Mercurial > mplayer.hg
annotate libmpdemux/demux_rtp.cpp @ 30935:59f25d383d12
rename eatgv -> ffeatgv, lpcm -> fflpcm
for consitently naming all ffmpeg codecs
requested by compn
author | attila |
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date | Wed, 31 Mar 2010 06:33:08 +0000 |
parents | 390c8d36d463 |
children | cc43f6d8bb6b |
rev | line source |
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1 /* |
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2 * routines (with C-linkage) that interface between MPlayer |
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3 * and the "LIVE555 Streaming Media" libraries |
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4 * |
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5 * This file is part of MPlayer. |
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6 * |
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7 * MPlayer is free software; you can redistribute it and/or modify |
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8 * it under the terms of the GNU General Public License as published by |
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9 * the Free Software Foundation; either version 2 of the License, or |
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10 * (at your option) any later version. |
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11 * |
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12 * MPlayer is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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15 * GNU General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU General Public License along |
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18 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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19 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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20 */ |
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22 extern "C" { |
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23 // on MinGW, we must include windows.h before the things it conflicts |
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24 #ifdef __MINGW32__ // with. they are each protected from |
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25 #include <windows.h> // windows.h, but not the other way around. |
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26 #endif |
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27 #include "demux_rtp.h" |
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28 #include "stheader.h" |
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29 } |
9250 | 30 #include "demux_rtp_internal.h" |
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31 |
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32 #include "BasicUsageEnvironment.hh" |
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33 #include "liveMedia.hh" |
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34 #include "GroupsockHelper.hh" |
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35 #include <unistd.h> |
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36 |
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37 // A data structure representing input data for each stream: |
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38 class ReadBufferQueue { |
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39 public: |
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40 ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, |
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41 char const* tag); |
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42 virtual ~ReadBufferQueue(); |
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43 |
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44 FramedSource* readSource() const { return fReadSource; } |
7534 | 45 RTPSource* rtpSource() const { return fRTPSource; } |
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46 demuxer_t* ourDemuxer() const { return fOurDemuxer; } |
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47 char const* tag() const { return fTag; } |
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48 |
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49 char blockingFlag; // used to implement synchronous reads |
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50 |
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51 // For A/V synchronization: |
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52 Boolean prevPacketWasSynchronized; |
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53 float prevPacketPTS; |
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54 ReadBufferQueue** otherQueue; |
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55 |
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56 // The 'queue' actually consists of just a single "demux_packet_t" |
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57 // (because the underlying OS does the actual queueing/buffering): |
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58 demux_packet_t* dp; |
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59 |
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60 // However, we sometimes inspect buffers before delivering them. |
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61 // For this, we maintain a queue of pending buffers: |
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62 void savePendingBuffer(demux_packet_t* dp); |
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63 demux_packet_t* getPendingBuffer(); |
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64 |
22852 | 65 // For H264 over rtsp using AVParser, the next packet has to be saved |
66 demux_packet_t* nextpacket; | |
67 | |
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68 private: |
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69 demux_packet_t* pendingDPHead; |
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70 demux_packet_t* pendingDPTail; |
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71 |
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72 FramedSource* fReadSource; |
7534 | 73 RTPSource* fRTPSource; |
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74 demuxer_t* fOurDemuxer; |
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75 char const* fTag; // used for debugging |
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76 }; |
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77 |
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78 // A structure of RTP-specific state, kept so that we can cleanly |
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79 // reclaim it: |
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80 typedef struct RTPState { |
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81 char const* sdpDescription; |
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82 RTSPClient* rtspClient; |
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83 SIPClient* sipClient; |
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84 MediaSession* mediaSession; |
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85 ReadBufferQueue* audioBufferQueue; |
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86 ReadBufferQueue* videoBufferQueue; |
9250 | 87 unsigned flags; |
7534 | 88 struct timeval firstSyncTime; |
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89 }; |
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90 |
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91 extern "C" char* network_username; |
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92 extern "C" char* network_password; |
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93 static char* openURL_rtsp(RTSPClient* client, char const* url) { |
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94 // If we were given a user name (and optional password), then use them: |
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95 if (network_username != NULL) { |
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96 char const* password = network_password == NULL ? "" : network_password; |
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97 return client->describeWithPassword(url, network_username, password); |
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98 } else { |
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99 return client->describeURL(url); |
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100 } |
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101 } |
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102 |
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103 static char* openURL_sip(SIPClient* client, char const* url) { |
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104 // If we were given a user name (and optional password), then use them: |
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105 if (network_username != NULL) { |
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106 char const* password = network_password == NULL ? "" : network_password; |
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107 return client->inviteWithPassword(url, network_username, password); |
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108 } else { |
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109 return client->invite(url); |
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110 } |
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111 } |
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112 |
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113 #ifdef CONFIG_LIBNEMESI |
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114 extern int rtsp_transport_tcp; |
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115 #else |
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116 int rtsp_transport_tcp = 0; |
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117 #endif |
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118 |
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119 extern int rtsp_port; |
29453 | 120 #ifdef CONFIG_LIBAVCODEC |
121 extern AVCodecContext *avcctx; | |
122 #endif | |
7664 | 123 |
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124 extern "C" int audio_id, video_id, dvdsub_id; |
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125 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { |
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126 Boolean success = False; |
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127 do { |
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128 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); |
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129 if (scheduler == NULL) break; |
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130 UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); |
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131 if (env == NULL) break; |
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132 |
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133 RTSPClient* rtspClient = NULL; |
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134 SIPClient* sipClient = NULL; |
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135 |
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136 if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen |
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137 demuxer->stream->eof = 0; // just in case |
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138 |
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139 // Look at the stream's 'priv' field to see if we were initiated |
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140 // via a SDP description: |
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141 char* sdpDescription = (char*)(demuxer->stream->priv); |
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142 if (sdpDescription == NULL) { |
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143 // We weren't given a SDP description directly, so assume that |
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144 // we were given a RTSP or SIP URL: |
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145 char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; |
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146 char const* url = demuxer->stream->streaming_ctrl->url->url; |
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147 extern int verbose; |
10054
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148 if (strcmp(protocol, "rtsp") == 0) { |
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149 rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer"); |
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150 if (rtspClient == NULL) { |
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151 fprintf(stderr, "Failed to create RTSP client: %s\n", |
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152 env->getResultMsg()); |
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153 break; |
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154 } |
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155 sdpDescription = openURL_rtsp(rtspClient, url); |
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156 } else { // SIP |
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157 unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM) |
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158 sipClient = SIPClient::createNew(*env, desiredAudioType, NULL, |
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159 verbose, "MPlayer"); |
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160 if (sipClient == NULL) { |
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161 fprintf(stderr, "Failed to create SIP client: %s\n", |
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162 env->getResultMsg()); |
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163 break; |
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164 } |
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165 sipClient->setClientStartPortNum(8000); |
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166 sdpDescription = openURL_sip(sipClient, url); |
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|
167 } |
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168 |
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169 if (sdpDescription == NULL) { |
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170 fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", |
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171 url, env->getResultMsg()); |
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172 break; |
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173 } |
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174 } |
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175 |
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176 // Now that we have a SDP description, create a MediaSession from it: |
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177 MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); |
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178 if (mediaSession == NULL) break; |
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179 |
9250 | 180 |
181 // Create a 'RTPState' structure containing the state that we just created, | |
182 // and store it in the demuxer's 'priv' field, for future reference: | |
183 RTPState* rtpState = new RTPState; | |
184 rtpState->sdpDescription = sdpDescription; | |
185 rtpState->rtspClient = rtspClient; | |
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186 rtpState->sipClient = sipClient; |
9250 | 187 rtpState->mediaSession = mediaSession; |
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188 rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; |
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189 rtpState->flags = 0; |
9250 | 190 rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; |
191 demuxer->priv = rtpState; | |
192 | |
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193 int audiofound = 0, videofound = 0; |
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194 // Create RTP receivers (sources) for each subsession: |
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195 MediaSubsessionIterator iter(*mediaSession); |
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|
196 MediaSubsession* subsession; |
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197 unsigned desiredReceiveBufferSize; |
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|
198 while ((subsession = iter.next()) != NULL) { |
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199 // Ignore any subsession that's not audio or video: |
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200 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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201 if (audiofound) { |
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202 fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName()); |
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203 continue; |
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204 } |
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|
205 desiredReceiveBufferSize = 100000; |
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diff
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|
206 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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207 if (videofound) { |
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|
208 fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName()); |
963cd0542cec
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|
209 continue; |
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Fix playback of streams with more than one video track (only one supported).
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|
210 } |
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changeset
|
211 desiredReceiveBufferSize = 2000000; |
6910
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arpi
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diff
changeset
|
212 } else { |
1a747aee653b
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arpi
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diff
changeset
|
213 continue; |
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applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
214 } |
1a747aee653b
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arpi
parents:
diff
changeset
|
215 |
17650
d6e575c86bb8
Allows the LIVE555 library to forces the client's port to be used
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diff
changeset
|
216 if (rtsp_port) |
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Allows the LIVE555 library to forces the client's port to be used
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diff
changeset
|
217 subsession->setClientPortNum (rtsp_port); |
29263
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changeset
|
218 |
6910
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arpi
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diff
changeset
|
219 if (!subsession->initiate()) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
220 fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
221 } else { |
17650
d6e575c86bb8
Allows the LIVE555 library to forces the client's port to be used
bertrand
parents:
16572
diff
changeset
|
222 fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); |
6910
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arpi
parents:
diff
changeset
|
223 |
9565
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parents:
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changeset
|
224 // Set the OS's socket receive buffer sufficiently large to avoid |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
225 // incoming packets getting dropped between successive reads from this |
e74916774667
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parents:
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diff
changeset
|
226 // subsession's demuxer. Depending on the bitrate(s) that you expect, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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changeset
|
227 // you may wish to tweak the "desiredReceiveBufferSize" values above. |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
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diff
changeset
|
228 int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
229 int receiveBufferSize |
e74916774667
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parents:
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diff
changeset
|
230 = increaseReceiveBufferTo(*env, rtpSocketNum, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
231 desiredReceiveBufferSize); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
232 if (verbose > 0) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
233 fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n", |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
234 subsession->mediumName(), receiveBufferSize); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
235 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
236 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
237 if (rtspClient != NULL) { |
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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
|
238 // Issue a RTSP "SETUP" command on the chosen subsession: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
239 if (!rtspClient->setupMediaSubsession(*subsession, False, |
29280
99063eff7512
Unify over rtsp_transport_tcp as var to hold the -rtsp-stream-over-tcp flag and make sure libnemesi only builds are fine
lu_zero
parents:
29263
diff
changeset
|
240 rtsp_transport_tcp)) break; |
24598
5411b440eb24
Fix playback of streams with more than one audio track (only one supported).
cehoyos
parents:
23569
diff
changeset
|
241 if (!strcmp(subsession->mediumName(), "audio")) |
5411b440eb24
Fix playback of streams with more than one audio track (only one supported).
cehoyos
parents:
23569
diff
changeset
|
242 audiofound = 1; |
25237
963cd0542cec
Fix playback of streams with more than one video track (only one supported).
cehoyos
parents:
24598
diff
changeset
|
243 if (!strcmp(subsession->mediumName(), "video")) |
963cd0542cec
Fix playback of streams with more than one video track (only one supported).
cehoyos
parents:
24598
diff
changeset
|
244 videofound = 1; |
6910
1a747aee653b
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arpi
parents:
diff
changeset
|
245 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
246 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
247 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
248 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
249 if (rtspClient != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
250 // 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
|
251 if (!rtspClient->playMediaSession(*mediaSession)) break; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
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diff
changeset
|
252 } else if (sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
253 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
|
254 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
255 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
256 // Now that the session is ready to be read, do additional |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
257 // 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
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258 iter.reset(); |
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259 while ((subsession = iter.next()) != NULL) { |
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260 if (subsession->readSource() == NULL) continue; // not reading this |
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261 |
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262 unsigned flags = 0; |
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263 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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264 rtpState->audioBufferQueue |
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265 = new ReadBufferQueue(subsession, demuxer, "audio"); |
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266 rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); |
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267 rtpCodecInitialize_audio(demuxer, subsession, flags); |
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268 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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269 rtpState->videoBufferQueue |
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270 = new ReadBufferQueue(subsession, demuxer, "video"); |
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271 rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); |
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272 rtpCodecInitialize_video(demuxer, subsession, flags); |
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273 } |
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274 rtpState->flags |= flags; |
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275 } |
12043
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Fixed a bug that was accidentally introduced by the addition of MPEG Transport
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276 success = True; |
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277 } while (0); |
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278 if (!success) return NULL; // an error occurred |
11984
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Added support for multiplexed audio+video RTP streams.
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279 |
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280 // Hack: If audio and video are demuxed together on a single RTP stream, |
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281 // then create a new "demuxer_t" structure to allow the higher-level |
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282 // code to recognize this: |
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283 if (demux_is_multiplexed_rtp_stream(demuxer)) { |
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284 stream_t* s = new_ds_stream(demuxer->video); |
12521
27ed4672c619
Added "audio_id", "video_id", "dvdsub_id" to the call to "demux_open()".
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285 demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN, |
27ed4672c619
Added "audio_id", "video_id", "dvdsub_id" to the call to "demux_open()".
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286 audio_id, video_id, dvdsub_id, NULL); |
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287 demuxer = new_demuxers_demuxer(od, od, od); |
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288 } |
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289 |
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290 return demuxer; |
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291 } |
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292 |
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293 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) { |
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294 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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295 RTPState* rtpState = (RTPState*)(demuxer->priv); |
9250 | 296 |
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297 return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0; |
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298 } |
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299 |
10477
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300 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) { |
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301 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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302 RTPState* rtpState = (RTPState*)(demuxer->priv); |
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303 |
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304 return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0; |
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305 } |
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306 |
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307 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
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308 Boolean mustGetNewData, |
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309 float& ptsBehind); // forward |
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310 |
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311 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { |
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312 // Get a filled-in "demux_packet" from the RTP source, and deliver it. |
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313 // Note that this is called as a synchronous read operation, so it needs |
1a747aee653b
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314 // to block in the (hopefully infrequent) case where no packet is |
1a747aee653b
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315 // immediately available. |
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316 |
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317 while (1) { |
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318 float ptsBehind; |
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319 demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking |
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320 if (dp == NULL) return 0; |
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321 |
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322 if (demuxer->stream->eof) return 0; // source stream has closed down |
29263
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diego
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323 |
9565
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324 // Before using this packet, check to make sure that its presentation |
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325 // time is not far behind the other stream (if any). If it is, |
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326 // then we discard this packet, and get another instead. (The rest of |
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327 // MPlayer doesn't always do a good job of synchronizing when the |
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|
328 // audio and video streams get this far apart.) |
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329 // (We don't do this when streaming over TCP, because then the audio and |
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|
330 // video streams are interleaved.) |
12007
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
parents:
11984
diff
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|
331 // (Also, if the stream is *excessively* far behind, then we allow |
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
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11984
diff
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|
332 // the packet, because in this case it probably means that there was |
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
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11984
diff
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|
333 // an error in the source's timestamp synchronization.) |
9565
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|
334 const float ptsBehindThreshold = 1.0; // seconds |
12007
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
parents:
11984
diff
changeset
|
335 const float ptsBehindLimit = 60.0; // seconds |
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
parents:
11984
diff
changeset
|
336 if (ptsBehind < ptsBehindThreshold || |
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
parents:
11984
diff
changeset
|
337 ptsBehind > ptsBehindLimit || |
29280
99063eff7512
Unify over rtsp_transport_tcp as var to hold the -rtsp-stream-over-tcp flag and make sure libnemesi only builds are fine
lu_zero
parents:
29263
diff
changeset
|
338 rtsp_transport_tcp) { // packet's OK |
9565
e74916774667
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diff
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|
339 ds_add_packet(ds, dp); |
e74916774667
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|
340 break; |
e74916774667
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|
341 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
342 |
9786
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
parents:
9565
diff
changeset
|
343 #ifdef DEBUG_PRINT_DISCARDED_PACKETS |
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
parents:
9565
diff
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|
344 RTPState* rtpState = (RTPState*)(demuxer->priv); |
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
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diff
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|
345 ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; |
177a3393057c
Added some optional debugging printfs (disabled by default)
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|
346 fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind); |
177a3393057c
Added some optional debugging printfs (disabled by default)
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|
347 #endif |
9565
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|
348 free_demux_packet(dp); // give back this packet, and get another one |
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|
349 } |
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|
350 |
e74916774667
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351 return 1; |
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|
352 } |
e74916774667
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changeset
|
353 |
e74916774667
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|
354 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds, |
e74916774667
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|
355 unsigned char*& packetData, unsigned& packetDataLen, |
e74916774667
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|
356 float& pts) { |
e74916774667
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|
357 // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet" |
e74916774667
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358 // is not delivered to the "demux_stream". |
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|
359 float ptsBehind; |
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|
360 demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking |
e74916774667
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|
361 if (dp == NULL) return False; |
e74916774667
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rsf
parents:
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changeset
|
362 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
363 packetData = dp->buffer; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
364 packetDataLen = dp->len; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
365 pts = dp->pts; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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parents:
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diff
changeset
|
366 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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|
367 return True; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
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|
368 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
369 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
370 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
|
371 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
372 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
|
373 // 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
|
374 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
375 // 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
|
376 RTPState* rtpState = (RTPState*)(demuxer->priv); |
7143
fdfc10c2e2d2
- Tell the RTSP client code to use the string "mplayer" in RTSP
arpi
parents:
7008
diff
changeset
|
377 if (rtpState == NULL) return; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
378 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
379 teardownRTSPorSIPSession(rtpState); |
9565
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|
380 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
381 UsageEnvironment* env = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
382 TaskScheduler* scheduler = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
383 if (rtpState->mediaSession != NULL) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
384 env = &(rtpState->mediaSession->envir()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
385 scheduler = &(env->taskScheduler()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
386 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
387 Medium::close(rtpState->mediaSession); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
388 Medium::close(rtpState->rtspClient); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
389 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
|
390 delete rtpState->audioBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
391 delete rtpState->videoBufferQueue; |
29585
a17d4f8040f1
free(), delete and delete[] are all different and can't just be used at random,
reimar
parents:
29460
diff
changeset
|
392 delete[] rtpState->sdpDescription; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
393 delete rtpState; |
29451
6aca83f5ba73
Fix a crash when playing some H264 over rtsp streams: Do pass an
cehoyos
parents:
29280
diff
changeset
|
394 #ifdef CONFIG_LIBAVCODEC |
29460 | 395 av_freep(&avcctx); |
29451
6aca83f5ba73
Fix a crash when playing some H264 over rtsp streams: Do pass an
cehoyos
parents:
29280
diff
changeset
|
396 #endif |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
397 |
12866
332f24a1dd5c
Updated to conform to a small change in the LIVE.COM API.
rsf
parents:
12531
diff
changeset
|
398 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
|
399 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
400 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
401 ////////// 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
|
402 |
27776
ef0b0f9291a2
increase the max RTP packet size to 5MB as modern Elphel
attila
parents:
27341
diff
changeset
|
403 #define MAX_RTP_FRAME_SIZE 5000000 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
404 // >= 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
|
405 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
406 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
|
407 unsigned /*numTruncatedBytes*/, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
408 struct timeval presentationTime, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
409 unsigned /*durationInMicroseconds*/) { |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
410 int headersize = 0; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
411 if (frameSize >= MAX_RTP_FRAME_SIZE) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
412 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
|
413 MAX_RTP_FRAME_SIZE); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
414 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
415 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
|
416 demuxer_t* demuxer = bufferQueue->ourDemuxer(); |
7534 | 417 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
|
418 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
419 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
|
420 |
22851
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
421 demux_packet_t* dp = bufferQueue->dp; |
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
422 |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
423 if (bufferQueue->readSource()->isAMRAudioSource()) |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
424 headersize = 1; |
22852 | 425 else if (bufferQueue == rtpState->videoBufferQueue && |
426 ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) { | |
427 dp->buffer[0]=0x00; | |
428 dp->buffer[1]=0x00; | |
429 dp->buffer[2]=0x01; | |
430 headersize = 3; | |
431 } | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
432 |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
433 resize_demux_packet(dp, frameSize + headersize); |
7534 | 434 |
435 // Set the packet's presentation time stamp, depending on whether or | |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
436 // 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
|
437 Boolean hasBeenSynchronized |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
438 = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
439 if (hasBeenSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
440 if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
441 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
|
442 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
443 } |
7534 | 444 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
445 struct timeval* fst = &(rtpState->firstSyncTime); // abbrev |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
446 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
|
447 *fst = presentationTime; |
7534 | 448 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
449 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
450 // 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
|
451 // 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
|
452 // 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
|
453 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
|
454 + (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
|
455 bufferQueue->prevPacketPTS = dp->pts; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
456 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
457 if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
458 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
|
459 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
460 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
461 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
462 // 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
|
463 dp->pts = bufferQueue->prevPacketPTS; |
7534 | 464 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
465 bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized; |
7534 | 466 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
467 dp->pos = demuxer->filepos; |
22759 | 468 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
|
469 |
9250 | 470 // 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
|
471 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
472 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
473 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
474 static void onSourceClosure(void* clientData) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
475 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
|
476 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
|
477 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
478 demuxer->stream->eof = 1; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
479 |
9250 | 480 // 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
|
481 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
482 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
483 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
484 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
|
485 Boolean mustGetNewData, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
486 float& ptsBehind) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
487 // 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
|
488 // (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
|
489 // the demuxer's 'priv' field) |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
490 RTPState* rtpState = (RTPState*)(demuxer->priv); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
491 ReadBufferQueue* bufferQueue = NULL; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
492 int headersize = 0; |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
493 TaskToken task; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
494 |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
495 if (demuxer->stream->eof) return NULL; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
496 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
497 if (ds == demuxer->video) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
498 bufferQueue = rtpState->videoBufferQueue; |
22852 | 499 if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4')) |
500 headersize = 3; | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
501 } else if (ds == demuxer->audio) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
502 bufferQueue = rtpState->audioBufferQueue; |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
503 if (bufferQueue->readSource()->isAMRAudioSource()) |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
504 headersize = 1; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
505 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
506 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
|
507 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
508 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
509 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
510 if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
511 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
|
512 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
513 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
514 |
23555 | 515 demux_packet_t* dp = NULL; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
516 if (!mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
517 // 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
|
518 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
|
519 if (dp != NULL) { |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
520 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
|
521 return dp; |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
522 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
523 } |
9250 | 524 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
525 // Allocate a new packet buffer, and arrange to read into it: |
22852 | 526 if (!bufferQueue->nextpacket) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
527 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
|
528 bufferQueue->dp = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
529 if (dp == NULL) return NULL; |
22852 | 530 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
531 |
27341
e7c989f7a7c9
Start unifying names of internal preprocessor directives.
diego
parents:
25270
diff
changeset
|
532 #ifdef CONFIG_LIBAVCODEC |
22852 | 533 extern AVCodecParserContext * h264parserctx; |
534 int consumed, poutbuf_size = 1; | |
23224 | 535 const uint8_t *poutbuf = NULL; |
23555 | 536 float lastpts = 0.0; |
22852 | 537 |
538 do { | |
539 if (!bufferQueue->nextpacket) { | |
540 #endif | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
541 // Schedule the read operation: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
542 bufferQueue->blockingFlag = 0; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
543 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
|
544 afterReading, bufferQueue, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
545 onSourceClosure, bufferQueue); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
546 // Block ourselves until data becomes available: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
547 TaskScheduler& scheduler |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
548 = bufferQueue->readSource()->envir().taskScheduler(); |
23569 | 549 int delay = 10000000; |
550 if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime()) | |
551 delay /= 10; | |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
552 task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue); |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
553 scheduler.doEventLoop(&bufferQueue->blockingFlag); |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
554 scheduler.unscheduleDelayedTask(task); |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
555 if (demuxer->stream->eof) { |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
556 free_demux_packet(dp); |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
557 return NULL; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
558 } |
9250 | 559 |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
560 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
|
561 dp->buffer[0] = |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
562 ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader(); |
27341
e7c989f7a7c9
Start unifying names of internal preprocessor directives.
diego
parents:
25270
diff
changeset
|
563 #ifdef CONFIG_LIBAVCODEC |
22852 | 564 } else { |
565 bufferQueue->dp = dp = bufferQueue->nextpacket; | |
566 bufferQueue->nextpacket = NULL; | |
567 } | |
568 if (headersize == 3 && h264parserctx) { // h264 | |
569 consumed = h264parserctx->parser->parser_parse(h264parserctx, | |
29453 | 570 avcctx, |
22852 | 571 &poutbuf, &poutbuf_size, |
572 dp->buffer, dp->len); | |
573 | |
574 if (!consumed && !poutbuf_size) | |
575 return NULL; | |
576 | |
577 if (!poutbuf_size) { | |
578 lastpts=dp->pts; | |
579 free_demux_packet(dp); | |
580 bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE); | |
581 } else { | |
582 bufferQueue->nextpacket = dp; | |
583 bufferQueue->dp = dp = new_demux_packet(poutbuf_size); | |
584 memcpy(dp->buffer, poutbuf, poutbuf_size); | |
585 dp->pts=lastpts; | |
586 } | |
587 } | |
588 } while (!poutbuf_size); | |
589 #endif | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
590 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
591 // 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
|
592 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
|
593 && bufferQueue->prevPacketWasSynchronized |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
594 && *(bufferQueue->otherQueue) != NULL |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
595 && (*(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
|
596 && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
597 ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
598 - bufferQueue->prevPacketPTS; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
599 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
600 ptsBehind = 0.0; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
601 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
602 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
603 if (mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
604 // Save this buffer for future reads: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
605 bufferQueue->savePendingBuffer(dp); |
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 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
608 return dp; |
9250 | 609 } |
610 | |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
611 static void teardownRTSPorSIPSession(RTPState* rtpState) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
612 MediaSession* mediaSession = rtpState->mediaSession; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
613 if (mediaSession == NULL) return; |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
614 if (rtpState->rtspClient != NULL) { |
22893
777031848801
Use teardownMediaSession() instead of teardownMediaSubsession()
cehoyos
parents:
22852
diff
changeset
|
615 rtpState->rtspClient->teardownMediaSession(*mediaSession); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
616 } else if (rtpState->sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
617 rtpState->sipClient->sendBYE(); |
9250 | 618 } |
619 } | |
620 | |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
621 ////////// "ReadBuffer" and "ReadBufferQueue" implementation: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
622 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
623 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
624 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
|
625 : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL), |
23555 | 626 dp(NULL), nextpacket(NULL), |
627 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
|
628 fReadSource(subsession == NULL ? NULL : subsession->readSource()), |
7534 | 629 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
|
630 fOurDemuxer(demuxer), fTag(strdup(tag)) { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
631 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
632 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
633 ReadBufferQueue::~ReadBufferQueue() { |
29585
a17d4f8040f1
free(), delete and delete[] are all different and can't just be used at random,
reimar
parents:
29460
diff
changeset
|
634 free((void *)fTag); |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
635 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
636 // 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
|
637 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
638 while (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
639 demux_packet_t* dpNext = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
640 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
641 free_demux_packet(dp); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
642 dp = dpNext; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
643 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
644 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
645 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
646 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
|
647 // 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
|
648 if (pendingDPTail == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
649 pendingDPHead = pendingDPTail = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
650 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
651 pendingDPTail->next = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
652 pendingDPTail = dp; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
653 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
654 dp->next = NULL; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
655 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
656 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
657 demux_packet_t* ReadBufferQueue::getPendingBuffer() { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
658 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
659 if (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
660 pendingDPHead = dp->next; |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
661 if (pendingDPHead == NULL) pendingDPTail = NULL; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
662 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
663 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
664 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
665 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
666 return dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
667 } |
16175 | 668 |
30644
390c8d36d463
Drop weird and unnecessary _s/_st suffixes from demuxer struct names.
diego
parents:
29585
diff
changeset
|
669 static int demux_rtp_control(struct demuxer *demuxer, int cmd, void *arg) { |
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
|
670 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
|
671 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
672 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
|
673 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
|
674 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
|
675 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
|
676 *((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
|
677 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
|
678 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
679 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
|
680 if (endpts <= 0) |
20be7be45f47
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681 return DEMUXER_CTRL_DONTKNOW; |
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682 *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts); |
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683 return DEMUXER_CTRL_OK; |
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684 |
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685 default: |
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implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
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686 return DEMUXER_CTRL_NOTIMPL; |
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687 } |
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688 } |
16175 | 689 |
690 demuxer_desc_t demuxer_desc_rtp = { | |
16572
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691 "LIVE555 RTP demuxer", |
25270 | 692 "live555", |
16175 | 693 "", |
694 "Ross Finlayson", | |
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695 "requires LIVE555 Streaming Media library", |
16175 | 696 DEMUXER_TYPE_RTP, |
697 0, // no autodetect | |
698 NULL, | |
699 demux_rtp_fill_buffer, | |
700 demux_open_rtp, | |
701 demux_close_rtp, | |
702 NULL, | |
22355
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703 demux_rtp_control |
16175 | 704 }; |