Mercurial > libavformat.hg
annotate rtpdec.c @ 4105:8f778b7c07d0 libavformat
oggflac: fix bitstream reader usage
author | mru |
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date | Sun, 14 Dec 2008 17:30:18 +0000 |
parents | af2c0aef892b |
children | 77e0c7511d41 |
rev | line source |
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0 | 1 /* |
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2 * RTP input format |
0 | 3 * Copyright (c) 2002 Fabrice Bellard. |
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 |
1419 | 32 #include "rtp_internal.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)); |
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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 | |
294
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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, |
294
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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->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; |
1445
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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)); |
1445
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456 return -1; |
0 | 457 } |
1445
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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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Make rtp_parse_packet() always call the vfunc of the dynamic payload handler
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474 } else if (s->parse_packet) { |
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475 rv = s->parse_packet(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. |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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479 switch(st->codec->codec_id) { |
294
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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; |
1427
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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
bcoudurier
parents:
1460
diff
changeset
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511 case CODEC_ID_AAC: |
1427
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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
lucabe
parents:
3977
diff
changeset
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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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changeset
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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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changeset
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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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changeset
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524 memcpy(pkt->data, buf, infos->au_headers[0].size); |
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parents:
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changeset
|
525 buf += infos->au_headers[0].size; |
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changeset
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526 len -= infos->au_headers[0].size; |
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changeset
|
527 } |
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changeset
|
528 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
529 rv= 0; |
1427
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parents:
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diff
changeset
|
530 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
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
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
534 break; |
0 | 535 } |
885 | 536 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
537 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
538 finalize_packet(s, pkt, timestamp); |
0 | 539 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
540 return rv; |
0 | 541 } |
542 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
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
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
547 mpegts_parse_close(s->ts); |
0 | 548 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
549 av_free(s); |
0 | 550 } |