/dports/devel/mutagen/mutagen-0.11.8/vendor/github.com/pion/webrtc/v2/ |
H A D | mediaengine.go | 63 payloadType := uint8(pt) 101 if codec.PayloadType == payloadType { 150 payloadType, 162 payloadType, 174 payloadType, 186 payloadType, 198 payloadType, 210 payloadType, 222 payloadType, 278 payloadType uint8, [all …]
|
/dports/net/h323plus/h323plus-1_27_2/src/ |
H A D | rtp2wav.cxx | 52 payloadType = RTP_DataFrame::IllegalPayloadType; in OpalRtpToWavFile() 67 payloadType = RTP_DataFrame::IllegalPayloadType; in OpalRtpToWavFile() 85 payloadType = frame.GetPayloadType(); in OnFirstPacket() 87 if (payloadType >= PARRAYSIZE(SupportedTypes) || SupportedTypes[payloadType] == 0) { in OnFirstPacket() 88 PTRACE(1, "rtp2wav\tUnsupported payload type: " << payloadType); in OnFirstPacket() 92 if (!SetFormat(SupportedTypes[payloadType])) { in OnFirstPacket() 93 PTRACE(1, "rtp2wav\tCould not set WAV file format: " << SupportedTypes[payloadType]); in OnFirstPacket() 102 PTRACE(3, "rtp2wav\tStarted recording payload type " << payloadType in OnFirstPacket() 112 if (payloadType == RTP_DataFrame::IllegalPayloadType) { in ReceivedPacket() 120 if (payloadType != frame.GetPayloadType()) in ReceivedPacket()
|
/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 22 uint8_t payloadType, in SendData() argument 35 rtpInfo.header.payloadType = payloadType; in SendData() 149 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 205 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 218 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 245 _payloadStats[n].payloadType = -1; in Channel() 279 _payloadStats[n].payloadType = -1; in ResetStats() 298 payloadStats.payloadType = -1; in Stats() 305 if (payloadStats.payloadType == -1) { in Stats() 356 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
H A D | RTPFile.cc | 32 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader() 43 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, in MakeRTPheader() argument 47 rtpHeader[1] = payloadType; in MakeRTPheader() 60 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, in RTPPacket() argument 63 : payloadType(payloadType), in RTPPacket() 86 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 89 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 103 rtpInfo->header.payloadType = packet->payloadType; in Read() 169 void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 174 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 22 uint8_t payloadType, in SendData() argument 35 rtpInfo.header.payloadType = payloadType; in SendData() 149 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 205 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 218 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 245 _payloadStats[n].payloadType = -1; in Channel() 279 _payloadStats[n].payloadType = -1; in ResetStats() 298 payloadStats.payloadType = -1; in Stats() 305 if (payloadStats.payloadType == -1) { in Stats() 356 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_coding/main/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 36 rtpInfo.header.payloadType = payloadType; in SendData() 150 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 206 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 219 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 247 _payloadStats[n].payloadType = -1; in Channel() 282 _payloadStats[n].payloadType = -1; in ResetStats() 301 payloadStats.payloadType = -1; in Stats() 308 if (payloadStats.payloadType == -1) { in Stats() 359 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
H A D | RTPFile.cc | 32 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader() 43 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, in MakeRTPheader() argument 47 rtpHeader[1] = payloadType; in MakeRTPheader() 60 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, in RTPPacket() argument 63 : payloadType(payloadType), in RTPPacket() 86 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 89 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 103 rtpInfo->header.payloadType = packet->payloadType; in Read() 169 void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 174 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 22 uint8_t payloadType, in SendData() argument 35 rtpInfo.header.payloadType = payloadType; in SendData() 149 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 205 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 218 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 245 _payloadStats[n].payloadType = -1; in Channel() 279 _payloadStats[n].payloadType = -1; in ResetStats() 298 payloadStats.payloadType = -1; in Stats() 305 if (payloadStats.payloadType == -1) { in Stats() 356 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
H A D | RTPFile.cc | 32 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader() 43 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, in MakeRTPheader() argument 47 rtpHeader[1] = payloadType; in MakeRTPheader() 60 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, in RTPPacket() argument 63 : payloadType(payloadType), in RTPPacket() 86 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 89 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 103 rtpInfo->header.payloadType = packet->payloadType; in Read() 169 void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 174 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 22 uint8_t payloadType, in SendData() argument 35 rtpInfo.header.payloadType = payloadType; in SendData() 149 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 205 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 218 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 245 _payloadStats[n].payloadType = -1; in Channel() 279 _payloadStats[n].payloadType = -1; in ResetStats() 298 payloadStats.payloadType = -1; in Stats() 305 if (payloadStats.payloadType == -1) { in Stats() 356 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/main/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 36 rtpInfo.header.payloadType = payloadType; in SendData() 150 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 206 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 219 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() 247 _payloadStats[n].payloadType = -1; in Channel() 282 _payloadStats[n].payloadType = -1; in ResetStats() 301 payloadStats.payloadType = -1; in Stats() 308 if (payloadStats.payloadType == -1) { in Stats() 359 payloadType[k] = (uint8_t) _payloadStats[k].payloadType; in Stats() [all …]
|
/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 38 rtp_header.payloadType = payloadType; in SendData() 96 if ((rtp_header.payloadType != _lastPayloadType) && in CalcStatistics() 103 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics() 109 _lastPayloadType = rtp_header.payloadType; in CalcStatistics() 114 if (rtp_header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 171 currentPayloadStr->payloadType = rtp_header.payloadType; in CalcStatistics() 177 while (_payloadStats[n].payloadType != -1) { in CalcStatistics() 184 _payloadStats[n].payloadType = rtp_header.payloadType; in CalcStatistics() 211 _payloadStats[n].payloadType = -1; in Channel() [all …]
|
H A D | RTPFile.cc | 30 rtp_header->payloadType = rtpHeader[1]; in ParseRTPHeader() 44 uint8_t payloadType, in MakeRTPheader() argument 49 rtpHeader[1] = payloadType; in MakeRTPheader() 62 RTPPacket::RTPPacket(uint8_t payloadType, in RTPPacket() argument 68 : payloadType(payloadType), in RTPPacket() 91 void RTPBuffer::Write(const uint8_t payloadType, in Write() argument 97 RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 113 rtp_header->payloadType = packet->payloadType; in Read() 179 void RTPFile::Write(const uint8_t payloadType, in Write() argument 187 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_coding/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 38 rtp_header.payloadType = payloadType; in SendData() 96 if ((rtp_header.payloadType != _lastPayloadType) && in CalcStatistics() 103 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics() 109 _lastPayloadType = rtp_header.payloadType; in CalcStatistics() 114 if (rtp_header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 171 currentPayloadStr->payloadType = rtp_header.payloadType; in CalcStatistics() 177 while (_payloadStats[n].payloadType != -1) { in CalcStatistics() 184 _payloadStats[n].payloadType = rtp_header.payloadType; in CalcStatistics() 211 _payloadStats[n].payloadType = -1; in Channel() [all …]
|
H A D | RTPFile.cc | 30 rtp_header->payloadType = rtpHeader[1]; in ParseRTPHeader() 44 uint8_t payloadType, in MakeRTPheader() argument 49 rtpHeader[1] = payloadType; in MakeRTPheader() 62 RTPPacket::RTPPacket(uint8_t payloadType, in RTPPacket() argument 68 : payloadType(payloadType), in RTPPacket() 83 void RTPBuffer::Write(const uint8_t payloadType, in Write() argument 89 RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 106 rtp_header->payloadType = packet->payloadType; in Read() 170 void RTPFile::Write(const uint8_t payloadType, in Write() argument 178 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 38 rtp_header.payloadType = payloadType; in SendData() 96 if ((rtp_header.payloadType != _lastPayloadType) && in CalcStatistics() 103 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics() 109 _lastPayloadType = rtp_header.payloadType; in CalcStatistics() 114 if (rtp_header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 171 currentPayloadStr->payloadType = rtp_header.payloadType; in CalcStatistics() 177 while (_payloadStats[n].payloadType != -1) { in CalcStatistics() 184 _payloadStats[n].payloadType = rtp_header.payloadType; in CalcStatistics() 211 _payloadStats[n].payloadType = -1; in Channel() [all …]
|
H A D | RTPFile.cc | 30 rtp_header->payloadType = rtpHeader[1]; in ParseRTPHeader() 44 uint8_t payloadType, in MakeRTPheader() argument 49 rtpHeader[1] = payloadType; in MakeRTPheader() 62 RTPPacket::RTPPacket(uint8_t payloadType, in RTPPacket() argument 68 : payloadType(payloadType), in RTPPacket() 83 void RTPBuffer::Write(const uint8_t payloadType, in Write() argument 89 RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 106 rtp_header->payloadType = packet->payloadType; in Read() 170 void RTPFile::Write(const uint8_t payloadType, in Write() argument 178 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_coding/test/ |
H A D | Channel.cc | 23 uint8_t payloadType, in SendData() argument 38 rtp_header.payloadType = payloadType; in SendData() 96 if ((rtp_header.payloadType != _lastPayloadType) && in CalcStatistics() 103 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics() 109 _lastPayloadType = rtp_header.payloadType; in CalcStatistics() 114 if (rtp_header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 171 currentPayloadStr->payloadType = rtp_header.payloadType; in CalcStatistics() 177 while (_payloadStats[n].payloadType != -1) { in CalcStatistics() 184 _payloadStats[n].payloadType = rtp_header.payloadType; in CalcStatistics() 211 _payloadStats[n].payloadType = -1; in Channel() [all …]
|
H A D | RTPFile.cc | 30 rtp_header->payloadType = rtpHeader[1]; in ParseRTPHeader() 44 uint8_t payloadType, in MakeRTPheader() argument 49 rtpHeader[1] = payloadType; in MakeRTPheader() 62 RTPPacket::RTPPacket(uint8_t payloadType, in RTPPacket() argument 68 : payloadType(payloadType), in RTPPacket() 91 void RTPBuffer::Write(const uint8_t payloadType, in Write() argument 97 RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 113 rtp_header->payloadType = packet->payloadType; in Read() 179 void RTPFile::Write(const uint8_t payloadType, in Write() argument 187 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
|
/dports/multimedia/vvdec/vvdec-1.1.2/source/Lib/CommonLib/ |
H A D | SEI_internal.cpp | 54 const char *SEI_internal::getSEIMessageString( vvdecSEIPayloadType payloadType) in getSEIMessageString() argument 56 switch (payloadType) in getSEIMessageString() 93 if ( s->payloadType == seiType) in getSeisByType() 107 if ((*it)->payloadType == seiType) in extractSeisByType() 126 if( sei->payloadType == VVDEC_SCALABLE_NESTING ) in deleteSEIs() 153 vvdecSEI* SEI_internal::allocSEI( vvdecSEIPayloadType payloadType ) in allocSEI() argument 160 sei->payloadType = (vvdecSEIPayloadType)payloadType; in allocSEI() 183 int size = userDefSize>0 ? userDefSize : getPayloadSize( sei->payloadType ); in allocSEIPayload() 196 int SEI_internal::getPayloadSize(vvdecSEIPayloadType payloadType) in getPayloadSize() argument 198 switch (payloadType) in getPayloadSize()
|
/dports/graphics/libbpg/libbpg-0.9.8/jctvc/TLibCommon/ |
H A D | SEI.h | 94 virtual PayloadType payloadType() const = 0; 160 PayloadType payloadType() const { return BUFFERING_PERIOD; } in payloadType() function 189 PayloadType payloadType() const { return PICTURE_TIMING; } in payloadType() function 228 PayloadType payloadType() const { return DECODING_UNIT_INFO; } in payloadType() function 246 PayloadType payloadType() const { return RECOVERY_POINT; } in payloadType() function 259 PayloadType payloadType() const { return FRAME_PACKING; } in payloadType() function 349 PayloadType payloadType() const { return NO_DISPLAY; } in payloadType() function 362 PayloadType payloadType() const { return SOP_DESCRIPTION; } in payloadType() function 379 PayloadType payloadType() const { return TONE_MAPPING_INFO; } in payloadType() function 493 PayloadType payloadType() const { return SCALABLE_NESTING; } in payloadType() function [all …]
|
/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 72 const int8_t payloadType, in RegisterAudioPayload() argument 82 _cngNBPayloadType = payloadType; in RegisterAudioPayload() 85 _cngWBPayloadType = payloadType; in RegisterAudioPayload() 88 _cngSWBPayloadType = payloadType; in RegisterAudioPayload() 91 _cngFBPayloadType = payloadType; in RegisterAudioPayload() 100 _dtmfPayloadType = payloadType; in RegisterAudioPayload() 165 const int8_t payloadType, in SendAudio() argument 359 _lastPayloadType = payloadType; in SendAudio() 394 if(payloadType < -1 ) in SetRED() 399 _REDPayloadType = payloadType; in SetRED() [all …]
|
/dports/security/snowflake-tor/snowflake-ead5a960d7fa19dc890ccbfc0765c5ab6629eaa9/vendor/github.com/pion/sctp/ |
H A D | reassembly_queue_test.go | 20 payloadType: orgPpi, 32 payloadType: orgPpi, 62 payloadType: orgPpi, 75 payloadType: orgPpi, 87 payloadType: orgPpi, 118 payloadType: orgPpi, 131 payloadType: orgPpi, 176 payloadType: orgPpi, 189 payloadType: orgPpi, 202 payloadType: orgPpi, [all …]
|
/dports/devel/mutagen/mutagen-0.11.8/vendor/github.com/pion/sctp/ |
H A D | reassembly_queue_test.go | 21 payloadType: orgPpi, 33 payloadType: orgPpi, 63 payloadType: orgPpi, 76 payloadType: orgPpi, 88 payloadType: orgPpi, 119 payloadType: orgPpi, 132 payloadType: orgPpi, 177 payloadType: orgPpi, 190 payloadType: orgPpi, 203 payloadType: orgPpi, [all …]
|
/dports/net-im/jitsi-videobridge/jitsi-videobridge-dbddd16/src/main/java/org/jitsi/videobridge/cc/ |
H A D | AdaptiveTrackProjection.java | 245 int payloadType = rtpPacket.getPayloadType(); in getContext() local 247 if (context == null || contextPayloadType != payloadType) in getContext() 249 payloadTypeObject = payloadTypes.get((byte)payloadType); in getContext() 252 … logger.error("No payload type object signalled for payload type " + payloadType + " yet, " + in getContext() 289 + payloadType + in getContext() 293 contextPayloadType = payloadType; in getContext() 311 + payloadType + in getContext() 317 contextPayloadType = payloadType; in getContext() 323 else if (context == null || contextPayloadType != payloadType) in getContext() 331 " generic context for payload type " + payloadType); in getContext() [all …]
|