Mercurial > libavformat.hg
annotate rtpdec.c @ 2892:0d82fdf4fa94 libavformat
Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies
author | lucabe |
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date | Fri, 04 Jan 2008 20:09:48 +0000 |
parents | a6c922b05571 |
children | 62ff44e23c10 |
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 */ |
21 #include "avformat.h" | |
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22 #include "mpegts.h" |
774 | 23 #include "bitstream.h" |
0 | 24 |
25 #include <unistd.h> | |
1754 | 26 #include "network.h" |
0 | 27 |
1419 | 28 #include "rtp_internal.h" |
1460 | 29 #include "rtp_h264.h" |
1419 | 30 |
0 | 31 //#define DEBUG |
32 | |
33 /* TODO: - add RTCP statistics reporting (should be optional). | |
34 | |
35 - add support for h263/mpeg4 packetized output : IDEA: send a | |
36 buffer to 'rtp_write_packet' contains all the packets for ONE | |
37 frame. Each packet should have a four byte header containing | |
38 the length in big endian format (same trick as | |
885 | 39 'url_open_dyn_packet_buf') |
0 | 40 */ |
41 | |
1419 | 42 /* statistics functions */ |
43 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; | |
44 | |
45 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4}; | |
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46 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC}; |
0 | 47 |
1419 | 48 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) |
49 { | |
50 handler->next= RTPFirstDynamicPayloadHandler; | |
51 RTPFirstDynamicPayloadHandler= handler; | |
52 } | |
885 | 53 |
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54 void av_register_rtp_dynamic_payload_handlers(void) |
1419 | 55 { |
56 register_dynamic_payload_handler(&mp4v_es_handler); | |
57 register_dynamic_payload_handler(&mpeg4_generic_handler); | |
58 register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
59 } | |
0 | 60 |
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61 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
0 | 62 { |
63 if (buf[1] != 200) | |
64 return -1; | |
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65 s->last_rtcp_ntp_time = AV_RB64(buf + 8); |
173 | 66 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) |
67 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
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68 s->last_rtcp_timestamp = AV_RB32(buf + 16); |
0 | 69 return 0; |
70 } | |
71 | |
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72 #define RTP_SEQ_MOD (1<<16) |
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73 |
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74 /** |
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75 * called on parse open packet |
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76 */ |
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77 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet. |
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78 { |
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79 memset(s, 0, sizeof(RTPStatistics)); |
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80 s->max_seq= base_sequence; |
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81 s->probation= 1; |
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82 } |
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83 |
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84 /** |
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85 * called whenever there is a large jump in sequence numbers, or when they get out of probation... |
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86 */ |
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87 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) |
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88 { |
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89 s->max_seq= seq; |
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90 s->cycles= 0; |
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91 s->base_seq= seq -1; |
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92 s->bad_seq= RTP_SEQ_MOD + 1; |
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93 s->received= 0; |
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94 s->expected_prior= 0; |
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95 s->received_prior= 0; |
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96 s->jitter= 0; |
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97 s->transit= 0; |
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98 } |
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99 |
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100 /** |
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101 * returns 1 if we should handle this packet. |
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102 */ |
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103 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) |
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104 { |
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105 uint16_t udelta= seq - s->max_seq; |
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106 const int MAX_DROPOUT= 3000; |
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107 const int MAX_MISORDER = 100; |
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108 const int MIN_SEQUENTIAL = 2; |
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109 |
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110 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */ |
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111 if(s->probation) |
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112 { |
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113 if(seq==s->max_seq + 1) { |
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114 s->probation--; |
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115 s->max_seq= seq; |
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116 if(s->probation==0) { |
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117 rtp_init_sequence(s, seq); |
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118 s->received++; |
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119 return 1; |
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120 } |
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121 } else { |
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122 s->probation= MIN_SEQUENTIAL - 1; |
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123 s->max_seq = seq; |
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124 } |
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125 } else if (udelta < MAX_DROPOUT) { |
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126 // in order, with permissible gap |
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127 if(seq < s->max_seq) { |
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128 //sequence number wrapped; count antother 64k cycles |
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129 s->cycles += RTP_SEQ_MOD; |
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130 } |
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131 s->max_seq= seq; |
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132 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { |
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133 // sequence made a large jump... |
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134 if(seq==s->bad_seq) { |
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135 // two sequential packets-- assume that the other side restarted without telling us; just resync. |
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136 rtp_init_sequence(s, seq); |
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137 } else { |
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138 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1); |
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139 return 0; |
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140 } |
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141 } else { |
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142 // duplicate or reordered packet... |
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143 } |
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144 s->received++; |
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145 return 1; |
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146 } |
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147 |
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148 #if 0 |
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149 /** |
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150 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the |
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151 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values |
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152 * never change. I left this in in case someone else can see a way. (rdm) |
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153 */ |
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154 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp) |
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155 { |
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156 uint32_t transit= arrival_timestamp - sent_timestamp; |
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157 int d; |
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158 s->transit= transit; |
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159 d= FFABS(transit - s->transit); |
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160 s->jitter += d - ((s->jitter + 8)>>4); |
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161 } |
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162 #endif |
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163 |
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164 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) |
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165 { |
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166 ByteIOContext *pb; |
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167 uint8_t *buf; |
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168 int len; |
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169 int rtcp_bytes; |
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170 RTPStatistics *stats= &s->statistics; |
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171 uint32_t lost; |
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172 uint32_t extended_max; |
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173 uint32_t expected_interval; |
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174 uint32_t received_interval; |
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175 uint32_t lost_interval; |
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176 uint32_t expected; |
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177 uint32_t fraction; |
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178 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time? |
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179 |
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180 if (!s->rtp_ctx || (count < 1)) |
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181 return -1; |
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182 |
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183 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ |
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184 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ |
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185 s->octet_count += count; |
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186 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
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187 RTCP_TX_RATIO_DEN; |
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188 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? |
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189 if (rtcp_bytes < 28) |
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190 return -1; |
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191 s->last_octet_count = s->octet_count; |
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192 |
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193 if (url_open_dyn_buf(&pb) < 0) |
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194 return -1; |
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195 |
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196 // Receiver Report |
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197 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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198 put_byte(pb, 201); |
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199 put_be16(pb, 7); /* length in words - 1 */ |
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200 put_be32(pb, s->ssrc); // our own SSRC |
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201 put_be32(pb, s->ssrc); // XXX: should be the server's here! |
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202 // some placeholders we should really fill... |
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203 // RFC 1889/p64 |
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204 extended_max= stats->cycles + stats->max_seq; |
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205 expected= extended_max - stats->base_seq + 1; |
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206 lost= expected - stats->received; |
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207 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... |
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208 expected_interval= expected - stats->expected_prior; |
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209 stats->expected_prior= expected; |
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210 received_interval= stats->received - stats->received_prior; |
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211 stats->received_prior= stats->received; |
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212 lost_interval= expected_interval - received_interval; |
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213 if (expected_interval==0 || lost_interval<=0) fraction= 0; |
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214 else fraction = (lost_interval<<8)/expected_interval; |
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215 |
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216 fraction= (fraction<<24) | lost; |
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217 |
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218 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ |
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219 put_be32(pb, extended_max); /* max sequence received */ |
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220 put_be32(pb, stats->jitter>>4); /* jitter */ |
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221 |
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222 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE) |
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223 { |
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224 put_be32(pb, 0); /* last SR timestamp */ |
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225 put_be32(pb, 0); /* delay since last SR */ |
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226 } else { |
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227 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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228 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time; |
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229 |
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230 put_be32(pb, middle_32_bits); /* last SR timestamp */ |
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231 put_be32(pb, delay_since_last); /* delay since last SR */ |
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232 } |
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233 |
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234 // CNAME |
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235 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ |
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236 put_byte(pb, 202); |
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237 len = strlen(s->hostname); |
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238 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ |
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239 put_be32(pb, s->ssrc); |
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240 put_byte(pb, 0x01); |
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241 put_byte(pb, len); |
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242 put_buffer(pb, s->hostname, len); |
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243 // padding |
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244 for (len = (6 + len) % 4; len % 4; len++) { |
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245 put_byte(pb, 0); |
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246 } |
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247 |
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248 put_flush_packet(pb); |
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249 len = url_close_dyn_buf(pb, &buf); |
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250 if ((len > 0) && buf) { |
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251 int result; |
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252 #if defined(DEBUG) |
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253 printf("sending %d bytes of RR\n", len); |
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254 #endif |
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255 result= url_write(s->rtp_ctx, buf, len); |
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256 #if defined(DEBUG) |
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257 printf("result from url_write: %d\n", result); |
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258 #endif |
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259 av_free(buf); |
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260 } |
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261 return 0; |
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262 } |
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263 |
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264 /** |
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265 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
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266 * MPEG2TS streams to indicate that they should be demuxed inside the |
885 | 267 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
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268 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
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269 */ |
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270 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data) |
294
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271 { |
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272 RTPDemuxContext *s; |
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273 |
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274 s = av_mallocz(sizeof(RTPDemuxContext)); |
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275 if (!s) |
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276 return NULL; |
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277 s->payload_type = payload_type; |
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278 s->last_rtcp_ntp_time = AV_NOPTS_VALUE; |
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279 s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
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280 s->ic = s1; |
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281 s->st = st; |
774 | 282 s->rtp_payload_data = rtp_payload_data; |
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283 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? |
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284 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { |
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285 s->ts = mpegts_parse_open(s->ic); |
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286 if (s->ts == NULL) { |
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287 av_free(s); |
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288 return NULL; |
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289 } |
305 | 290 } else { |
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291 switch(st->codec->codec_id) { |
305 | 292 case CODEC_ID_MPEG1VIDEO: |
293 case CODEC_ID_MPEG2VIDEO: | |
294 case CODEC_ID_MP2: | |
295 case CODEC_ID_MP3: | |
296 case CODEC_ID_MPEG4: | |
1419 | 297 case CODEC_ID_H264: |
2023 | 298 st->need_parsing = AVSTREAM_PARSE_FULL; |
305 | 299 break; |
300 default: | |
301 break; | |
302 } | |
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303 } |
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304 // needed to send back RTCP RR in RTSP sessions |
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305 s->rtp_ctx = rtpc; |
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306 gethostname(s->hostname, sizeof(s->hostname)); |
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307 return s; |
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308 } |
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309 |
774 | 310 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
311 { | |
312 int au_headers_length, au_header_size, i; | |
313 GetBitContext getbitcontext; | |
314 rtp_payload_data_t *infos; | |
315 | |
316 infos = s->rtp_payload_data; | |
317 | |
318 if (infos == NULL) | |
319 return -1; | |
320 | |
321 /* decode the first 2 bytes where are stored the AUHeader sections | |
322 length in bits */ | |
1673 | 323 au_headers_length = AV_RB16(buf); |
774 | 324 |
325 if (au_headers_length > RTP_MAX_PACKET_LENGTH) | |
326 return -1; | |
327 | |
328 infos->au_headers_length_bytes = (au_headers_length + 7) / 8; | |
329 | |
330 /* skip AU headers length section (2 bytes) */ | |
331 buf += 2; | |
332 | |
333 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); | |
334 | |
335 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ | |
336 au_header_size = infos->sizelength + infos->indexlength; | |
337 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) | |
338 return -1; | |
339 | |
340 infos->nb_au_headers = au_headers_length / au_header_size; | |
341 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); | |
342 | |
343 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) | |
2164 | 344 In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
774 | 345 but does when sending the whole as one big packet... */ |
346 infos->au_headers[0].size = 0; | |
347 infos->au_headers[0].index = 0; | |
348 for (i = 0; i < infos->nb_au_headers; ++i) { | |
349 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); | |
350 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); | |
351 } | |
352 | |
353 infos->nb_au_headers = 1; | |
354 | |
355 return 0; | |
356 } | |
357 | |
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358 /** |
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359 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. |
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360 */ |
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361 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) |
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362 { |
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363 switch(s->st->codec->codec_id) { |
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364 case CODEC_ID_MP2: |
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365 case CODEC_ID_MPEG1VIDEO: |
2760 | 366 case CODEC_ID_MPEG2VIDEO: |
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367 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { |
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368 int64_t addend; |
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369 |
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370 int delta_timestamp; |
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371 /* XXX: is it really necessary to unify the timestamp base ? */ |
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372 /* compute pts from timestamp with received ntp_time */ |
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373 delta_timestamp = timestamp - s->last_rtcp_timestamp; |
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374 /* convert to 90 kHz without overflow */ |
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375 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; |
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376 addend = (addend * 5625) >> 14; |
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377 pkt->pts = addend + delta_timestamp; |
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378 } |
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379 break; |
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380 case CODEC_ID_AAC: |
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381 case CODEC_ID_H264: |
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382 case CODEC_ID_MPEG4: |
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383 pkt->pts = timestamp; |
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384 break; |
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385 default: |
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386 /* no timestamp info yet */ |
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387 break; |
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388 } |
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389 pkt->stream_index = s->st->index; |
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390 } |
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391 |
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392 /** |
885 | 393 * Parse an RTP or RTCP packet directly sent as a buffer. |
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394 * @param s RTP parse context. |
0 | 395 * @param pkt returned packet |
294
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396 * @param buf input buffer or NULL to read the next packets |
0 | 397 * @param len buffer len |
885 | 398 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow |
294
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399 * (use buf as NULL to read the next). -1 if no packet (error or no more packet). |
0 | 400 */ |
885 | 401 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, |
294
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402 const uint8_t *buf, int len) |
0 | 403 { |
404 unsigned int ssrc, h; | |
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405 int payload_type, seq, ret; |
0 | 406 AVStream *st; |
65 | 407 uint32_t timestamp; |
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408 int rv= 0; |
885 | 409 |
294
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410 if (!buf) { |
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411 /* return the next packets, if any */ |
1419 | 412 if(s->st && s->parse_packet) { |
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413 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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414 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0); |
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415 finalize_packet(s, pkt, timestamp); |
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416 return rv; |
1419 | 417 } else { |
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418 // TODO: Move to a dynamic packet handler (like above) |
1424
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419 if (s->read_buf_index >= s->read_buf_size) |
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420 return -1; |
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421 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
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422 s->read_buf_size - s->read_buf_index); |
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423 if (ret < 0) |
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424 return -1; |
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425 s->read_buf_index += ret; |
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426 if (s->read_buf_index < s->read_buf_size) |
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427 return 1; |
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428 else |
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429 return 0; |
1419 | 430 } |
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431 } |
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432 |
0 | 433 if (len < 12) |
434 return -1; | |
435 | |
436 if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
437 return -1; | |
438 if (buf[1] >= 200 && buf[1] <= 204) { | |
294
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439 rtcp_parse_packet(s, buf, len); |
0 | 440 return -1; |
441 } | |
442 payload_type = buf[1] & 0x7f; | |
2222 | 443 seq = AV_RB16(buf + 2); |
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444 timestamp = AV_RB32(buf + 4); |
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445 ssrc = AV_RB32(buf + 8); |
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446 /* store the ssrc in the RTPDemuxContext */ |
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447 s->ssrc = ssrc; |
885 | 448 |
0 | 449 /* NOTE: we can handle only one payload type */ |
450 if (s->payload_type != payload_type) | |
451 return -1; | |
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452 |
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|
453 st = s->st; |
1445
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454 // only do something with this if all the rtp checks pass... |
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455 if(!rtp_valid_packet_in_sequence(&s->statistics, seq)) |
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456 { |
986
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mmu_man
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diff
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|
457 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", |
0 | 458 payload_type, seq, ((s->seq + 1) & 0xffff)); |
1445
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459 return -1; |
0 | 460 } |
1445
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|
461 |
0 | 462 s->seq = seq; |
463 len -= 12; | |
464 buf += 12; | |
294
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465 |
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466 if (!st) { |
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467 /* specific MPEG2TS demux support */ |
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468 ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
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469 if (ret < 0) |
0 | 470 return -1; |
294
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471 if (ret < len) { |
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472 s->read_buf_size = len - ret; |
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473 memcpy(s->buf, buf + ret, s->read_buf_size); |
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474 s->read_buf_index = 0; |
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475 return 1; |
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476 } |
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477 } else { |
1419 | 478 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise. |
820
feca73904e67
changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
michael
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777
diff
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479 switch(st->codec->codec_id) { |
294
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480 case CODEC_ID_MP2: |
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481 /* better than nothing: skip mpeg audio RTP header */ |
0 | 482 if (len <= 4) |
483 return -1; | |
2167
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484 h = AV_RB32(buf); |
294
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485 len -= 4; |
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486 buf += 4; |
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487 av_new_packet(pkt, len); |
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488 memcpy(pkt->data, buf, len); |
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489 break; |
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490 case CODEC_ID_MPEG1VIDEO: |
2760 | 491 case CODEC_ID_MPEG2VIDEO: |
305 | 492 /* better than nothing: skip mpeg video RTP header */ |
294
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493 if (len <= 4) |
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494 return -1; |
2167
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diff
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|
495 h = AV_RB32(buf); |
0 | 496 buf += 4; |
497 len -= 4; | |
294
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498 if (h & (1 << 26)) { |
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499 /* mpeg2 */ |
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500 if (len <= 4) |
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501 return -1; |
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502 buf += 4; |
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503 len -= 4; |
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|
504 } |
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|
505 av_new_packet(pkt, len); |
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506 memcpy(pkt->data, buf, len); |
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507 break; |
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508 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles |
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509 // timestamps. |
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510 // TODO: Put this into a dynamic packet handler... |
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511 case CODEC_ID_AAC: |
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512 if (rtp_parse_mp4_au(s, buf)) |
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513 return -1; |
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514 { |
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515 rtp_payload_data_t *infos = s->rtp_payload_data; |
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516 if (infos == NULL) |
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517 return -1; |
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518 buf += infos->au_headers_length_bytes + 2; |
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519 len -= infos->au_headers_length_bytes + 2; |
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520 |
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521 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
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522 one au_header */ |
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523 av_new_packet(pkt, infos->au_headers[0].size); |
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524 memcpy(pkt->data, buf, infos->au_headers[0].size); |
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525 buf += infos->au_headers[0].size; |
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526 len -= infos->au_headers[0].size; |
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527 } |
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528 s->read_buf_size = len; |
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529 rv= 0; |
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530 break; |
294
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531 default: |
1419 | 532 if(s->parse_packet) { |
1431
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533 rv= s->parse_packet(s, pkt, ×tamp, buf, len); |
1419 | 534 } else { |
1424
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535 av_new_packet(pkt, len); |
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536 memcpy(pkt->data, buf, len); |
1419 | 537 } |
294
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538 break; |
0 | 539 } |
885 | 540 |
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541 // now perform timestamp things.... |
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542 finalize_packet(s, pkt, timestamp); |
0 | 543 } |
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544 return rv; |
0 | 545 } |
546 | |
294
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547 void rtp_parse_close(RTPDemuxContext *s) |
0 | 548 { |
1419 | 549 // TODO: fold this into the protocol specific data fields. |
2759
b252e318023a
Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP
lucabe
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2706
diff
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550 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
294
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551 mpegts_parse_close(s->ts); |
0 | 552 } |
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553 av_free(s); |
0 | 554 } |