| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_quality_test.cc | 44 rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0,
59 rtp_generator_->set_drift_factor(drift_factor_);
68 rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
|
| neteq_quality_test.h | 55 // Transmit() uses |rtp_generator_| to generate a packet and passes it to
88 scoped_ptr<RtpGenerator> rtp_generator_; member in class:webrtc::test::NetEqQualityTest
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_external_decoder_unittest.cc | 49 rtp_generator_(samples_per_ms_),
99 int next_send_time = rtp_generator_.GetRtpHeader(kPayloadType,
194 test::RtpGenerator rtp_generator_; member in class:webrtc::NetEqExternalDecoderTest
|
| neteq_stereo_unittest.cc | 58 rtp_generator_(samples_per_ms_),
160 rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
250 test::RtpGenerator rtp_generator_; member in class:webrtc::NetEqStereoTest
|
