| /external/webrtc/webrtc/modules/ |
| D | modules_unittests.isolate | 13 '<(DEPTH)/data/audio_processing/output_data_fixed.pb',
20 '<(DEPTH)/data/audio_processing/output_data_float.pb',
21 '<(DEPTH)/data/audio_processing/output_data_mac.pb',
38 '<(DEPTH)/resources/audio_processing/agc/agc_audio.pcm',
39 '<(DEPTH)/resources/audio_processing/agc/agc_no_circular_buffer.dat',
40 '<(DEPTH)/resources/audio_processing/agc/agc_pitch_gain.dat',
41 '<(DEPTH)/resources/audio_processing/agc/agc_pitch_lag.dat',
42 '<(DEPTH)/resources/audio_processing/agc/agc_spectral_peak.dat',
43 '<(DEPTH)/resources/audio_processing/agc/agc_vad.dat',
44 '<(DEPTH)/resources/audio_processing/agc/agc_voicing_prob.dat',
[all …]
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| D | modules.gyp | 15 'audio_processing/audio_processing.gypi',
34 'audio_processing/audio_processing_tests.gypi',
121 'audio_processing',
222 'audio_processing/aec/echo_cancellation_unittest.cc',
223 'audio_processing/aec/system_delay_unittest.cc',
224 'audio_processing/agc/agc_manager_direct_unittest.cc',
226 # 'audio_processing/agc/agc_unittest.cc',
227 'audio_processing/agc/histogram_unittest.cc',
228 'audio_processing/agc/mock_agc.h',
229 'audio_processing/beamformer/array_util_unittest.cc',
[all …]
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| D | Android.bp | 4 "audio_processing",
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| /external/webrtc/webrtc/voice_engine/ |
| D | voe_audio_processing_impl.cc | 87 nsLevel = _shared->audio_processing()->noise_suppression()->level(); in SetNsStatus()
106 if (_shared->audio_processing()->noise_suppression()->set_level(nsLevel) != in SetNsStatus()
112 if (_shared->audio_processing()->noise_suppression()->Enable(enable) != 0) { in SetNsStatus()
133 enabled = _shared->audio_processing()->noise_suppression()->is_enabled(); in GetNsStatus()
135 _shared->audio_processing()->noise_suppression()->level(); in GetNsStatus()
182 agcMode = _shared->audio_processing()->gain_control()->mode(); in SetAgcStatus()
195 if (_shared->audio_processing()->gain_control()->set_mode(agcMode) != 0) { in SetAgcStatus()
200 if (_shared->audio_processing()->gain_control()->Enable(enable) != 0) { in SetAgcStatus()
232 enabled = _shared->audio_processing()->gain_control()->is_enabled(); in GetAgcStatus()
234 _shared->audio_processing()->gain_control()->mode(); in GetAgcStatus()
[all …]
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| D | voe_base_unittest.cc | 27 EXPECT_EQ(audioproc, base_->audio_processing()); in TEST_F()
32 EXPECT_EQ(nullptr, base_->audio_processing()); in TEST_F()
34 EXPECT_NE(nullptr, base_->audio_processing()); in TEST_F()
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| D | shared_data.h | 42 AudioProcessing* audio_processing() { return audioproc_.get(); } in audio_processing() function
43 void set_audio_processing(AudioProcessing* audio_processing);
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| D | voe_base_impl.h | 32 AudioProcessing* audio_processing() override { in audio_processing() function
33 return shared_->audio_processing(); in audio_processing()
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| D | voice_engine.gyp | 23 '<(webrtc_root)/modules/modules.gyp:audio_processing',
122 '<(webrtc_root)/modules/modules.gyp:audio_processing',
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| D | BUILD.gn | 106 "../modules/audio_processing",
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| D | voe_base_impl.cc | 693 if (shared_->audio_processing()) { in TerminateInternal()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | audio_processing_tests.gypi | 30 '<(webrtc_root)/modules/modules.gyp:audio_processing',
42 '<(webrtc_root)/modules/modules.gyp:audio_processing',
56 '<(webrtc_root)/modules/modules.gyp:audio_processing',
69 '<(webrtc_root)/modules/modules.gyp:audio_processing',
88 'proto_out_protected': 'webrtc/audio_processing',
108 'audio_processing',
123 'audio_processing',
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| D | audio_processing.gypi | 11 'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
15 'target_name': 'audio_processing',
85 'include/audio_processing.h',
232 'proto_out_protected': 'webrtc/audio_processing',
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| D | BUILD.gn | 23 source_set("audio_processing") {
75 "include/audio_processing.h",
233 proto_out_dir = "webrtc/audio_processing"
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| /external/webrtc/ |
| D | WATCHLISTS | 68 'audio_processing': {
69 'filepath': 'webrtc/modules/audio_processing/.*',
152 'audio_processing': ['aluebs@webrtc.org',
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| /external/webrtc/webrtc/modules/audio_conference_mixer/ |
| D | audio_conference_mixer.gypi | 15 'audio_processing',
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| D | BUILD.gn | 47 "../audio_processing",
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| /external/webrtc/webrtc/voice_engine/include/ |
| D | voe_base.h | 133 virtual AudioProcessing* audio_processing() = 0;
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| /external/webrtc/webrtc/modules/audio_coding/ |
| D | audio_coding_tests.gypi | 20 'audio_processing',
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| /external/webrtc/webrtc/ |
| D | call.h | 89 AudioProcessing* audio_processing = nullptr; member
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| D | webrtc_tests.gypi | 225 'modules/audio_processing/audio_processing_performance_unittest.cc',
232 '<(webrtc_root)/modules/modules.gyp:audio_processing',
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| D | BUILD.gn | 190 "modules/audio_processing",
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| D | LICENSE_THIRD_PARTY | 17 modules/audio_processing/aec/aec_rdft.c
234 modules/audio_processing/aec/aec_rdft.c
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| /external/webrtc/webrtc/tools/ |
| D | tools.gyp | 133 '<(webrtc_root)/modules/modules.gyp:audio_processing',
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| /external/webrtc/webrtc/test/ |
| D | mock_voice_engine.h | 105 MOCK_METHOD0(audio_processing, AudioProcessing*());
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| /external/webrtc/talk/media/webrtc/ |
| D | webrtcvoiceengine.cc | 837 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); in ApplyOptions()
875 webrtc::AudioProcessing* ap = voe()->base()->audio_processing(); in SetDefaultDevices()
2288 webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing(); in MuteStream()
|