view rtp.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.

/*
 * RTP definitions
 * Copyright (c) 2002 Fabrice Bellard
 *
 * 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_RTP_H
#define AVFORMAT_RTP_H

#include "libavcodec/avcodec.h"

/**
 * Return the payload type for a given codec.
 *
 * @param codec The context of the codec
 * @return In case of unknown payload type or dynamic payload type, a
 * negative value is returned; otherwise, the payload type (the 'PT' field
 * in the RTP header) is returned.
 */
int ff_rtp_get_payload_type(AVCodecContext *codec);

/**
 * Initialize a codec context based on the payload type.
 *
 * Fill the codec_type and codec_id fields of a codec context with
 * information depending on the payload type; for audio codecs, the
 * channels and sample_rate fields are also filled.
 *
 * @param codec The context of the codec
 * @param payload_type The payload type (the 'PT' field in the RTP header)
 * @return In case of unknown payload type or dynamic payload type, a
 * negative value is returned; otherwise, 0 is returned
 */
int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type);

/**
 * Return the encoding name (as defined in
 * http://www.iana.org/assignments/rtp-parameters) for a given payload type.
 *
 * @param payload_type The payload type (the 'PT' field in the RTP header)
 * @return In case of unknown payload type or dynamic payload type, a pointer
 * to an empty string is returned; otherwise, a pointer to a string containing
 * the encoding name is returned
 */
const char *ff_rtp_enc_name(int payload_type);

/**
 * Return the codec id for the given encoding name and codec type.
 *
 * @param buf A pointer to the string containing the encoding name
 * @param codec_type The codec type
 * @return In case of unknown encoding name, CODEC_ID_NONE is returned;
 * otherwise, the codec id is returned
 */
enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type);

#define RTP_PT_PRIVATE 96
#define RTP_VERSION 2
#define RTP_MAX_SDES 256   /**< maximum text length for SDES */

/* RTCP paquets use 0.5 % of the bandwidth */
#define RTCP_TX_RATIO_NUM 5
#define RTCP_TX_RATIO_DEN 1000

#endif /* AVFORMAT_RTP_H */