/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | generic_encoder.h | 52 void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; }; argument
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H A D | packet.h | 34 uint8_t payloadType; member in class:webrtc::VCMPacket
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H A D | video_sender.cc | 185 uint8_t payloadType, 192 _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec); 200 externalEncoder, payloadType, internalSource); 184 RegisterExternalEncoder(VideoEncoder* externalEncoder, uint8_t payloadType, bool internalSource ) argument
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H A D | video_receiver.cc | 332 uint8_t payloadType, 338 return _codecDataBase.DeregisterExternalDecoder(payloadType) ? 0 : -1; 341 externalDecoder, payloadType, internalRenderTiming) 331 RegisterExternalDecoder(VideoDecoder* externalDecoder, uint8_t payloadType, bool internalRenderTiming) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | RTPFile.h | 29 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, 40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo, 48 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, 54 uint8_t payloadType; member in class:webrtc::RTPPacket 68 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, 100 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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H A D | Channel.cc | 21 int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType, argument 33 rtpInfo.header.payloadType = payloadType; 76 rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0]; 126 if ((rtpInfo.header.payloadType != _lastPayloadType) 133 if (_lastPayloadType == _payloadStats[n].payloadType) { 139 _lastPayloadType = rtpInfo.header.payloadType; 144 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { 191 currentPayloadStr->payloadType 332 Stats(uint8_t* payloadType, uint32_t* payloadLenByte) argument [all...] |
H A D | Channel.h | 41 int16_t payloadType; member in struct:webrtc::ACMTestPayloadStats 54 const FrameType frameType, const uint8_t payloadType, 67 void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
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H A D | EncodeDecodeTest.cc | 38 const FrameType /* frameType */, const uint8_t payloadType, 42 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, 37 SendData( const FrameType , const uint8_t payloadType, const uint32_t timeStamp, const uint8_t* payloadData, const uint16_t payloadSize, const RTPFragmentationHeader* ) argument
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H A D | RTPFile.cc | 31 rtpInfo->header.payloadType = rtpHeader[1]; 42 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, argument 46 rtpHeader[1] = (unsigned char) (payloadType & 0xFF); 62 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, argument 65 : payloadType(payloadType), 88 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, argument 91 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, 105 rtpInfo->header.payloadType = packet->payloadType; 181 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const uint16_t payloadSize, uint32_t frequency) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | coder.cc | 61 const uint8_t payloadType = _receiveCodec.pltype; local 65 payloadType, 109 uint8_t /* payloadType */,
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H A D | rtp_dump_impl.cc | 210 const uint8_t payloadType = packet[1]; local 213 switch(payloadType)
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H A D | video_coder.cc | 112 const uint8_t payloadType, 124 _videoEncodedData->payloadType = payloadType; 110 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* ) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 88 const int8_t payloadType, 98 _cngNBPayloadType = payloadType; 101 _cngWBPayloadType = payloadType; 104 _cngSWBPayloadType = payloadType; 107 _cngFBPayloadType = payloadType; 116 _dtmfPayloadType = payloadType; 132 const int8_t payloadType) 138 if(_lastPayloadType != payloadType) 143 if(_cngNBPayloadType == payloadType) 152 if(_cngWBPayloadType == payloadType) 86 RegisterAudioPayload( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t frequency, const uint8_t channels, const uint32_t rate, RtpUtility::Payload*& payload) argument 131 MarkerBit(const FrameType frameType, const int8_t payloadType) argument 231 SendAudio( const FrameType frameType, const int8_t payloadType, const uint32_t captureTimeStamp, const uint8_t* payloadData, const uint32_t dataSize, const RTPFragmentationHeader* fragmentation) argument 475 SetRED(const int8_t payloadType) argument [all...] |
H A D | rtp_sender_video.cc | 78 const int8_t payloadType, 270 const int8_t payloadType, 294 payloadType, 319 const int8_t payloadType, 354 dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms); 76 RegisterVideoPayload( const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int8_t payloadType, const uint32_t maxBitRate, RtpUtility::Payload*& payload) argument 268 SendVideo(const RtpVideoCodecTypes videoType, const FrameType frameType, const int8_t payloadType, const uint32_t captureTimeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader* fragmentation, VideoCodecInformation* codecInfo, const RTPVideoTypeHeader* rtpTypeHdr) argument 317 Send(const RtpVideoCodecTypes videoType, const FrameType frameType, const int8_t payloadType, const uint32_t captureTimeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoTypeHeader* rtpTypeHdr) argument
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H A D | rtp_utility.cc | 250 const uint8_t payloadType = _ptrRTPDataBegin[1]; local 252 switch (payloadType) { 297 header->payloadType = PT; 348 header.payloadType = PT;
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H A D | rtcp_sender.cc | 974 const uint8_t payloadType) 1025 rtcpbuffer[pos] = payloadType; 1859 const int8_t payloadType = feedback_state.send_payload_type; local 1860 if (payloadType == -1) { 1864 (uint8_t)payloadType); 971 BuildRPSI(uint8_t* rtcpbuffer, int& pos, const uint64_t pictureID, const uint8_t payloadType) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api_audio.cc | 32 if (rtpHeader->header.payloadType == 98 || 33 rtpHeader->header.payloadType == 99) { 45 if (rtpHeader->header.payloadType == 100 || 46 rtpHeader->header.payloadType == 101 || 47 rtpHeader->header.payloadType == 102) { 66 const int8_t payloadType, 71 if (payloadType == 96) { 73 "The rate should be 64K for this payloadType"; 64 OnInitializeDecoder( const int32_t id, const int8_t payloadType, const char payloadName[RTP_PAYLOAD_NAME_SIZE], const int frequency, const uint8_t channels, const uint32_t rate) argument
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | generic_codec_test.cc | 537 const uint8_t payloadType, 549 rtpInfo.header.payloadType = payloadType; 535 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& , const webrtc::RTPVideoHeader* ) argument
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H A D | normal_test.cc | 75 const uint8_t payloadType, 108 rtpInfo.header.payloadType = payloadType; 73 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& , const webrtc::RTPVideoHeader* videoHdr) argument
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H A D | test_callbacks.cc | 56 const uint8_t payloadType, 90 rtpInfo.header.payloadType = payloadType; 148 const uint8_t payloadType, 160 payloadType, 306 header.payloadType, &payload_specific)) { 54 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* videoHdr) argument 146 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* videoHdr) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
H A D | NETEQTEST_RTPpacket.cc | 159 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) 199 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) 289 rtp_header->header.payloadType = _rtpInfo.header.payloadType; 353 uint8_t NETEQTEST_RTPpacket::payloadType() const function in class:NETEQTEST_RTPpacket 366 return tempRTPinfo.header.payloadType; 445 _rtpInfo.header.payloadType = pt; 553 RTPinfo->header.payloadType, 627 void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqNo, uint32_t timestamp, uint32_t ssrc, uint8_t markerBit) const argument 638 rtp_data[1]=(unsigned char)(payloadType [all...] |
H A D | RTPencode.cc | 79 void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc); 80 int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen, 247 int payloadType; local 422 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed); 526 printf("Payload type: %i\n",payloadType); 648 red_PT[1] = payloadType; 664 red_PT[1] = payloadType; 678 makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc); 782 if (payloadType==NETEQ_CODEC_G722_PT) 1686 void makeRTPheader(unsigned char* rtp_data, int payloadType, in argument 1706 makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen, int seqNo, uint32_t ssrc) argument [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/include/ |
H A D | voe_rtp_rtcp.h | 277 int channel, unsigned char payloadType, bool markerBit, 276 InsertExtraRTPPacket( int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize) argument
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
H A D | H264TrackImpl.java | 563 int payloadType = 0; field in class:H264TrackImpl.SEIMessage 593 payloadType = 0; 598 payloadType += last_payload_type_bytes; 602 payloadType += last_payload_type_bytes; 613 if (payloadType == 1) { // pic_timing is what we are interested in! 710 "payloadType=" + payloadType + 712 if (payloadType == 1) {
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/external/chromium_org/third_party/webrtc/modules/interface/ |
H A D | module_common_types.h | 290 : payloadType(0), 305 payloadType = data.payloadType; 332 payloadType = data.payloadType; 362 uint8_t payloadType; member in class:webrtc::EncodedVideoData
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