view audiointerleave.h @ 4633:0c69b895a01f libavformat

Don't let finalize_packet() touch pkt->stream_index. Instead, let individual payload handlers take care of that themselves at their own option. What this patch really does is "fix" a bug in MS-RTSP protocol where incoming packets are always coming in over the connection (UDP) or interleave-id (TCP) of the stream-id of the first ASF packet in the RTP packet. However, RTP packets may contain multiple ASF packets (and usually do, from what I can see), and therefore this leads to playback bugs. The intended stream-id per ASF packet is given in the respective ASF packet header. The ASF demuxer will correctly read this and set pkt->stream_index, but since the "stream" parameter can not be known to rtpdec.c or any of the RTP/RTSP code, the "st" parameter in all these functions is basically invalid. Therefore, using st->id as pkt->stream_index leads to various playback bugs. The result of this patch is that pkt->stream_index is left untouched for RTP/ASF (and possibly for other payloads that have similar behaviour). The patch was discussed in the "[PATCH] rtpdec.c: don't overwrite pkt->stream_index in finalize_packet()" thread on the mailinglist.

/*
 * audio interleaving prototypes and declarations
 *
 * 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
 */

#ifndef AVFORMAT_AUDIOINTERLEAVE_H
#define AVFORMAT_AUDIOINTERLEAVE_H

#include "libavutil/fifo.h"
#include "avformat.h"

typedef struct {
    AVFifoBuffer fifo;
    unsigned fifo_size;           ///< size of currently allocated FIFO
    uint64_t dts;                 ///< current dts
    int sample_size;              ///< size of one sample all channels included
    const int *samples_per_frame; ///< must be 0-terminated
    const int *samples;           ///< current samples per frame, pointer to samples_per_frame
    AVRational time_base;         ///< time base of output audio packets
} AudioInterleaveContext;

int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base);
void ff_audio_interleave_close(AVFormatContext *s);

int ff_interleave_compare_dts(AVFormatContext *s, AVPacket *next, AVPacket *pkt);
/**
 * Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame
 * and interleave them correctly.
 * The first element of AVStream->priv_data must be AudioInterleaveContext
 * when using this function.
 *
 * @param get_packet function will output a packet when streams are correctly interleaved.
 * @param compare_ts function will compare AVPackets and decide interleaving order.
 */
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 *));

#endif /* AVFORMAT_AUDIOINTERLEAVE_H */