aic->samples_per_frame = samples_per_frame; + aic->samples = aic->samples_per_frame; + aic->time_base = time_base; + + aic->fifo_size = 100* *aic->samples; + aic->fifo= av_fifo_alloc(100 * *aic->samples); + } + } + + return 0; +} + +static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, + int stream_index, int flush) +{ + AVStream *st = s->streams[stream_index]; + AudioInterleaveContext *aic = st->priv_data; + + int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size); + if (!size || (!flush && size == av_fifo_size(aic->fifo))) + return 0; + + if (av_new_packet(pkt, size) < 0) + return AVERROR(ENOMEM); + av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); + + pkt->dts = pkt->pts = aic->dts; + pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); + pkt->stream_index = stream_index; + aic->dts += pkt->duration; + + aic->samples++; + if (!*aic->samples) + aic->samples = aic->samples_per_frame; + + return size; +} + +int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, + int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), + int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) +{ + int i; + + if (pkt) { + AVStream *st = s->streams[pkt->stream_index]; + AudioInterleaveContext *aic = st->priv_data; + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; + if (new_size > aic->fifo_size) { + if (av_fifo_realloc2(aic->fifo, new_size) < 0) + return -1; + aic->fifo_size = new_size; + } + av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); + } else { + int ret; + // rewrite pts and dts to be decoded time line position + pkt->pts = pkt->dts = aic->dts; + aic->dts += pkt->duration; + ret = ff_interleave_add_packet(s, pkt, compare_ts); + if (ret < 0) + return ret; + } + pkt = NULL; + } + + for (i = 0; i < s->nb_streams; i++) { + AVStream *st = s->streams[i]; + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + AVPacket new_pkt; + int ret; + while ((ret = ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { + ret = ff_interleave_add_packet(s, &new_pkt, compare_ts); + if (ret < 0) + return ret; + } + if (ret < 0) + return ret; + } + } + + return get_packet(s, out, NULL, flush); +} |