863b536100
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process. This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper. BUG=2567 R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
114 lines
3.6 KiB
C++
114 lines
3.6 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_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
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#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
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#include "webrtc/voice_engine/include/voe_audio_processing.h"
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#include "webrtc/voice_engine/shared_data.h"
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namespace webrtc {
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class VoEAudioProcessingImpl : public VoEAudioProcessing {
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public:
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virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged);
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virtual int GetNsStatus(bool& enabled, NsModes& mode);
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virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged);
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virtual int GetAgcStatus(bool& enabled, AgcModes& mode);
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virtual int SetAgcConfig(AgcConfig config);
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virtual int GetAgcConfig(AgcConfig& config);
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virtual int SetRxNsStatus(int channel,
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bool enable,
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NsModes mode = kNsUnchanged);
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virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode);
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virtual int SetRxAgcStatus(int channel,
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bool enable,
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AgcModes mode = kAgcUnchanged);
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virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode);
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virtual int SetRxAgcConfig(int channel, AgcConfig config);
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virtual int GetRxAgcConfig(int channel, AgcConfig& config);
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virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged);
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virtual int GetEcStatus(bool& enabled, EcModes& mode);
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virtual int EnableDriftCompensation(bool enable);
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virtual bool DriftCompensationEnabled();
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virtual void SetDelayOffsetMs(int offset);
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virtual int DelayOffsetMs();
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virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
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bool enableCNG = true);
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virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG);
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virtual int EnableHighPassFilter(bool enable);
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virtual bool IsHighPassFilterEnabled();
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virtual int RegisterRxVadObserver(int channel,
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VoERxVadCallback& observer);
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virtual int DeRegisterRxVadObserver(int channel);
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virtual int VoiceActivityIndicator(int channel);
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virtual int SetEcMetricsStatus(bool enable);
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virtual int GetEcMetricsStatus(bool& enabled);
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virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP);
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virtual int GetEcDelayMetrics(int& delay_median, int& delay_std);
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virtual int StartDebugRecording(const char* fileNameUTF8);
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virtual int StartDebugRecording(FILE* file_handle);
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virtual int StopDebugRecording();
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virtual int SetTypingDetectionStatus(bool enable);
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virtual int GetTypingDetectionStatus(bool& enabled);
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virtual int TimeSinceLastTyping(int &seconds);
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// TODO(niklase) Remove default argument as soon as libJingle is updated!
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virtual int SetTypingDetectionParameters(int timeWindow,
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int costPerTyping,
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int reportingThreshold,
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int penaltyDecay,
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int typeEventDelay = 0);
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virtual void EnableStereoChannelSwapping(bool enable);
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virtual bool IsStereoChannelSwappingEnabled();
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protected:
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VoEAudioProcessingImpl(voe::SharedData* shared);
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virtual ~VoEAudioProcessingImpl();
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private:
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bool _isAecMode;
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voe::SharedData* _shared;
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};
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
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