Skip to content
This repository

HTTPS clone URL

Subversion checkout URL

You can clone with HTTPS or Subversion.

Download ZIP
tree: 1e9caf9c92
Fetching contributors…

Cannot retrieve contributors at this time

file 145 lines (126 sloc) 4.856 kb
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
/*
* 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_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;

    if (!time_base.num) {
        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
        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 == 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 -1;
            }
            aic->samples_per_frame = samples_per_frame;
            aic->samples = aic->samples_per_frame;
            aic->time_base = time_base;

            aic->fifo_size = 100* *aic->samples;
            aic->fifo= av_fifo_alloc(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_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;

    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 -1;
                aic->fifo_size = new_size;
            }
            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
        } else {
            int ret;
            // rewrite pts and dts to be decoded time line position
            pkt->pts = pkt->dts = aic->dts;
            aic->dts += pkt->duration;
            ret = ff_interleave_add_packet(s, pkt, compare_ts);
            if (ret < 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;
            int ret;
            while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush)) {
                ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
                if (ret < 0)
                    return ret;
            }
        }
    }

    return get_packet(s, out, pkt, flush);
}
Something went wrong with that request. Please try again.