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https://git.videolan.org/git/ffmpeg.git
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![Michael Niedermayer](/assets/img/avatar_default.png)
fixes the random dts/pts during encoding asf preroll fix no more initial zero frames for b frame encoding mpeg-es dts during demuxing fixed .ffm timestamp scale fixed, ffm is still broken though Originally committed as revision 3168 to svn://svn.ffmpeg.org/ffmpeg/trunk
752 lines
21 KiB
C
752 lines
21 KiB
C
/*
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* RTP input/output format
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* Copyright (c) 2002 Fabrice Bellard.
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*
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* This library 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 of the License, or (at your option) any later version.
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*
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* This library 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 this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "avformat.h"
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#include "mpegts.h"
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#include <unistd.h>
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#include <sys/types.h>
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#include <sys/socket.h>
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#include <netinet/in.h>
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#ifndef __BEOS__
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# include <arpa/inet.h>
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#else
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# include "barpainet.h"
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#endif
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#include <netdb.h>
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//#define DEBUG
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'url_open_dyn_packet_buf')
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*/
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#define RTP_VERSION 2
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#define RTP_MAX_SDES 256 /* maximum text length for SDES */
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/* RTCP paquets use 0.5 % of the bandwidth */
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#define RTCP_TX_RATIO_NUM 5
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#define RTCP_TX_RATIO_DEN 1000
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typedef enum {
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RTCP_SR = 200,
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RTCP_RR = 201,
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RTCP_SDES = 202,
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RTCP_BYE = 203,
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RTCP_APP = 204
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} rtcp_type_t;
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typedef enum {
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RTCP_SDES_END = 0,
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RTCP_SDES_CNAME = 1,
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RTCP_SDES_NAME = 2,
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RTCP_SDES_EMAIL = 3,
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RTCP_SDES_PHONE = 4,
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RTCP_SDES_LOC = 5,
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RTCP_SDES_TOOL = 6,
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RTCP_SDES_NOTE = 7,
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RTCP_SDES_PRIV = 8,
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RTCP_SDES_IMG = 9,
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RTCP_SDES_DOOR = 10,
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RTCP_SDES_SOURCE = 11
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} rtcp_sdes_type_t;
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struct RTPDemuxContext {
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AVFormatContext *ic;
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AVStream *st;
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int payload_type;
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uint32_t ssrc;
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uint16_t seq;
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uint32_t timestamp;
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uint32_t base_timestamp;
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uint32_t cur_timestamp;
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int max_payload_size;
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MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
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int read_buf_index;
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int read_buf_size;
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/* rtcp sender statistics receive */
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int64_t last_rtcp_ntp_time;
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int64_t first_rtcp_ntp_time;
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uint32_t last_rtcp_timestamp;
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/* rtcp sender statistics */
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unsigned int packet_count;
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unsigned int octet_count;
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unsigned int last_octet_count;
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int first_packet;
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/* buffer for output */
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uint8_t buf[RTP_MAX_PACKET_LENGTH];
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uint8_t *buf_ptr;
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};
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int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
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{
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switch(payload_type) {
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case RTP_PT_ULAW:
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_id = CODEC_ID_PCM_MULAW;
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codec->channels = 1;
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codec->sample_rate = 8000;
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break;
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case RTP_PT_ALAW:
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_id = CODEC_ID_PCM_ALAW;
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codec->channels = 1;
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codec->sample_rate = 8000;
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break;
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case RTP_PT_S16BE_STEREO:
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_id = CODEC_ID_PCM_S16BE;
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codec->channels = 2;
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codec->sample_rate = 44100;
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break;
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case RTP_PT_S16BE_MONO:
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_id = CODEC_ID_PCM_S16BE;
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codec->channels = 1;
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codec->sample_rate = 44100;
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break;
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case RTP_PT_MPEGAUDIO:
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_id = CODEC_ID_MP2;
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break;
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case RTP_PT_JPEG:
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codec->codec_type = CODEC_TYPE_VIDEO;
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codec->codec_id = CODEC_ID_MJPEG;
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break;
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case RTP_PT_MPEGVIDEO:
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codec->codec_type = CODEC_TYPE_VIDEO;
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codec->codec_id = CODEC_ID_MPEG1VIDEO;
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break;
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case RTP_PT_MPEG2TS:
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codec->codec_type = CODEC_TYPE_DATA;
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codec->codec_id = CODEC_ID_MPEG2TS;
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break;
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default:
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return -1;
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}
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return 0;
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}
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/* return < 0 if unknown payload type */
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int rtp_get_payload_type(AVCodecContext *codec)
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{
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int payload_type;
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/* compute the payload type */
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payload_type = -1;
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switch(codec->codec_id) {
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case CODEC_ID_PCM_MULAW:
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payload_type = RTP_PT_ULAW;
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break;
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case CODEC_ID_PCM_ALAW:
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payload_type = RTP_PT_ALAW;
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break;
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case CODEC_ID_PCM_S16BE:
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if (codec->channels == 1) {
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payload_type = RTP_PT_S16BE_MONO;
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} else if (codec->channels == 2) {
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payload_type = RTP_PT_S16BE_STEREO;
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}
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break;
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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payload_type = RTP_PT_MPEGAUDIO;
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break;
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case CODEC_ID_MJPEG:
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payload_type = RTP_PT_JPEG;
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break;
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case CODEC_ID_MPEG1VIDEO:
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payload_type = RTP_PT_MPEGVIDEO;
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break;
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case CODEC_ID_MPEG2TS:
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payload_type = RTP_PT_MPEG2TS;
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break;
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default:
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break;
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}
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return payload_type;
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}
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static inline uint32_t decode_be32(const uint8_t *p)
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{
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return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
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}
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static inline uint64_t decode_be64(const uint8_t *p)
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{
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return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
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}
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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{
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if (buf[1] != 200)
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return -1;
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s->last_rtcp_ntp_time = decode_be64(buf + 8);
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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s->last_rtcp_timestamp = decode_be32(buf + 16);
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return 0;
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}
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/**
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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*/
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
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{
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RTPDemuxContext *s;
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s = av_mallocz(sizeof(RTPDemuxContext));
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if (!s)
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return NULL;
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s->payload_type = payload_type;
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->ic = s1;
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s->st = st;
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if (payload_type == RTP_PT_MPEG2TS) {
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s->ts = mpegts_parse_open(s->ic);
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if (s->ts == NULL) {
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av_free(s);
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return NULL;
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}
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} else {
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switch(st->codec.codec_id) {
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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case CODEC_ID_MPEG4:
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st->need_parsing = 1;
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break;
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default:
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break;
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}
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}
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return s;
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}
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/**
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* Parse an RTP or RTCP packet directly sent as a buffer.
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* @param s RTP parse context.
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* @param pkt returned packet
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* @param buf input buffer or NULL to read the next packets
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* @param len buffer len
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* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
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* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
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*/
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len)
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{
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unsigned int ssrc, h;
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int payload_type, seq, delta_timestamp, ret;
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AVStream *st;
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uint32_t timestamp;
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if (!buf) {
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/* return the next packets, if any */
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if (s->read_buf_index >= s->read_buf_size)
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return -1;
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ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
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s->read_buf_size - s->read_buf_index);
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if (ret < 0)
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return -1;
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s->read_buf_index += ret;
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if (s->read_buf_index < s->read_buf_size)
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return 1;
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else
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return 0;
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}
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if (len < 12)
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return -1;
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if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
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return -1;
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if (buf[1] >= 200 && buf[1] <= 204) {
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rtcp_parse_packet(s, buf, len);
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return -1;
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}
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payload_type = buf[1] & 0x7f;
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seq = (buf[2] << 8) | buf[3];
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timestamp = decode_be32(buf + 4);
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ssrc = decode_be32(buf + 8);
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/* NOTE: we can handle only one payload type */
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if (s->payload_type != payload_type)
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return -1;
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#if defined(DEBUG) || 1
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if (seq != ((s->seq + 1) & 0xffff)) {
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av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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payload_type, seq, ((s->seq + 1) & 0xffff));
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}
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s->seq = seq;
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#endif
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len -= 12;
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buf += 12;
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st = s->st;
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if (!st) {
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/* specific MPEG2TS demux support */
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ret = mpegts_parse_packet(s->ts, pkt, buf, len);
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if (ret < 0)
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return -1;
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if (ret < len) {
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s->read_buf_size = len - ret;
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memcpy(s->buf, buf + ret, s->read_buf_size);
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s->read_buf_index = 0;
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return 1;
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}
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} else {
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switch(st->codec.codec_id) {
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case CODEC_ID_MP2:
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/* better than nothing: skip mpeg audio RTP header */
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if (len <= 4)
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return -1;
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h = decode_be32(buf);
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len -= 4;
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buf += 4;
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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case CODEC_ID_MPEG1VIDEO:
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/* better than nothing: skip mpeg video RTP header */
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if (len <= 4)
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return -1;
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h = decode_be32(buf);
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buf += 4;
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len -= 4;
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if (h & (1 << 26)) {
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/* mpeg2 */
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if (len <= 4)
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return -1;
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buf += 4;
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len -= 4;
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}
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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default:
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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break;
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}
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switch(st->codec.codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MPEG1VIDEO:
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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int64_t addend;
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/* XXX: is it really necessary to unify the timestamp base ? */
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to 90 kHz without overflow */
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addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
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addend = (addend * 5625) >> 14;
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pkt->pts = addend + delta_timestamp;
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}
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break;
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default:
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/* no timestamp info yet */
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break;
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}
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pkt->stream_index = s->st->index;
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}
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return 0;
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}
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void rtp_parse_close(RTPDemuxContext *s)
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{
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if (s->payload_type == RTP_PT_MPEG2TS) {
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mpegts_parse_close(s->ts);
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}
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av_free(s);
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}
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/* rtp output */
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static int rtp_write_header(AVFormatContext *s1)
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{
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RTPDemuxContext *s = s1->priv_data;
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int payload_type, max_packet_size, n;
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AVStream *st;
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if (s1->nb_streams != 1)
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return -1;
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st = s1->streams[0];
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payload_type = rtp_get_payload_type(&st->codec);
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if (payload_type < 0)
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payload_type = RTP_PT_PRIVATE; /* private payload type */
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s->payload_type = payload_type;
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s->base_timestamp = random();
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s->timestamp = s->base_timestamp;
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s->ssrc = random();
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s->first_packet = 1;
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max_packet_size = url_fget_max_packet_size(&s1->pb);
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if (max_packet_size <= 12)
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return AVERROR_IO;
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s->max_payload_size = max_packet_size - 12;
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switch(st->codec.codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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s->buf_ptr = s->buf + 4;
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s->cur_timestamp = 0;
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break;
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case CODEC_ID_MPEG1VIDEO:
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s->cur_timestamp = 0;
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break;
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case CODEC_ID_MPEG2TS:
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n = s->max_payload_size / TS_PACKET_SIZE;
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if (n < 1)
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n = 1;
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s->max_payload_size = n * TS_PACKET_SIZE;
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s->buf_ptr = s->buf;
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break;
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default:
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s->buf_ptr = s->buf;
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break;
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}
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return 0;
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}
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/* send an rtcp sender report packet */
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static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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{
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RTPDemuxContext *s = s1->priv_data;
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#if defined(DEBUG)
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printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
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#endif
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put_byte(&s1->pb, (RTP_VERSION << 6));
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put_byte(&s1->pb, 200);
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put_be16(&s1->pb, 6); /* length in words - 1 */
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put_be32(&s1->pb, s->ssrc);
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put_be64(&s1->pb, ntp_time);
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put_be32(&s1->pb, s->timestamp);
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put_be32(&s1->pb, s->packet_count);
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put_be32(&s1->pb, s->octet_count);
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put_flush_packet(&s1->pb);
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}
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/* send an rtp packet. sequence number is incremented, but the caller
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must update the timestamp itself */
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static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
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{
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RTPDemuxContext *s = s1->priv_data;
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#ifdef DEBUG
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printf("rtp_send_data size=%d\n", len);
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#endif
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/* build the RTP header */
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put_byte(&s1->pb, (RTP_VERSION << 6));
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put_byte(&s1->pb, s->payload_type & 0x7f);
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put_be16(&s1->pb, s->seq);
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put_be32(&s1->pb, s->timestamp);
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put_be32(&s1->pb, s->ssrc);
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put_buffer(&s1->pb, buf1, len);
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put_flush_packet(&s1->pb);
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s->seq++;
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s->octet_count += len;
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s->packet_count++;
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}
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/* send an integer number of samples and compute time stamp and fill
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the rtp send buffer before sending. */
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static void rtp_send_samples(AVFormatContext *s1,
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const uint8_t *buf1, int size, int sample_size)
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{
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RTPDemuxContext *s = s1->priv_data;
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int len, max_packet_size, n;
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max_packet_size = (s->max_payload_size / sample_size) * sample_size;
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/* not needed, but who nows */
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if ((size % sample_size) != 0)
|
|
av_abort();
|
|
while (size > 0) {
|
|
len = (max_packet_size - (s->buf_ptr - s->buf));
|
|
if (len > size)
|
|
len = size;
|
|
|
|
/* copy data */
|
|
memcpy(s->buf_ptr, buf1, len);
|
|
s->buf_ptr += len;
|
|
buf1 += len;
|
|
size -= len;
|
|
n = (s->buf_ptr - s->buf);
|
|
/* if buffer full, then send it */
|
|
if (n >= max_packet_size) {
|
|
rtp_send_data(s1, s->buf, n);
|
|
s->buf_ptr = s->buf;
|
|
/* update timestamp */
|
|
s->timestamp += n / sample_size;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* NOTE: we suppose that exactly one frame is given as argument here */
|
|
/* XXX: test it */
|
|
static void rtp_send_mpegaudio(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int len, count, max_packet_size;
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
/* test if we must flush because not enough space */
|
|
len = (s->buf_ptr - s->buf);
|
|
if ((len + size) > max_packet_size) {
|
|
if (len > 4) {
|
|
rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
|
|
s->buf_ptr = s->buf + 4;
|
|
/* 90 KHz time stamp */
|
|
s->timestamp = s->base_timestamp +
|
|
(s->cur_timestamp * 90000LL) / st->codec.sample_rate;
|
|
}
|
|
}
|
|
|
|
/* add the packet */
|
|
if (size > max_packet_size) {
|
|
/* big packet: fragment */
|
|
count = 0;
|
|
while (size > 0) {
|
|
len = max_packet_size - 4;
|
|
if (len > size)
|
|
len = size;
|
|
/* build fragmented packet */
|
|
s->buf[0] = 0;
|
|
s->buf[1] = 0;
|
|
s->buf[2] = count >> 8;
|
|
s->buf[3] = count;
|
|
memcpy(s->buf + 4, buf1, len);
|
|
rtp_send_data(s1, s->buf, len + 4);
|
|
size -= len;
|
|
buf1 += len;
|
|
count += len;
|
|
}
|
|
} else {
|
|
if (s->buf_ptr == s->buf + 4) {
|
|
/* no fragmentation possible */
|
|
s->buf[0] = 0;
|
|
s->buf[1] = 0;
|
|
s->buf[2] = 0;
|
|
s->buf[3] = 0;
|
|
}
|
|
memcpy(s->buf_ptr, buf1, size);
|
|
s->buf_ptr += size;
|
|
}
|
|
s->cur_timestamp += st->codec.frame_size;
|
|
}
|
|
|
|
/* NOTE: a single frame must be passed with sequence header if
|
|
needed. XXX: use slices. */
|
|
static void rtp_send_mpegvideo(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int len, h, max_packet_size;
|
|
uint8_t *q;
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
while (size > 0) {
|
|
/* XXX: more correct headers */
|
|
h = 0;
|
|
if (st->codec.sub_id == 2)
|
|
h |= 1 << 26; /* mpeg 2 indicator */
|
|
q = s->buf;
|
|
*q++ = h >> 24;
|
|
*q++ = h >> 16;
|
|
*q++ = h >> 8;
|
|
*q++ = h;
|
|
|
|
if (st->codec.sub_id == 2) {
|
|
h = 0;
|
|
*q++ = h >> 24;
|
|
*q++ = h >> 16;
|
|
*q++ = h >> 8;
|
|
*q++ = h;
|
|
}
|
|
|
|
len = max_packet_size - (q - s->buf);
|
|
if (len > size)
|
|
len = size;
|
|
|
|
memcpy(q, buf1, len);
|
|
q += len;
|
|
|
|
/* 90 KHz time stamp */
|
|
s->timestamp = s->base_timestamp +
|
|
av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
|
|
rtp_send_data(s1, s->buf, q - s->buf);
|
|
|
|
buf1 += len;
|
|
size -= len;
|
|
}
|
|
s->cur_timestamp++;
|
|
}
|
|
|
|
static void rtp_send_raw(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int len, max_packet_size;
|
|
|
|
max_packet_size = s->max_payload_size;
|
|
|
|
while (size > 0) {
|
|
len = max_packet_size;
|
|
if (len > size)
|
|
len = size;
|
|
|
|
/* 90 KHz time stamp */
|
|
s->timestamp = s->base_timestamp +
|
|
av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
|
|
rtp_send_data(s1, buf1, len);
|
|
|
|
buf1 += len;
|
|
size -= len;
|
|
}
|
|
s->cur_timestamp++;
|
|
}
|
|
|
|
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
|
|
static void rtp_send_mpegts_raw(AVFormatContext *s1,
|
|
const uint8_t *buf1, int size)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
int len, out_len;
|
|
|
|
while (size >= TS_PACKET_SIZE) {
|
|
len = s->max_payload_size - (s->buf_ptr - s->buf);
|
|
if (len > size)
|
|
len = size;
|
|
memcpy(s->buf_ptr, buf1, len);
|
|
buf1 += len;
|
|
size -= len;
|
|
s->buf_ptr += len;
|
|
|
|
out_len = s->buf_ptr - s->buf;
|
|
if (out_len >= s->max_payload_size) {
|
|
rtp_send_data(s1, s->buf, out_len);
|
|
s->buf_ptr = s->buf;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* write an RTP packet. 'buf1' must contain a single specific frame. */
|
|
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
{
|
|
RTPDemuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int rtcp_bytes;
|
|
int64_t ntp_time;
|
|
int size= pkt->size;
|
|
uint8_t *buf1= pkt->data;
|
|
|
|
#ifdef DEBUG
|
|
printf("%d: write len=%d\n", pkt->stream_index, size);
|
|
#endif
|
|
|
|
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
|
|
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
|
|
RTCP_TX_RATIO_DEN;
|
|
if (s->first_packet || rtcp_bytes >= 28) {
|
|
/* compute NTP time */
|
|
/* XXX: 90 kHz timestamp hardcoded */
|
|
ntp_time = (pkt->pts << 28) / 5625;
|
|
rtcp_send_sr(s1, ntp_time);
|
|
s->last_octet_count = s->octet_count;
|
|
s->first_packet = 0;
|
|
}
|
|
|
|
switch(st->codec.codec_id) {
|
|
case CODEC_ID_PCM_MULAW:
|
|
case CODEC_ID_PCM_ALAW:
|
|
case CODEC_ID_PCM_U8:
|
|
case CODEC_ID_PCM_S8:
|
|
rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
|
|
break;
|
|
case CODEC_ID_PCM_U16BE:
|
|
case CODEC_ID_PCM_U16LE:
|
|
case CODEC_ID_PCM_S16BE:
|
|
case CODEC_ID_PCM_S16LE:
|
|
rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
|
|
break;
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MP3:
|
|
rtp_send_mpegaudio(s1, buf1, size);
|
|
break;
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
rtp_send_mpegvideo(s1, buf1, size);
|
|
break;
|
|
case CODEC_ID_MPEG2TS:
|
|
rtp_send_mpegts_raw(s1, buf1, size);
|
|
break;
|
|
default:
|
|
/* better than nothing : send the codec raw data */
|
|
rtp_send_raw(s1, buf1, size);
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_write_trailer(AVFormatContext *s1)
|
|
{
|
|
// RTPDemuxContext *s = s1->priv_data;
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat rtp_mux = {
|
|
"rtp",
|
|
"RTP output format",
|
|
NULL,
|
|
NULL,
|
|
sizeof(RTPDemuxContext),
|
|
CODEC_ID_PCM_MULAW,
|
|
CODEC_ID_NONE,
|
|
rtp_write_header,
|
|
rtp_write_packet,
|
|
rtp_write_trailer,
|
|
};
|
|
|
|
int rtp_init(void)
|
|
{
|
|
av_register_output_format(&rtp_mux);
|
|
return 0;
|
|
}
|