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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_freep(&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 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];
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 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 
79  int stream_index, int flush)
80 {
81  AVStream *st = s->streams[stream_index];
83  int ret;
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  ret = av_new_packet(pkt, size);
89  if (ret < 0)
90  return ret;
91  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
92 
93  pkt->dts = pkt->pts = aic->dts;
94  pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
95  pkt->stream_index = stream_index;
96  aic->dts += pkt->duration;
97 
98  aic->samples++;
99  if (!*aic->samples)
100  aic->samples = aic->samples_per_frame;
101 
102  return size;
103 }
104 
106  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
107  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
108 {
109  int i, ret;
110 
111  if (pkt) {
112  AVStream *st = s->streams[pkt->stream_index];
114  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
115  unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
116  if (new_size > aic->fifo_size) {
117  if (av_fifo_realloc2(aic->fifo, new_size) < 0)
118  return AVERROR(ENOMEM);
119  aic->fifo_size = new_size;
120  }
121  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
122  } else {
123  // rewrite pts and dts to be decoded time line position
124  pkt->pts = pkt->dts = aic->dts;
125  aic->dts += pkt->duration;
126  if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 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  while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
137  if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
138  return ret;
139  }
140  if (ret < 0)
141  return ret;
142  }
143  }
144 
145  return get_packet(s, out, NULL, flush);
146 }