110 lines
3.4 KiB
C
110 lines
3.4 KiB
C
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#include <list>
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#include "audio_processing.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class FileWrapper;
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class AudioBuffer;
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class EchoCancellationImpl;
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class EchoControlMobileImpl;
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class GainControlImpl;
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class HighPassFilterImpl;
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class LevelEstimatorImpl;
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class NoiseSuppressionImpl;
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class ProcessingComponent;
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class VoiceDetectionImpl;
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class AudioProcessingImpl : public AudioProcessing {
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public:
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enum {
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kSampleRate8kHz = 8000,
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kSampleRate16kHz = 16000,
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kSampleRate32kHz = 32000
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};
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explicit AudioProcessingImpl(int id);
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virtual ~AudioProcessingImpl();
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CriticalSectionWrapper* crit() const;
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int split_sample_rate_hz() const;
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bool was_stream_delay_set() const;
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// AudioProcessing methods.
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virtual int Initialize();
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virtual int InitializeLocked();
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virtual int set_sample_rate_hz(int rate);
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virtual int sample_rate_hz() const;
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virtual int set_num_channels(int input_channels, int output_channels);
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virtual int num_input_channels() const;
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virtual int num_output_channels() const;
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virtual int set_num_reverse_channels(int channels);
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virtual int num_reverse_channels() const;
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virtual int ProcessStream(AudioFrame* frame);
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virtual int AnalyzeReverseStream(AudioFrame* frame);
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virtual int set_stream_delay_ms(int delay);
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virtual int stream_delay_ms() const;
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virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
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virtual int StopDebugRecording();
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virtual EchoCancellation* echo_cancellation() const;
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virtual EchoControlMobile* echo_control_mobile() const;
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virtual GainControl* gain_control() const;
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virtual HighPassFilter* high_pass_filter() const;
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virtual LevelEstimator* level_estimator() const;
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virtual NoiseSuppression* noise_suppression() const;
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virtual VoiceDetection* voice_detection() const;
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// Module methods.
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virtual WebRtc_Word32 Version(WebRtc_Word8* version,
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WebRtc_UWord32& remainingBufferInBytes,
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WebRtc_UWord32& position) const;
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
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private:
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int id_;
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EchoCancellationImpl* echo_cancellation_;
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EchoControlMobileImpl* echo_control_mobile_;
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GainControlImpl* gain_control_;
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HighPassFilterImpl* high_pass_filter_;
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LevelEstimatorImpl* level_estimator_;
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NoiseSuppressionImpl* noise_suppression_;
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VoiceDetectionImpl* voice_detection_;
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std::list<ProcessingComponent*> component_list_;
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FileWrapper* debug_file_;
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CriticalSectionWrapper* crit_;
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AudioBuffer* render_audio_;
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AudioBuffer* capture_audio_;
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int sample_rate_hz_;
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int split_sample_rate_hz_;
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int samples_per_channel_;
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int stream_delay_ms_;
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bool was_stream_delay_set_;
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int num_render_input_channels_;
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int num_capture_input_channels_;
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int num_capture_output_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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