| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | 395 rtc::scoped_ptr<AudioProcessing> apm_; member in class:webrtc::__anon27882::ApmTest
430 apm_.reset(AudioProcessing::Create(config));
434 ASSERT_TRUE(apm_.get() != NULL);
495 Init(apm_.get());
528 EnableAllAPComponents(apm_.get());
575 EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
576 apm_->echo_cancellation()->set_stream_drift_samples(0);
577 EXPECT_EQ(apm_->kNoError,
578 apm_->gain_control()->set_stream_analog_level(127))
[all...] |
| debug_dump_test.cc | 81 AudioProcessing* apm() const { return apm_.get(); }
108 rtc::scoped_ptr<AudioProcessing> apm_; member in class:webrtc::test::__anon27885::DebugDumpGenerator
134 apm_(AudioProcessing::Create(config)),
184 apm_->StartDebugRecording(dump_file_name_.c_str());
193 RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
194 apm_->set_stream_key_pressed(i % 10 == 9);
196 apm_->ProcessStream(input_->channels(), input_config_,
200 apm_->ProcessReverseStream(reverse_->channels(),
208 apm_->StopDebugRecording();
257 rtc::scoped_ptr<AudioProcessing> apm_; member in class:webrtc::test::DebugDumpTest
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_processing_impl_locking_unittest.cc | 334 AudioProcessing* const apm_ = nullptr; member in class:webrtc::__anon27864::CaptureProcessor
349 AudioProcessing* apm_ = nullptr; member in class:webrtc::__anon27864::StatsProcessor
380 AudioProcessing* const apm_ = nullptr; member in class:webrtc::__anon27864::RenderProcessor
446 rtc::scoped_ptr<AudioProcessing> apm_; member in class:webrtc::__anon27864::AudioProcessingImplLockTest
497 apm_(AudioProcessingImpl::Create()),
505 apm_.get()),
513 apm_.get()),
514 stats_thread_state_(&rand_gen_, &test_config_, apm_.get()) {}
534 ASSERT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true))
[all...] |
| audio_processing_performance_unittest.cc | 246 apm_(apm),
330 apm_->set_stream_delay_ms(30);
334 const int result = apm_->ProcessStream(
362 const int result = apm_->ProcessReverseStream(
442 AudioProcessing* apm_ = nullptr; member in class:webrtc::__anon27865::TimedThreadApiProcessor
576 apm_.reset(AudioProcessingImpl::Create());
577 ASSERT_TRUE(!!apm_);
578 set_default_mobile_apm_runtime_settings(apm_.get());
584 apm_.reset(AudioProcessingImpl::Create(config));
585 ASSERT_TRUE(!!apm_);
677 rtc::scoped_ptr<AudioProcessing> apm_; member in class:webrtc::__anon27865::CallSimulator
[all...] |
| echo_control_mobile_impl.cc | 74 apm_(apm),
101 assert(audio->num_channels() == apm_->num_reverse_channels());
107 for (size_t i = 0; i < apm_->num_output_channels(); i++) {
153 (apm_->num_output_channels() * apm_->num_reverse_channels());
154 for (size_t i = 0; i < apm_->num_output_channels(); i++) {
155 for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
174 if (!apm_->was_stream_delay_set()) {
179 assert(audio->num_channels() == apm_->num_output_channels());
194 for (size_t j = 0; j < apm_->num_reverse_channels(); j++)
[all...] |
| echo_cancellation_impl.cc | 68 apm_(apm),
95 assert(audio->num_channels() == apm_->num_reverse_channels());
102 for (size_t i = 0; i < apm_->num_output_channels(); i++) {
148 (apm_->num_output_channels() * apm_->num_reverse_channels());
149 for (size_t i = 0; i < apm_->num_output_channels(); i++) {
150 for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
168 if (!apm_->was_stream_delay_set()) {
177 assert(audio->num_channels() == apm_->num_proc_channels());
185 for (size_t j = 0; j < apm_->num_reverse_channels(); j++)
[all...] |
| echo_control_mobile_impl.h | 67 const AudioProcessing* apm_; member in class:webrtc::EchoControlMobileImpl
|
| echo_cancellation_impl.h | 81 const AudioProcessing* apm_; member in class:webrtc::EchoCancellationImpl
|
| gain_control_impl.cc | 49 apm_(apm),
209 apm_->echo_cancellation()->stream_has_echo(),
417 apm_->proc_sample_rate_hz());
438 return apm_->num_proc_channels();
|
| gain_control_impl.h | 77 const AudioProcessing* apm_; member in class:webrtc::GainControlImpl
|
