Searched defs:rtpStream (Results 1 - 2 of 2) sorted by relevance
/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | PacketLossTest.cc | 28 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
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H A D | EncodeDecodeTest.cc | 28 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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