Mercurial > libavformat.hg
annotate rtpdec.c @ 4387:5c42816e12c6 libavformat
Add "AVFormatContext *ctx" (that being the RTSP demuxer's) as first argument
to the parse_packet() function pointer in RTPDynamicProtocolHandlers. This
allows these functions to peek back and retrieve values from the demuxer's
context (or RTSPState). The ASF/RTP payload parser will use this to be able
to parse SDP values (which occur even before the payload ID is given), store
them in the RTSPState and then retrieve them while parsing payload data. See
"[PATCH] RTSP-MS 13/15: add RTSP demuxer AVFormatContext to parse_packet()
function pointer (was: transport context)" mailinglist thread.
author | rbultje |
---|---|
date | Fri, 06 Feb 2009 01:37:19 +0000 |
parents | f49e5d92ab26 |
children | 80f21f72d7d6 |
rev | line source |
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0 | 1 /* |
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2 * RTP input format |
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3 * Copyright (c) 2002 Fabrice Bellard |
0 | 4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
3286 | 21 |
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22 /* needed for gethostname() */ |
3955 | 23 #define _XOPEN_SOURCE 600 |
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24 |
3286 | 25 #include "libavcodec/bitstream.h" |
0 | 26 #include "avformat.h" |
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27 #include "mpegts.h" |
0 | 28 |
29 #include <unistd.h> | |
1754 | 30 #include "network.h" |
0 | 31 |
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32 #include "rtp.h" |
1460 | 33 #include "rtp_h264.h" |
1419 | 34 |
0 | 35 //#define DEBUG |
36 | |
37 /* TODO: - add RTCP statistics reporting (should be optional). | |
38 | |
39 - add support for h263/mpeg4 packetized output : IDEA: send a | |
40 buffer to 'rtp_write_packet' contains all the packets for ONE | |
41 frame. Each packet should have a four byte header containing | |
42 the length in big endian format (same trick as | |
885 | 43 'url_open_dyn_packet_buf') |
0 | 44 */ |
45 | |
1419 | 46 /* statistics functions */ |
47 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
48 | |
49 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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50 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 51 |
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52 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
1419 | 53 { |
54 handler->next= RTPFirstDynamicPayloadHandler; | |
55 RTPFirstDynamicPayloadHandler= handler; | |
56 } | |
885 | 57 |
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58 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 59 { |
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60 ff_register_dynamic_payload_handler(&mp4v_es_handler); |
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61 ff_register_dynamic_payload_handler(&mpeg4_generic_handler); |
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62 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); |
1419 | 63 } |
0 | 64 |
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65 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 66 { |
67 if (buf[1] != 200) | |
68 return -1; | |
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69 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
173 | 70 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
71 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
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72 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 73 return 0; |
74 } | |
75 | |
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76 #define RTP_SEQ_MOD (1<<16) |
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77 |
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78 /** |
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79 * called on parse open packet |
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80 */ |
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81 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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82 { |
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83 memset(s, 0, sizeof(RTPStatistics)); |
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84 s->max_seq= base_sequence; |
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85 s->probation= 1; |
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86 } |
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87 |
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88 /** |
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89 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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90 */ |
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91 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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92 { |
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93 s->max_seq= seq; |
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94 s->cycles= 0; |
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95 s->base_seq= seq -1; |
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96 s->bad_seq= RTP_SEQ_MOD + 1; |
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97 s->received= 0; |
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98 s->expected_prior= 0; |
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99 s->received_prior= 0; |
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100 s->jitter= 0; |
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101 s->transit= 0; |
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102 } |
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103 |
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104 /** |
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105 * returns 1 if we should handle this packet. |
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106 */ |
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107 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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108 { |
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109 uint16_t udelta= seq - s->max_seq; |
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110 const int MAX_DROPOUT= 3000; |
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111 const int MAX_MISORDER = 100; |
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112 const int MIN_SEQUENTIAL = 2; |
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113 |
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114 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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115 if(s->probation) |
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116 { |
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117 if(seq==s->max_seq + 1) { |
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118 s->probation--; |
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119 s->max_seq= seq; |
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120 if(s->probation==0) { |
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121 rtp_init_sequence(s, seq); |
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122 s->received++; |
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123 return 1; |
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124 } |
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125 } else { |
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126 s->probation= MIN_SEQUENTIAL - 1; |
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127 s->max_seq = seq; |
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128 } |
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129 } else if (udelta < MAX_DROPOUT) { |
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130 // in order, with permissible gap |
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131 if(seq < s->max_seq) { |
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132 //sequence number wrapped; count antother 64k cycles |
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133 s->cycles += RTP_SEQ_MOD; |
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134 } |
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135 s->max_seq= seq; |
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136 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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137 // sequence made a large jump... |
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138 if(seq==s->bad_seq) { |
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139 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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140 rtp_init_sequence(s, seq); |
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141 } else { |
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142 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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143 return 0; |
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144 } |
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145 } else { |
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146 // duplicate or reordered packet... |
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147 } |
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148 s->received++; |
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149 return 1; |
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150 } |
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151 |
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152 #if 0 |
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153 /** |
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154 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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155 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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156 * never change. I left this in in case someone else can see a way. (rdm) |
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157 */ |
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158 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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159 { |
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160 uint32_t transit= arrival_timestamp - sent_timestamp; |
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161 int d; |
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162 s->transit= transit; |
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163 d= FFABS(transit - s->transit); |
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164 s->jitter += d - ((s->jitter + 8)>>4); |
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165 } |
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166 #endif |
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167 |
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168 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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169 { |
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170 ByteIOContext *pb; |
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171 uint8_t *buf; |
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172 int len; |
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173 int rtcp_bytes; |
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174 RTPStatistics *stats= &s->statistics; |
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175 uint32_t lost; |
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176 uint32_t extended_max; |
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177 uint32_t expected_interval; |
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178 uint32_t received_interval; |
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179 uint32_t lost_interval; |
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180 uint32_t expected; |
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181 uint32_t fraction; |
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182 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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183 |
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184 if (!s->rtp_ctx || (count < 1)) |
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185 return -1; |
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186 |
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187 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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188 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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189 s->octet_count += count; |
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190 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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191 RTCP_TX_RATIO_DEN; |
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192 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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193 if (rtcp_bytes < 28) |
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194 return -1; |
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195 s->last_octet_count = s->octet_count; |
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196 |
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197 if (url_open_dyn_buf(&pb) < 0) |
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198 return -1; |
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199 |
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200 // Receiver Report |
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201 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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202 put_byte(pb, 201); |
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203 put_be16(pb, 7); /* length in words - 1 */ |
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204 put_be32(pb, s->ssrc); // our own SSRC |
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205 put_be32(pb, s->ssrc); // XXX: should be the server's here! |
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206 // some placeholders we should really fill... |
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207 // RFC 1889/p64 |
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208 extended_max= stats->cycles + stats->max_seq; |
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209 expected= extended_max - stats->base_seq + 1; |
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210 lost= expected - stats->received; |
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211 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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212 expected_interval= expected - stats->expected_prior; |
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213 stats->expected_prior= expected; |
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214 received_interval= stats->received - stats->received_prior; |
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215 stats->received_prior= stats->received; |
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216 lost_interval= expected_interval - received_interval; |
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217 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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218 else fraction = (lost_interval<<8)/expected_interval; |
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219 |
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220 fraction= (fraction<<24) | lost; |
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221 |
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222 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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223 put_be32(pb, extended_max); /* max sequence received */ |
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224 put_be32(pb, stats->jitter>>4); /* jitter */ |
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225 |
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226 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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227 { |
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228 put_be32(pb, 0); /* last SR timestamp */ |
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229 put_be32(pb, 0); /* delay since last SR */ |
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230 } else { |
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231 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special? |
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232 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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233 |
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234 put_be32(pb, middle_32_bits); /* last SR timestamp */ |
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235 put_be32(pb, delay_since_last); /* delay since last SR */ |
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236 } |
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237 |
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238 // CNAME |
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239 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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240 put_byte(pb, 202); |
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241 len = strlen(s->hostname); |
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242 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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243 put_be32(pb, s->ssrc); |
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244 put_byte(pb, 0x01); |
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245 put_byte(pb, len); |
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246 put_buffer(pb, s->hostname, len); |
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247 // padding |
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248 for (len = (6 + len) % 4; len % 4; len++) { |
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249 put_byte(pb, 0); |
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250 } |
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251 |
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252 put_flush_packet(pb); |
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253 len = url_close_dyn_buf(pb, &buf); |
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254 if ((len > 0) && buf) { |
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255 int result; |
3579 | 256 dprintf(s->ic, "sending %d bytes of RR\n", len); |
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257 result= url_write(s->rtp_ctx, buf, len); |
3579 | 258 dprintf(s->ic, "result from url_write: %d\n", result); |
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259 av_free(buf); |
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260 } |
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261 return 0; |
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262 } |
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263 |
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264 /** |
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265 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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266 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 267 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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268 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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269 */ |
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270 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data) |
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271 { |
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272 RTPDemuxContext *s; |
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273 |
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274 s = av_mallocz(sizeof(RTPDemuxContext)); |
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275 if (!s) |
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276 return NULL; |
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277 s->payload_type = payload_type; |
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278 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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279 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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280 s->ic = s1; |
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281 s->st = st; |
774 | 282 s->rtp_payload_data = rtp_payload_data; |
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283 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
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284 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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285 s->ts = mpegts_parse_open(s->ic); |
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286 if (s->ts == NULL) { |
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287 av_free(s); |
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288 return NULL; |
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289 } |
305 | 290 } else { |
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291 av_set_pts_info(st, 32, 1, 90000); |
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292 switch(st->codec->codec_id) { |
305 | 293 case CODEC_ID_MPEG1VIDEO: |
294 case CODEC_ID_MPEG2VIDEO: | |
295 case CODEC_ID_MP2: | |
296 case CODEC_ID_MP3: | |
297 case CODEC_ID_MPEG4: | |
1419 | 298 case CODEC_ID_H264: |
2023 | 299 st->need_parsing = AVSTREAM_PARSE_FULL; |
305 | 300 break; |
301 default: | |
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302 if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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303 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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304 } |
305 | 305 break; |
306 } | |
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307 } |
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308 // needed to send back RTCP RR in RTSP sessions |
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309 s->rtp_ctx = rtpc; |
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310 gethostname(s->hostname, sizeof(s->hostname)); |
294
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311 return s; |
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312 } |
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313 |
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314 void |
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315 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
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316 RTPDynamicProtocolHandler *handler) |
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317 { |
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318 s->dynamic_protocol_context = ctx; |
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319 s->parse_packet = handler->parse_packet; |
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320 } |
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321 |
774 | 322 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
323 { | |
324 int au_headers_length, au_header_size, i; | |
325 GetBitContext getbitcontext; | |
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326 RTPPayloadData *infos; |
774 | 327 |
328 infos = s->rtp_payload_data; | |
329 | |
330 if (infos == NULL) | |
331 return -1; | |
332 | |
3142 | 333 /* decode the first 2 bytes where the AUHeader sections are stored |
774 | 334 length in bits */ |
1673 | 335 au_headers_length = AV_RB16(buf); |
774 | 336 |
337 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
338 return -1; | |
339 | |
340 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
341 | |
342 /* skip AU headers length section (2 bytes) */ | |
343 buf += 2; | |
344 | |
345 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
346 | |
347 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
348 au_header_size = infos->sizelength + infos->indexlength; | |
349 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
350 return -1; | |
351 | |
352 infos->nb_au_headers = au_headers_length / au_header_size; | |
353 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
354 | |
355 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 356 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 357 but does when sending the whole as one big packet... */ |
358 infos->au_headers[0].size = 0; | |
359 infos->au_headers[0].index = 0; | |
360 for (i = 0; i < infos->nb_au_headers; ++i) { | |
361 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
362 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
363 } | |
364 | |
365 infos->nb_au_headers = 1; | |
366 | |
367 return 0; | |
368 } | |
369 | |
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370 /** |
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371 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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372 */ |
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373 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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374 { |
3535 | 375 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
376 int64_t addend; | |
377 int delta_timestamp; | |
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378 |
3535 | 379 /* compute pts from timestamp with received ntp_time */ |
380 delta_timestamp = timestamp - s->last_rtcp_timestamp; | |
381 /* convert to the PTS timebase */ | |
382 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32); | |
383 pkt->pts = addend + delta_timestamp; | |
384 } | |
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385 pkt->stream_index = s->st->index; |
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386 } |
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387 |
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388 /** |
885 | 389 * Parse an RTP or RTCP packet directly sent as a buffer. |
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390 * @param s RTP parse context. |
0 | 391 * @param pkt returned packet |
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392 * @param buf input buffer or NULL to read the next packets |
0 | 393 * @param len buffer len |
885 | 394 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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395 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 396 */ |
885 | 397 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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398 const uint8_t *buf, int len) |
0 | 399 { |
400 unsigned int ssrc, h; | |
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401 int payload_type, seq, ret, flags = 0; |
0 | 402 AVStream *st; |
65 | 403 uint32_t timestamp; |
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404 int rv= 0; |
885 | 405 |
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406 if (!buf) { |
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407 /* return the next packets, if any */ |
1419 | 408 if(s->st && s->parse_packet) { |
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409 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... |
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410 rv= s->parse_packet(s->ic, s->dynamic_protocol_context, |
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411 s->st, pkt, ×tamp, NULL, 0, flags); |
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412 finalize_packet(s, pkt, timestamp); |
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413 return rv; |
1419 | 414 } else { |
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415 // TODO: Move to a dynamic packet handler (like above) |
1424
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416 if (s->read_buf_index >= s->read_buf_size) |
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417 return -1; |
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418 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
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419 s->read_buf_size - s->read_buf_index); |
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420 if (ret < 0) |
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421 return -1; |
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422 s->read_buf_index += ret; |
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423 if (s->read_buf_index < s->read_buf_size) |
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424 return 1; |
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425 else |
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426 return 0; |
1419 | 427 } |
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428 } |
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429 |
0 | 430 if (len < 12) |
431 return -1; | |
432 | |
433 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
434 return -1; | |
435 if (buf[1] >= 200 && buf[1] <= 204) { | |
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436 rtcp_parse_packet(s, buf, len); |
0 | 437 return -1; |
438 } | |
439 payload_type = buf[1] & 0x7f; | |
2222 | 440 seq = AV_RB16(buf + 2); |
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441 timestamp = AV_RB32(buf + 4); |
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442 ssrc = AV_RB32(buf + 8); |
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443 /* store the ssrc in the RTPDemuxContext */ |
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444 s->ssrc = ssrc; |
885 | 445 |
0 | 446 /* NOTE: we can handle only one payload type */ |
447 if (s->payload_type != payload_type) | |
448 return -1; | |
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449 |
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450 st = s->st; |
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451 // only do something with this if all the rtp checks pass... |
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452 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
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453 { |
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454 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 455 payload_type, seq, ((s->seq + 1) & 0xffff)); |
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456 return -1; |
0 | 457 } |
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458 |
0 | 459 s->seq = seq; |
460 len -= 12; | |
461 buf += 12; | |
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462 |
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463 if (!st) { |
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464 /* specific MPEG2TS demux support */ |
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465 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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466 if (ret < 0) |
0 | 467 return -1; |
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468 if (ret < len) { |
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469 s->read_buf_size = len - ret; |
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470 memcpy(s->buf, buf + ret, s->read_buf_size); |
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471 s->read_buf_index = 0; |
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472 return 1; |
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473 } |
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474 } else if (s->parse_packet) { |
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475 rv = s->parse_packet(s->ic, s->dynamic_protocol_context, |
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476 s->st, pkt, ×tamp, buf, len, flags); |
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477 } else { |
1419 | 478 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise. |
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479 switch(st->codec->codec_id) { |
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480 case CODEC_ID_MP2: |
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481 /* better than nothing: skip mpeg audio RTP header */ |
0 | 482 if (len <= 4) |
483 return -1; | |
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484 h = AV_RB32(buf); |
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485 len -= 4; |
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486 buf += 4; |
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487 av_new_packet(pkt, len); |
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488 memcpy(pkt->data, buf, len); |
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489 break; |
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490 case CODEC_ID_MPEG1VIDEO: |
2760 | 491 case CODEC_ID_MPEG2VIDEO: |
305 | 492 /* better than nothing: skip mpeg video RTP header */ |
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493 if (len <= 4) |
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494 return -1; |
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495 h = AV_RB32(buf); |
0 | 496 buf += 4; |
497 len -= 4; | |
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498 if (h & (1 << 26)) { |
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499 /* mpeg2 */ |
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500 if (len <= 4) |
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501 return -1; |
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502 buf += 4; |
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503 len -= 4; |
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504 } |
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505 av_new_packet(pkt, len); |
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506 memcpy(pkt->data, buf, len); |
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507 break; |
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508 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles |
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509 // timestamps. |
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510 // TODO: Put this into a dynamic packet handler... |
1472
49d5a5ca2987
get rid of CODEC_ID_MPEG4AAC after next version bump, and change it to CODEC_ID_AAC where used
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511 case CODEC_ID_AAC: |
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512 if (rtp_parse_mp4_au(s, buf)) |
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513 return -1; |
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514 { |
4101
af2c0aef892b
Rename rtp_payload_data_t to avoid clashes with the POSIX namespace
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515 RTPPayloadData *infos = s->rtp_payload_data; |
1427
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516 if (infos == NULL) |
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517 return -1; |
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518 buf += infos->au_headers_length_bytes + 2; |
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519 len -= infos->au_headers_length_bytes + 2; |
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520 |
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521 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
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522 one au_header */ |
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523 av_new_packet(pkt, infos->au_headers[0].size); |
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524 memcpy(pkt->data, buf, infos->au_headers[0].size); |
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gpoirier
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changeset
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525 buf += infos->au_headers[0].size; |
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parents:
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changeset
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526 len -= infos->au_headers[0].size; |
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gpoirier
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527 } |
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528 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
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1427
diff
changeset
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529 rv= 0; |
1427
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changeset
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530 break; |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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changeset
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531 default: |
2918
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
532 av_new_packet(pkt, len); |
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
533 memcpy(pkt->data, buf, len); |
294
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534 break; |
0 | 535 } |
885 | 536 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
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diff
changeset
|
537 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
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diff
changeset
|
538 finalize_packet(s, pkt, timestamp); |
0 | 539 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
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diff
changeset
|
540 return rv; |
0 | 541 } |
542 | |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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543 void rtp_parse_close(RTPDemuxContext *s) |
0 | 544 { |
1419 | 545 // TODO: fold this into the protocol specific data fields. |
2759
b252e318023a
Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP
lucabe
parents:
2706
diff
changeset
|
546 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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547 mpegts_parse_close(s->ts); |
0 | 548 } |
294
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added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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549 av_free(s); |
0 | 550 } |