Searched refs:_rtpStream (Results 1 – 3 of 3) sorted by relevance
| /external/webrtc/modules/audio_coding/test/ |
| D | EncodeDecodeTest.cc | 34 : _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {} in TestPacketization()
44 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, in SendData()
142 _rtpStream = rtpStream; in Setup()
160 if (!_rtpStream->EndOfFile()) { in IncomingPacket()
163 _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload, in IncomingPacket()
166 if (_rtpStream->EndOfFile()) { in IncomingPacket()
177 _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload, in IncomingPacket()
179 if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) { in IncomingPacket()
223 if (_rtpStream->EndOfFile()) { in Run()
|
| D | PacketLossTest.cc | 45 if (!_rtpStream->EndOfFile()) { in IncomingPacket()
47 _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload, in IncomingPacket()
50 if (_rtpStream->EndOfFile()) { in IncomingPacket()
63 _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload, in IncomingPacket()
65 if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) { in IncomingPacket()
|
| D | EncodeDecodeTest.h | 44 RTPStream* _rtpStream; variable
95 RTPStream* _rtpStream; variable
|