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 Libav.
7  *
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.
12  *
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.
17  *
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
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  for (i = 0; i < s->nb_streams; i++) {
51  AVStream *st = s->streams[i];
53 
54  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
55  aic->sample_size = (st->codec->channels *
57  if (!aic->sample_size) {
58  av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
59  return -1;
60  }
61  aic->samples_per_frame = samples_per_frame;
62  aic->samples = aic->samples_per_frame;
63  aic->time_base = time_base;
64 
65  aic->fifo_size = 100* *aic->samples;
66  aic->fifo= av_fifo_alloc(100 * *aic->samples);
67  }
68  }
69 
70  return 0;
71 }
72 
74  int stream_index, int flush)
75 {
76  AVStream *st = s->streams[stream_index];
78 
79  int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
80  if (!size || (!flush && size == av_fifo_size(aic->fifo)))
81  return 0;
82 
83  av_new_packet(pkt, size);
84  av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
85 
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;
90 
91  aic->samples++;
92  if (!*aic->samples)
93  aic->samples = aic->samples_per_frame;
94 
95  return size;
96 }
97 
99  int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
100  int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
101 {
102  int i;
103 
104  if (pkt) {
105  AVStream *st = s->streams[pkt->stream_index];
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)
111  return -1;
112  aic->fifo_size = new_size;
113  }
114  av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
115  } else {
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);
120  }
121  pkt = NULL;
122  }
123 
124  for (i = 0; i < s->nb_streams; i++) {
125  AVStream *st = s->streams[i];
126  if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
127  AVPacket new_pkt;
128  while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
129  ff_interleave_add_packet(s, &new_pkt, compare_ts);
130  }
131  }
132 
133  return get_packet(s, out, pkt, flush);
134 }