view audiointerleave.c @ 5854:3dc78fb78b2a libavformat

Fix erroneous behaviour when format probe hits end of file If the format probe hits end of file, do not add the error code to the buffer position. This is obviously wrong, and with a small input file would cause a negative buffer overflow. Fixes issue 1818.

/*
 * 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);
}