| /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,
|
| 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,
|
| EncodeDecodeTest.h | 33 const uint32_t timeStamp, const uint8_t* payloadData,
|
| 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);
|
| /external/chromium_org/third_party/webrtc/modules/utility/source/ |
| video_coder.cc | 115 const uint8_t* payloadData,
127 memcpy(_videoEncodedData->payloadData, payloadData,
|
| coder.h | 48 const uint8_t* payloadData,
|
| video_coder.h | 55 const uint8_t* payloadData,
|
| coder.cc | 111 const uint8_t* payloadData,
115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
|
| /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);
|
| /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);
|
| rtp_sender_video.h | 53 const uint8_t* payloadData,
109 const uint8_t* payloadData,
|
| 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;
|
| rtp_sender_audio.h | 40 const uint8_t* payloadData,
|
| /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,
|
| test_callbacks.h | 51 const uint8_t* payloadData,
108 const uint8_t* payloadData,
|
| generic_codec_test.h | 98 const uint8_t* payloadData,
|
| /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;
|
| /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,
|
| 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));
|
| /external/chromium_org/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ |
| GAEChannelClient.m | 133 NSDictionary* payloadData =
135 [self.delegate onMessage:payloadData];
|
| /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,
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/interface/ |
| video_coding_defines.h | 73 const uint8_t* payloadData,
|
