/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
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; 69 const int16_t seqNo, const uint8_t* payloadData, 72 uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, 101 const int16_t seqNo, const uint8_t* payloadData, 104 uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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RTPFile.cc | 62 const uint8_t* payloadData, uint16_t payloadSize, 70 this->payloadData = new uint8_t[payloadSize]; 71 memcpy(this->payloadData, payloadData, payloadSize); 76 delete[] payloadData; 88 const int16_t seqNo, const uint8_t* payloadData, 90 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, 97 uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, 109 memcpy(payloadData, packet->payloadData, packet->payloadSize) [all...] |
Channel.h | 54 const uint32_t timeStamp, const uint8_t* payloadData,
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EncodeDecodeTest.h | 33 const uint32_t timeStamp, const uint8_t* payloadData,
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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);
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
video_coder.cc | 115 const uint8_t* payloadData, 127 memcpy(_videoEncodedData->payloadData, payloadData,
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coder.h | 48 const uint8_t* payloadData,
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video_coder.h | 55 const uint8_t* payloadData,
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coder.cc | 111 const uint8_t* payloadData, 115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
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/external/webrtc/src/modules/interface/ |
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, data.payloadSize) [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api.h | 90 const uint8_t* payloadData, 94 memcpy(_payloadData, payloadData, payloadSize);
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/external/chromium_org/third_party/webrtc/modules/interface/ |
module_common_types.h | 305 payloadData(NULL), 325 payloadData = new uint8_t[data.payloadSize]; 326 memcpy(payloadData, data.payloadData, data.payloadSize); 328 payloadData = NULL; 333 delete[] payloadData; 352 delete[] payloadData; 353 payloadData = new uint8_t[data.payloadSize]; 354 memcpy(payloadData, data.payloadData, data.payloadSize) [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_sender_audio.cc | 235 const uint8_t* payloadData, 334 if (payloadSize == 0 || payloadData == NULL) { 398 payloadData + fragmentation->fragmentationOffset[1], 404 payloadData + fragmentation->fragmentationOffset[0], 414 payloadData + fragmentation->fragmentationOffset[0], 425 payloadData + fragmentation->fragmentationOffset[0], 431 memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
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rtp_sender_video.h | 53 const uint8_t* payloadData, 109 const uint8_t* payloadData,
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rtp_sender_video.cc | 282 const uint8_t* payloadData, 310 capture_time_ms, payloadData, payloadSize); 317 payloadData, 413 const uint8_t* payloadData, 421 const uint8_t* data = payloadData;
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rtp_sender_audio.h | 40 const uint8_t* payloadData,
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
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,
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test_callbacks.h | 51 const uint8_t* payloadData, 108 const uint8_t* payloadData,
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generic_codec_test.h | 98 const uint8_t* payloadData,
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/external/chromium_org/third_party/WebKit/Source/devtools/front_end/network/ |
ResourceWebSocketFrameView.js | 58 length: typeof payload.payloadData === "undefined" ? payload.errorMessage.length.toString() : payload.payloadData.length.toString(), 70 row.data = payload.payloadData;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
TestSenderReceiver.h | 99 virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData, 127 const uint8_t* payloadData,
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TestSenderReceiver.cc | 309 int32_t TestSenderReceiver::OnReceivedPayloadData(const uint8_t* payloadData, 405 const uint8_t* payloadData, 409 return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
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/external/chromium_org/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ |
GAEChannelClient.m | 133 NSDictionary* payloadData = 135 [self.delegate onMessage:payloadData];
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/H264/ |
rtp_sender_h264.h | 28 const uint8_t* payloadData, 35 const uint8_t* payloadData,
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/interface/ |
video_coding_defines.h | 73 const uint8_t* payloadData,
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