2 * Audio Interleaving functions
4 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
6 * This file is part of FFmpeg.
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.
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.
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
23 #include "libavutil/fifo.h"
24 #include "libavutil/mathematics.h"
26 #include "audiointerleave.h"
29 void ff_audio_interleave_close(AVFormatContext *s)
32 for (i = 0; i < s->nb_streams; i++) {
33 AVStream *st = s->streams[i];
34 AudioInterleaveContext *aic = st->priv_data;
36 if (aic && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
37 av_fifo_freep(&aic->fifo);
41 int ff_audio_interleave_init(AVFormatContext *s,
42 const int samples_per_frame,
48 av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
49 return AVERROR(EINVAL);
51 for (i = 0; i < s->nb_streams; i++) {
52 AVStream *st = s->streams[i];
53 AudioInterleaveContext *aic = st->priv_data;
55 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
56 int max_samples = samples_per_frame ? samples_per_frame :
57 av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP);
58 aic->sample_size = (st->codecpar->channels *
59 av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
60 if (!aic->sample_size) {
61 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
62 return AVERROR(EINVAL);
64 aic->samples_per_frame = samples_per_frame;
65 aic->time_base = time_base;
67 if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
68 return AVERROR(ENOMEM);
69 aic->fifo_size = 100 * max_samples;
76 static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
77 int stream_index, int flush)
79 AVStream *st = s->streams[stream_index];
80 AudioInterleaveContext *aic = st->priv_data;
82 int nb_samples = aic->samples_per_frame ? aic->samples_per_frame :
83 (av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples);
84 int frame_size = nb_samples * aic->sample_size;
85 int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
86 if (!size || (!flush && size == av_fifo_size(aic->fifo)))
89 ret = av_new_packet(pkt, frame_size);
92 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
95 memset(pkt->data + size, 0, pkt->size - size);
97 pkt->dts = pkt->pts = aic->dts;
98 pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base);
99 pkt->stream_index = stream_index;
100 aic->dts += pkt->duration;
101 aic->nb_samples += nb_samples;
107 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
108 int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
109 int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
114 AVStream *st = s->streams[pkt->stream_index];
115 AudioInterleaveContext *aic = st->priv_data;
116 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
117 unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
118 if (new_size > aic->fifo_size) {
119 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
120 return AVERROR(ENOMEM);
121 aic->fifo_size = new_size;
123 av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
125 // rewrite pts and dts to be decoded time line position
126 pkt->pts = pkt->dts = aic->dts;
127 aic->dts += pkt->duration;
128 if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
134 for (i = 0; i < s->nb_streams; i++) {
135 AVStream *st = s->streams[i];
136 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
138 while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
139 if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
147 return get_packet(s, out, NULL, flush);