/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | RTPFile.h | 30 const int16_t seqNo, const uint8_t* payloadData, 35 virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, 49 const uint8_t* payloadData, uint16_t payloadSize, 57 uint8_t* payloadData; member in class:webrtc::RTPPacket 69 const int16_t seqNo, const uint8_t* payloadData, 72 virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, 101 const int16_t seqNo, const uint8_t* payloadData, 104 virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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H A D | Channel.cc | 22 const uint32_t timeStamp, const uint8_t* payloadData, 64 payloadData + fragmentation->fragmentationOffset[1], 68 payloadData + fragmentation->fragmentationOffset[0], 73 memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0], 79 memcpy(_payloadData, payloadData, payloadDataSize); 95 //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile); 21 SendData(const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, const uint8_t* payloadData, const uint16_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
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H A D | EncodeDecodeTest.cc | 39 const uint32_t timeStamp, const uint8_t* payloadData, 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 | 63 const uint8_t* payloadData, uint16_t payloadSize, 71 this->payloadData = new uint8_t[payloadSize]; 72 memcpy(this->payloadData, payloadData, payloadSize); 77 delete[] payloadData; 89 const int16_t seqNo, const uint8_t* payloadData, 91 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, 98 uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, argument 110 memcpy(payloadData, packet->payloadData, packe 62 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, const uint8_t* payloadData, uint16_t payloadSize, uint32_t frequency) argument 88 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 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 210 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, uint16_t payloadSize, uint32_t* offset) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | coder.cc | 111 const uint8_t* payloadData, 115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 107 SendData( FrameType , uint8_t , uint32_t , const uint8_t* payloadData, uint16_t payloadSize, const RTPFragmentationHeader* ) argument
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H A D | video_coder.cc | 115 const uint8_t* payloadData, 127 memcpy(_videoEncodedData->payloadData, payloadData, 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/test/testAPI/ |
H A D | test_api.h | 92 const uint8_t* payloadData, 96 memcpy(_payloadData, payloadData, payloadSize); 91 OnReceivedPayloadData( const uint8_t* payloadData, const uint16_t payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) argument
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H A D | test_api_audio.cc | 29 const uint8_t* payloadData, 36 memcpy(str, payloadData, payloadSize); 49 if (payloadData[0] == 0xff) { 28 OnReceivedPayloadData( const uint8_t* payloadData, const uint16_t payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_sender_unittest.cc | 264 virtual int OnReceivedPayloadData(const uint8_t* payloadData, argument
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H A D | rtp_sender_audio.cc | 235 const uint8_t* payloadData, 336 if (payloadSize == 0 || payloadData == NULL) { 400 payloadData + fragmentation->fragmentationOffset[1], 406 payloadData + fragmentation->fragmentationOffset[0], 416 payloadData + fragmentation->fragmentationOffset[0], 427 payloadData + fragmentation->fragmentationOffset[0], 433 memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize); 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
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H A D | rtp_sender_video.cc | 273 const uint8_t* payloadData, 297 payloadData, 322 const uint8_t* payloadData, 328 const uint8_t* data = payloadData; 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 | rtcp_receiver_unittest.cc | 53 virtual int OnReceivedPayloadData(const uint8_t* payloadData, argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
H A D | TestLoadGenerator.cc | 133 const uint8_t* payloadData, 138 return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType)); 132 sendPayload(const uint32_t timeStamp, const uint8_t* payloadData, const uint32_t payloadSize, const webrtc::FrameType frameType ) argument
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H A D | TestSenderReceiver.cc | 309 int32_t TestSenderReceiver::OnReceivedPayloadData(const uint8_t* payloadData, argument 405 const uint8_t* payloadData, 409 return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize)); 404 SendOutgoingData(const uint32_t timeStamp, const uint8_t* payloadData, const uint32_t payloadSize, const webrtc::FrameType frameType ) argument
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | generic_codec_test.cc | 540 const uint8_t* payloadData, 557 return _vcm.IncomingPacket(payloadData, payloadSize, rtpInfo); 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 | 78 const uint8_t* payloadData, 87 if (fwrite(payloadData, 1, payloadSize, _encodedFile) != payloadSize) { 122 _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo); 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 | 59 const uint8_t* payloadData, 67 if (fwrite(payloadData, 1, payloadSize, _encodedFile) != payloadSize) { 100 int ret = _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo); 151 const uint8_t* payloadData, 163 payloadData, 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/voice_engine/include/ |
H A D | voe_rtp_rtcp.h | 278 const char* payloadData, unsigned short payloadSize) { return -1; }; 276 InsertExtraRTPPacket( int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize) argument
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/external/chromium_org/third_party/webrtc/modules/interface/ |
H A D | module_common_types.h | 297 payloadData(NULL), 317 payloadData = new uint8_t[data.payloadSize]; 318 memcpy(payloadData, data.payloadData, data.payloadSize); 320 payloadData = NULL; 325 delete[] payloadData; 344 delete[] payloadData; 345 payloadData = new uint8_t[data.payloadSize]; 346 memcpy(payloadData, data.payloadData, dat 369 uint8_t* payloadData; member in class:webrtc::EncodedVideoData [all...] |
/external/webrtc/src/modules/interface/ |
H A D | module_common_types.h | 308 payloadData(NULL), 328 payloadData = new WebRtc_UWord8[data.payloadSize]; 329 memcpy(payloadData, data.payloadData, data.payloadSize); 333 payloadData = NULL; 340 delete [] payloadData; 362 delete [] payloadData; 363 payloadData = new WebRtc_UWord8[data.payloadSize]; 364 memcpy(payloadData, data.payloadData, dat 389 WebRtc_UWord8* payloadData; member in class:webrtc::EncodedVideoData 792 UpdateFrame( const WebRtc_Word32 id, const WebRtc_UWord32 timeStamp, const WebRtc_Word16* payloadData, const WebRtc_UWord16 payloadDataLengthInSamples, const int frequencyInHz, const SpeechType speechType, const VADActivity vadActivity, const WebRtc_UWord8 audioChannel, const WebRtc_Word32 volume, const WebRtc_Word32 energy) argument [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | channel.cc | 114 const uint8_t* payloadData, 141 payloadData, 509 Channel::OnReceivedPayloadData(const uint8_t* payloadData, argument 533 if (audio_coding_->IncomingPacket(payloadData, 111 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, uint16_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
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