Mercurial > libavformat.hg
view audiointerleave.c @ 5075:9a49ff4cdb6e libavformat
Remove inconsistent use of a @brief command in av_write_trailer()
doxy.
author | stefano |
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date | Sun, 28 Jun 2009 21:05:46 +0000 |
parents | fc0a165de804 |
children | 536e5527c1e0 |
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/* * 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 "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 == CODEC_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; for (i = 0; i < s->nb_streams; i++) { AVStream *st = s->streams[i]; AudioInterleaveContext *aic = st->priv_data; if (st->codec->codec_type == CODEC_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; av_new_packet(pkt, size); 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 == CODEC_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 { // rewrite pts and dts to be decoded time line position pkt->pts = pkt->dts = aic->dts; aic->dts += pkt->duration; ff_interleave_add_packet(s, pkt, compare_ts); } pkt = NULL; } for (i = 0; i < s->nb_streams; i++) { AVStream *st = s->streams[i]; if (st->codec->codec_type == CODEC_TYPE_AUDIO) { AVPacket new_pkt; while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) ff_interleave_add_packet(s, &new_pkt, compare_ts); } } return get_packet(s, out, pkt, flush); }