| /external/webrtc/modules/audio_processing/agc2/ |
| D | input_volume_controller.cc | 258 int applied_input_volume = recommended_input_volume_; in SetInputVolume() local
259 if (applied_input_volume == 0) { in SetInputVolume()
264 if (applied_input_volume < 0 || applied_input_volume > kMaxInputVolume) { in SetInputVolume()
266 << applied_input_volume; in SetInputVolume()
273 if (applied_input_volume > input_volume_ + kVolumeQuantizationSlack || in SetInputVolume()
274 applied_input_volume < input_volume_ - kVolumeQuantizationSlack) { in SetInputVolume()
278 << input_volume_ << " to " << applied_input_volume; in SetInputVolume()
279 input_volume_ = applied_input_volume; in SetInputVolume()
298 RTC_DLOG(LS_INFO) << "[AGC2] Applied input volume: " << applied_input_volume in SetInputVolume()
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| D | input_volume_controller_unittest.cc | 256 int applied_input_volume, in UpdateRecommendedInputVolume() argument
259 controller.set_stream_analog_level(applied_input_volume); in UpdateRecommendedInputVolume()
260 EXPECT_EQ(controller.recommended_analog_level(), applied_input_volume); in UpdateRecommendedInputVolume()
311 int CallAgcSequence(int applied_input_volume, in CallAgcSequence() argument
314 controller.set_stream_analog_level(applied_input_volume); in CallAgcSequence()
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| /external/webrtc/modules/audio_processing/ |
| D | gain_controller2.cc | 137 void GainController2::Analyze(int applied_input_volume, in Analyze() argument
139 RTC_DCHECK_GE(applied_input_volume, 0); in Analyze()
140 RTC_DCHECK_LE(applied_input_volume, 255); in Analyze()
143 input_volume_controller_->set_stream_analog_level(applied_input_volume); in Analyze()
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| D | audio_processing_impl.cc | 1426 capture_.applied_input_volume.value_or(kUnspecifiedDataDumpInputVolume)); in ProcessCaptureStreamLocked()
1463 if (capture_.applied_input_volume.has_value()) { in ProcessCaptureStreamLocked()
1465 *capture_.applied_input_volume); in ProcessCaptureStreamLocked()
1501 RTC_DCHECK(capture_.applied_input_volume.has_value()); in ProcessCaptureStreamLocked()
1502 if (capture_.applied_input_volume.has_value()) { in ProcessCaptureStreamLocked()
1503 submodules_.gain_controller2->Analyze(*capture_.applied_input_volume, in ProcessCaptureStreamLocked()
1937 capture_.applied_input_volume.has_value() && in set_stream_analog_level_locked()
1938 *capture_.applied_input_volume != level; in set_stream_analog_level_locked()
1939 capture_.applied_input_volume = level; in set_stream_analog_level_locked()
1959 if (!capture_.applied_input_volume.has_value()) { in recommended_stream_analog_level()
[all …]
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| D | audio_processing_impl_unittest.cc | 994 const int applied_input_volume = GetVolume(); in TEST_P() local
995 const int expected_volume = std::max(applied_input_volume, GetMinVolume()); in TEST_P()
1004 ProcessInputVolume(*apm, /*num_frames=*/1, applied_input_volume); in TEST_P()
1006 ASSERT_NE(applied_input_volume, 0); in TEST_P()
1011 ASSERT_EQ(recommended_input_volume, applied_input_volume); in TEST_P()
1018 ASSERT_EQ(recommended_input_volume, applied_input_volume); in TEST_P()
1055 const int applied_input_volume = GetVolume(); in TEST_P() local
1064 ProcessInputVolume(*apm, /*num_frames=*/400, applied_input_volume); in TEST_P()
1066 ASSERT_NE(applied_input_volume, 0); in TEST_P()
1071 ASSERT_EQ(recommended_input_volume, applied_input_volume); in TEST_P()
[all …]
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| D | gain_controller2.h | 59 void Analyze(int applied_input_volume, const AudioBuffer& audio_buffer);
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| D | debug.proto | 38 optional int32 applied_input_volume = 5; field
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| D | audio_processing_impl.h | 485 absl::optional<int> applied_input_volume; member
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| D | audio_processing_unittest.cc | 360 EXPECT_EQ(actual.applied_input_volume(), expected.applied_input_volume()); in ExpectStreamFieldsEq()
1521 apm_->set_stream_analog_level(msg.applied_input_volume()); in ProcessDebugDump()
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| /external/webrtc/modules/audio_processing/aec_dump/ |
| D | capture_stream_info.cc | 56 if (state.applied_input_volume.has_value()) { in AddAudioProcessingState()
57 stream->set_applied_input_volume(*state.applied_input_volume); in AddAudioProcessingState()
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| /external/webrtc/modules/audio_processing/include/ |
| D | aec_dump.h | 71 absl::optional<int> applied_input_volume; member
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| /external/webrtc/modules/audio_processing/test/ |
| D | debug_dump_replayer.cc | 125 apm_->set_stream_analog_level(msg.applied_input_volume()); in OnStreamEvent()
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| D | aec_dump_based_simulator.cc | 180 ? absl::optional<int>(msg.applied_input_volume()) in PrepareProcessStreamCall()
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| /external/webrtc/rtc_tools/unpack_aecdump/ |
| D | unpack.cc | 474 int32_t level = msg.applied_input_volume(); in do_main()
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| /external/webrtc/modules/audio_processing/agc/ |
| D | agc_manager_direct_unittest.cc | 354 int CallAgcSequence(int applied_input_volume, in CallAgcSequence() argument
357 manager.set_stream_analog_level(applied_input_volume); in CallAgcSequence()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | debug.pb.h | 786 int32_t applied_input_volume() const; in Swap()
2549 inline int32_t Stream::applied_input_volume() const { in applied_input_volume() function
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