Mercurial > libavformat.hg
annotate rtpdec.c @ 3485:99565673f94c libavformat
Check if there is at least a stream before writing trailer.
Patch by Art Clarke: aclarke vlideshow com
author | benoit |
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date | Fri, 20 Jun 2008 07:27:44 +0000 |
parents | 6f61c3b36632 |
children | e04ae2357e66 |
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 |
22 #include "libavcodec/bitstream.h" | |
0 | 23 #include "avformat.h" |
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24 #include "mpegts.h" |
0 | 25 |
26 #include <unistd.h> | |
1754 | 27 #include "network.h" |
0 | 28 |
1419 | 29 #include "rtp_internal.h" |
1460 | 30 #include "rtp_h264.h" |
1419 | 31 |
0 | 32 //#define DEBUG |
33 | |
34 /* TODO: - add RTCP statistics reporting (should be optional). | |
35 | |
36 - add support for h263/mpeg4 packetized output : IDEA: send a | |
37 buffer to 'rtp_write_packet' contains all the packets for ONE | |
38 frame. Each packet should have a four byte header containing | |
39 the length in big endian format (same trick as | |
885 | 40 'url_open_dyn_packet_buf') |
0 | 41 */ |
42 | |
1419 | 43 /* statistics functions */ |
44 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
45 | |
46 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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47 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 48 |
1419 | 49 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
50 { | |
51 handler->next= RTPFirstDynamicPayloadHandler; | |
52 RTPFirstDynamicPayloadHandler= handler; | |
53 } | |
885 | 54 |
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55 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 56 { |
57 register_dynamic_payload_handler(&mp4v_es_handler); | |
58 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
59 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
60 } | |
0 | 61 |
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62 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 63 { |
64 if (buf[1] != 200) | |
65 return -1; | |
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66 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
173 | 67 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
68 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
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69 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 70 return 0; |
71 } | |
72 | |
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73 #define RTP_SEQ_MOD (1<<16) |
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74 |
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75 /** |
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76 * called on parse open packet |
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77 */ |
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78 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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79 { |
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80 memset(s, 0, sizeof(RTPStatistics)); |
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81 s->max_seq= base_sequence; |
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82 s->probation= 1; |
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83 } |
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84 |
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85 /** |
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86 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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87 */ |
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88 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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89 { |
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90 s->max_seq= seq; |
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91 s->cycles= 0; |
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92 s->base_seq= seq -1; |
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93 s->bad_seq= RTP_SEQ_MOD + 1; |
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94 s->received= 0; |
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95 s->expected_prior= 0; |
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96 s->received_prior= 0; |
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97 s->jitter= 0; |
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98 s->transit= 0; |
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99 } |
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100 |
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101 /** |
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102 * returns 1 if we should handle this packet. |
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103 */ |
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104 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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105 { |
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106 uint16_t udelta= seq - s->max_seq; |
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107 const int MAX_DROPOUT= 3000; |
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108 const int MAX_MISORDER = 100; |
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109 const int MIN_SEQUENTIAL = 2; |
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110 |
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111 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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112 if(s->probation) |
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113 { |
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114 if(seq==s->max_seq + 1) { |
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115 s->probation--; |
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116 s->max_seq= seq; |
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117 if(s->probation==0) { |
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118 rtp_init_sequence(s, seq); |
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119 s->received++; |
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120 return 1; |
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121 } |
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122 } else { |
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123 s->probation= MIN_SEQUENTIAL - 1; |
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124 s->max_seq = seq; |
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125 } |
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126 } else if (udelta < MAX_DROPOUT) { |
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127 // in order, with permissible gap |
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128 if(seq < s->max_seq) { |
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129 //sequence number wrapped; count antother 64k cycles |
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130 s->cycles += RTP_SEQ_MOD; |
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131 } |
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132 s->max_seq= seq; |
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133 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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134 // sequence made a large jump... |
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135 if(seq==s->bad_seq) { |
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136 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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137 rtp_init_sequence(s, seq); |
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138 } else { |
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139 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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140 return 0; |
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141 } |
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142 } else { |
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143 // duplicate or reordered packet... |
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144 } |
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145 s->received++; |
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146 return 1; |
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147 } |
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148 |
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149 #if 0 |
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150 /** |
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151 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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152 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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153 * never change. I left this in in case someone else can see a way. (rdm) |
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154 */ |
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155 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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156 { |
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157 uint32_t transit= arrival_timestamp - sent_timestamp; |
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158 int d; |
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159 s->transit= transit; |
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160 d= FFABS(transit - s->transit); |
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161 s->jitter += d - ((s->jitter + 8)>>4); |
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162 } |
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163 #endif |
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164 |
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165 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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166 { |
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167 ByteIOContext *pb; |
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168 uint8_t *buf; |
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169 int len; |
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170 int rtcp_bytes; |
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171 RTPStatistics *stats= &s->statistics; |
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172 uint32_t lost; |
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173 uint32_t extended_max; |
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174 uint32_t expected_interval; |
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175 uint32_t received_interval; |
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176 uint32_t lost_interval; |
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177 uint32_t expected; |
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178 uint32_t fraction; |
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179 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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180 |
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181 if (!s->rtp_ctx || (count < 1)) |
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182 return -1; |
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183 |
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184 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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185 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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186 s->octet_count += count; |
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187 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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188 RTCP_TX_RATIO_DEN; |
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189 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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190 if (rtcp_bytes < 28) |
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191 return -1; |
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192 s->last_octet_count = s->octet_count; |
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193 |
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194 if (url_open_dyn_buf(&pb) < 0) |
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195 return -1; |
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196 |
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197 // Receiver Report |
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198 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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199 put_byte(pb, 201); |
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200 put_be16(pb, 7); /* length in words - 1 */ |
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201 put_be32(pb, s->ssrc); // our own SSRC |
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202 put_be32(pb, s->ssrc); // XXX: should be the server's here! |
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203 // some placeholders we should really fill... |
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204 // RFC 1889/p64 |
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205 extended_max= stats->cycles + stats->max_seq; |
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206 expected= extended_max - stats->base_seq + 1; |
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207 lost= expected - stats->received; |
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208 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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209 expected_interval= expected - stats->expected_prior; |
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210 stats->expected_prior= expected; |
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211 received_interval= stats->received - stats->received_prior; |
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212 stats->received_prior= stats->received; |
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213 lost_interval= expected_interval - received_interval; |
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214 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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215 else fraction = (lost_interval<<8)/expected_interval; |
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216 |
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217 fraction= (fraction<<24) | lost; |
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218 |
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219 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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220 put_be32(pb, extended_max); /* max sequence received */ |
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221 put_be32(pb, stats->jitter>>4); /* jitter */ |
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222 |
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223 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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224 { |
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225 put_be32(pb, 0); /* last SR timestamp */ |
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226 put_be32(pb, 0); /* delay since last SR */ |
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227 } else { |
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228 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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229 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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230 |
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231 put_be32(pb, middle_32_bits); /* last SR timestamp */ |
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232 put_be32(pb, delay_since_last); /* delay since last SR */ |
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233 } |
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234 |
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235 // CNAME |
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236 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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237 put_byte(pb, 202); |
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238 len = strlen(s->hostname); |
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239 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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240 put_be32(pb, s->ssrc); |
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241 put_byte(pb, 0x01); |
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242 put_byte(pb, len); |
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243 put_buffer(pb, s->hostname, len); |
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244 // padding |
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245 for (len = (6 + len) % 4; len % 4; len++) { |
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246 put_byte(pb, 0); |
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247 } |
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248 |
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249 put_flush_packet(pb); |
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250 len = url_close_dyn_buf(pb, &buf); |
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251 if ((len > 0) && buf) { |
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252 int result; |
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253 #if defined(DEBUG) |
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254 printf("sending %d bytes of RR\n", len); |
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255 #endif |
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256 result= url_write(s->rtp_ctx, buf, len); |
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257 #if defined(DEBUG) |
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258 printf("result from url_write: %d\n", result); |
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259 #endif |
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260 av_free(buf); |
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261 } |
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262 return 0; |
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263 } |
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264 |
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265 /** |
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266 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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267 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 268 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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269 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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270 */ |
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271 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data) |
294
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272 { |
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273 RTPDemuxContext *s; |
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274 |
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275 s = av_mallocz(sizeof(RTPDemuxContext)); |
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276 if (!s) |
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277 return NULL; |
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278 s->payload_type = payload_type; |
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279 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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280 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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281 s->ic = s1; |
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282 s->st = st; |
774 | 283 s->rtp_payload_data = rtp_payload_data; |
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284 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
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285 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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286 s->ts = mpegts_parse_open(s->ic); |
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287 if (s->ts == NULL) { |
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288 av_free(s); |
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289 return NULL; |
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290 } |
305 | 291 } else { |
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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: | |
302 break; | |
303 } | |
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304 } |
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305 // needed to send back RTCP RR in RTSP sessions |
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306 s->rtp_ctx = rtpc; |
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307 gethostname(s->hostname, sizeof(s->hostname)); |
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308 return s; |
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309 } |
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310 |
774 | 311 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
312 { | |
313 int au_headers_length, au_header_size, i; | |
314 GetBitContext getbitcontext; | |
315 rtp_payload_data_t *infos; | |
316 | |
317 infos = s->rtp_payload_data; | |
318 | |
319 if (infos == NULL) | |
320 return -1; | |
321 | |
3142 | 322 /* decode the first 2 bytes where the AUHeader sections are stored |
774 | 323 length in bits */ |
1673 | 324 au_headers_length = AV_RB16(buf); |
774 | 325 |
326 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
327 return -1; | |
328 | |
329 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
330 | |
331 /* skip AU headers length section (2 bytes) */ | |
332 buf += 2; | |
333 | |
334 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
335 | |
336 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
337 au_header_size = infos->sizelength + infos->indexlength; | |
338 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
339 return -1; | |
340 | |
341 infos->nb_au_headers = au_headers_length / au_header_size; | |
342 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
343 | |
344 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 345 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 346 but does when sending the whole as one big packet... */ |
347 infos->au_headers[0].size = 0; | |
348 infos->au_headers[0].index = 0; | |
349 for (i = 0; i < infos->nb_au_headers; ++i) { | |
350 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
351 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
352 } | |
353 | |
354 infos->nb_au_headers = 1; | |
355 | |
356 return 0; | |
357 } | |
358 | |
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359 /** |
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360 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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361 */ |
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362 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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363 { |
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364 switch(s->st->codec->codec_id) { |
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365 case CODEC_ID_MP2: |
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366 case CODEC_ID_MPEG1VIDEO: |
2760 | 367 case CODEC_ID_MPEG2VIDEO: |
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368 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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369 int64_t addend; |
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370 |
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371 int delta_timestamp; |
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372 /* XXX: is it really necessary to unify the timestamp base ? */ |
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373 /* compute pts from timestamp with received ntp_time */ |
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374 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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375 /* convert to 90 kHz without overflow */ |
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376 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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377 addend = (addend * 5625) >> 14; |
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378 pkt->pts = addend + delta_timestamp; |
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379 } |
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380 break; |
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381 case CODEC_ID_AAC: |
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382 case CODEC_ID_H264: |
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383 case CODEC_ID_MPEG4: |
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384 pkt->pts = timestamp; |
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385 break; |
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386 default: |
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387 /* no timestamp info yet */ |
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388 break; |
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389 } |
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390 pkt->stream_index = s->st->index; |
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391 } |
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392 |
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393 /** |
885 | 394 * Parse an RTP or RTCP packet directly sent as a buffer. |
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395 * @param s RTP parse context. |
0 | 396 * @param pkt returned packet |
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397 * @param buf input buffer or NULL to read the next packets |
0 | 398 * @param len buffer len |
885 | 399 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
294
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400 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 401 */ |
885 | 402 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
294
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403 const uint8_t *buf, int len) |
0 | 404 { |
405 unsigned int ssrc, h; | |
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406 int payload_type, seq, ret, flags = 0; |
0 | 407 AVStream *st; |
65 | 408 uint32_t timestamp; |
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409 int rv= 0; |
885 | 410 |
294
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411 if (!buf) { |
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412 /* return the next packets, if any */ |
1419 | 413 if(s->st && s->parse_packet) { |
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414 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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415 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0, flags); |
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416 finalize_packet(s, pkt, timestamp); |
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417 return rv; |
1419 | 418 } else { |
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419 // TODO: Move to a dynamic packet handler (like above) |
1424
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420 if (s->read_buf_index >= s->read_buf_size) |
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421 return -1; |
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422 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
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423 s->read_buf_size - s->read_buf_index); |
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424 if (ret < 0) |
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425 return -1; |
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426 s->read_buf_index += ret; |
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427 if (s->read_buf_index < s->read_buf_size) |
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428 return 1; |
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gpoirier
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429 else |
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430 return 0; |
1419 | 431 } |
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432 } |
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433 |
0 | 434 if (len < 12) |
435 return -1; | |
436 | |
437 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
438 return -1; | |
439 if (buf[1] >= 200 && buf[1] <= 204) { | |
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440 rtcp_parse_packet(s, buf, len); |
0 | 441 return -1; |
442 } | |
443 payload_type = buf[1] & 0x7f; | |
2222 | 444 seq = AV_RB16(buf + 2); |
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445 timestamp = AV_RB32(buf + 4); |
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446 ssrc = AV_RB32(buf + 8); |
1425
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447 /* store the ssrc in the RTPDemuxContext */ |
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448 s->ssrc = ssrc; |
885 | 449 |
0 | 450 /* NOTE: we can handle only one payload type */ |
451 if (s->payload_type != payload_type) | |
452 return -1; | |
986
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mmu_man
parents:
896
diff
changeset
|
453 |
f5194ed612b5
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mmu_man
parents:
896
diff
changeset
|
454 st = s->st; |
1445
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gpoirier
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|
455 // only do something with this if all the rtp checks pass... |
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|
456 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
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457 { |
986
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mmu_man
parents:
896
diff
changeset
|
458 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 459 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
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gpoirier
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diff
changeset
|
460 return -1; |
0 | 461 } |
1445
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|
462 |
0 | 463 s->seq = seq; |
464 len -= 12; | |
465 buf += 12; | |
294
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466 |
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467 if (!st) { |
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468 /* specific MPEG2TS demux support */ |
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469 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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470 if (ret < 0) |
0 | 471 return -1; |
294
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472 if (ret < len) { |
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473 s->read_buf_size = len - ret; |
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474 memcpy(s->buf, buf + ret, s->read_buf_size); |
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475 s->read_buf_index = 0; |
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476 return 1; |
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|
477 } |
2917
62ff44e23c10
Make rtp_parse_packet() always call the vfunc of the dynamic payload handler
rbultje
parents:
2891
diff
changeset
|
478 } else if (s->parse_packet) { |
2941
6da0564c9d02
Add a flags field to the RTPDynamicPayloadPacketHandlerProc (PKT_FLAG_*).
rbultje
parents:
2918
diff
changeset
|
479 rv = s->parse_packet(s, pkt, ×tamp, buf, len, flags); |
294
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480 } else { |
1419 | 481 // 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
michael
parents:
777
diff
changeset
|
482 switch(st->codec->codec_id) { |
294
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483 case CODEC_ID_MP2: |
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484 /* better than nothing: skip mpeg audio RTP header */ |
0 | 485 if (len <= 4) |
486 return -1; | |
2167
c4244900f986
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benoit
parents:
2164
diff
changeset
|
487 h = AV_RB32(buf); |
294
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488 len -= 4; |
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489 buf += 4; |
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490 av_new_packet(pkt, len); |
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491 memcpy(pkt->data, buf, len); |
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492 break; |
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493 case CODEC_ID_MPEG1VIDEO: |
2760 | 494 case CODEC_ID_MPEG2VIDEO: |
305 | 495 /* better than nothing: skip mpeg video RTP header */ |
294
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496 if (len <= 4) |
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497 return -1; |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
498 h = AV_RB32(buf); |
0 | 499 buf += 4; |
500 len -= 4; | |
294
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501 if (h & (1 << 26)) { |
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|
502 /* mpeg2 */ |
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|
503 if (len <= 4) |
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504 return -1; |
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505 buf += 4; |
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506 len -= 4; |
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507 } |
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508 av_new_packet(pkt, len); |
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509 memcpy(pkt->data, buf, len); |
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510 break; |
1427
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511 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles |
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512 // timestamps. |
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513 // 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
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1460
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514 case CODEC_ID_AAC: |
1427
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515 if (rtp_parse_mp4_au(s, buf)) |
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516 return -1; |
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changeset
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517 { |
f1614c754d5b
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518 rtp_payload_data_t *infos = s->rtp_payload_data; |
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519 if (infos == NULL) |
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diff
changeset
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520 return -1; |
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diff
changeset
|
521 buf += infos->au_headers_length_bytes + 2; |
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gpoirier
parents:
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diff
changeset
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522 len -= infos->au_headers_length_bytes + 2; |
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gpoirier
parents:
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diff
changeset
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523 |
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gpoirier
parents:
1425
diff
changeset
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524 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
f1614c754d5b
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gpoirier
parents:
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diff
changeset
|
525 one au_header */ |
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gpoirier
parents:
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diff
changeset
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526 av_new_packet(pkt, infos->au_headers[0].size); |
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move up (in the file) AAC audio handling moved from below
gpoirier
parents:
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diff
changeset
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527 memcpy(pkt->data, buf, infos->au_headers[0].size); |
f1614c754d5b
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gpoirier
parents:
1425
diff
changeset
|
528 buf += infos->au_headers[0].size; |
f1614c754d5b
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gpoirier
parents:
1425
diff
changeset
|
529 len -= infos->au_headers[0].size; |
f1614c754d5b
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gpoirier
parents:
1425
diff
changeset
|
530 } |
f1614c754d5b
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gpoirier
parents:
1425
diff
changeset
|
531 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
532 rv= 0; |
1427
f1614c754d5b
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gpoirier
parents:
1425
diff
changeset
|
533 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
534 default: |
2918
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
535 av_new_packet(pkt, len); |
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
536 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
|
537 break; |
0 | 538 } |
885 | 539 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
540 // now perform timestamp things.... |
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
541 finalize_packet(s, pkt, timestamp); |
0 | 542 } |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
543 return rv; |
0 | 544 } |
545 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
546 void rtp_parse_close(RTPDemuxContext *s) |
0 | 547 { |
1419 | 548 // 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
|
549 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
|
550 mpegts_parse_close(s->ts); |
0 | 551 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
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diff
changeset
|
552 av_free(s); |
0 | 553 } |