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Searched refs:rtpStream (Results 1 – 4 of 4) sorted by relevance

/external/webrtc/modules/audio_coding/test/
DEncodeDecodeTest.cc33 TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency) in TestPacketization() argument
34 : _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {} in TestPacketization()
53 RTPStream* rtpStream, in Setup() argument
71 _packetization = new TestPacketization(rtpStream, format.clockrate_hz); in Setup()
105 RTPStream* rtpStream, in Setup() argument
142 _rtpStream = rtpStream; in Setup()
DPacketLossTest.cc30 RTPStream* rtpStream, in Setup() argument
41 Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num); in Setup()
91 RTPStream* rtpStream, in Setup() argument
96 Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type, in Setup()
DEncodeDecodeTest.h29 TestPacketization(RTPStream* rtpStream, uint16_t frequency);
53 RTPStream* rtpStream,
76 RTPStream* rtpStream,
DPacketLossTest.h24 RTPStream* rtpStream,
45 RTPStream* rtpStream,