| /external/webrtc/webrtc/modules/audio_processing/ |
| D | audio_buffer.cc | 46 AudioBuffer::AudioBuffer(size_t input_num_frames, in AudioBuffer() function in webrtc::AudioBuffer
103 AudioBuffer::~AudioBuffer() {} in ~AudioBuffer()
105 void AudioBuffer::CopyFrom(const float* const* data, in CopyFrom()
150 void AudioBuffer::CopyTo(const StreamConfig& stream_config, in CopyTo()
183 void AudioBuffer::InitForNewData() { in InitForNewData()
191 const int16_t* const* AudioBuffer::channels_const() const { in channels_const()
195 int16_t* const* AudioBuffer::channels() { in channels()
200 const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const { in split_bands_const()
206 int16_t* const* AudioBuffer::split_bands(size_t channel) { in split_bands()
213 const int16_t* const* AudioBuffer::split_channels_const(Band band) const { in split_channels_const()
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| D | gain_control_impl.h | 25 class AudioBuffer; variable
35 int ProcessRenderAudio(AudioBuffer* audio);
36 int AnalyzeCaptureAudio(AudioBuffer* audio);
37 int ProcessCaptureAudio(AudioBuffer* audio);
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| D | noise_suppression_impl.h | 21 class AudioBuffer; variable
30 void AnalyzeCaptureAudio(AudioBuffer* audio);
31 void ProcessCaptureAudio(AudioBuffer* audio);
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| D | echo_control_mobile_impl.h | 22 class AudioBuffer; variable
33 int ProcessRenderAudio(const AudioBuffer* audio);
34 int ProcessCaptureAudio(AudioBuffer* audio);
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| D | echo_cancellation_impl.h | 22 class AudioBuffer; variable
32 int ProcessRenderAudio(const AudioBuffer* audio);
33 int ProcessCaptureAudio(AudioBuffer* audio);
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| D | audio_buffer.h | 33 class AudioBuffer {
36 AudioBuffer(size_t input_num_frames,
41 virtual ~AudioBuffer();
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| D | level_estimator_impl.h | 21 class AudioBuffer; variable
31 void ProcessStream(AudioBuffer* audio);
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| D | high_pass_filter_impl.h | 21 class AudioBuffer; variable
30 void ProcessCaptureAudio(AudioBuffer* audio);
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| D | voice_detection_impl.h | 21 class AudioBuffer; variable
30 void ProcessCaptureAudio(AudioBuffer* audio);
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| D | noise_suppression_impl.cc | 70 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { in AnalyzeCaptureAudio()
87 void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | gain_control_impl.cc | 69 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { in ProcessRenderAudio()
127 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { in AnalyzeCaptureAudio()
179 int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | level_estimator_impl.cc | 31 void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { in ProcessStream()
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| D | audio_processing_impl.h | 310 rtc::scoped_ptr<AudioBuffer> capture_audio;
335 rtc::scoped_ptr<AudioBuffer> render_audio;
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| D | echo_control_mobile_impl.cc | 93 int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) { in ProcessRenderAudio()
167 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | high_pass_filter_impl.cc | 104 void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | voice_detection_impl.cc | 55 void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | echo_cancellation_impl.cc | 88 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) { in ProcessRenderAudio()
162 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) { in ProcessCaptureAudio()
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| D | audio_processing_impl.cc | 356 render_.render_audio.reset(new AudioBuffer( in InitializeLocked()
376 new AudioBuffer(formats_.api_format.input_stream().num_frames(), in InitializeLocked()
758 AudioBuffer* ca = capture_.capture_audio.get(); // For brevity. in ProcessStreamLocked()
980 AudioBuffer* ra = render_.render_audio.get(); // For brevity. in ProcessReverseStreamLocked()
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| /external/drrickorang/LoopbackApp/app/src/main/cpp/lb2/ |
| D | audio_buffer.h | 104 class AudioBuffer : public AudioBufferBase<T> {
107 constexpr AudioBuffer(): AudioBufferBase<T>(nullptr, 0, 1), mBuffer() {} in AudioBuffer() function
108 AudioBuffer(size_t frameCount, int channelCount) in AudioBuffer() function
113 AudioBuffer(const AudioBuffer<T>&) = delete;
114 AudioBuffer(AudioBuffer<T>&&) = default;
115 AudioBuffer<T>& operator=(const AudioBuffer<T>&) = delete;
116 AudioBuffer<T>& operator=(AudioBuffer<T>&&) = default;
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| D | test_context.h | 52 AudioBuffer<sample_t> createAudioBuffer() const { in createAudioBuffer()
53 return AudioBuffer<sample_t>(getFrameCount(), getChannelCount()); in createAudioBuffer()
118 AudioBuffer<sample_t> mSineBuffer;
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| D | loopback_test.cpp | 50 AudioBuffer<sample_t> readBuffer(mTestCtx->createAudioBuffer()); in collectRecording()
149 return AudioBuffer<sample_t>(); in writeCallback()
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| D | sound_system_aaudio.cpp | 345 std::unique_ptr<AudioBuffer<sample_t>> mConversionBuffer;
381 mConversionBuffer.reset(new AudioBuffer<sample_t>( in init()
390 AudioBuffer<sample_t> drainBuffer(mStream->getFramesPerBurst(), mStream->getChannelCount()); in drain()
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| D | loopback_test.h | 47 AudioBuffer<sample_t> mReadBuffer;
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| D | sound_system_echo.cpp | 77 AudioBuffer<sample_t> drainBuffer( in drainInput()
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| /external/webrtc/webrtc/modules/audio_device/ios/ |
| D | audio_device_ios.mm | 730 // AudioBuffer structure, which holds a pointer to the actual data buffer
736 AudioBuffer* audio_buffer = &audio_record_buffer_list_.mBuffers[0];
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