Mercurial > mplayer.hg
annotate libmpdemux/demux_rtp.cpp @ 22885:2fb21c753674
remove useless ()
author | aurel |
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date | Sun, 01 Apr 2007 21:58:45 +0000 |
parents | d7f4da5e9ee4 |
children | 777031848801 |
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 |
22852 | 47 // For H264 over rtsp using AVParser, the next packet has to be saved |
48 demux_packet_t* nextpacket; | |
49 | |
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50 private: |
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51 demux_packet_t* pendingDPHead; |
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52 demux_packet_t* pendingDPTail; |
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53 |
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54 FramedSource* fReadSource; |
7534 | 55 RTPSource* fRTPSource; |
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56 demuxer_t* fOurDemuxer; |
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57 char const* fTag; // used for debugging |
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58 }; |
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59 |
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60 // A structure of RTP-specific state, kept so that we can cleanly |
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61 // reclaim it: |
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62 typedef struct RTPState { |
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63 char const* sdpDescription; |
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64 RTSPClient* rtspClient; |
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65 SIPClient* sipClient; |
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66 MediaSession* mediaSession; |
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67 ReadBufferQueue* audioBufferQueue; |
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68 ReadBufferQueue* videoBufferQueue; |
9250 | 69 unsigned flags; |
7534 | 70 struct timeval firstSyncTime; |
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71 }; |
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72 |
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73 extern "C" char* network_username; |
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74 extern "C" char* network_password; |
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75 static char* openURL_rtsp(RTSPClient* client, char const* url) { |
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76 // If we were given a user name (and optional password), then use them: |
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77 if (network_username != NULL) { |
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78 char const* password = network_password == NULL ? "" : network_password; |
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79 return client->describeWithPassword(url, network_username, password); |
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80 } else { |
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81 return client->describeURL(url); |
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82 } |
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83 } |
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84 |
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85 static char* openURL_sip(SIPClient* client, char const* url) { |
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86 // If we were given a user name (and optional password), then use them: |
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87 if (network_username != NULL) { |
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88 char const* password = network_password == NULL ? "" : network_password; |
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89 return client->inviteWithPassword(url, network_username, password); |
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90 } else { |
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91 return client->invite(url); |
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92 } |
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93 } |
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94 |
7664 | 95 int rtspStreamOverTCP = 0; |
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96 extern int rtsp_port; |
7664 | 97 |
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98 extern "C" int audio_id, video_id, dvdsub_id; |
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99 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { |
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100 Boolean success = False; |
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101 do { |
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102 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); |
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103 if (scheduler == NULL) break; |
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104 UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); |
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105 if (env == NULL) break; |
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106 |
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107 RTSPClient* rtspClient = NULL; |
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108 SIPClient* sipClient = NULL; |
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109 |
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110 if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen |
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111 demuxer->stream->eof = 0; // just in case |
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112 |
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113 // Look at the stream's 'priv' field to see if we were initiated |
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114 // via a SDP description: |
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115 char* sdpDescription = (char*)(demuxer->stream->priv); |
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116 if (sdpDescription == NULL) { |
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117 // We weren't given a SDP description directly, so assume that |
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118 // we were given a RTSP or SIP URL: |
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119 char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; |
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120 char const* url = demuxer->stream->streaming_ctrl->url->url; |
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121 extern int verbose; |
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122 if (strcmp(protocol, "rtsp") == 0) { |
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123 rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer"); |
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124 if (rtspClient == NULL) { |
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125 fprintf(stderr, "Failed to create RTSP client: %s\n", |
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126 env->getResultMsg()); |
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127 break; |
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128 } |
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129 sdpDescription = openURL_rtsp(rtspClient, url); |
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130 } else { // SIP |
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131 unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM) |
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132 sipClient = SIPClient::createNew(*env, desiredAudioType, NULL, |
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133 verbose, "MPlayer"); |
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134 if (sipClient == NULL) { |
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135 fprintf(stderr, "Failed to create SIP client: %s\n", |
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|
136 env->getResultMsg()); |
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137 break; |
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|
138 } |
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139 sipClient->setClientStartPortNum(8000); |
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140 sdpDescription = openURL_sip(sipClient, url); |
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|
141 } |
1a747aee653b
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142 |
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143 if (sdpDescription == NULL) { |
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144 fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", |
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145 url, env->getResultMsg()); |
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146 break; |
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147 } |
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148 } |
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149 |
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150 // Now that we have a SDP description, create a MediaSession from it: |
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151 MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); |
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152 if (mediaSession == NULL) break; |
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153 |
9250 | 154 |
155 // Create a 'RTPState' structure containing the state that we just created, | |
156 // and store it in the demuxer's 'priv' field, for future reference: | |
157 RTPState* rtpState = new RTPState; | |
158 rtpState->sdpDescription = sdpDescription; | |
159 rtpState->rtspClient = rtspClient; | |
10054
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160 rtpState->sipClient = sipClient; |
9250 | 161 rtpState->mediaSession = mediaSession; |
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162 rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; |
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163 rtpState->flags = 0; |
9250 | 164 rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; |
165 demuxer->priv = rtpState; | |
166 | |
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167 // Create RTP receivers (sources) for each subsession: |
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168 MediaSubsessionIterator iter(*mediaSession); |
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169 MediaSubsession* subsession; |
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170 unsigned desiredReceiveBufferSize; |
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171 while ((subsession = iter.next()) != NULL) { |
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172 // Ignore any subsession that's not audio or video: |
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173 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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174 desiredReceiveBufferSize = 100000; |
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175 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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176 desiredReceiveBufferSize = 2000000; |
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177 } else { |
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178 continue; |
1a747aee653b
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|
179 } |
1a747aee653b
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180 |
17650
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Allows the LIVE555 library to forces the client's port to be used
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181 if (rtsp_port) |
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182 subsession->setClientPortNum (rtsp_port); |
d6e575c86bb8
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183 |
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diff
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184 if (!subsession->initiate()) { |
1a747aee653b
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diff
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185 fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); |
1a747aee653b
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|
186 } else { |
17650
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187 fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); |
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188 |
9565
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189 // Set the OS's socket receive buffer sufficiently large to avoid |
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190 // incoming packets getting dropped between successive reads from this |
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191 // subsession's demuxer. Depending on the bitrate(s) that you expect, |
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192 // you may wish to tweak the "desiredReceiveBufferSize" values above. |
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193 int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); |
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194 int receiveBufferSize |
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195 = increaseReceiveBufferTo(*env, rtpSocketNum, |
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196 desiredReceiveBufferSize); |
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197 if (verbose > 0) { |
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198 fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n", |
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changeset
|
199 subsession->mediumName(), receiveBufferSize); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
200 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
201 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
202 if (rtspClient != NULL) { |
e74916774667
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parents:
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diff
changeset
|
203 // Issue a RTSP "SETUP" command on the chosen subsession: |
e74916774667
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diff
changeset
|
204 if (!rtspClient->setupMediaSubsession(*subsession, False, |
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Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
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diff
changeset
|
205 rtspStreamOverTCP)) break; |
6910
1a747aee653b
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arpi
parents:
diff
changeset
|
206 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
207 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
208 } |
9565
e74916774667
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rsf
parents:
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diff
changeset
|
209 |
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rsf
parents:
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diff
changeset
|
210 if (rtspClient != NULL) { |
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parents:
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diff
changeset
|
211 // Issue a RTSP aggregate "PLAY" command on the whole session: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
212 if (!rtspClient->playMediaSession(*mediaSession)) break; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
213 } else if (sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
214 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
|
215 } |
e74916774667
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rsf
parents:
9457
diff
changeset
|
216 |
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diff
changeset
|
217 // Now that the session is ready to be read, do additional |
e74916774667
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parents:
9457
diff
changeset
|
218 // 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
|
219 iter.reset(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
220 while ((subsession = iter.next()) != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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parents:
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diff
changeset
|
221 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
|
222 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
223 unsigned flags = 0; |
e74916774667
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rsf
parents:
9457
diff
changeset
|
224 if (strcmp(subsession->mediumName(), "audio") == 0) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
225 rtpState->audioBufferQueue |
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parents:
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diff
changeset
|
226 = new ReadBufferQueue(subsession, demuxer, "audio"); |
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diff
changeset
|
227 rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); |
e74916774667
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parents:
9457
diff
changeset
|
228 rtpCodecInitialize_audio(demuxer, subsession, flags); |
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diff
changeset
|
229 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
e74916774667
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rsf
parents:
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diff
changeset
|
230 rtpState->videoBufferQueue |
e74916774667
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diff
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|
231 = new ReadBufferQueue(subsession, demuxer, "video"); |
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changeset
|
232 rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); |
e74916774667
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changeset
|
233 rtpCodecInitialize_video(demuxer, subsession, flags); |
e74916774667
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diff
changeset
|
234 } |
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rsf
parents:
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diff
changeset
|
235 rtpState->flags |= flags; |
e74916774667
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rsf
parents:
9457
diff
changeset
|
236 } |
12043
8d45dcfc016f
Fixed a bug that was accidentally introduced by the addition of MPEG Transport
rsf
parents:
12007
diff
changeset
|
237 success = True; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
238 } while (0); |
12043
8d45dcfc016f
Fixed a bug that was accidentally introduced by the addition of MPEG Transport
rsf
parents:
12007
diff
changeset
|
239 if (!success) return NULL; // an error occurred |
11984
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
240 |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
241 // Hack: If audio and video are demuxed together on a single RTP stream, |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
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changeset
|
242 // 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
|
243 // code to recognize this: |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
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diff
changeset
|
244 if (demux_is_multiplexed_rtp_stream(demuxer)) { |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
245 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
|
246 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
|
247 audio_id, video_id, dvdsub_id, NULL); |
11984
94e1953fbe8e
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rsf
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diff
changeset
|
248 demuxer = new_demuxers_demuxer(od, od, od); |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
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diff
changeset
|
249 } |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
250 |
94e1953fbe8e
Added support for multiplexed audio+video RTP streams.
rsf
parents:
11755
diff
changeset
|
251 return demuxer; |
6910
1a747aee653b
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arpi
parents:
diff
changeset
|
252 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
253 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
254 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
|
255 // Get the RTP state that was stored in the demuxer's 'priv' field: |
1a747aee653b
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256 RTPState* rtpState = (RTPState*)(demuxer->priv); |
9250 | 257 |
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258 return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0; |
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|
259 } |
1a747aee653b
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260 |
10477
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261 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) { |
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262 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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263 RTPState* rtpState = (RTPState*)(demuxer->priv); |
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|
264 |
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265 return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0; |
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266 } |
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267 |
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268 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
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269 Boolean mustGetNewData, |
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270 float& ptsBehind); // forward |
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271 |
1a747aee653b
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|
272 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { |
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273 // Get a filled-in "demux_packet" from the RTP source, and deliver it. |
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|
274 // Note that this is called as a synchronous read operation, so it needs |
1a747aee653b
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275 // to block in the (hopefully infrequent) case where no packet is |
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276 // immediately available. |
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277 |
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278 while (1) { |
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279 float ptsBehind; |
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280 demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking |
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281 if (dp == NULL) return 0; |
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282 |
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283 if (demuxer->stream->eof) return 0; // source stream has closed down |
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284 |
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285 // Before using this packet, check to make sure that its presentation |
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286 // time is not far behind the other stream (if any). If it is, |
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287 // then we discard this packet, and get another instead. (The rest of |
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288 // MPlayer doesn't always do a good job of synchronizing when the |
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289 // audio and video streams get this far apart.) |
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290 // (We don't do this when streaming over TCP, because then the audio and |
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291 // video streams are interleaved.) |
12007
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292 // (Also, if the stream is *excessively* far behind, then we allow |
b41cc72bb8ad
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293 // the packet, because in this case it probably means that there was |
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294 // an error in the source's timestamp synchronization.) |
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295 const float ptsBehindThreshold = 1.0; // seconds |
12007
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296 const float ptsBehindLimit = 60.0; // seconds |
b41cc72bb8ad
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297 if (ptsBehind < ptsBehindThreshold || |
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|
298 ptsBehind > ptsBehindLimit || |
b41cc72bb8ad
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|
299 rtspStreamOverTCP) { // packet's OK |
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300 ds_add_packet(ds, dp); |
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|
301 break; |
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|
302 } |
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|
303 |
9786
177a3393057c
Added some optional debugging printfs (disabled by default)
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|
304 #ifdef DEBUG_PRINT_DISCARDED_PACKETS |
177a3393057c
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|
305 RTPState* rtpState = (RTPState*)(demuxer->priv); |
177a3393057c
Added some optional debugging printfs (disabled by default)
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|
306 ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; |
177a3393057c
Added some optional debugging printfs (disabled by default)
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307 fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind); |
177a3393057c
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|
308 #endif |
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309 free_demux_packet(dp); // give back this packet, and get another one |
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310 } |
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311 |
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312 return 1; |
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|
313 } |
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314 |
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315 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds, |
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316 unsigned char*& packetData, unsigned& packetDataLen, |
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|
317 float& pts) { |
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318 // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet" |
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319 // is not delivered to the "demux_stream". |
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|
320 float ptsBehind; |
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|
321 demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking |
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|
322 if (dp == NULL) return False; |
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|
323 |
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|
324 packetData = dp->buffer; |
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|
325 packetDataLen = dp->len; |
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|
326 pts = dp->pts; |
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|
327 |
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|
328 return True; |
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|
329 } |
e74916774667
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changeset
|
330 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
331 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
|
332 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
333 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
|
334 // 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
|
335 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
336 // 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
|
337 RTPState* rtpState = (RTPState*)(demuxer->priv); |
7143
fdfc10c2e2d2
- Tell the RTSP client code to use the string "mplayer" in RTSP
arpi
parents:
7008
diff
changeset
|
338 if (rtpState == NULL) return; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
339 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
340 teardownRTSPorSIPSession(rtpState); |
9565
e74916774667
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|
341 |
6910
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applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
342 UsageEnvironment* env = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
343 TaskScheduler* scheduler = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
344 if (rtpState->mediaSession != NULL) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
345 env = &(rtpState->mediaSession->envir()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
346 scheduler = &(env->taskScheduler()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
347 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
348 Medium::close(rtpState->mediaSession); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
349 Medium::close(rtpState->rtspClient); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
350 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
|
351 delete rtpState->audioBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
352 delete rtpState->videoBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
353 delete rtpState->sdpDescription; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
354 delete rtpState; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
355 |
12866
332f24a1dd5c
Updated to conform to a small change in the LIVE.COM API.
rsf
parents:
12531
diff
changeset
|
356 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
|
357 } |
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 ////////// 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
|
360 |
9565
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|
361 #define MAX_RTP_FRAME_SIZE 50000 |
e74916774667
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|
362 // >= 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
|
363 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
364 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
|
365 unsigned /*numTruncatedBytes*/, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
366 struct timeval presentationTime, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
367 unsigned /*durationInMicroseconds*/) { |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
368 int headersize = 0; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
369 if (frameSize >= MAX_RTP_FRAME_SIZE) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
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changeset
|
370 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:
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diff
changeset
|
371 MAX_RTP_FRAME_SIZE); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
372 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
373 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
|
374 demuxer_t* demuxer = bufferQueue->ourDemuxer(); |
7534 | 375 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
|
376 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
377 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
|
378 |
22851
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
379 demux_packet_t* dp = bufferQueue->dp; |
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
380 |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
381 if (bufferQueue->readSource()->isAMRAudioSource()) |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
382 headersize = 1; |
22852 | 383 else if (bufferQueue == rtpState->videoBufferQueue && |
384 ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) { | |
385 dp->buffer[0]=0x00; | |
386 dp->buffer[1]=0x00; | |
387 dp->buffer[2]=0x01; | |
388 headersize = 3; | |
389 } | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
390 |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
391 resize_demux_packet(dp, frameSize + headersize); |
7534 | 392 |
393 // Set the packet's presentation time stamp, depending on whether or | |
394 // 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
|
395 Boolean hasBeenSynchronized |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
396 = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
397 if (hasBeenSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
398 if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
399 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
|
400 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
401 } |
7534 | 402 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
403 struct timeval* fst = &(rtpState->firstSyncTime); // abbrev |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
404 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
|
405 *fst = presentationTime; |
7534 | 406 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
407 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
408 // 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
|
409 // 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
|
410 // 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
|
411 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
|
412 + (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
|
413 bufferQueue->prevPacketPTS = dp->pts; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
414 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
415 if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
416 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
|
417 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
418 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
419 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
420 // use the previous packet's "pts" once again: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
421 dp->pts = bufferQueue->prevPacketPTS; |
7534 | 422 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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parents:
9457
diff
changeset
|
423 bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized; |
7534 | 424 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
425 dp->pos = demuxer->filepos; |
22759 | 426 demuxer->filepos += frameSize + headersize; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
427 |
9250 | 428 // 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
|
429 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
430 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
431 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
432 static void onSourceClosure(void* clientData) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
433 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
|
434 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
|
435 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
436 demuxer->stream->eof = 1; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
437 |
9250 | 438 // 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
|
439 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
440 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
441 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
442 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
e74916774667
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rsf
parents:
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diff
changeset
|
443 Boolean mustGetNewData, |
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Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
444 float& ptsBehind) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
445 // 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
|
446 // (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
|
447 // the demuxer's 'priv' field) |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
448 RTPState* rtpState = (RTPState*)(demuxer->priv); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
449 ReadBufferQueue* bufferQueue = NULL; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
450 int headersize = 0; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
451 if (ds == demuxer->video) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
452 bufferQueue = rtpState->videoBufferQueue; |
22852 | 453 if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4')) |
454 headersize = 3; | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
455 } else if (ds == demuxer->audio) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
456 bufferQueue = rtpState->audioBufferQueue; |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
457 if (bufferQueue->readSource()->isAMRAudioSource()) |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
458 headersize = 1; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
459 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
460 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
|
461 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
462 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
463 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
464 if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
465 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
|
466 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
467 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
468 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
469 demux_packet_t* dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
470 if (!mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
471 // 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
|
472 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
|
473 if (dp != NULL) { |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
474 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
|
475 return dp; |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
476 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
477 } |
9250 | 478 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
479 // Allocate a new packet buffer, and arrange to read into it: |
22852 | 480 if (!bufferQueue->nextpacket) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
481 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
|
482 bufferQueue->dp = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
483 if (dp == NULL) return NULL; |
22852 | 484 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
485 |
22852 | 486 #ifdef USE_LIBAVCODEC |
487 extern AVCodecParserContext * h264parserctx; | |
488 int consumed, poutbuf_size = 1; | |
489 uint8_t *poutbuf = NULL; | |
490 float lastpts; | |
491 | |
492 do { | |
493 if (!bufferQueue->nextpacket) { | |
494 #endif | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
495 // Schedule the read operation: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
496 bufferQueue->blockingFlag = 0; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
497 bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize, |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
498 afterReading, bufferQueue, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
499 onSourceClosure, bufferQueue); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
500 // Block ourselves until data becomes available: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
501 TaskScheduler& scheduler |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
502 = bufferQueue->readSource()->envir().taskScheduler(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
503 scheduler.doEventLoop(&bufferQueue->blockingFlag); |
9250 | 504 |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
505 if (headersize == 1) // amr |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
506 dp->buffer[0] = |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
507 ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader(); |
22852 | 508 #ifdef USE_LIBAVCODEC |
509 } else { | |
510 bufferQueue->dp = dp = bufferQueue->nextpacket; | |
511 bufferQueue->nextpacket = NULL; | |
512 } | |
513 if (headersize == 3 && h264parserctx) { // h264 | |
514 consumed = h264parserctx->parser->parser_parse(h264parserctx, | |
515 NULL, | |
516 &poutbuf, &poutbuf_size, | |
517 dp->buffer, dp->len); | |
518 | |
519 if (!consumed && !poutbuf_size) | |
520 return NULL; | |
521 | |
522 if (!poutbuf_size) { | |
523 lastpts=dp->pts; | |
524 free_demux_packet(dp); | |
525 bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE); | |
526 } else { | |
527 bufferQueue->nextpacket = dp; | |
528 bufferQueue->dp = dp = new_demux_packet(poutbuf_size); | |
529 memcpy(dp->buffer, poutbuf, poutbuf_size); | |
530 dp->pts=lastpts; | |
531 } | |
532 } | |
533 } while (!poutbuf_size); | |
534 #endif | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
535 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
536 // 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
|
537 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
|
538 && bufferQueue->prevPacketWasSynchronized |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
539 && *(bufferQueue->otherQueue) != NULL |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
540 && (*(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
|
541 && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
542 ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
543 - bufferQueue->prevPacketPTS; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
544 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
545 ptsBehind = 0.0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
546 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
547 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
548 if (mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
549 // Save this buffer for future reads: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
550 bufferQueue->savePendingBuffer(dp); |
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 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
553 return dp; |
9250 | 554 } |
555 | |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
556 static void teardownRTSPorSIPSession(RTPState* rtpState) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
557 MediaSession* mediaSession = rtpState->mediaSession; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
558 if (mediaSession == NULL) return; |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
559 if (rtpState->rtspClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
560 MediaSubsessionIterator iter(*mediaSession); |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
561 MediaSubsession* subsession; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
562 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
563 while ((subsession = iter.next()) != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
564 rtpState->rtspClient->teardownMediaSubsession(*subsession); |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
565 } |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
566 } else if (rtpState->sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
567 rtpState->sipClient->sendBYE(); |
9250 | 568 } |
569 } | |
570 | |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
571 ////////// "ReadBuffer" and "ReadBufferQueue" implementation: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
572 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
573 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
574 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
|
575 : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL), |
22852 | 576 nextpacket(NULL), |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
577 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
|
578 fReadSource(subsession == NULL ? NULL : subsession->readSource()), |
7534 | 579 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
|
580 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
|
581 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
582 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
583 ReadBufferQueue::~ReadBufferQueue() { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
584 delete fTag; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
585 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
586 // 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
|
587 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
588 while (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
589 demux_packet_t* dpNext = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
590 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
591 free_demux_packet(dp); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
592 dp = dpNext; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
593 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
594 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
595 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
596 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
|
597 // 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
|
598 if (pendingDPTail == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
599 pendingDPHead = pendingDPTail = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
600 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
601 pendingDPTail->next = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
602 pendingDPTail = dp; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
603 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
604 dp->next = NULL; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
605 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
606 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
607 demux_packet_t* ReadBufferQueue::getPendingBuffer() { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
608 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
609 if (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
610 pendingDPHead = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
611 if (pendingDPHead == NULL) pendingDPTail = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
612 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
613 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
614 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
615 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
616 return dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
617 } |
16175 | 618 |
22355
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
619 static int demux_rtp_control(struct demuxer_st *demuxer, int cmd, void *arg) { |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
620 double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime(); |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
621 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
622 switch(cmd) { |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
623 case DEMUXER_CTRL_GET_TIME_LENGTH: |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
624 if (endpts <= 0) |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
625 return DEMUXER_CTRL_DONTKNOW; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
626 *((double *)arg) = endpts; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
627 return DEMUXER_CTRL_OK; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
628 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
629 case DEMUXER_CTRL_GET_PERCENT_POS: |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
630 if (endpts <= 0) |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
631 return DEMUXER_CTRL_DONTKNOW; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
632 *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts); |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
633 return DEMUXER_CTRL_OK; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
634 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
635 default: |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
636 return DEMUXER_CTRL_NOTIMPL; |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
637 } |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
638 } |
16175 | 639 |
640 demuxer_desc_t demuxer_desc_rtp = { | |
16572
56a5f69e9b35
"LIVE.COM Streaming Media" is now called "LIVE555 Streaming Media".
rsf
parents:
16175
diff
changeset
|
641 "LIVE555 RTP demuxer", |
16175 | 642 "rtp", |
643 "", | |
644 "Ross Finlayson", | |
16572
56a5f69e9b35
"LIVE.COM Streaming Media" is now called "LIVE555 Streaming Media".
rsf
parents:
16175
diff
changeset
|
645 "requires LIVE555 Streaming Media library", |
16175 | 646 DEMUXER_TYPE_RTP, |
647 0, // no autodetect | |
648 NULL, | |
649 demux_rtp_fill_buffer, | |
650 demux_open_rtp, | |
651 demux_close_rtp, | |
652 NULL, | |
22355
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
653 demux_rtp_control |
16175 | 654 }; |