1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
|
/*
* Audio Interleaving functions
*
* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/fifo.h"
#include "avformat.h"
#include "audiointerleave.h"
#include "internal.h"
void ff_audio_interleave_close(AVFormatContext *s)
{
int i;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO)
av_fifo_free(&aic->fifo);
}
}
int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
AVRational time_base)
{
int i;
if (!samples_per_frame)
return -1;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
aic->sample_size = (st->codec->channels *
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return -1;
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
av_fifo_init(&aic->fifo, 100 * *aic->samples);
}
}
return 0;
}
static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
int stream_index, int flush)
{
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
return 0;
av_new_packet(pkt, size);
av_fifo_read(&aic->fifo, pkt->data, size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return size;
}
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
{
int i;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
} else {
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
ff_interleave_add_packet(s, pkt, compare_ts);
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
AVPacket new_pkt;
while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
ff_interleave_add_packet(s, &new_pkt, compare_ts);
}
}
return get_packet(s, out, pkt, flush);
}
|
