/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | 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; member in class:webrtc::RTPPacket 71 const uint8_t* payloadData, 76 uint8_t* payloadData, 108 const uint8_t* payloadData, 113 uint8_t* payloadData,
|
H A D | 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, argument 108 memcpy(payloadData, packet->payloadData, packe 60 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, const uint8_t* payloadData, size_t payloadSize, uint32_t frequency) argument 86 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument 169 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument 189 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, size_t payloadSize, uint32_t* offset) argument [all...] |
H A D | Channel.h | 56 const uint8_t* payloadData,
|
H A D | EncodeDecodeTest.h | 35 const uint8_t* payloadData,
|
H A D | 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); 22 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
|
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 size_t payloadSize, const RTPFragmentationHeader* ) argument
|
/external/webrtc/webrtc/modules/utility/source/ |
H A D | coder.h | 44 const uint8_t* payloadData,
|
H A D | coder.cc | 104 const uint8_t* payloadData, 108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 100 SendData( FrameType , uint8_t , uint32_t , const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* ) argument
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | 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); 152 SendAudio(FrameType frameType, int8_t payloadType, uint32_t captureTimeStamp, const uint8_t* payloadData, size_t dataSize, const RTPFragmentationHeader* fragmentation) argument
|
H A D | rtp_sender_audio.h | 39 const uint8_t* payloadData,
|
H A D | rtp_sender_video.h | 52 const uint8_t* payloadData,
|
H A D | rtp_sender_video.cc | 230 const uint8_t* payloadData, 262 const uint8_t* data = payloadData; 225 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 size_t payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpHdr) argument
|
H A D | rtcp_receiver_unittest.cc | 60 int OnReceivedPayloadData(const uint8_t* payloadData,
|
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | 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/ |
H A D | rtp_rtcp_defines.h | 193 virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData, 350 int32_t OnReceivedPayloadData(const uint8_t* payloadData,
|
H A D | rtp_rtcp.h | 292 * payloadData - payload buffer of frame to send 304 const uint8_t* payloadData,
|
/external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
H A D | mock_rtp_rtcp.h | 30 int32_t(const uint8_t* payloadData, 128 const uint8_t* payloadData,
|
/external/webrtc/webrtc/voice_engine/ |
H A D | channel.h | 364 const uint8_t* payloadData, 374 int32_t OnReceivedPayloadData(const uint8_t* payloadData,
|
H A D | channel.cc | 251 const uint8_t* payloadData, 279 payloadData, 456 Channel::OnReceivedPayloadData(const uint8_t* payloadData, argument 480 if (audio_coding_->IncomingPacket(payloadData, 248 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
|