view audiointerleave.c @ 5836:2997c88028cd libavformat

Move the probe loop from av_open_input_file() into its own method av_probe_input_buffer() so that it can be reused. Here are a few differences to the original way things were probed: - maximum probe buffer size can be specified as a parameter. - offset within the stream to probe from can be specified as a parameter. - instead of seeking back to the start each time a probe fails, stream data is appended to the reallocated buffer. This lowers the amount of data read from the stream (there is no repetition) and results in fewer closed and reopened streams (when seeking fails). New attempt after r22296, which was revert in r22315 due to a FATE failure. See the thread: Subject: [FFmpeg-devel] [PATCH] Move av_open_input_file probe loop to its own method Date: 2010-03-05 03:23:57 GMT Patch by Micah F. Galizia printf("%s%s@%s.%s", "micah", "galizia", "gmail", "com").

/*
 * 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;

            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 == CODEC_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 {
            // 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);
}