audiointerleave.c 4.59 KB
Newer Older
1 2 3 4 5
/*
 * Audio Interleaving functions
 *
 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
 *
6
 * This file is part of Libav.
7
 *
8
 * Libav is free software; you can redistribute it and/or
9 10 11 12
 * 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.
 *
13
 * Libav is distributed in the hope that it will be useful,
14 15 16 17 18
 * 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
19
 * License along with Libav; if not, write to the Free Software
20 21 22 23
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/fifo.h"
24
#include "libavutil/mathematics.h"
25 26
#include "avformat.h"
#include "audiointerleave.h"
27
#include "internal.h"
28 29 30 31 32 33 34 35

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;

36
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
37
            av_fifo_free(aic->fifo);
38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53
    }
}

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;

54
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
55 56 57 58 59 60 61 62 63 64
            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;

65
            aic->fifo_size = 100* *aic->samples;
66
            aic->fifo= av_fifo_alloc(100 * *aic->samples);
67 68 69 70 71 72
        }
    }

    return 0;
}

73 74
static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
                                       int stream_index, int flush)
75 76 77
{
    AVStream *st = s->streams[stream_index];
    AudioInterleaveContext *aic = st->priv_data;
78
    int ret;
79 80
    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
81 82
        return 0;

83 84 85
    ret = av_new_packet(pkt, size);
    if (ret < 0)
        return ret;
86
    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
87 88 89 90 91 92 93 94 95 96 97 98 99

    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;
}

100
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
101 102 103
                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
                        int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
{
104
    int i, ret;
105 106 107 108

    if (pkt) {
        AVStream *st = s->streams[pkt->stream_index];
        AudioInterleaveContext *aic = st->priv_data;
109
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
110
            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
111
            if (new_size > aic->fifo_size) {
112
                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
113 114 115
                    return -1;
                aic->fifo_size = new_size;
            }
116
            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
117 118
        } else {
            // rewrite pts and dts to be decoded time line position
119
            pkt->pts = pkt->dts = aic->dts;
120
            aic->dts += pkt->duration;
121 122
            if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
                return ret;
123 124 125 126 127 128
        }
        pkt = NULL;
    }

    for (i = 0; i < s->nb_streams; i++) {
        AVStream *st = s->streams[i];
129
        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
130
            AVPacket new_pkt = { 0 };
131
            while (interleave_new_audio_packet(s, &new_pkt, i, flush))
132 133
                if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
                    return ret;
134 135 136
        }
    }

137
    return get_packet(s, out, NULL, flush);
138
}