/aosp_15_r20/external/webrtc/modules/audio_processing/capture_levels_adjuster/ |
H A D | audio_samples_scaler_unittest.cc | 27 void PopulateBuffer(AudioBuffer& audio_buffer) { in PopulateBuffer() 54 AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), in TEST_P() local 75 AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), in TEST_P() local 154 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST() local 183 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST() local
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H A D | capture_levels_adjuster_unittest.cc | 28 void PopulateBuffer(AudioBuffer& audio_buffer) { in PopulateBuffer() 86 AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), in TEST_P() local 136 AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(), in TEST_P() local
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H A D | capture_levels_adjuster.cc | 61 void CaptureLevelsAdjuster::ApplyPreLevelAdjustment(AudioBuffer& audio_buffer) { in ApplyPreLevelAdjustment() 66 AudioBuffer& audio_buffer) { in ApplyPostLevelAdjustment()
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H A D | audio_samples_scaler.cc | 24 void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) { in Process()
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/aosp_15_r20/external/webrtc/modules/audio_processing/agc/ |
H A D | agc_manager_direct_unittest.cc | 111 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in CallPreProcessAudioBuffer() local 174 AudioBuffer& audio_buffer) { in WriteAudioBufferSamples() 197 const AudioBuffer& audio_buffer, in CallPreProcessAndProcess() 430 AudioBuffer audio_buffer; member in webrtc::AgcManagerDirectTestHelper 480 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST_P() local 1523 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1580 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1646 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1713 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1816 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST_P() local [all …]
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H A D | agc_manager_direct.cc | 541 void AgcManagerDirect::AnalyzePreProcess(const AudioBuffer& audio_buffer) { in AnalyzePreProcess() 621 void AgcManagerDirect::Process(const AudioBuffer& audio_buffer) { in Process() 626 void AgcManagerDirect::Process(const AudioBuffer& audio_buffer, in Process()
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/aosp_15_r20/external/webrtc/modules/audio_processing/agc2/ |
H A D | input_volume_controller_unittest.cc | 95 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in CallPreProcessAudioBuffer() local 156 AudioBuffer& audio_buffer) { in WriteAudioBufferSamples() 179 const AudioBuffer& audio_buffer, in CallPreProcessAndProcess() 375 AudioBuffer audio_buffer; member in webrtc::InputVolumeControllerTestHelper 860 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 916 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 975 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1040 AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz, in TEST() local 1115 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST_P() local 1138 AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz, in TEST_P() local
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H A D | input_volume_controller.cc | 426 void InputVolumeController::AnalyzePreProcess(const AudioBuffer& audio_buffer) { in AnalyzePreProcess()
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/aosp_15_r20/external/webrtc/modules/audio_device/ |
H A D | audio_device_buffer.cc | 232 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer, in SetRecordedBuffer() 237 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer, in SetRecordedBuffer() 347 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) { in GetPlayoutData()
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H A D | fine_audio_buffer.cc | 65 void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer, in GetPlayoutData() 109 rtc::ArrayView<const int16_t> audio_buffer, in DeliverRecordedData()
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H A D | audio_device_unittest.cc | 367 int32_t RealRecordedDataIsAvailable(const void* audio_buffer, in RealRecordedDataIsAvailable() 413 void* audio_buffer, in RealNeedMorePlayData()
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/aosp_15_r20/external/webrtc/modules/audio_processing/ |
H A D | gain_controller2_unittest.cc | 426 AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, in TEST() local 485 AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, in TEST() local 554 AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, in TEST() local 626 AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, in TEST() local
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H A D | gain_controller2.cc | 138 const AudioBuffer& audio_buffer) { in Analyze()
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H A D | high_pass_filter_unittest.cc | 29 AudioBuffer audio_buffer( in ProcessOneFrameAsAudioBuffer() local
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/aosp_15_r20/external/webrtc/modules/audio_processing/test/ |
H A D | audio_buffer_tools.cc | 54 void FillBuffer(float value, AudioBuffer& audio_buffer) { in FillBuffer() 60 void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer) { in FillBufferChannel()
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/aosp_15_r20/external/webrtc/pc/test/ |
H A D | fake_audio_capture_module_unittest.cc | 106 size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) { in GenerateZeroBuffer() 110 size_t CopyFromRecBuffer(void* audio_buffer, size_t audio_buffer_size) { in CopyFromRecBuffer()
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/aosp_15_r20/external/webrtc/modules/audio_processing/transient/ |
H A D | click_annotate.cc | 69 std::unique_ptr<float[]> audio_buffer(new float[audio_buffer_length]); in main() local
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H A D | transient_suppression_test.cc | 74 int16_t* audio_buffer, in ReadBuffers()
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/aosp_15_r20/external/webrtc/modules/audio_processing/aec3/ |
H A D | block_delay_buffer_unittest.cc | 77 AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate, in TEST_P() local
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/aosp_15_r20/external/webrtc/modules/audio_device/win/ |
H A D | core_audio_output_win.cc | 86 void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer()
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H A D | core_audio_input_win.cc | 89 void CoreAudioInput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer()
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/aosp_15_r20/external/webrtc/modules/audio_device/android/ |
H A D | opensles_recorder.cc | 167 void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer()
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/aosp_15_r20/external/webrtc/sdk/android/src/jni/audio_device/ |
H A D | opensles_recorder.cc | 178 void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { in AttachAudioBuffer()
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/aosp_15_r20/external/webrtc/audio/ |
H A D | audio_state_unittest.cc | 349 int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels]; in TEST_P() local
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