| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 514 int playout_sample_rate() const { return playout_parameters_.sample_rate(); } function in class:webrtc::AudioDeviceTest
663 playout_channels(), playout_sample_rate(),
685 playout_channels(), playout_sample_rate(),
706 playout_sample_rate(), NotNull(), _, _, _))
739 playout_sample_rate(), NotNull(), _, _, _))
763 std::string file_name = GetFileName(playout_sample_rate());
765 new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
796 EXPECT_EQ(record_sample_rate(), playout_sample_rate());
825 EXPECT_EQ(record_sample_rate(), playout_sample_rate());
[all...] |
| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_device_unittest.cc | 518 int playout_sample_rate() const { function in class:webrtc::AudioDeviceTest
864 playout_sample_rate(),
[all...] |
| /external/webrtc/talk/media/base/ |
| mediachannel.h | 126 SetFrom(&playout_sample_rate, change.playout_sample_rate);
153 playout_sample_rate == o.playout_sample_rate &&
184 ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
221 rtc::Optional<uint32_t> playout_sample_rate; member in struct:cricket::AudioOptions
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvoiceengine.cc | 851 if (options.playout_sample_rate) {
852 LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
854 *options.playout_sample_rate)) {
855 LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
[all...] |
| webrtcvoiceengine_unittest.cc | 2213 unsigned int recording_sample_rate, playout_sample_rate; local
[all...] |
