/external/webrtc/webrtc/modules/audio_coding/neteq/ |
neteq_network_stats_unittest.cc | 116 rtp_generator_(new test::RtpGenerator(samples_per_ms_)), 185 rtp_generator_->GetRtpHeader(kPayloadType, frame_size_samples_, 190 next_send_time = rtp_generator_->GetRtpHeader(kPayloadType, 266 rtc::scoped_ptr<test::RtpGenerator> rtp_generator_; member in class:webrtc::test::NetEqNetworkStatsTest
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neteq_external_decoder_unittest.cc | 36 rtp_generator_(new test::RtpGenerator(samples_per_ms_)), 76 int next_send_time = rtp_generator_->GetRtpHeader( 143 rtp_generator_.reset(rtp_generator); 151 rtc::scoped_ptr<test::RtpGenerator> rtp_generator_; member in class:webrtc::NetEqExternalDecoderUnitTest
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neteq_stereo_unittest.cc | 58 rtp_generator_(samples_per_ms_), 159 rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_, 251 test::RtpGenerator rtp_generator_; member in class:webrtc::NetEqStereoTest
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_quality_test.h | 91 // Transmit() uses |rtp_generator_| to generate a packet and passes it to 126 rtc::scoped_ptr<RtpGenerator> rtp_generator_; member in class:webrtc::test::NetEqQualityTest
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neteq_quality_test.cc | 229 rtp_generator_( 297 rtp_generator_->set_drift_factor(drift_factor_); 372 rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
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