aboutsummaryrefslogtreecommitdiffstats
path: root/libavformat/audiointerleave.c
blob: b64010ff44c940a629f9e453eda3d6248bf9691c (plain) (blame)
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
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
/*
 * 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 "libavutil/mathematics.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 == AVMEDIA_TYPE_AUDIO)
            av_fifo_freep(&aic->fifo);
    }
}

int ff_audio_interleave_init(AVFormatContext *s,
                             const int *samples_per_frame,
                             AVRational time_base)
{
    int i;

    if (!samples_per_frame)
        return AVERROR(EINVAL);

    if (!time_base.num) {
        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
        return AVERROR(EINVAL);
    }
    for (i = 0; i < s->nb_streams; i++) {
        AVStream *st = s->streams[i];
        AudioInterleaveContext *aic = st->priv_data;

        if (st->codec->codec_type == AVMEDIA_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 AVERROR(EINVAL);
            }
            aic->samples_per_frame = samples_per_frame;
            aic->samples = aic->samples_per_frame;
            aic->time_base = time_base;

            aic->fifo_size = 100* *aic->samples;
            if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
                return AVERROR(ENOMEM);
        }
    }

    return 0;
}

static int 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 ret;
    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
        return 0;

    ret = av_new_packet(pkt, size);
    if (ret < 0)
        return ret;
    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);

    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, ret;

    if (pkt) {
        AVStream *st = s->streams[pkt->stream_index];
        AudioInterleaveContext *aic = st->priv_data;
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
            if (new_size > aic->fifo_size) {
                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
                    return AVERROR(ENOMEM);
                aic->fifo_size = new_size;
            }
            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;
            if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
                return ret;
        }
        pkt = NULL;
    }

    for (i = 0; i < s->nb_streams; i++) {
        AVStream *st = s->streams[i];
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            AVPacket new_pkt;
            while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
                if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
                    return ret;
            }
            if (ret < 0)
                return ret;
        }
    }

    return get_packet(s, out, NULL, flush);
}