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audiointerleave.c
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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 
30 {
31  int i;
32  for (i = 0; i < s->nb_streams; i++) {
33  AVStream *st = s->streams[i];
35 
37  av_fifo_free(aic->fifo);
38  }
39 }
40 
42  const int *samples_per_frame,
43  AVRational time_base)
44 {
45  int i;
46 
47  if (!samples_per_frame)
48  return -1;
49 
50  if (!time_base.num) {
51  av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
52  return -1;
53  }
54  for (i = 0; i < s->nb_streams; i++) {
55  AVStream *st = s->streams[i];
57 
58  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
59  aic->sample_size = (st->codec->channels *
61  if (!aic->sample_size) {
62  av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
63  return -1;
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  aic->fifo= av_fifo_alloc(100 * *aic->samples);
71  }
72  }
73 
74  return 0;
75 }
76 
78  int stream_index, int flush)
79 {
80  AVStream *st = s->streams[stream_index];
82 
83  int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
84  if (!size || (!flush && size == av_fifo_size(aic->fifo)))
85  return 0;
86 
87  if (av_new_packet(pkt, size) < 0)
88  return AVERROR(ENOMEM);
89  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
90 
91  pkt->dts = pkt->pts = aic->dts;
92  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
93  pkt->stream_index = stream_index;
94  aic->dts += pkt->duration;
95 
96  aic->samples++;
97  if (!*aic->samples)
98  aic->samples = aic->samples_per_frame;
99 
100  return size;
101 }
102 
104  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
105  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
106 {
107  int i;
108 
109  if (pkt) {
110  AVStream *st = s->streams[pkt->stream_index];
112  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
113  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
114  if (new_size > aic->fifo_size) {
115  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
116  return -1;
117  aic->fifo_size = new_size;
118  }
119  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
120  } else {
121  int ret;
122  // rewrite pts and dts to be decoded time line position
123  pkt->pts = pkt->dts = aic->dts;
124  aic->dts += pkt->duration;
125  ret = ff_interleave_add_packet(s, pkt, compare_ts);
126  if (ret < 0)
127  return ret;
128  }
129  pkt = NULL;
130  }
131 
132  for (i = 0; i < s->nb_streams; i++) {
133  AVStream *st = s->streams[i];
134  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
135  AVPacket new_pkt;
136  int ret;
137  while ((ret = ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
138  ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
139  if (ret < 0)
140  return ret;
141  }
142  if (ret < 0)
143  return ret;
144  }
145  }
146 
147  return get_packet(s, out, NULL, flush);
148 }