Mercurial > libavformat.hg
annotate rtpdec.c @ 5914:ae90b07a6f29 libavformat
Add AVFMT_FLAG_NOFILLIN and AVFMT_FLAG_NOPARSE.
author | michael |
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date | Wed, 31 Mar 2010 12:55:16 +0000 |
parents | 536e5527c1e0 |
children | e3b4d7181606 |
rev | line source |
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0 | 1 /* |
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Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies
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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 |
4872 | 25 #include "libavcodec/get_bits.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 |
4388 | 32 #include "rtpdec.h" |
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33 #include "rtpdec_amr.h" |
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34 #include "rtpdec_asf.h" |
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35 #include "rtpdec_h263.h" |
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36 #include "rtpdec_h264.h" |
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37 #include "rtpdec_vorbis.h" |
5866 | 38 #include "rtpdec_theora.h" |
1419 | 39 |
0 | 40 //#define DEBUG |
41 | |
42 /* TODO: - add RTCP statistics reporting (should be optional). | |
43 | |
44 - add support for h263/mpeg4 packetized output : IDEA: send a | |
45 buffer to 'rtp_write_packet' contains all the packets for ONE | |
46 frame. Each packet should have a four byte header containing | |
47 the length in big endian format (same trick as | |
885 | 48 'url_open_dyn_packet_buf') |
0 | 49 */ |
50 | |
1419 | 51 /* statistics functions */ |
52 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
53 | |
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54 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", AVMEDIA_TYPE_VIDEO, CODEC_ID_MPEG4}; |
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55 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 56 |
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57 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
1419 | 58 { |
59 handler->next= RTPFirstDynamicPayloadHandler; | |
60 RTPFirstDynamicPayloadHandler= handler; | |
61 } | |
885 | 62 |
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63 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 64 { |
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65 ff_register_dynamic_payload_handler(&mp4v_es_handler); |
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66 ff_register_dynamic_payload_handler(&mpeg4_generic_handler); |
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67 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler); |
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68 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler); |
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69 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler); |
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70 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler); |
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71 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); |
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72 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); |
5866 | 73 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); |
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74 |
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75 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); |
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76 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler); |
1419 | 77 } |
0 | 78 |
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79 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 80 { |
81 if (buf[1] != 200) | |
82 return -1; | |
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83 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
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84 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 85 return 0; |
86 } | |
87 | |
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88 #define RTP_SEQ_MOD (1<<16) |
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89 |
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90 /** |
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91 * called on parse open packet |
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92 */ |
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93 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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94 { |
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95 memset(s, 0, sizeof(RTPStatistics)); |
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96 s->max_seq= base_sequence; |
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97 s->probation= 1; |
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98 } |
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99 |
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100 /** |
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101 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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102 */ |
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103 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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104 { |
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105 s->max_seq= seq; |
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106 s->cycles= 0; |
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107 s->base_seq= seq -1; |
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108 s->bad_seq= RTP_SEQ_MOD + 1; |
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109 s->received= 0; |
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110 s->expected_prior= 0; |
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111 s->received_prior= 0; |
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112 s->jitter= 0; |
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113 s->transit= 0; |
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114 } |
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115 |
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116 /** |
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117 * returns 1 if we should handle this packet. |
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118 */ |
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119 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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120 { |
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121 uint16_t udelta= seq - s->max_seq; |
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122 const int MAX_DROPOUT= 3000; |
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123 const int MAX_MISORDER = 100; |
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124 const int MIN_SEQUENTIAL = 2; |
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125 |
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126 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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127 if(s->probation) |
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128 { |
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129 if(seq==s->max_seq + 1) { |
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130 s->probation--; |
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131 s->max_seq= seq; |
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132 if(s->probation==0) { |
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133 rtp_init_sequence(s, seq); |
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134 s->received++; |
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135 return 1; |
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136 } |
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137 } else { |
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138 s->probation= MIN_SEQUENTIAL - 1; |
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139 s->max_seq = seq; |
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140 } |
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141 } else if (udelta < MAX_DROPOUT) { |
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142 // in order, with permissible gap |
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143 if(seq < s->max_seq) { |
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144 //sequence number wrapped; count antother 64k cycles |
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145 s->cycles += RTP_SEQ_MOD; |
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146 } |
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147 s->max_seq= seq; |
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148 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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149 // sequence made a large jump... |
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150 if(seq==s->bad_seq) { |
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151 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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152 rtp_init_sequence(s, seq); |
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153 } else { |
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154 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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155 return 0; |
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156 } |
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157 } else { |
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158 // duplicate or reordered packet... |
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159 } |
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160 s->received++; |
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161 return 1; |
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162 } |
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163 |
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164 #if 0 |
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165 /** |
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166 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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167 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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168 * never change. I left this in in case someone else can see a way. (rdm) |
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169 */ |
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170 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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171 { |
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172 uint32_t transit= arrival_timestamp - sent_timestamp; |
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173 int d; |
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174 s->transit= transit; |
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175 d= FFABS(transit - s->transit); |
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176 s->jitter += d - ((s->jitter + 8)>>4); |
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177 } |
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178 #endif |
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179 |
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180 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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181 { |
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182 ByteIOContext *pb; |
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183 uint8_t *buf; |
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184 int len; |
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185 int rtcp_bytes; |
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186 RTPStatistics *stats= &s->statistics; |
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187 uint32_t lost; |
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188 uint32_t extended_max; |
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189 uint32_t expected_interval; |
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190 uint32_t received_interval; |
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191 uint32_t lost_interval; |
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192 uint32_t expected; |
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193 uint32_t fraction; |
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194 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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195 |
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196 if (!s->rtp_ctx || (count < 1)) |
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197 return -1; |
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198 |
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199 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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200 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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201 s->octet_count += count; |
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202 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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203 RTCP_TX_RATIO_DEN; |
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204 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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205 if (rtcp_bytes < 28) |
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206 return -1; |
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207 s->last_octet_count = s->octet_count; |
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208 |
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209 if (url_open_dyn_buf(&pb) < 0) |
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210 return -1; |
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211 |
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212 // Receiver Report |
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213 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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214 put_byte(pb, 201); |
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215 put_be16(pb, 7); /* length in words - 1 */ |
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216 put_be32(pb, s->ssrc); // our own SSRC |
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217 put_be32(pb, s->ssrc); // XXX: should be the server's here! |
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218 // some placeholders we should really fill... |
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219 // RFC 1889/p64 |
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220 extended_max= stats->cycles + stats->max_seq; |
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221 expected= extended_max - stats->base_seq + 1; |
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222 lost= expected - stats->received; |
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223 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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224 expected_interval= expected - stats->expected_prior; |
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225 stats->expected_prior= expected; |
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226 received_interval= stats->received - stats->received_prior; |
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227 stats->received_prior= stats->received; |
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228 lost_interval= expected_interval - received_interval; |
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229 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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230 else fraction = (lost_interval<<8)/expected_interval; |
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231 |
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232 fraction= (fraction<<24) | lost; |
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233 |
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234 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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235 put_be32(pb, extended_max); /* max sequence received */ |
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236 put_be32(pb, stats->jitter>>4); /* jitter */ |
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237 |
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238 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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239 { |
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240 put_be32(pb, 0); /* last SR timestamp */ |
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241 put_be32(pb, 0); /* delay since last SR */ |
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242 } else { |
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243 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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244 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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245 |
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246 put_be32(pb, middle_32_bits); /* last SR timestamp */ |
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247 put_be32(pb, delay_since_last); /* delay since last SR */ |
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248 } |
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249 |
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250 // CNAME |
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251 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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252 put_byte(pb, 202); |
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253 len = strlen(s->hostname); |
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254 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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255 put_be32(pb, s->ssrc); |
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256 put_byte(pb, 0x01); |
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257 put_byte(pb, len); |
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258 put_buffer(pb, s->hostname, len); |
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259 // padding |
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260 for (len = (6 + len) % 4; len % 4; len++) { |
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261 put_byte(pb, 0); |
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262 } |
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263 |
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264 put_flush_packet(pb); |
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265 len = url_close_dyn_buf(pb, &buf); |
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266 if ((len > 0) && buf) { |
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267 int result; |
3579 | 268 dprintf(s->ic, "sending %d bytes of RR\n", len); |
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269 result= url_write(s->rtp_ctx, buf, len); |
3579 | 270 dprintf(s->ic, "result from url_write: %d\n", result); |
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271 av_free(buf); |
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272 } |
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273 return 0; |
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274 } |
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275 |
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276 void rtp_send_punch_packets(URLContext* rtp_handle) |
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277 { |
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278 ByteIOContext *pb; |
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279 uint8_t *buf; |
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280 int len; |
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281 |
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282 /* Send a small RTP packet */ |
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283 if (url_open_dyn_buf(&pb) < 0) |
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284 return; |
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285 |
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286 put_byte(pb, (RTP_VERSION << 6)); |
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287 put_byte(pb, 0); /* Payload type */ |
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288 put_be16(pb, 0); /* Seq */ |
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289 put_be32(pb, 0); /* Timestamp */ |
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290 put_be32(pb, 0); /* SSRC */ |
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291 |
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292 put_flush_packet(pb); |
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293 len = url_close_dyn_buf(pb, &buf); |
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294 if ((len > 0) && buf) |
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295 url_write(rtp_handle, buf, len); |
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296 av_free(buf); |
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297 |
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298 /* Send a minimal RTCP RR */ |
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299 if (url_open_dyn_buf(&pb) < 0) |
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300 return; |
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301 |
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302 put_byte(pb, (RTP_VERSION << 6)); |
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303 put_byte(pb, 201); /* receiver report */ |
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304 put_be16(pb, 1); /* length in words - 1 */ |
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305 put_be32(pb, 0); /* our own SSRC */ |
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306 |
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307 put_flush_packet(pb); |
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308 len = url_close_dyn_buf(pb, &buf); |
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309 if ((len > 0) && buf) |
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310 url_write(rtp_handle, buf, len); |
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311 av_free(buf); |
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312 } |
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313 |
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314 |
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315 /** |
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316 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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317 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 318 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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319 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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320 */ |
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321 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data) |
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322 { |
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323 RTPDemuxContext *s; |
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324 |
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325 s = av_mallocz(sizeof(RTPDemuxContext)); |
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326 if (!s) |
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327 return NULL; |
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328 s->payload_type = payload_type; |
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329 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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330 s->ic = s1; |
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331 s->st = st; |
774 | 332 s->rtp_payload_data = rtp_payload_data; |
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333 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
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334 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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335 s->ts = ff_mpegts_parse_open(s->ic); |
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336 if (s->ts == NULL) { |
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337 av_free(s); |
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338 return NULL; |
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339 } |
305 | 340 } else { |
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341 av_set_pts_info(st, 32, 1, 90000); |
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342 switch(st->codec->codec_id) { |
305 | 343 case CODEC_ID_MPEG1VIDEO: |
344 case CODEC_ID_MPEG2VIDEO: | |
345 case CODEC_ID_MP2: | |
346 case CODEC_ID_MP3: | |
347 case CODEC_ID_MPEG4: | |
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348 case CODEC_ID_H263: |
1419 | 349 case CODEC_ID_H264: |
2023 | 350 st->need_parsing = AVSTREAM_PARSE_FULL; |
305 | 351 break; |
352 default: | |
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353 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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354 av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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355 } |
305 | 356 break; |
357 } | |
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358 } |
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359 // needed to send back RTCP RR in RTSP sessions |
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360 s->rtp_ctx = rtpc; |
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361 gethostname(s->hostname, sizeof(s->hostname)); |
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362 return s; |
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363 } |
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364 |
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365 void |
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366 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, |
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367 RTPDynamicProtocolHandler *handler) |
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368 { |
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369 s->dynamic_protocol_context = ctx; |
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370 s->parse_packet = handler->parse_packet; |
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371 } |
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372 |
774 | 373 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
374 { | |
375 int au_headers_length, au_header_size, i; | |
376 GetBitContext getbitcontext; | |
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377 RTPPayloadData *infos; |
774 | 378 |
379 infos = s->rtp_payload_data; | |
380 | |
381 if (infos == NULL) | |
382 return -1; | |
383 | |
3142 | 384 /* decode the first 2 bytes where the AUHeader sections are stored |
774 | 385 length in bits */ |
1673 | 386 au_headers_length = AV_RB16(buf); |
774 | 387 |
388 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
389 return -1; | |
390 | |
391 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
392 | |
393 /* skip AU headers length section (2 bytes) */ | |
394 buf += 2; | |
395 | |
396 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
397 | |
398 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
399 au_header_size = infos->sizelength + infos->indexlength; | |
400 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
401 return -1; | |
402 | |
403 infos->nb_au_headers = au_headers_length / au_header_size; | |
404 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
405 | |
406 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 407 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 408 but does when sending the whole as one big packet... */ |
409 infos->au_headers[0].size = 0; | |
410 infos->au_headers[0].index = 0; | |
411 for (i = 0; i < infos->nb_au_headers; ++i) { | |
412 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
413 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
414 } | |
415 | |
416 infos->nb_au_headers = 1; | |
417 | |
418 return 0; | |
419 } | |
420 | |
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421 /** |
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422 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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423 */ |
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424 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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425 { |
3535 | 426 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
427 int64_t addend; | |
428 int delta_timestamp; | |
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429 |
3535 | 430 /* compute pts from timestamp with received ntp_time */ |
431 delta_timestamp = timestamp - s->last_rtcp_timestamp; | |
432 /* convert to the PTS timebase */ | |
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433 addend = av_rescale(s->last_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32); |
3535 | 434 pkt->pts = addend + delta_timestamp; |
435 } | |
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436 } |
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437 |
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438 /** |
885 | 439 * Parse an RTP or RTCP packet directly sent as a buffer. |
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440 * @param s RTP parse context. |
0 | 441 * @param pkt returned packet |
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442 * @param buf input buffer or NULL to read the next packets |
0 | 443 * @param len buffer len |
885 | 444 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
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445 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 446 */ |
885 | 447 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
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448 const uint8_t *buf, int len) |
0 | 449 { |
450 unsigned int ssrc, h; | |
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451 int payload_type, seq, ret, flags = 0; |
0 | 452 AVStream *st; |
65 | 453 uint32_t timestamp; |
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454 int rv= 0; |
885 | 455 |
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456 if (!buf) { |
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457 /* return the next packets, if any */ |
1419 | 458 if(s->st && s->parse_packet) { |
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459 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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460 rv= s->parse_packet(s->ic, s->dynamic_protocol_context, |
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461 s->st, pkt, ×tamp, NULL, 0, flags); |
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462 finalize_packet(s, pkt, timestamp); |
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463 return rv; |
1419 | 464 } else { |
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465 // TODO: Move to a dynamic packet handler (like above) |
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466 if (s->read_buf_index >= s->read_buf_size) |
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467 return -1; |
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468 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
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469 s->read_buf_size - s->read_buf_index); |
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470 if (ret < 0) |
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471 return -1; |
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472 s->read_buf_index += ret; |
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473 if (s->read_buf_index < s->read_buf_size) |
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474 return 1; |
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475 else |
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476 return 0; |
1419 | 477 } |
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478 } |
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479 |
0 | 480 if (len < 12) |
481 return -1; | |
482 | |
483 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
484 return -1; | |
485 if (buf[1] >= 200 && buf[1] <= 204) { | |
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486 rtcp_parse_packet(s, buf, len); |
0 | 487 return -1; |
488 } | |
489 payload_type = buf[1] & 0x7f; | |
4590
85c7d517d556
Implement marker bit, which is used for several RTP payloads currently
rbultje
parents:
4388
diff
changeset
|
490 if (buf[1] & 0x80) |
85c7d517d556
Implement marker bit, which is used for several RTP payloads currently
rbultje
parents:
4388
diff
changeset
|
491 flags |= RTP_FLAG_MARKER; |
2222 | 492 seq = AV_RB16(buf + 2); |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
493 timestamp = AV_RB32(buf + 4); |
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
494 ssrc = AV_RB32(buf + 8); |
1425
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
495 /* store the ssrc in the RTPDemuxContext */ |
00d9393a126f
make ffmpeg able to send back a RTCP receiver report.
gpoirier
parents:
1424
diff
changeset
|
496 s->ssrc = ssrc; |
885 | 497 |
0 | 498 /* NOTE: we can handle only one payload type */ |
499 if (s->payload_type != payload_type) | |
500 return -1; | |
986
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
parents:
896
diff
changeset
|
501 |
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
parents:
896
diff
changeset
|
502 st = s->st; |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
503 // only do something with this if all the rtp checks pass... |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
504 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
505 { |
986
f5194ed612b5
fix debug code: do not try to take st->codec if st is NULL. This makes ffplay give me pics from FreeBoxTV \o/
mmu_man
parents:
896
diff
changeset
|
506 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 507 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
508 return -1; |
0 | 509 } |
1445
db97355877b1
add valid statistics for the RTCP receiver report.
gpoirier
parents:
1443
diff
changeset
|
510 |
0 | 511 s->seq = seq; |
512 len -= 12; | |
513 buf += 12; | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
514 |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
515 if (!st) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
516 /* specific MPEG2TS demux support */ |
5619
072cb3a1d4a7
Fix warnings about implicit function declaration when compiling rtpdec.c
cehoyos
parents:
5601
diff
changeset
|
517 ret = ff_mpegts_parse_packet(s->ts, pkt, 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
|
518 if (ret < 0) |
0 | 519 return -1; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
520 if (ret < len) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
521 s->read_buf_size = len - ret; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
522 memcpy(s->buf, buf + ret, s->read_buf_size); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
523 s->read_buf_index = 0; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
524 return 1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
525 } |
4631
4bbf1fe0796b
In the current implementation of rtp_parse_packet(), finalize_packet() is
rbultje
parents:
4590
diff
changeset
|
526 return 0; |
2917
62ff44e23c10
Make rtp_parse_packet() always call the vfunc of the dynamic payload handler
rbultje
parents:
2891
diff
changeset
|
527 } else if (s->parse_packet) { |
4387
5c42816e12c6
Add "AVFormatContext *ctx" (that being the RTSP demuxer's) as first argument
rbultje
parents:
4291
diff
changeset
|
528 rv = s->parse_packet(s->ic, s->dynamic_protocol_context, |
3976
64056a0c38ce
Change function prototype of RTPDynamicPayloadHandler.parse_packet() to
rbultje
parents:
3955
diff
changeset
|
529 s->st, pkt, ×tamp, buf, len, flags); |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
530 } else { |
1419 | 531 // 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
|
532 switch(st->codec->codec_id) { |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
533 case CODEC_ID_MP2: |
5472 | 534 case CODEC_ID_MP3: |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
535 /* better than nothing: skip mpeg audio RTP header */ |
0 | 536 if (len <= 4) |
537 return -1; | |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
538 h = AV_RB32(buf); |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
539 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
540 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
541 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
542 memcpy(pkt->data, buf, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
543 break; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
544 case CODEC_ID_MPEG1VIDEO: |
2760 | 545 case CODEC_ID_MPEG2VIDEO: |
305 | 546 /* better than nothing: skip mpeg video RTP header */ |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
547 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
548 return -1; |
2167
c4244900f986
remove decode_be32/64 and replace them by AV_RB32/64
benoit
parents:
2164
diff
changeset
|
549 h = AV_RB32(buf); |
0 | 550 buf += 4; |
551 len -= 4; | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
552 if (h & (1 << 26)) { |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
553 /* mpeg2 */ |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
554 if (len <= 4) |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
555 return -1; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
556 buf += 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
557 len -= 4; |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
558 } |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
559 av_new_packet(pkt, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
560 memcpy(pkt->data, buf, len); |
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
561 break; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
562 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
563 // timestamps. |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
564 // 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
|
565 case CODEC_ID_AAC: |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
566 if (rtp_parse_mp4_au(s, buf)) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
567 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
568 { |
4101
af2c0aef892b
Rename rtp_payload_data_t to avoid clashes with the POSIX namespace
lucabe
parents:
3977
diff
changeset
|
569 RTPPayloadData *infos = s->rtp_payload_data; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
570 if (infos == NULL) |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
571 return -1; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
572 buf += infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
573 len -= infos->au_headers_length_bytes + 2; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
574 |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
575 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
576 one au_header */ |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
577 av_new_packet(pkt, infos->au_headers[0].size); |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
578 memcpy(pkt->data, buf, infos->au_headers[0].size); |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
579 buf += infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
580 len -= infos->au_headers[0].size; |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
581 } |
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
582 s->read_buf_size = len; |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
583 rv= 0; |
1427
f1614c754d5b
move up (in the file) AAC audio handling moved from below
gpoirier
parents:
1425
diff
changeset
|
584 break; |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
585 default: |
2918
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
586 av_new_packet(pkt, len); |
9e85e46922b5
Reindent after r11493 (always use parse_packet() vfunc in rtp_parse_packet()),
rbultje
parents:
2917
diff
changeset
|
587 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
|
588 break; |
0 | 589 } |
4633
0c69b895a01f
Don't let finalize_packet() touch pkt->stream_index. Instead, let individual
rbultje
parents:
4632
diff
changeset
|
590 |
0c69b895a01f
Don't let finalize_packet() touch pkt->stream_index. Instead, let individual
rbultje
parents:
4632
diff
changeset
|
591 pkt->stream_index = st->index; |
4631
4bbf1fe0796b
In the current implementation of rtp_parse_packet(), finalize_packet() is
rbultje
parents:
4590
diff
changeset
|
592 } |
885 | 593 |
4632 | 594 // now perform timestamp things.... |
595 finalize_packet(s, pkt, timestamp); | |
4631
4bbf1fe0796b
In the current implementation of rtp_parse_packet(), finalize_packet() is
rbultje
parents:
4590
diff
changeset
|
596 |
1431
2d8a17631520
fix more dynamic protocol stuff, needed by the forthcoming h264
gpoirier
parents:
1427
diff
changeset
|
597 return rv; |
0 | 598 } |
599 | |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
600 void rtp_parse_close(RTPDemuxContext *s) |
0 | 601 { |
1419 | 602 // 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
|
603 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
5619
072cb3a1d4a7
Fix warnings about implicit function declaration when compiling rtpdec.c
cehoyos
parents:
5601
diff
changeset
|
604 ff_mpegts_parse_close(s->ts); |
0 | 605 } |
294
6091b76cfc2a
added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API
bellard
parents:
241
diff
changeset
|
606 av_free(s); |
0 | 607 } |