| /external/webrtc/webrtc/modules/audio_coding/test/ |
| RTPFile.h | 30 const int16_t seqNo, const uint8_t* payloadData,
35 virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
49 const uint8_t* payloadData, size_t payloadSize,
57 uint8_t* payloadData;
71 const uint8_t* payloadData,
76 uint8_t* payloadData,
108 const uint8_t* payloadData,
113 uint8_t* payloadData,
|
| RTPFile.cc | 61 const uint8_t* payloadData, size_t payloadSize,
69 this->payloadData = new uint8_t[payloadSize];
70 memcpy(this->payloadData, payloadData, payloadSize);
75 delete[] payloadData;
87 const int16_t seqNo, const uint8_t* payloadData,
89 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
96 size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
108 memcpy(payloadData, packet->payloadData, packet->payloadSize)
[all...] |
| Channel.h | 56 const uint8_t* payloadData,
|
| EncodeDecodeTest.h | 35 const uint8_t* payloadData,
|
| Channel.cc | 25 const uint8_t* payloadData,
69 payloadData + fragmentation->fragmentationOffset[1],
73 payloadData + fragmentation->fragmentationOffset[0],
78 memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
84 memcpy(_payloadData, payloadData, payloadDataSize);
100 //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
|
| EncodeDecodeTest.cc | 39 const uint32_t timeStamp, const uint8_t* payloadData,
42 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
|
| /external/webrtc/webrtc/modules/utility/source/ |
| coder.h | 44 const uint8_t* payloadData,
|
| coder.cc | 104 const uint8_t* payloadData,
108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 155 const uint8_t* payloadData,
251 if (payloadSize == 0 || payloadData == NULL) {
312 payloadData + fragmentation->fragmentationOffset[1],
318 payloadData + fragmentation->fragmentationOffset[0],
327 payloadData + fragmentation->fragmentationOffset[0],
337 payloadData + fragmentation->fragmentationOffset[0],
342 memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
|
| rtp_sender_audio.h | 39 const uint8_t* payloadData,
|
| rtp_sender_video.h | 52 const uint8_t* payloadData,
|
| rtp_sender_video.cc | 230 const uint8_t* payloadData,
262 const uint8_t* data = payloadData;
|
| rtcp_receiver_unittest.cc | 60 int OnReceivedPayloadData(const uint8_t* payloadData,
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api_audio.cc | 29 const uint8_t* payloadData,
36 memcpy(str, payloadData, payloadSize);
49 if (payloadData[0] == 0xff) {
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp_defines.h | 193 virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
350 int32_t OnReceivedPayloadData(const uint8_t* payloadData,
|
| rtp_rtcp.h | 292 * payloadData - payload buffer of frame to send
304 const uint8_t* payloadData,
|
| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 30 int32_t(const uint8_t* payloadData,
128 const uint8_t* payloadData,
|
| /external/webrtc/webrtc/voice_engine/ |
| channel.h | 364 const uint8_t* payloadData,
374 int32_t OnReceivedPayloadData(const uint8_t* payloadData,
|
| channel.cc | 251 const uint8_t* payloadData,
279 payloadData,
456 Channel::OnReceivedPayloadData(const uint8_t* payloadData,
480 if (audio_coding_->IncomingPacket(payloadData,
[all...] |
| /device/linaro/bootloader/edk2/EdkCompatibilityPkg/Foundation/Include/IndustryStandard/ |
| Tpm12.h | [all...] |
| /device/linaro/bootloader/edk2/MdePkg/Include/IndustryStandard/ |
| Tpm12.h | [all...] |
| /prebuilts/tools/common/m2/repository/com/firebase/firebase-client-jvm/2.5.2/ |
| firebase-client-jvm-2.5.2.jar | |
