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ffmpeg / libavformat / audiointerleave.c @ bd51c013

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/*
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 * Audio Interleaving functions
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 *
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 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/fifo.h"
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#include "avformat.h"
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#include "audiointerleave.h"
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#include "internal.h"
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void ff_audio_interleave_close(AVFormatContext *s)
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{
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    int i;
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    for (i = 0; i < s->nb_streams; i++) {
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        AVStream *st = s->streams[i];
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        AudioInterleaveContext *aic = st->priv_data;
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO)
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            av_fifo_free(aic->fifo);
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    }
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}
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int ff_audio_interleave_init(AVFormatContext *s,
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                             const int *samples_per_frame,
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                             AVRational time_base)
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{
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    int i;
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    if (!samples_per_frame)
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        return -1;
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    for (i = 0; i < s->nb_streams; i++) {
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        AVStream *st = s->streams[i];
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        AudioInterleaveContext *aic = st->priv_data;
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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            aic->sample_size = (st->codec->channels *
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                                av_get_bits_per_sample(st->codec->codec_id)) / 8;
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            if (!aic->sample_size) {
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                av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
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                return -1;
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            }
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            aic->samples_per_frame = samples_per_frame;
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            aic->samples = aic->samples_per_frame;
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            aic->time_base = time_base;
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            aic->fifo_size = 100* *aic->samples;
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            aic->fifo= av_fifo_alloc(100 * *aic->samples);
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        }
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    }
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    return 0;
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}
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static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
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                                   int stream_index, int flush)
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{
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    AVStream *st = s->streams[stream_index];
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    AudioInterleaveContext *aic = st->priv_data;
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    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
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    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
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        return 0;
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    av_new_packet(pkt, size);
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    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
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    pkt->dts = pkt->pts = aic->dts;
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    pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
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    pkt->stream_index = stream_index;
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    aic->dts += pkt->duration;
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    aic->samples++;
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    if (!*aic->samples)
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        aic->samples = aic->samples_per_frame;
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    return size;
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}
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int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
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                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
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                        int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
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{
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    int i;
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    if (pkt) {
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        AVStream *st = s->streams[pkt->stream_index];
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        AudioInterleaveContext *aic = st->priv_data;
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
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            if (new_size > aic->fifo_size) {
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                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
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                    return -1;
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                aic->fifo_size = new_size;
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            }
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            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
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        } else {
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            // rewrite pts and dts to be decoded time line position
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            pkt->pts = pkt->dts = aic->dts;
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            aic->dts += pkt->duration;
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            ff_interleave_add_packet(s, pkt, compare_ts);
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        }
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        pkt = NULL;
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    }
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    for (i = 0; i < s->nb_streams; i++) {
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        AVStream *st = s->streams[i];
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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            AVPacket new_pkt;
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            while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
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                ff_interleave_add_packet(s, &new_pkt, compare_ts);
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        }
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    }
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    return get_packet(s, out, pkt, flush);
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}