Searched defs:rtpStream (Results 1 - 2 of 2) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/test/
H A DPacketLossTest.cc28 RTPStream *rtpStream,
38 Receiver::Setup(acm, rtpStream, ss.str(), channels);
89 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, argument
92 Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
27 Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length) argument
H A DEncodeDecodeTest.cc28 TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency) argument
29 : _rtpStream(rtpStream),
54 void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream, argument
91 _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
125 void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream, argument
149 _rtpStream = rtpStream;

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