Mercurial > mplayer.hg
annotate libmpdemux/demux_rtp.cpp @ 18077:cede662f40a5
timestamps are 33 bits long, so they don't fit in an unsigned int; 10l
author | nicodvb |
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date | Tue, 11 Apr 2006 20:05:24 +0000 |
parents | d6e575c86bb8 |
children | 7d60e0c9f58a |
rev | line source |
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9250 | 1 ////////// Routines (with C-linkage) that interface between "MPlayer" |
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2 ////////// and the "LIVE555 Streaming Media" libraries: |
9250 | 3 |
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4 extern "C" { |
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5 // on MinGW, we must include windows.h before the things it conflicts |
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6 #ifdef __MINGW32__ // with. they are each protected from |
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7 #include <windows.h> // windows.h, but not the other way around. |
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8 #endif |
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9 #include "demux_rtp.h" |
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10 #include "stheader.h" |
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11 } |
9250 | 12 #include "demux_rtp_internal.h" |
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13 |
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14 #include "BasicUsageEnvironment.hh" |
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15 #include "liveMedia.hh" |
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16 #include "GroupsockHelper.hh" |
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17 #include <unistd.h> |
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18 |
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19 // A data structure representing input data for each stream: |
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20 class ReadBufferQueue { |
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21 public: |
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22 ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, |
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23 char const* tag); |
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24 virtual ~ReadBufferQueue(); |
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25 |
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26 FramedSource* readSource() const { return fReadSource; } |
7534 | 27 RTPSource* rtpSource() const { return fRTPSource; } |
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28 demuxer_t* ourDemuxer() const { return fOurDemuxer; } |
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29 char const* tag() const { return fTag; } |
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30 |
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31 char blockingFlag; // used to implement synchronous reads |
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32 |
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33 // For A/V synchronization: |
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34 Boolean prevPacketWasSynchronized; |
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35 float prevPacketPTS; |
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36 ReadBufferQueue** otherQueue; |
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37 |
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38 // The 'queue' actually consists of just a single "demux_packet_t" |
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39 // (because the underlying OS does the actual queueing/buffering): |
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40 demux_packet_t* dp; |
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41 |
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42 // However, we sometimes inspect buffers before delivering them. |
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43 // For this, we maintain a queue of pending buffers: |
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44 void savePendingBuffer(demux_packet_t* dp); |
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45 demux_packet_t* getPendingBuffer(); |
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46 |
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47 private: |
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48 demux_packet_t* pendingDPHead; |
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49 demux_packet_t* pendingDPTail; |
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50 |
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51 FramedSource* fReadSource; |
7534 | 52 RTPSource* fRTPSource; |
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53 demuxer_t* fOurDemuxer; |
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54 char const* fTag; // used for debugging |
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55 }; |
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56 |
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57 // A structure of RTP-specific state, kept so that we can cleanly |
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58 // reclaim it: |
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59 typedef struct RTPState { |
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60 char const* sdpDescription; |
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61 RTSPClient* rtspClient; |
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62 SIPClient* sipClient; |
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63 MediaSession* mediaSession; |
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64 ReadBufferQueue* audioBufferQueue; |
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65 ReadBufferQueue* videoBufferQueue; |
9250 | 66 unsigned flags; |
7534 | 67 struct timeval firstSyncTime; |
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68 }; |
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69 |
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70 extern "C" char* network_username; |
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71 extern "C" char* network_password; |
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72 static char* openURL_rtsp(RTSPClient* client, char const* url) { |
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73 // If we were given a user name (and optional password), then use them: |
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74 if (network_username != NULL) { |
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75 char const* password = network_password == NULL ? "" : network_password; |
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76 return client->describeWithPassword(url, network_username, password); |
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77 } else { |
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78 return client->describeURL(url); |
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79 } |
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80 } |
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81 |
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82 static char* openURL_sip(SIPClient* client, char const* url) { |
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83 // If we were given a user name (and optional password), then use them: |
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84 if (network_username != NULL) { |
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85 char const* password = network_password == NULL ? "" : network_password; |
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86 return client->inviteWithPassword(url, network_username, password); |
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87 } else { |
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88 return client->invite(url); |
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89 } |
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90 } |
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91 |
7664 | 92 int rtspStreamOverTCP = 0; |
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93 int rtsp_port = 0; |
7664 | 94 |
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95 extern "C" int audio_id, video_id, dvdsub_id; |
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96 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { |
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97 Boolean success = False; |
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98 do { |
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99 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); |
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100 if (scheduler == NULL) break; |
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101 UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); |
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102 if (env == NULL) break; |
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103 |
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104 RTSPClient* rtspClient = NULL; |
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105 SIPClient* sipClient = NULL; |
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106 |
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107 if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen |
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108 demuxer->stream->eof = 0; // just in case |
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109 |
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110 // Look at the stream's 'priv' field to see if we were initiated |
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111 // via a SDP description: |
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112 char* sdpDescription = (char*)(demuxer->stream->priv); |
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113 if (sdpDescription == NULL) { |
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114 // We weren't given a SDP description directly, so assume that |
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115 // we were given a RTSP or SIP URL: |
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116 char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; |
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117 char const* url = demuxer->stream->streaming_ctrl->url->url; |
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118 extern int verbose; |
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119 if (strcmp(protocol, "rtsp") == 0) { |
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120 rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer"); |
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121 if (rtspClient == NULL) { |
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122 fprintf(stderr, "Failed to create RTSP client: %s\n", |
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123 env->getResultMsg()); |
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124 break; |
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125 } |
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126 sdpDescription = openURL_rtsp(rtspClient, url); |
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127 } else { // SIP |
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|
128 unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM) |
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129 sipClient = SIPClient::createNew(*env, desiredAudioType, NULL, |
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130 verbose, "MPlayer"); |
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131 if (sipClient == NULL) { |
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132 fprintf(stderr, "Failed to create SIP client: %s\n", |
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|
133 env->getResultMsg()); |
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134 break; |
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|
135 } |
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136 sipClient->setClientStartPortNum(8000); |
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137 sdpDescription = openURL_sip(sipClient, url); |
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|
138 } |
1a747aee653b
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139 |
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140 if (sdpDescription == NULL) { |
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141 fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", |
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142 url, env->getResultMsg()); |
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143 break; |
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144 } |
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145 } |
1a747aee653b
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146 |
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147 // Now that we have a SDP description, create a MediaSession from it: |
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148 MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); |
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149 if (mediaSession == NULL) break; |
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150 |
9250 | 151 |
152 // Create a 'RTPState' structure containing the state that we just created, | |
153 // and store it in the demuxer's 'priv' field, for future reference: | |
154 RTPState* rtpState = new RTPState; | |
155 rtpState->sdpDescription = sdpDescription; | |
156 rtpState->rtspClient = rtspClient; | |
10054
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157 rtpState->sipClient = sipClient; |
9250 | 158 rtpState->mediaSession = mediaSession; |
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159 rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; |
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160 rtpState->flags = 0; |
9250 | 161 rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; |
162 demuxer->priv = rtpState; | |
163 | |
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164 // Create RTP receivers (sources) for each subsession: |
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165 MediaSubsessionIterator iter(*mediaSession); |
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166 MediaSubsession* subsession; |
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167 unsigned desiredReceiveBufferSize; |
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168 while ((subsession = iter.next()) != NULL) { |
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169 // Ignore any subsession that's not audio or video: |
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170 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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171 desiredReceiveBufferSize = 100000; |
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172 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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173 desiredReceiveBufferSize = 2000000; |
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174 } else { |
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175 continue; |
1a747aee653b
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|
176 } |
1a747aee653b
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177 |
17650
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178 if (rtsp_port) |
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179 subsession->setClientPortNum (rtsp_port); |
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180 |
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181 if (!subsession->initiate()) { |
1a747aee653b
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182 fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); |
1a747aee653b
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|
183 } else { |
17650
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184 fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); |
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185 |
9565
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186 // Set the OS's socket receive buffer sufficiently large to avoid |
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187 // incoming packets getting dropped between successive reads from this |
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188 // subsession's demuxer. Depending on the bitrate(s) that you expect, |
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189 // you may wish to tweak the "desiredReceiveBufferSize" values above. |
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190 int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); |
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191 int receiveBufferSize |
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192 = increaseReceiveBufferTo(*env, rtpSocketNum, |
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193 desiredReceiveBufferSize); |
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194 if (verbose > 0) { |
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195 fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n", |
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changeset
|
196 subsession->mediumName(), receiveBufferSize); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
197 } |
1a747aee653b
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arpi
parents:
diff
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|
198 |
9565
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diff
changeset
|
199 if (rtspClient != NULL) { |
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parents:
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diff
changeset
|
200 // Issue a RTSP "SETUP" command on the chosen subsession: |
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diff
changeset
|
201 if (!rtspClient->setupMediaSubsession(*subsession, False, |
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diff
changeset
|
202 rtspStreamOverTCP)) break; |
6910
1a747aee653b
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arpi
parents:
diff
changeset
|
203 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
204 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
205 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
206 |
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rsf
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diff
changeset
|
207 if (rtspClient != NULL) { |
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diff
changeset
|
208 // Issue a RTSP aggregate "PLAY" command on the whole session: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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9457
diff
changeset
|
209 if (!rtspClient->playMediaSession(*mediaSession)) break; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
210 } else if (sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
211 sipClient->sendACK(); // to start the stream flowing |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
212 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
213 |
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diff
changeset
|
214 // Now that the session is ready to be read, do additional |
e74916774667
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parents:
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diff
changeset
|
215 // MPlayer codec-specific initialization on each subsession: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
216 iter.reset(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
217 while ((subsession = iter.next()) != NULL) { |
e74916774667
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diff
changeset
|
218 if (subsession->readSource() == NULL) continue; // not reading this |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
219 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
220 unsigned flags = 0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
221 if (strcmp(subsession->mediumName(), "audio") == 0) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
222 rtpState->audioBufferQueue |
e74916774667
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rsf
parents:
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diff
changeset
|
223 = new ReadBufferQueue(subsession, demuxer, "audio"); |
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Improved RTP packet buffering, by relying on the underlying OS's UDP
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parents:
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diff
changeset
|
224 rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); |
e74916774667
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parents:
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diff
changeset
|
225 rtpCodecInitialize_audio(demuxer, subsession, flags); |
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diff
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|
226 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
e74916774667
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rsf
parents:
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diff
changeset
|
227 rtpState->videoBufferQueue |
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diff
changeset
|
228 = new ReadBufferQueue(subsession, demuxer, "video"); |
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diff
changeset
|
229 rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); |
e74916774667
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diff
changeset
|
230 rtpCodecInitialize_video(demuxer, subsession, flags); |
e74916774667
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diff
changeset
|
231 } |
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rsf
parents:
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diff
changeset
|
232 rtpState->flags |= flags; |
e74916774667
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rsf
parents:
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diff
changeset
|
233 } |
12043
8d45dcfc016f
Fixed a bug that was accidentally introduced by the addition of MPEG Transport
rsf
parents:
12007
diff
changeset
|
234 success = True; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
235 } while (0); |
12043
8d45dcfc016f
Fixed a bug that was accidentally introduced by the addition of MPEG Transport
rsf
parents:
12007
diff
changeset
|
236 if (!success) return NULL; // an error occurred |
11984
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
237 |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
238 // Hack: If audio and video are demuxed together on a single RTP stream, |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
239 // then create a new "demuxer_t" structure to allow the higher-level |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
240 // code to recognize this: |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
241 if (demux_is_multiplexed_rtp_stream(demuxer)) { |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
242 stream_t* s = new_ds_stream(demuxer->video); |
12521
27ed4672c619
Added "audio_id", "video_id", "dvdsub_id" to the call to "demux_open()".
rsf
parents:
12043
diff
changeset
|
243 demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN, |
27ed4672c619
Added "audio_id", "video_id", "dvdsub_id" to the call to "demux_open()".
rsf
parents:
12043
diff
changeset
|
244 audio_id, video_id, dvdsub_id, NULL); |
11984
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
245 demuxer = new_demuxers_demuxer(od, od, od); |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
246 } |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
247 |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
248 return demuxer; |
6910
1a747aee653b
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arpi
parents:
diff
changeset
|
249 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
250 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
251 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
252 // Get the RTP state that was stored in the demuxer's 'priv' field: |
1a747aee653b
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253 RTPState* rtpState = (RTPState*)(demuxer->priv); |
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255 return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0; |
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256 } |
1a747aee653b
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257 |
10477
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258 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) { |
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259 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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260 RTPState* rtpState = (RTPState*)(demuxer->priv); |
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261 |
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262 return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0; |
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263 } |
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264 |
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265 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
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266 Boolean mustGetNewData, |
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267 float& ptsBehind); // forward |
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268 |
1a747aee653b
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|
269 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { |
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270 // Get a filled-in "demux_packet" from the RTP source, and deliver it. |
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271 // Note that this is called as a synchronous read operation, so it needs |
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272 // to block in the (hopefully infrequent) case where no packet is |
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273 // immediately available. |
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274 |
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275 while (1) { |
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276 float ptsBehind; |
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277 demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking |
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278 if (dp == NULL) return 0; |
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279 |
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280 if (demuxer->stream->eof) return 0; // source stream has closed down |
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281 |
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282 // Before using this packet, check to make sure that its presentation |
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283 // time is not far behind the other stream (if any). If it is, |
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284 // then we discard this packet, and get another instead. (The rest of |
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285 // MPlayer doesn't always do a good job of synchronizing when the |
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286 // audio and video streams get this far apart.) |
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287 // (We don't do this when streaming over TCP, because then the audio and |
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288 // video streams are interleaved.) |
12007
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289 // (Also, if the stream is *excessively* far behind, then we allow |
b41cc72bb8ad
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290 // the packet, because in this case it probably means that there was |
b41cc72bb8ad
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291 // an error in the source's timestamp synchronization.) |
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292 const float ptsBehindThreshold = 1.0; // seconds |
12007
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293 const float ptsBehindLimit = 60.0; // seconds |
b41cc72bb8ad
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294 if (ptsBehind < ptsBehindThreshold || |
b41cc72bb8ad
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|
295 ptsBehind > ptsBehindLimit || |
b41cc72bb8ad
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|
296 rtspStreamOverTCP) { // packet's OK |
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297 ds_add_packet(ds, dp); |
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298 break; |
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|
299 } |
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|
300 |
9786
177a3393057c
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|
301 #ifdef DEBUG_PRINT_DISCARDED_PACKETS |
177a3393057c
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|
302 RTPState* rtpState = (RTPState*)(demuxer->priv); |
177a3393057c
Added some optional debugging printfs (disabled by default)
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|
303 ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; |
177a3393057c
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304 fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind); |
177a3393057c
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|
305 #endif |
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306 free_demux_packet(dp); // give back this packet, and get another one |
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307 } |
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|
308 |
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309 return 1; |
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|
310 } |
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|
311 |
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312 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds, |
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313 unsigned char*& packetData, unsigned& packetDataLen, |
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|
314 float& pts) { |
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|
315 // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet" |
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|
316 // is not delivered to the "demux_stream". |
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|
317 float ptsBehind; |
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|
318 demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking |
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|
319 if (dp == NULL) return False; |
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|
320 |
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|
321 packetData = dp->buffer; |
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|
322 packetDataLen = dp->len; |
e74916774667
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|
323 pts = dp->pts; |
e74916774667
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|
324 |
e74916774667
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|
325 return True; |
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|
326 } |
e74916774667
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changeset
|
327 |
e74916774667
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|
328 Boolean insertRTPData(demuxer_t* demuxer, demux_stream_t* ds, |
e74916774667
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|
329 unsigned char* data, unsigned dataLen) { |
e74916774667
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diff
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|
330 // Begin by finding the buffer queue that we want to add data to. |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
331 // (Get this from the RTP state, which we stored in |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
332 // the demuxer's 'priv' field) |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
333 RTPState* rtpState = (RTPState*)(demuxer->priv); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
334 ReadBufferQueue* bufferQueue = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
335 if (ds == demuxer->video) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
336 bufferQueue = rtpState->videoBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
337 } else if (ds == demuxer->audio) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
338 bufferQueue = rtpState->audioBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
339 } else { |
9565
e74916774667
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parents:
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diff
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|
340 fprintf(stderr, "(demux_rtp)insertRTPData: internal error: unknown stream\n"); |
9250 | 341 return False; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
342 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
343 |
9565
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|
344 if (data == NULL || dataLen == 0) return False; |
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|
345 |
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346 demux_packet_t* dp = new_demux_packet(dataLen); |
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347 if (dp == NULL) return False; |
9250 | 348 |
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349 // Copy our data into the buffer, and save it: |
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|
350 memmove(dp->buffer, data, dataLen); |
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|
351 dp->len = dataLen; |
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|
352 dp->pts = 0; |
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parents:
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|
353 bufferQueue->savePendingBuffer(dp); |
9904
afbbf103a662
Access-controlled RTSP sessions can now be played, if the user uses the
rsf
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9786
diff
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|
354 return True; |
9565
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|
355 } |
9250 | 356 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
357 static void teardownRTSPorSIPSession(RTPState* rtpState); // forward |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
358 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
359 extern "C" void demux_close_rtp(demuxer_t* demuxer) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
360 // Reclaim all RTP-related state: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
361 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
362 // Get the RTP state that was stored in the demuxer's 'priv' field: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
363 RTPState* rtpState = (RTPState*)(demuxer->priv); |
7143
fdfc10c2e2d2
- Tell the RTSP client code to use the string "mplayer" in RTSP
arpi
parents:
7008
diff
changeset
|
364 if (rtpState == NULL) return; |
9565
e74916774667
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parents:
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|
365 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
366 teardownRTSPorSIPSession(rtpState); |
9565
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367 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
368 UsageEnvironment* env = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
369 TaskScheduler* scheduler = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
370 if (rtpState->mediaSession != NULL) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
371 env = &(rtpState->mediaSession->envir()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
372 scheduler = &(env->taskScheduler()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
373 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
374 Medium::close(rtpState->mediaSession); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
375 Medium::close(rtpState->rtspClient); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
376 Medium::close(rtpState->sipClient); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
377 delete rtpState->audioBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
378 delete rtpState->videoBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
379 delete rtpState->sdpDescription; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
380 delete rtpState; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
381 |
12866
332f24a1dd5c
Updated to conform to a small change in the LIVE.COM API.
rsf
parents:
12531
diff
changeset
|
382 env->reclaim(); delete scheduler; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
383 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
384 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
385 ////////// Extra routines that help implement the above interface functions: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
386 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
387 #define MAX_RTP_FRAME_SIZE 50000 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
388 // >= the largest conceivable frame composed from one or more RTP packets |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
389 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
390 static void afterReading(void* clientData, unsigned frameSize, |
11755
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
391 unsigned /*numTruncatedBytes*/, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
392 struct timeval presentationTime, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
393 unsigned /*durationInMicroseconds*/) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
394 if (frameSize >= MAX_RTP_FRAME_SIZE) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
395 fprintf(stderr, "Saw an input frame too large (>=%d). Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n", |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
396 MAX_RTP_FRAME_SIZE); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
397 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
398 ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
399 demuxer_t* demuxer = bufferQueue->ourDemuxer(); |
7534 | 400 RTPState* rtpState = (RTPState*)(demuxer->priv); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
401 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
402 if (frameSize > 0) demuxer->stream->eof = 0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
403 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
404 demux_packet_t* dp = bufferQueue->dp; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
405 dp->len = frameSize; |
7534 | 406 |
407 // Set the packet's presentation time stamp, depending on whether or | |
408 // not our RTP source's timestamps have been synchronized yet: | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
409 Boolean hasBeenSynchronized |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
410 = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
411 if (hasBeenSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
412 if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
413 fprintf(stderr, "%s stream has been synchronized using RTCP \n", |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
414 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
415 } |
7534 | 416 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
417 struct timeval* fst = &(rtpState->firstSyncTime); // abbrev |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
418 if (fst->tv_sec == 0 && fst->tv_usec == 0) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
419 *fst = presentationTime; |
7534 | 420 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
421 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
422 // For the "pts" field, use the time differential from the first |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
423 // synchronized time, rather than absolute time, in order to avoid |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
424 // round-off errors when converting to a float: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
425 dp->pts = presentationTime.tv_sec - fst->tv_sec |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
426 + (presentationTime.tv_usec - fst->tv_usec)/1000000.0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
427 bufferQueue->prevPacketPTS = dp->pts; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
428 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
429 if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
430 fprintf(stderr, "%s stream is no longer RTCP-synchronized \n", |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
431 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
432 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
433 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
434 // use the previous packet's "pts" once again: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
435 dp->pts = bufferQueue->prevPacketPTS; |
7534 | 436 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
437 bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized; |
7534 | 438 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
439 dp->pos = demuxer->filepos; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
440 demuxer->filepos += frameSize; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
441 |
9250 | 442 // Signal any pending 'doEventLoop()' call on this queue: |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
443 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
444 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
445 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
446 static void onSourceClosure(void* clientData) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
447 ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
448 demuxer_t* demuxer = bufferQueue->ourDemuxer(); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
449 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
450 demuxer->stream->eof = 1; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
451 |
9250 | 452 // Signal any pending 'doEventLoop()' call on this queue: |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
453 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
454 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
455 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
456 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
457 Boolean mustGetNewData, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
458 float& ptsBehind) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
459 // Begin by finding the buffer queue that we want to read from: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
460 // (Get this from the RTP state, which we stored in |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
461 // the demuxer's 'priv' field) |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
462 RTPState* rtpState = (RTPState*)(demuxer->priv); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
463 ReadBufferQueue* bufferQueue = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
464 if (ds == demuxer->video) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
465 bufferQueue = rtpState->videoBufferQueue; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
466 } else if (ds == demuxer->audio) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
467 bufferQueue = rtpState->audioBufferQueue; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
468 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
469 fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n"); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
470 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
471 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
472 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
473 if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
474 fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n"); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
475 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
476 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
477 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
478 demux_packet_t* dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
479 if (!mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
480 // Check whether we have a previously-saved buffer that we can use: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
481 dp = bufferQueue->getPendingBuffer(); |
9911
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
482 if (dp != NULL) { |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
483 ptsBehind = 0.0; // so that we always accept this data |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
484 return dp; |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
485 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
486 } |
9250 | 487 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
488 // Allocate a new packet buffer, and arrange to read into it: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
489 dp = new_demux_packet(MAX_RTP_FRAME_SIZE); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
490 bufferQueue->dp = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
491 if (dp == NULL) return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
492 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
493 // Schedule the read operation: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
494 bufferQueue->blockingFlag = 0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
495 bufferQueue->readSource()->getNextFrame(dp->buffer, MAX_RTP_FRAME_SIZE, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
496 afterReading, bufferQueue, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
497 onSourceClosure, bufferQueue); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
498 // Block ourselves until data becomes available: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
499 TaskScheduler& scheduler |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
500 = bufferQueue->readSource()->envir().taskScheduler(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
501 scheduler.doEventLoop(&bufferQueue->blockingFlag); |
9250 | 502 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
503 // Set the "ptsBehind" result parameter: |
10937
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
504 if (bufferQueue->prevPacketPTS != 0.0 |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
505 && bufferQueue->prevPacketWasSynchronized |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
506 && *(bufferQueue->otherQueue) != NULL |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
507 && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0 |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
508 && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
509 ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
510 - bufferQueue->prevPacketPTS; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
511 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
512 ptsBehind = 0.0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
513 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
514 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
515 if (mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
516 // Save this buffer for future reads: |
e74916774667
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rsf
parents:
9457
diff
changeset
|
517 bufferQueue->savePendingBuffer(dp); |
e74916774667
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rsf
parents:
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diff
changeset
|
518 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
519 |
e74916774667
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rsf
parents:
9457
diff
changeset
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520 return dp; |
9250 | 521 } |
522 | |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
523 static void teardownRTSPorSIPSession(RTPState* rtpState) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
524 MediaSession* mediaSession = rtpState->mediaSession; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
525 if (mediaSession == NULL) return; |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
526 if (rtpState->rtspClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
527 MediaSubsessionIterator iter(*mediaSession); |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
528 MediaSubsession* subsession; |
9565
e74916774667
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rsf
parents:
9457
diff
changeset
|
529 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
530 while ((subsession = iter.next()) != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
531 rtpState->rtspClient->teardownMediaSubsession(*subsession); |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
532 } |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
533 } else if (rtpState->sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
534 rtpState->sipClient->sendBYE(); |
9250 | 535 } |
536 } | |
537 | |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
538 ////////// "ReadBuffer" and "ReadBufferQueue" implementation: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
539 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
540 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
541 demuxer_t* demuxer, char const* tag) |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
542 : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL), |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
543 dp(NULL), pendingDPHead(NULL), pendingDPTail(NULL), |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
544 fReadSource(subsession == NULL ? NULL : subsession->readSource()), |
7534 | 545 fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()), |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
546 fOurDemuxer(demuxer), fTag(strdup(tag)) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
547 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
548 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
549 ReadBufferQueue::~ReadBufferQueue() { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
550 delete fTag; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
551 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
552 // Free any pending buffers (that never got delivered): |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
553 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
554 while (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
555 demux_packet_t* dpNext = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
556 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
557 free_demux_packet(dp); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
558 dp = dpNext; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
559 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
560 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
561 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
562 void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
563 // Keep this buffer around, until MPlayer asks for it later: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
564 if (pendingDPTail == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
565 pendingDPHead = pendingDPTail = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
566 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
567 pendingDPTail->next = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
568 pendingDPTail = dp; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
569 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
570 dp->next = NULL; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
571 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
572 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
573 demux_packet_t* ReadBufferQueue::getPendingBuffer() { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
574 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
575 if (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
576 pendingDPHead = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
577 if (pendingDPHead == NULL) pendingDPTail = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
578 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
579 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
580 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
581 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
582 return dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
583 } |
16175 | 584 |
585 | |
586 demuxer_desc_t demuxer_desc_rtp = { | |
16572
56a5f69e9b35
"LIVE.COM Streaming Media" is now called "LIVE555 Streaming Media".
rsf
parents:
16175
diff
changeset
|
587 "LIVE555 RTP demuxer", |
16175 | 588 "rtp", |
589 "", | |
590 "Ross Finlayson", | |
16572
56a5f69e9b35
"LIVE.COM Streaming Media" is now called "LIVE555 Streaming Media".
rsf
parents:
16175
diff
changeset
|
591 "requires LIVE555 Streaming Media library", |
16175 | 592 DEMUXER_TYPE_RTP, |
593 0, // no autodetect | |
594 NULL, | |
595 demux_rtp_fill_buffer, | |
596 demux_open_rtp, | |
597 demux_close_rtp, | |
598 NULL, | |
599 NULL | |
600 }; |