Reorder arguments for av_fifo_generic_read to be more logical and
[libav.git] / libavformat / audiointerleave.c
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 "avformat.h"
25 #include "audiointerleave.h"
26 #include "internal.h"
27
28 void ff_audio_interleave_close(AVFormatContext *s)
29 {
30 int i;
31 for (i = 0; i < s->nb_streams; i++) {
32 AVStream *st = s->streams[i];
33 AudioInterleaveContext *aic = st->priv_data;
34
35 if (st->codec->codec_type == CODEC_TYPE_AUDIO)
36 av_fifo_free(aic->fifo);
37 }
38 }
39
40 int ff_audio_interleave_init(AVFormatContext *s,
41 const int *samples_per_frame,
42 AVRational time_base)
43 {
44 int i;
45
46 if (!samples_per_frame)
47 return -1;
48
49 for (i = 0; i < s->nb_streams; i++) {
50 AVStream *st = s->streams[i];
51 AudioInterleaveContext *aic = st->priv_data;
52
53 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
54 aic->sample_size = (st->codec->channels *
55 av_get_bits_per_sample(st->codec->codec_id)) / 8;
56 if (!aic->sample_size) {
57 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
58 return -1;
59 }
60 aic->samples_per_frame = samples_per_frame;
61 aic->samples = aic->samples_per_frame;
62 aic->time_base = time_base;
63
64 aic->fifo_size = 100* *aic->samples;
65 aic->fifo= av_fifo_alloc(100 * *aic->samples);
66 }
67 }
68
69 return 0;
70 }
71
72 static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
73 int stream_index, int flush)
74 {
75 AVStream *st = s->streams[stream_index];
76 AudioInterleaveContext *aic = st->priv_data;
77
78 int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
79 if (!size || (!flush && size == av_fifo_size(aic->fifo)))
80 return 0;
81
82 av_new_packet(pkt, size);
83 av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
84
85 pkt->dts = pkt->pts = aic->dts;
86 pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
87 pkt->stream_index = stream_index;
88 aic->dts += pkt->duration;
89
90 aic->samples++;
91 if (!*aic->samples)
92 aic->samples = aic->samples_per_frame;
93
94 return size;
95 }
96
97 int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
98 int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
99 int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
100 {
101 int i;
102
103 if (pkt) {
104 AVStream *st = s->streams[pkt->stream_index];
105 AudioInterleaveContext *aic = st->priv_data;
106 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
107 unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
108 if (new_size > aic->fifo_size) {
109 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
110 return -1;
111 aic->fifo_size = new_size;
112 }
113 av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
114 } else {
115 // rewrite pts and dts to be decoded time line position
116 pkt->pts = pkt->dts = aic->dts;
117 aic->dts += pkt->duration;
118 ff_interleave_add_packet(s, pkt, compare_ts);
119 }
120 pkt = NULL;
121 }
122
123 for (i = 0; i < s->nb_streams; i++) {
124 AVStream *st = s->streams[i];
125 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
126 AVPacket new_pkt;
127 while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
128 ff_interleave_add_packet(s, &new_pkt, compare_ts);
129 }
130 }
131
132 return get_packet(s, out, pkt, flush);
133 }