/external/webrtc/webrtc/modules/audio_processing/test/ |
D | debug_dump_test.cc | 244 void OnInitEvent(const audioproc::Init& msg); 245 void OnStreamEvent(const audioproc::Stream& msg); 246 void OnReverseStreamEvent(const audioproc::ReverseStream& msg); 247 void OnConfigEvent(const audioproc::Config& msg); 249 void MaybeRecreateApm(const audioproc::Config& msg); 250 void ConfigureApm(const audioproc::Config& msg); 274 audioproc::Event event_msg; in VerifyDebugDump() 278 case audioproc::Event::INIT: in VerifyDebugDump() 281 case audioproc::Event::STREAM: in VerifyDebugDump() 284 case audioproc::Event::REVERSE_STREAM: in VerifyDebugDump() [all …]
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D | audio_file_processor.h | 121 void HandleMessage(const webrtc::audioproc::Init& msg); 122 void HandleMessage(const webrtc::audioproc::Stream& msg); 123 void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
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D | audio_file_processor.cc | 22 using webrtc::audioproc::Event; 23 using webrtc::audioproc::Init; 24 using webrtc::audioproc::ReverseStream; 25 using webrtc::audioproc::Stream;
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D | unpack.cc | 51 using audioproc::Event; 52 using audioproc::ReverseStream; 53 using audioproc::Stream; 54 using audioproc::Init; 234 const audioproc::Config msg = event_msg.config(); in do_main()
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D | audio_processing_unittest.cc | 208 const audioproc::Test::Statistic& reference) { in TestStats() 216 audioproc::Test::Statistic* msg) { in WriteStatsMessage() 1716 audioproc::Event event_msg; in ProcessDebugDump() 1720 if (event_msg.type() == audioproc::Event::INIT) { in ProcessDebugDump() 1721 const audioproc::Init msg = event_msg.init(); in ProcessDebugDump() 1745 } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) { in ProcessDebugDump() 1746 const audioproc::ReverseStream msg = event_msg.reverse_stream(); in ProcessDebugDump() 1765 } else if (event_msg.type() == audioproc::Event::STREAM) { in ProcessDebugDump() 1766 const audioproc::Stream msg = event_msg.stream(); in ProcessDebugDump() 1923 audioproc::OutputData ref_data; in TEST_F() [all …]
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D | unittest.proto | 3 package webrtc.audioproc;
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D | process_test.cc | 41 using webrtc::audioproc::Event; 42 using webrtc::audioproc::Init; 43 using webrtc::audioproc::ReverseStream; 44 using webrtc::audioproc::Stream;
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/external/webrtc/webrtc/voice_engine/ |
D | voe_base_unittest.cc | 25 AudioProcessing* audioproc = AudioProcessing::Create(); in TEST_F() local 26 EXPECT_EQ(0, base_->Init(&adm_, audioproc)); in TEST_F() 27 EXPECT_EQ(audioproc, base_->audio_processing()); in TEST_F() 51 AudioProcessing* audioproc = AudioProcessing::Create(); in TEST_F() local 52 EXPECT_EQ(0, base_->Init(&adm_, audioproc)); in TEST_F()
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D | shared_data.cc | 69 void SharedData::set_audio_processing(AudioProcessing* audioproc) { in set_audio_processing() argument 70 audioproc_.reset(audioproc); in set_audio_processing() 71 _transmitMixerPtr->SetAudioProcessingModule(audioproc); in set_audio_processing() 72 _outputMixerPtr->SetAudioProcessingModule(audioproc); in set_audio_processing()
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D | voe_base_impl.cc | 218 AudioProcessing* audioproc) { in Init() argument 328 if (!audioproc) { in Init() 329 audioproc = AudioProcessing::Create(); in Init() 330 if (!audioproc) { in Init() 336 shared_->set_audio_processing(audioproc); in Init() 341 if (audioproc->high_pass_filter()->Enable(true) != 0) { in Init() 345 if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) { in Init() 349 if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) { in Init() 354 GainControl* agc = audioproc->gain_control(); in Init()
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D | voe_base_impl.h | 31 AudioProcessing* audioproc = nullptr) override;
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/external/webrtc/webrtc/tools/agc/ |
D | agc_harness.cc | 109 AudioProcessing* audioproc = AudioProcessing::Create(config); in SetUp() local 110 RTC_CHECK_EQ(0, base_->Init(nullptr, audioproc)); in SetUp() 113 audioproc->gain_control()->Enable(true); in SetUp() 114 audioproc->high_pass_filter()->Enable(FLAGS_highpass); in SetUp() 115 audioproc->noise_suppression()->Enable(FLAGS_ns); in SetUp() 116 audioproc->echo_cancellation()->Enable(FLAGS_aec); in SetUp()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
D | after_initialization_fixture.cc | 28 webrtc::AudioProcessing* audioproc = webrtc::AudioProcessing::Create(config); in AfterInitializationFixture() local 30 EXPECT_EQ(0, voe_base_->Init(NULL, audioproc)); in AfterInitializationFixture()
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/external/webrtc/webrtc/modules/audio_processing/ |
D | audio_processing_impl.cc | 627 debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM); in ProcessStream() 628 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); in ProcessStream() 643 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); in ProcessStream() 717 debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM); in ProcessStream() 718 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); in ProcessStream() 732 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); in ProcessStream() 747 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); in ProcessStreamLocked() 895 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); in AnalyzeReverseStreamLocked() 896 audioproc::ReverseStream* msg = in AnalyzeReverseStreamLocked() 965 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); in AnalyzeReverseStream() [all …]
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D | audio_processing_impl.h | 137 ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {} in ApmDebugDumpThreadState() 138 rtc::scoped_ptr<audioproc::Event> event_msg; // Protobuf message.
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D | debug.proto | 3 package webrtc.audioproc;
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D | audio_processing_tests.gypi | 105 'target_name': 'audioproc',
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/external/webrtc/webrtc/modules/audio_processing/vad/ |
D | vad_audio_proc_unittest.cc | 30 VadAudioProc audioproc; in TEST() local 48 audioproc.ExtractFeatures(data, kDataLength, &features); in TEST()
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/external/webrtc/webrtc/voice_engine/include/ |
D | voe_base.h | 130 AudioProcessing* audioproc = NULL) = 0;
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/external/webrtc/webrtc/test/ |
D | mock_voice_engine.h | 104 AudioProcessing* audioproc));
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvoiceengine.cc | 837 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); in ApplyOptions() local 838 if (audioproc) { in ApplyOptions() 839 audioproc->SetExtraOptions(config); in ApplyOptions()
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D | fakewebrtcvoiceengine.h | 325 webrtc::AudioProcessing* audioproc)) {
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