| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | 161 EXPECT_NOERR(ap->gain_control()->set_mode(GainControl::kAdaptiveDigital));
162 EXPECT_NOERR(ap->gain_control()->Enable(true));
169 EXPECT_NOERR(ap->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
170 EXPECT_NOERR(ap->gain_control()->set_analog_level_limits(0, 255));
171 EXPECT_NOERR(ap->gain_control()->Enable(true));
508 apm_->gain_control()->set_stream_analog_level(127));
628 EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
634 apm_->gain_control()->set_stream_analog_level(127));
647 EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
664 EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true))
[all...] |
| process_test.cc | 296 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
299 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
301 apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
304 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
306 apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
309 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
311 apm->gain_control()->set_mode(GainControl::kFixedDigital));
318 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
320 apm->gain_control()->set_target_level_dbfs(level));
327 ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true))
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| voe_audio_processing_impl.cc | 188 agcMode = _shared->audio_processing()->gain_control()->mode();
201 if (_shared->audio_processing()->gain_control()->set_mode(agcMode) != 0) {
206 if (_shared->audio_processing()->gain_control()->Enable(enable) != 0) {
240 enabled = _shared->audio_processing()->gain_control()->is_enabled();
242 _shared->audio_processing()->gain_control()->mode();
275 if (_shared->audio_processing()->gain_control()->set_target_level_dbfs(
282 if (_shared->audio_processing()->gain_control()->set_compression_gain_db(
289 if (_shared->audio_processing()->gain_control()->enable_limiter(
314 _shared->audio_processing()->gain_control()->target_level_dbfs();
316 _shared->audio_processing()->gain_control()->compression_gain_db()
[all...] |
| channel.cc | [all...] |
| transmit_mixer.cc | [all...] |
| voe_base_impl.cc | 459 GainControl* agc = audioproc->gain_control();
[all...] |
| /external/chromium_org/content/renderer/media/ |
| media_stream_audio_processor_options.cc | 276 int err = audio_processing->gain_control()->set_mode(mode);
277 err |= audio_processing->gain_control()->Enable(true);
|
| media_stream_audio_processor_unittest.cc | 140 EXPECT_TRUE(audio_processing->gain_control()->is_enabled());
142 EXPECT_TRUE(audio_processing->gain_control()->mode() ==
146 EXPECT_TRUE(audio_processing->gain_control()->mode() ==
|
| media_stream_audio_processor.cc | 560 webrtc::GainControl* agc = ap->gain_control();
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| audio_processing_impl.h | 133 virtual GainControl* gain_control() const OVERRIDE;
|
| audio_processing_impl.cc | 750 GainControl* AudioProcessingImpl::gain_control() const { function in class:webrtc::AudioProcessingImpl
|
| /external/chromium_org/third_party/webrtc/modules/audio_conference_mixer/source/ |
| audio_conference_mixer_impl.cc | 166 if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
173 if(_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
176 if(_limiter->gain_control()->set_compression_gain_db(0)
180 if(_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
183 if(_limiter->gain_control()->Enable(true) != _limiter->kNoError)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_processing/include/ |
| audio_processing.h | 131 // apm->gain_control()->set_analog_level_limits(0, 255);
132 // apm->gain_control()->set_mode(kAdaptiveAnalog);
133 // apm->gain_control()->Enable(true);
145 // apm->gain_control()->set_stream_analog_level(analog_level);
150 // analog_level = apm->gain_control()->stream_analog_level();
345 virtual GainControl* gain_control() const = 0;
|
| mock_audio_processing.h | 250 virtual MockGainControl* gain_control() const { function in class:webrtc::MockAudioProcessing
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| fakewebrtcvoiceengine.h | 138 virtual webrtc::GainControl* gain_control() const OVERRIDE { return NULL; }
[all...] |
