| /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 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,
|
| 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,
110 memcpy(payloadData, packet->payloadData, packet->payloadSize)
[all...] |
| Channel.h | 55 const uint32_t timeStamp, const uint8_t* payloadData,
|
| EncodeDecodeTest.h | 34 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 | 49 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/source/ |
| rtp_sender_video.h | 50 const uint8_t* payloadData,
101 const uint8_t* payloadData,
|
| 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);
|
| rtp_sender_audio.h | 39 const uint8_t* payloadData,
|
| rtp_sender_video.cc | 273 const uint8_t* payloadData,
297 payloadData,
322 const uint8_t* payloadData,
328 const uint8_t* data = payloadData;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 92 const uint8_t* payloadData,
96 memcpy(_payloadData, payloadData, payloadSize);
|
| /external/chromium_org/third_party/webrtc/modules/interface/ |
| 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, data.payloadSize)
[all...] |
| /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,
|
| normal_test.h | 40 const uint8_t* payloadData,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| TestSenderReceiver.h | 102 const uint8_t* payloadData,
130 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));
|
| TestLoadGenerator.h | 41 const uint8_t* payloadData,
|
| /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/video_coding/main/interface/ |
| video_coding_defines.h | 74 const uint8_t* payloadData,
|
