1/* 2 * Audio Interleaving functions 3 * 4 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com> 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * FFmpeg is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with FFmpeg; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23#include "libavutil/fifo.h" 24#include "libavutil/mathematics.h" 25#include "avformat.h" 26#include "audiointerleave.h" 27#include "internal.h" 28 29void ff_audio_interleave_close(AVFormatContext *s) 30{ 31 int i; 32 for (i = 0; i < s->nb_streams; i++) { 33 AVStream *st = s->streams[i]; 34 AudioInterleaveContext *aic = st->priv_data; 35 36 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) 37 av_fifo_freep(&aic->fifo); 38 } 39} 40 41int ff_audio_interleave_init(AVFormatContext *s, 42 const int *samples_per_frame, 43 AVRational time_base) 44{ 45 int i; 46 47 if (!samples_per_frame) 48 return AVERROR(EINVAL); 49 50 if (!time_base.num) { 51 av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n"); 52 return AVERROR(EINVAL); 53 } 54 for (i = 0; i < s->nb_streams; i++) { 55 AVStream *st = s->streams[i]; 56 AudioInterleaveContext *aic = st->priv_data; 57 58 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { 59 aic->sample_size = (st->codec->channels * 60 av_get_bits_per_sample(st->codec->codec_id)) / 8; 61 if (!aic->sample_size) { 62 av_log(s, AV_LOG_ERROR, "could not compute sample size\n"); 63 return AVERROR(EINVAL); 64 } 65 aic->samples_per_frame = samples_per_frame; 66 aic->samples = aic->samples_per_frame; 67 aic->time_base = time_base; 68 69 aic->fifo_size = 100* *aic->samples; 70 if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples))) 71 return AVERROR(ENOMEM); 72 } 73 } 74 75 return 0; 76} 77 78static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt, 79 int stream_index, int flush) 80{ 81 AVStream *st = s->streams[stream_index]; 82 AudioInterleaveContext *aic = st->priv_data; 83 84 int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size); 85 if (!size || (!flush && size == av_fifo_size(aic->fifo))) 86 return 0; 87 88 if (av_new_packet(pkt, size) < 0) 89 return AVERROR(ENOMEM); 90 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); 91 92 pkt->dts = pkt->pts = aic->dts; 93 pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); 94 pkt->stream_index = stream_index; 95 aic->dts += pkt->duration; 96 97 aic->samples++; 98 if (!*aic->samples) 99 aic->samples = aic->samples_per_frame; 100 101 return size; 102} 103 104int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, 105 int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), 106 int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) 107{ 108 int i; 109 110 if (pkt) { 111 AVStream *st = s->streams[pkt->stream_index]; 112 AudioInterleaveContext *aic = st->priv_data; 113 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { 114 unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; 115 if (new_size > aic->fifo_size) { 116 if (av_fifo_realloc2(aic->fifo, new_size) < 0) 117 return AVERROR(ENOMEM); 118 aic->fifo_size = new_size; 119 } 120 av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); 121 } else { 122 int ret; 123 // rewrite pts and dts to be decoded time line position 124 pkt->pts = pkt->dts = aic->dts; 125 aic->dts += pkt->duration; 126 ret = ff_interleave_add_packet(s, pkt, compare_ts); 127 if (ret < 0) 128 return ret; 129 } 130 pkt = NULL; 131 } 132 133 for (i = 0; i < s->nb_streams; i++) { 134 AVStream *st = s->streams[i]; 135 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { 136 AVPacket new_pkt; 137 int ret; 138 while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { 139 ret = ff_interleave_add_packet(s, &new_pkt, compare_ts); 140 if (ret < 0) 141 return ret; 142 } 143 if (ret < 0) 144 return ret; 145 } 146 } 147 148 return get_packet(s, out, NULL, flush); 149} 150