ddbb8a2c24
Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
80 lines
3.0 KiB
C++
80 lines
3.0 KiB
C++
/*
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* Copyright (c) 2012 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_ECHO_CANCELLATION_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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namespace webrtc {
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class AudioBuffer;
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class CriticalSectionWrapper;
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class EchoCancellationImpl : public EchoCancellation,
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public ProcessingComponent {
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public:
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EchoCancellationImpl(const AudioProcessing* apm,
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CriticalSectionWrapper* crit);
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virtual ~EchoCancellationImpl();
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int ProcessRenderAudio(const AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio);
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// EchoCancellation implementation.
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virtual bool is_enabled() const OVERRIDE;
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virtual int stream_drift_samples() const OVERRIDE;
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// ProcessingComponent implementation.
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virtual int Initialize() OVERRIDE;
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virtual void SetExtraOptions(const Config& config) OVERRIDE;
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private:
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// EchoCancellation implementation.
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virtual int Enable(bool enable) OVERRIDE;
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virtual int enable_drift_compensation(bool enable) OVERRIDE;
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virtual bool is_drift_compensation_enabled() const OVERRIDE;
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virtual void set_stream_drift_samples(int drift) OVERRIDE;
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virtual int set_suppression_level(SuppressionLevel level) OVERRIDE;
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virtual SuppressionLevel suppression_level() const OVERRIDE;
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virtual int enable_metrics(bool enable) OVERRIDE;
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virtual bool are_metrics_enabled() const OVERRIDE;
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virtual bool stream_has_echo() const OVERRIDE;
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virtual int GetMetrics(Metrics* metrics) OVERRIDE;
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virtual int enable_delay_logging(bool enable) OVERRIDE;
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virtual bool is_delay_logging_enabled() const OVERRIDE;
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virtual int GetDelayMetrics(int* median, int* std) OVERRIDE;
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virtual struct AecCore* aec_core() const OVERRIDE;
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// ProcessingComponent implementation.
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virtual void* CreateHandle() const OVERRIDE;
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virtual int InitializeHandle(void* handle) const OVERRIDE;
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virtual int ConfigureHandle(void* handle) const OVERRIDE;
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virtual void DestroyHandle(void* handle) const OVERRIDE;
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virtual int num_handles_required() const OVERRIDE;
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virtual int GetHandleError(void* handle) const OVERRIDE;
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const AudioProcessing* apm_;
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CriticalSectionWrapper* crit_;
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bool drift_compensation_enabled_;
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bool metrics_enabled_;
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SuppressionLevel suppression_level_;
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int stream_drift_samples_;
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bool was_stream_drift_set_;
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bool stream_has_echo_;
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bool delay_logging_enabled_;
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bool delay_correction_enabled_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
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