Mercurial > mplayer.hg
annotate libmpdemux/demux_rtp.cpp @ 32074:9031c664ed05
Move v4l2_fd extern declaration to vo_v4l2.h, where it belongs.
author | diego |
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date | Sat, 11 Sep 2010 14:37:12 +0000 |
parents | a295c59e1a48 |
children | 048d9fffd740 |
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 */ |
9250 | 21 |
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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 |
31483 | 27 #include "demuxer.h" |
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28 #include "demux_rtp.h" |
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29 #include "stheader.h" |
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30 } |
9250 | 31 #include "demux_rtp_internal.h" |
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32 |
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33 #include "BasicUsageEnvironment.hh" |
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34 #include "liveMedia.hh" |
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35 #include "GroupsockHelper.hh" |
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36 #include <unistd.h> |
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37 |
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38 // A data structure representing input data for each stream: |
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39 class ReadBufferQueue { |
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40 public: |
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41 ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer, |
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42 char const* tag); |
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43 virtual ~ReadBufferQueue(); |
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44 |
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45 FramedSource* readSource() const { return fReadSource; } |
7534 | 46 RTPSource* rtpSource() const { return fRTPSource; } |
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47 demuxer_t* ourDemuxer() const { return fOurDemuxer; } |
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48 char const* tag() const { return fTag; } |
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49 |
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50 char blockingFlag; // used to implement synchronous reads |
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51 |
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52 // For A/V synchronization: |
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53 Boolean prevPacketWasSynchronized; |
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54 float prevPacketPTS; |
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55 ReadBufferQueue** otherQueue; |
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56 |
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57 // The 'queue' actually consists of just a single "demux_packet_t" |
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58 // (because the underlying OS does the actual queueing/buffering): |
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59 demux_packet_t* dp; |
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60 |
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61 // However, we sometimes inspect buffers before delivering them. |
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62 // For this, we maintain a queue of pending buffers: |
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63 void savePendingBuffer(demux_packet_t* dp); |
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64 demux_packet_t* getPendingBuffer(); |
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65 |
22852 | 66 // For H264 over rtsp using AVParser, the next packet has to be saved |
67 demux_packet_t* nextpacket; | |
68 | |
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69 private: |
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70 demux_packet_t* pendingDPHead; |
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71 demux_packet_t* pendingDPTail; |
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72 |
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73 FramedSource* fReadSource; |
7534 | 74 RTPSource* fRTPSource; |
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75 demuxer_t* fOurDemuxer; |
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76 char const* fTag; // used for debugging |
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77 }; |
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78 |
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79 // A structure of RTP-specific state, kept so that we can cleanly |
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80 // reclaim it: |
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81 typedef struct RTPState { |
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82 char const* sdpDescription; |
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83 RTSPClient* rtspClient; |
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84 SIPClient* sipClient; |
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85 MediaSession* mediaSession; |
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86 ReadBufferQueue* audioBufferQueue; |
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87 ReadBufferQueue* videoBufferQueue; |
9250 | 88 unsigned flags; |
7534 | 89 struct timeval firstSyncTime; |
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90 }; |
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91 |
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92 extern "C" char* network_username; |
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93 extern "C" char* network_password; |
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94 static char* openURL_rtsp(RTSPClient* client, char const* url) { |
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95 // If we were given a user name (and optional password), then use them: |
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96 if (network_username != NULL) { |
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97 char const* password = network_password == NULL ? "" : network_password; |
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98 return client->describeWithPassword(url, network_username, password); |
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99 } else { |
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100 return client->describeURL(url); |
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101 } |
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102 } |
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103 |
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104 static char* openURL_sip(SIPClient* client, char const* url) { |
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105 // If we were given a user name (and optional password), then use them: |
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106 if (network_username != NULL) { |
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107 char const* password = network_password == NULL ? "" : network_password; |
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108 return client->inviteWithPassword(url, network_username, password); |
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109 } else { |
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110 return client->invite(url); |
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111 } |
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112 } |
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113 |
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114 #ifdef CONFIG_LIBNEMESI |
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115 extern int rtsp_transport_tcp; |
31281 | 116 extern int rtsp_transport_http; |
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117 #else |
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118 int rtsp_transport_tcp = 0; |
31281 | 119 int rtsp_transport_http = 0; |
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120 #endif |
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121 |
29453 | 122 #ifdef CONFIG_LIBAVCODEC |
123 extern AVCodecContext *avcctx; | |
124 #endif | |
7664 | 125 |
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126 extern "C" int audio_id, video_id, dvdsub_id; |
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127 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) { |
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128 Boolean success = False; |
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129 do { |
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130 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); |
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131 if (scheduler == NULL) break; |
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132 UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler); |
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133 if (env == NULL) break; |
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134 |
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135 RTSPClient* rtspClient = NULL; |
10054
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136 SIPClient* sipClient = NULL; |
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137 |
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138 if (demuxer == NULL || demuxer->stream == NULL) break; // shouldn't happen |
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139 demuxer->stream->eof = 0; // just in case |
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140 |
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141 // Look at the stream's 'priv' field to see if we were initiated |
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142 // via a SDP description: |
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143 char* sdpDescription = (char*)(demuxer->stream->priv); |
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144 if (sdpDescription == NULL) { |
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145 // We weren't given a SDP description directly, so assume that |
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146 // we were given a RTSP or SIP URL: |
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147 char const* protocol = demuxer->stream->streaming_ctrl->url->protocol; |
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148 char const* url = demuxer->stream->streaming_ctrl->url->url; |
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149 extern int verbose; |
10054
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150 if (strcmp(protocol, "rtsp") == 0) { |
31281 | 151 if (rtsp_transport_http == 1) { |
152 rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port; | |
153 rtsp_transport_tcp = 1; | |
154 } | |
155 rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http); | |
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156 if (rtspClient == NULL) { |
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157 fprintf(stderr, "Failed to create RTSP client: %s\n", |
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158 env->getResultMsg()); |
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159 break; |
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160 } |
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161 sdpDescription = openURL_rtsp(rtspClient, url); |
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162 } else { // SIP |
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163 unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM) |
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164 sipClient = SIPClient::createNew(*env, desiredAudioType, NULL, |
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165 verbose, "MPlayer"); |
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166 if (sipClient == NULL) { |
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167 fprintf(stderr, "Failed to create SIP client: %s\n", |
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168 env->getResultMsg()); |
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169 break; |
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170 } |
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171 sipClient->setClientStartPortNum(8000); |
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172 sdpDescription = openURL_sip(sipClient, url); |
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|
173 } |
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174 |
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175 if (sdpDescription == NULL) { |
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176 fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n", |
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|
177 url, env->getResultMsg()); |
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178 break; |
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179 } |
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180 } |
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181 |
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182 // Now that we have a SDP description, create a MediaSession from it: |
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183 MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription); |
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184 if (mediaSession == NULL) break; |
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185 |
9250 | 186 |
187 // Create a 'RTPState' structure containing the state that we just created, | |
188 // and store it in the demuxer's 'priv' field, for future reference: | |
189 RTPState* rtpState = new RTPState; | |
190 rtpState->sdpDescription = sdpDescription; | |
191 rtpState->rtspClient = rtspClient; | |
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192 rtpState->sipClient = sipClient; |
9250 | 193 rtpState->mediaSession = mediaSession; |
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194 rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL; |
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195 rtpState->flags = 0; |
9250 | 196 rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0; |
197 demuxer->priv = rtpState; | |
198 | |
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199 int audiofound = 0, videofound = 0; |
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|
200 // Create RTP receivers (sources) for each subsession: |
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diff
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|
201 MediaSubsessionIterator iter(*mediaSession); |
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diff
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|
202 MediaSubsession* subsession; |
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|
203 unsigned desiredReceiveBufferSize; |
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|
204 while ((subsession = iter.next()) != NULL) { |
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205 // Ignore any subsession that's not audio or video: |
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diff
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206 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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207 if (audiofound) { |
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208 fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName()); |
5411b440eb24
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209 continue; |
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|
210 } |
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changeset
|
211 desiredReceiveBufferSize = 100000; |
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arpi
parents:
diff
changeset
|
212 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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|
213 if (videofound) { |
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changeset
|
214 fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName()); |
963cd0542cec
Fix playback of streams with more than one video track (only one supported).
cehoyos
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|
215 continue; |
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Fix playback of streams with more than one video track (only one supported).
cehoyos
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|
216 } |
9565
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changeset
|
217 desiredReceiveBufferSize = 2000000; |
6910
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arpi
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diff
changeset
|
218 } else { |
1a747aee653b
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arpi
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diff
changeset
|
219 continue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
220 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
221 |
17650
d6e575c86bb8
Allows the LIVE555 library to forces the client's port to be used
bertrand
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diff
changeset
|
222 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
|
223 subsession->setClientPortNum (rtsp_port); |
29263
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diego
parents:
29238
diff
changeset
|
224 |
6910
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arpi
parents:
diff
changeset
|
225 if (!subsession->initiate()) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
226 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
|
227 } else { |
17650
d6e575c86bb8
Allows the LIVE555 library to forces the client's port to be used
bertrand
parents:
16572
diff
changeset
|
228 fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum()); |
6910
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arpi
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diff
changeset
|
229 |
9565
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parents:
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changeset
|
230 // Set the OS's socket receive buffer sufficiently large to avoid |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
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diff
changeset
|
231 // incoming packets getting dropped between successive reads from this |
e74916774667
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rsf
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diff
changeset
|
232 // 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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|
233 // you may wish to tweak the "desiredReceiveBufferSize" values above. |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
234 int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
235 int receiveBufferSize |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
236 = increaseReceiveBufferTo(*env, rtpSocketNum, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
237 desiredReceiveBufferSize); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
238 if (verbose > 0) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
239 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
|
240 subsession->mediumName(), receiveBufferSize); |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
241 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
242 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
243 if (rtspClient != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
244 // 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
|
245 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
|
246 rtsp_transport_tcp)) break; |
24598
5411b440eb24
Fix playback of streams with more than one audio track (only one supported).
cehoyos
parents:
23569
diff
changeset
|
247 if (!strcmp(subsession->mediumName(), "audio")) |
5411b440eb24
Fix playback of streams with more than one audio track (only one supported).
cehoyos
parents:
23569
diff
changeset
|
248 audiofound = 1; |
25237
963cd0542cec
Fix playback of streams with more than one video track (only one supported).
cehoyos
parents:
24598
diff
changeset
|
249 if (!strcmp(subsession->mediumName(), "video")) |
963cd0542cec
Fix playback of streams with more than one video track (only one supported).
cehoyos
parents:
24598
diff
changeset
|
250 videofound = 1; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
251 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
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 } |
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 if (rtspClient != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
256 // 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
|
257 if (!rtspClient->playMediaSession(*mediaSession)) break; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
258 } else if (sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
259 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
|
260 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
261 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
262 // 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
|
263 // MPlayer codec-specific initialization on each subsession: |
e74916774667
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264 iter.reset(); |
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265 while ((subsession = iter.next()) != NULL) { |
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266 if (subsession->readSource() == NULL) continue; // not reading this |
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267 |
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268 unsigned flags = 0; |
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269 if (strcmp(subsession->mediumName(), "audio") == 0) { |
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270 rtpState->audioBufferQueue |
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271 = new ReadBufferQueue(subsession, demuxer, "audio"); |
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272 rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue); |
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273 rtpCodecInitialize_audio(demuxer, subsession, flags); |
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274 } else if (strcmp(subsession->mediumName(), "video") == 0) { |
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275 rtpState->videoBufferQueue |
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276 = new ReadBufferQueue(subsession, demuxer, "video"); |
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277 rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue); |
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278 rtpCodecInitialize_video(demuxer, subsession, flags); |
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279 } |
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280 rtpState->flags |= flags; |
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281 } |
12043
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Fixed a bug that was accidentally introduced by the addition of MPEG Transport
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282 success = True; |
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283 } while (0); |
12043
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Fixed a bug that was accidentally introduced by the addition of MPEG Transport
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284 if (!success) return NULL; // an error occurred |
11984
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Added support for multiplexed audio+video RTP streams.
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285 |
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286 // Hack: If audio and video are demuxed together on a single RTP stream, |
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287 // then create a new "demuxer_t" structure to allow the higher-level |
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288 // code to recognize this: |
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289 if (demux_is_multiplexed_rtp_stream(demuxer)) { |
94e1953fbe8e
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290 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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291 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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292 audio_id, video_id, dvdsub_id, NULL); |
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293 demuxer = new_demuxers_demuxer(od, od, od); |
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294 } |
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295 |
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296 return demuxer; |
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297 } |
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298 |
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299 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) { |
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300 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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301 RTPState* rtpState = (RTPState*)(demuxer->priv); |
9250 | 302 |
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303 return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0; |
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304 } |
1a747aee653b
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305 |
10477
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Added support for checking whether a RTP demuxer contains combined audio_video data.
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306 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) { |
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307 // Get the RTP state that was stored in the demuxer's 'priv' field: |
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308 RTPState* rtpState = (RTPState*)(demuxer->priv); |
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309 |
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310 return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0; |
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311 } |
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312 |
9565
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313 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds, |
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314 Boolean mustGetNewData, |
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315 float& ptsBehind); // forward |
6910
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316 |
1a747aee653b
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317 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) { |
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318 // Get a filled-in "demux_packet" from the RTP source, and deliver it. |
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319 // Note that this is called as a synchronous read operation, so it needs |
1a747aee653b
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arpi
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|
320 // to block in the (hopefully infrequent) case where no packet is |
1a747aee653b
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arpi
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321 // immediately available. |
1a747aee653b
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322 |
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323 while (1) { |
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324 float ptsBehind; |
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325 demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking |
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326 if (dp == NULL) return 0; |
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327 |
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328 if (demuxer->stream->eof) return 0; // source stream has closed down |
29263
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diego
parents:
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329 |
9565
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330 // Before using this packet, check to make sure that its presentation |
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331 // time is not far behind the other stream (if any). If it is, |
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332 // then we discard this packet, and get another instead. (The rest of |
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|
333 // MPlayer doesn't always do a good job of synchronizing when the |
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|
334 // audio and video streams get this far apart.) |
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|
335 // (We don't do this when streaming over TCP, because then the audio and |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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|
336 // 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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|
337 // (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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|
338 // 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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|
339 // an error in the source's timestamp synchronization.) |
9565
e74916774667
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|
340 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
|
341 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
|
342 if (ptsBehind < ptsBehindThreshold || |
b41cc72bb8ad
We now allow for the possibility of the RTCP audio/video synchronization being
rsf
parents:
11984
diff
changeset
|
343 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
|
344 rtsp_transport_tcp) { // packet's OK |
9565
e74916774667
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diff
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|
345 ds_add_packet(ds, dp); |
e74916774667
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|
346 break; |
e74916774667
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changeset
|
347 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
348 |
9786
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
parents:
9565
diff
changeset
|
349 #ifdef DEBUG_PRINT_DISCARDED_PACKETS |
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
parents:
9565
diff
changeset
|
350 RTPState* rtpState = (RTPState*)(demuxer->priv); |
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
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diff
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|
351 ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue; |
177a3393057c
Added some optional debugging printfs (disabled by default)
rsf
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diff
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|
352 fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind); |
177a3393057c
Added some optional debugging printfs (disabled by default)
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diff
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|
353 #endif |
9565
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|
354 free_demux_packet(dp); // give back this packet, and get another one |
e74916774667
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|
355 } |
e74916774667
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changeset
|
356 |
e74916774667
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|
357 return 1; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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changeset
|
358 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
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diff
changeset
|
359 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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changeset
|
360 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds, |
e74916774667
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|
361 unsigned char*& packetData, unsigned& packetDataLen, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
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|
362 float& pts) { |
e74916774667
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changeset
|
363 // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet" |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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changeset
|
364 // is not delivered to the "demux_stream". |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
365 float ptsBehind; |
e74916774667
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diff
changeset
|
366 demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
367 if (dp == NULL) return False; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
368 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
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diff
changeset
|
369 packetData = dp->buffer; |
e74916774667
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parents:
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diff
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|
370 packetDataLen = dp->len; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
371 pts = dp->pts; |
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:
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|
373 return True; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
374 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
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diff
changeset
|
375 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
376 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
|
377 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
378 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
|
379 // 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
|
380 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
381 // 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
|
382 RTPState* rtpState = (RTPState*)(demuxer->priv); |
7143
fdfc10c2e2d2
- Tell the RTSP client code to use the string "mplayer" in RTSP
arpi
parents:
7008
diff
changeset
|
383 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
|
384 |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
385 teardownRTSPorSIPSession(rtpState); |
9565
e74916774667
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parents:
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changeset
|
386 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
387 UsageEnvironment* env = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
388 TaskScheduler* scheduler = NULL; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
389 if (rtpState->mediaSession != NULL) { |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
390 env = &(rtpState->mediaSession->envir()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
391 scheduler = &(env->taskScheduler()); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
392 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
393 Medium::close(rtpState->mediaSession); |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
394 Medium::close(rtpState->rtspClient); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
395 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
|
396 delete rtpState->audioBufferQueue; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
397 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
|
398 delete[] rtpState->sdpDescription; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
399 delete rtpState; |
29451
6aca83f5ba73
Fix a crash when playing some H264 over rtsp streams: Do pass an
cehoyos
parents:
29280
diff
changeset
|
400 #ifdef CONFIG_LIBAVCODEC |
29460 | 401 av_freep(&avcctx); |
29451
6aca83f5ba73
Fix a crash when playing some H264 over rtsp streams: Do pass an
cehoyos
parents:
29280
diff
changeset
|
402 #endif |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
403 |
12866
332f24a1dd5c
Updated to conform to a small change in the LIVE.COM API.
rsf
parents:
12531
diff
changeset
|
404 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
|
405 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
406 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
407 ////////// 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
|
408 |
27776
ef0b0f9291a2
increase the max RTP packet size to 5MB as modern Elphel
attila
parents:
27341
diff
changeset
|
409 #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
|
410 // >= 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
|
411 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
412 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
|
413 unsigned /*numTruncatedBytes*/, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
414 struct timeval presentationTime, |
a79a318585ce
Changed to conform to recent changes to the "LIVE.COM Streaming Media" code.
rsf
parents:
10937
diff
changeset
|
415 unsigned /*durationInMicroseconds*/) { |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
416 int headersize = 0; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
417 if (frameSize >= MAX_RTP_FRAME_SIZE) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
418 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
|
419 MAX_RTP_FRAME_SIZE); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
420 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
421 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
|
422 demuxer_t* demuxer = bufferQueue->ourDemuxer(); |
7534 | 423 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
|
424 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
425 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
|
426 |
22851
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
427 demux_packet_t* dp = bufferQueue->dp; |
e878a71cb27f
Cosmetics: Move variable declaration and initialization up.
cehoyos
parents:
22759
diff
changeset
|
428 |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
429 if (bufferQueue->readSource()->isAMRAudioSource()) |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
430 headersize = 1; |
22852 | 431 else if (bufferQueue == rtpState->videoBufferQueue && |
432 ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) { | |
433 dp->buffer[0]=0x00; | |
434 dp->buffer[1]=0x00; | |
435 dp->buffer[2]=0x01; | |
436 headersize = 3; | |
437 } | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
438 |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
439 resize_demux_packet(dp, frameSize + headersize); |
7534 | 440 |
441 // Set the packet's presentation time stamp, depending on whether or | |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
442 // 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
|
443 Boolean hasBeenSynchronized |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
444 = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP(); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
445 if (hasBeenSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
446 if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
447 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
|
448 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
449 } |
7534 | 450 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
451 struct timeval* fst = &(rtpState->firstSyncTime); // abbrev |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
452 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
|
453 *fst = presentationTime; |
7534 | 454 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
455 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
456 // 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
|
457 // 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
|
458 // 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
|
459 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
|
460 + (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
|
461 bufferQueue->prevPacketPTS = dp->pts; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
462 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
463 if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
464 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
|
465 bufferQueue->tag()); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
466 } |
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 // 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
|
469 dp->pts = bufferQueue->prevPacketPTS; |
7534 | 470 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
471 bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized; |
7534 | 472 |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
473 dp->pos = demuxer->filepos; |
22759 | 474 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
|
475 |
9250 | 476 // 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
|
477 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
478 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
479 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
480 static void onSourceClosure(void* clientData) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
481 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
|
482 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
|
483 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
484 demuxer->stream->eof = 1; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
485 |
9250 | 486 // 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
|
487 bufferQueue->blockingFlag = ~0; |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
488 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
489 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
490 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
|
491 Boolean mustGetNewData, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
492 float& ptsBehind) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
493 // 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
|
494 // (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
|
495 // the demuxer's 'priv' field) |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
496 RTPState* rtpState = (RTPState*)(demuxer->priv); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
497 ReadBufferQueue* bufferQueue = NULL; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
498 int headersize = 0; |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
499 TaskToken task; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
500 |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
501 if (demuxer->stream->eof) return NULL; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
502 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
503 if (ds == demuxer->video) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
504 bufferQueue = rtpState->videoBufferQueue; |
22852 | 505 if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4')) |
506 headersize = 3; | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
507 } else if (ds == demuxer->audio) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
508 bufferQueue = rtpState->audioBufferQueue; |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
509 if (bufferQueue->readSource()->isAMRAudioSource()) |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
510 headersize = 1; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
511 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
512 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
|
513 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
514 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
515 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
516 if (bufferQueue == NULL || bufferQueue->readSource() == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
517 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
|
518 return NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
519 } |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
520 |
23555 | 521 demux_packet_t* dp = NULL; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
522 if (!mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
523 // 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
|
524 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
|
525 if (dp != NULL) { |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
526 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
|
527 return dp; |
fa45d335a43b
Fixed a bug that could sometimes cause the first video packet in a RTP stream
rsf
parents:
9904
diff
changeset
|
528 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
529 } |
9250 | 530 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
531 // Allocate a new packet buffer, and arrange to read into it: |
22852 | 532 if (!bufferQueue->nextpacket) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
533 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
|
534 bufferQueue->dp = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
535 if (dp == NULL) return NULL; |
22852 | 536 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
537 |
27341
e7c989f7a7c9
Start unifying names of internal preprocessor directives.
diego
parents:
25270
diff
changeset
|
538 #ifdef CONFIG_LIBAVCODEC |
22852 | 539 extern AVCodecParserContext * h264parserctx; |
540 int consumed, poutbuf_size = 1; | |
23224 | 541 const uint8_t *poutbuf = NULL; |
23555 | 542 float lastpts = 0.0; |
22852 | 543 |
544 do { | |
545 if (!bufferQueue->nextpacket) { | |
546 #endif | |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
547 // Schedule the read operation: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
548 bufferQueue->blockingFlag = 0; |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
549 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
|
550 afterReading, bufferQueue, |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
551 onSourceClosure, bufferQueue); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
552 // Block ourselves until data becomes available: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
553 TaskScheduler& scheduler |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
554 = bufferQueue->readSource()->envir().taskScheduler(); |
23569 | 555 int delay = 10000000; |
556 if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime()) | |
557 delay /= 10; | |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
558 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
|
559 scheduler.doEventLoop(&bufferQueue->blockingFlag); |
23554
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
560 scheduler.unscheduleDelayedTask(task); |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
561 if (demuxer->stream->eof) { |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
562 free_demux_packet(dp); |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
563 return NULL; |
046d7b14d7f1
Stop playing at eof or if no data arrives for ten seconds.
cehoyos
parents:
23224
diff
changeset
|
564 } |
9250 | 565 |
22519
5f3a0a712afb
cosmetics: renamed variable amr as headersize, more meaningful; patch by Carl Eugen Hoyos
nicodvb
parents:
22463
diff
changeset
|
566 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
|
567 dp->buffer[0] = |
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
568 ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader(); |
27341
e7c989f7a7c9
Start unifying names of internal preprocessor directives.
diego
parents:
25270
diff
changeset
|
569 #ifdef CONFIG_LIBAVCODEC |
22852 | 570 } else { |
571 bufferQueue->dp = dp = bufferQueue->nextpacket; | |
572 bufferQueue->nextpacket = NULL; | |
573 } | |
574 if (headersize == 3 && h264parserctx) { // h264 | |
575 consumed = h264parserctx->parser->parser_parse(h264parserctx, | |
29453 | 576 avcctx, |
22852 | 577 &poutbuf, &poutbuf_size, |
578 dp->buffer, dp->len); | |
579 | |
580 if (!consumed && !poutbuf_size) | |
581 return NULL; | |
582 | |
583 if (!poutbuf_size) { | |
584 lastpts=dp->pts; | |
585 free_demux_packet(dp); | |
586 bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE); | |
587 } else { | |
588 bufferQueue->nextpacket = dp; | |
589 bufferQueue->dp = dp = new_demux_packet(poutbuf_size); | |
590 memcpy(dp->buffer, poutbuf, poutbuf_size); | |
591 dp->pts=lastpts; | |
592 } | |
593 } | |
594 } while (!poutbuf_size); | |
595 #endif | |
22463
979b2aa16e80
support for AMR; it works inserting in the first byte of the demux_packet
nicodvb
parents:
22454
diff
changeset
|
596 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
597 // 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
|
598 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
|
599 && bufferQueue->prevPacketWasSynchronized |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
600 && *(bufferQueue->otherQueue) != NULL |
384f6a88a31d
Changed the criteria for when to drop RTP packets whose timestamp is too far
rsf
parents:
10477
diff
changeset
|
601 && (*(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
|
602 && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
603 ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
604 - bufferQueue->prevPacketPTS; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
605 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
606 ptsBehind = 0.0; |
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 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
609 if (mustGetNewData) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
610 // Save this buffer for future reads: |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
611 bufferQueue->savePendingBuffer(dp); |
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 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
614 return dp; |
9250 | 615 } |
616 | |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
617 static void teardownRTSPorSIPSession(RTPState* rtpState) { |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
618 MediaSession* mediaSession = rtpState->mediaSession; |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
619 if (mediaSession == NULL) return; |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
620 if (rtpState->rtspClient != NULL) { |
22893
777031848801
Use teardownMediaSession() instead of teardownMediaSubsession()
cehoyos
parents:
22852
diff
changeset
|
621 rtpState->rtspClient->teardownMediaSession(*mediaSession); |
10054
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
622 } else if (rtpState->sipClient != NULL) { |
1342075f81f0
Added SIP (IP telephony) client support. (This was already supported in the
rsf
parents:
9911
diff
changeset
|
623 rtpState->sipClient->sendBYE(); |
9250 | 624 } |
625 } | |
626 | |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
627 ////////// "ReadBuffer" and "ReadBufferQueue" implementation: |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
628 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
629 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession, |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
630 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
|
631 : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL), |
23555 | 632 dp(NULL), nextpacket(NULL), |
633 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
|
634 fReadSource(subsession == NULL ? NULL : subsession->readSource()), |
7534 | 635 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
|
636 fOurDemuxer(demuxer), fTag(strdup(tag)) { |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
637 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
638 |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
639 ReadBufferQueue::~ReadBufferQueue() { |
29585
a17d4f8040f1
free(), delete and delete[] are all different and can't just be used at random,
reimar
parents:
29460
diff
changeset
|
640 free((void *)fTag); |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
641 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
642 // 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
|
643 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
644 while (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
645 demux_packet_t* dpNext = dp->next; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
646 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
647 free_demux_packet(dp); |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
648 dp = dpNext; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
649 } |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
650 } |
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
651 |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
652 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
|
653 // 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
|
654 if (pendingDPTail == NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
655 pendingDPHead = pendingDPTail = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
656 } else { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
657 pendingDPTail->next = dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
658 pendingDPTail = dp; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
659 } |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
660 dp->next = NULL; |
6910
1a747aee653b
applied live.com streaming patch (-sdp and rtsp:// support) by Ross Finlayson <finlayson@live.com>
arpi
parents:
diff
changeset
|
661 } |
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 demux_packet_t* ReadBufferQueue::getPendingBuffer() { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
664 demux_packet_t* dp = pendingDPHead; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
665 if (dp != NULL) { |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
666 pendingDPHead = dp->next; |
29263
0f1b5b68af32
whitespace cosmetics: Remove all trailing whitespace.
diego
parents:
29238
diff
changeset
|
667 if (pendingDPHead == NULL) pendingDPTail = NULL; |
9565
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
668 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
669 dp->next = NULL; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
670 } |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
671 |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
672 return dp; |
e74916774667
Improved RTP packet buffering, by relying on the underlying OS's UDP
rsf
parents:
9457
diff
changeset
|
673 } |
16175 | 674 |
30644
390c8d36d463
Drop weird and unnecessary _s/_st suffixes from demuxer struct names.
diego
parents:
29585
diff
changeset
|
675 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
|
676 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
|
677 |
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 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
|
679 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
|
680 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
|
681 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
|
682 *((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
|
683 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
|
684 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
parents:
22353
diff
changeset
|
685 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
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686 if (endpts <= 0) |
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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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687 return DEMUXER_CTRL_DONTKNOW; |
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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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688 *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts); |
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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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689 return DEMUXER_CTRL_OK; |
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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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690 |
20be7be45f47
implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
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691 default: |
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implementation of DEMUXER_CTRL_GET_TIME_LENGTH and DEMUXER_CTRL_GET_PERCENT_POS; patch by the usual Hoyos
nicodvb
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692 return DEMUXER_CTRL_NOTIMPL; |
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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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693 } |
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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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694 } |
16175 | 695 |
696 demuxer_desc_t demuxer_desc_rtp = { | |
16572
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697 "LIVE555 RTP demuxer", |
25270 | 698 "live555", |
16175 | 699 "", |
700 "Ross Finlayson", | |
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701 "requires LIVE555 Streaming Media library", |
16175 | 702 DEMUXER_TYPE_RTP, |
703 0, // no autodetect | |
704 NULL, | |
705 demux_rtp_fill_buffer, | |
706 demux_open_rtp, | |
707 demux_close_rtp, | |
708 NULL, | |
22355
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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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709 demux_rtp_control |
16175 | 710 }; |