2 * Audio Interleaving functions
4 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
6 * This file is part of Libav.
8 * Libav 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 * Libav 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 Libav; 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 (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
37 av_fifo_free(aic->fifo);
41 int ff_audio_interleave_init(AVFormatContext *s,
42 const int *samples_per_frame,
47 if (!samples_per_frame)
50 for (i = 0; i < s->nb_streams; i++) {
51 AVStream *st = s->streams[i];
52 AudioInterleaveContext *aic = st->priv_data;
54 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
55 aic->sample_size = (st->codec->channels *
56 av_get_bits_per_sample(st->codec->codec_id)) / 8;
57 if (!aic->sample_size) {
58 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
61 aic->samples_per_frame = samples_per_frame;
62 aic->samples = aic->samples_per_frame;
63 aic->time_base = time_base;
65 aic->fifo_size = 100* *aic->samples;
66 aic->fifo= av_fifo_alloc(100 * *aic->samples);
73 static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
74 int stream_index, int flush)
76 AVStream *st = s->streams[stream_index];
77 AudioInterleaveContext *aic = st->priv_data;
79 int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
80 if (!size || (!flush && size == av_fifo_size(aic->fifo)))
83 av_new_packet(pkt, size);
84 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
86 pkt->dts = pkt->pts = aic->dts;
87 pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
88 pkt->stream_index = stream_index;
89 aic->dts += pkt->duration;
93 aic->samples = aic->samples_per_frame;
98 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
99 int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
100 int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
105 AVStream *st = s->streams[pkt->stream_index];
106 AudioInterleaveContext *aic = st->priv_data;
107 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
108 unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
109 if (new_size > aic->fifo_size) {
110 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
112 aic->fifo_size = new_size;
114 av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
116 // rewrite pts and dts to be decoded time line position
117 pkt->pts = pkt->dts = aic->dts;
118 aic->dts += pkt->duration;
119 ff_interleave_add_packet(s, pkt, compare_ts);
124 for (i = 0; i < s->nb_streams; i++) {
125 AVStream *st = s->streams[i];
126 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
128 while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
129 ff_interleave_add_packet(s, &new_pkt, compare_ts);
133 return get_packet(s, out, pkt, flush);