webrtc/voice_engine/include/voe_audio_processing.h

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

// This sub-API supports the following functionalities:
//
//  - Noise Suppression (NS).
//  - Automatic Gain Control (AGC).
//  - Echo Control (EC).
//  - Receiving side VAD, NS and AGC.
//  - Measurements of instantaneous speech, noise and echo levels.
//  - Generation of AP debug recordings.
//  - Detection of keyboard typing which can disrupt a voice conversation.
//
// Usage example, omitting error checking:
//
//  using namespace webrtc;
//  VoiceEngine* voe = VoiceEngine::Create();
//  VoEBase* base = VoEBase::GetInterface();
//  VoEAudioProcessing* ap = VoEAudioProcessing::GetInterface(voe);
//  base->Init();
//  ap->SetEcStatus(true, kAgcAdaptiveAnalog);
//  ...
//  base->Terminate();
//  base->Release();
//  ap->Release();
//  VoiceEngine::Delete(voe);
//
#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H

#include <stdio.h>

#include "webrtc/common_types.h"

namespace webrtc {

class VoiceEngine;

// VoERxVadCallback
class WEBRTC_DLLEXPORT VoERxVadCallback
{
public:
    virtual void OnRxVad(int channel, int vadDecision) = 0;

protected:
    virtual ~VoERxVadCallback() {}
};

// VoEAudioProcessing
class WEBRTC_DLLEXPORT VoEAudioProcessing
{
public:
    // Factory for the VoEAudioProcessing sub-API. Increases an internal
    // reference counter if successful. Returns NULL if the API is not
    // supported or if construction fails.
    static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);

    // Releases the VoEAudioProcessing sub-API and decreases an internal
    // reference counter. Returns the new reference count. This value should
    // be zero for all sub-API:s before the VoiceEngine object can be safely
    // deleted.
    virtual int Release() = 0;

    // Sets Noise Suppression (NS) status and mode.
    // The NS reduces noise in the microphone signal.
    virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;

    // Gets the NS status and mode.
    virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;

    // Sets the Automatic Gain Control (AGC) status and mode.
    // The AGC adjusts the microphone signal to an appropriate level.
    virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;

    // Gets the AGC status and mode.
    virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;

    // Sets the AGC configuration.
    // Should only be used in situations where the working environment
    // is well known.
    virtual int SetAgcConfig(AgcConfig config) = 0;

    // Gets the AGC configuration.
    virtual int GetAgcConfig(AgcConfig& config) = 0;

    // Sets the Echo Control (EC) status and mode.
    // The EC mitigates acoustic echo where a user can hear their own
    // speech repeated back due to an acoustic coupling between the
    // speaker and the microphone at the remote end.
    virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;

    // Gets the EC status and mode.
    virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;

    // Enables the compensation of clock drift between the capture and render
    // streams by the echo canceller (i.e. only using EcMode==kEcAec). It will
    // only be enabled if supported on the current platform; otherwise an error
    // will be returned. Check if the platform is supported by calling
    // |DriftCompensationSupported()|.
    virtual int EnableDriftCompensation(bool enable) = 0;
    virtual bool DriftCompensationEnabled() = 0;
    static bool DriftCompensationSupported();

    // Sets a delay |offset| in ms to add to the system delay reported by the
    // OS, which is used by the AEC to synchronize far- and near-end streams.
    // In some cases a system may introduce a delay which goes unreported by the
    // OS, but which is known to the user. This method can be used to compensate
    // for the unreported delay.
    virtual void SetDelayOffsetMs(int offset) = 0;
    virtual int DelayOffsetMs() = 0;

    // Modifies settings for the AEC designed for mobile devices (AECM).
    virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
                            bool enableCNG = true) = 0;

    // Gets settings for the AECM.
    virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;

    // Enables a high pass filter on the capture signal. This removes DC bias
    // and low-frequency noise. Recommended to be enabled.
    virtual int EnableHighPassFilter(bool enable) = 0;
    virtual bool IsHighPassFilterEnabled() = 0;

    // Sets status and mode of the receiving-side (Rx) NS.
    // The Rx NS reduces noise in the received signal for the specified
    // |channel|. Intended for advanced usage only.
    virtual int SetRxNsStatus(int channel,
                              bool enable,
                              NsModes mode = kNsUnchanged) = 0;

    // Gets status and mode of the receiving-side NS.
    virtual int GetRxNsStatus(int channel,
                              bool& enabled,
                              NsModes& mode) = 0;

    // Sets status and mode of the receiving-side (Rx) AGC.
    // The Rx AGC adjusts the received signal to an appropriate level
    // for the specified |channel|. Intended for advanced usage only.
    virtual int SetRxAgcStatus(int channel,
                               bool enable,
                               AgcModes mode = kAgcUnchanged) = 0;

    // Gets status and mode of the receiving-side AGC.
    virtual int GetRxAgcStatus(int channel,
                               bool& enabled,
                               AgcModes& mode) = 0;

    // Modifies the AGC configuration on the receiving side for the
    // specified |channel|.
    virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;

    // Gets the AGC configuration on the receiving side.
    virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;

    // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
    // notifications for the specified |channel|.
    virtual int RegisterRxVadObserver(int channel,
                                      VoERxVadCallback &observer) = 0;

    // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
    // notifications for the specified |channel|.
    virtual int DeRegisterRxVadObserver(int channel) = 0;

    // Gets the VAD/DTX activity for the specified |channel|.
    // The returned value is 1 if frames of audio contains speech
    // and 0 if silence. The output is always 1 if VAD is disabled.
    virtual int VoiceActivityIndicator(int channel) = 0;

    // Enables or disables the possibility to retrieve echo metrics and delay
    // logging values during an active call. The metrics are only supported in
    // AEC.
    virtual int SetEcMetricsStatus(bool enable) = 0;

    // Gets the current EC metric status.
    virtual int GetEcMetricsStatus(bool& enabled) = 0;

    // Gets the instantaneous echo level metrics.
    virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;

    // Gets the EC internal |delay_median| and |delay_std| in ms between
    // near-end and far-end. The metric |fraction_poor_delays| is the amount of
    // delay values that potentially can break the EC. The values are aggregated
    // over one second and the last updated metrics are returned.
    virtual int GetEcDelayMetrics(int& delay_median, int& delay_std,
                                  float& fraction_poor_delays) = 0;

    // Enables recording of Audio Processing (AP) debugging information.
    // The file can later be used for off-line analysis of the AP performance.
    virtual int StartDebugRecording(const char* fileNameUTF8) = 0;

    // Same as above but sets and uses an existing file handle. Takes ownership
    // of |file_handle| and passes it on to the audio processing module.
    virtual int StartDebugRecording(FILE* file_handle) = 0;

    // Disables recording of AP debugging information.
    virtual int StopDebugRecording() = 0;

    // Enables or disables detection of disturbing keyboard typing.
    // An error notification will be given as a callback upon detection.
    virtual int SetTypingDetectionStatus(bool enable) = 0;

    // Gets the current typing detection status.
    virtual int GetTypingDetectionStatus(bool& enabled) = 0;

    // Reports the lower of:
    // * Time in seconds since the last typing event.
    // * Time in seconds since the typing detection was enabled.
    // Returns error if typing detection is disabled.
    virtual int TimeSinceLastTyping(int &seconds) = 0;

    // Optional setting of typing detection parameters
    // Parameter with value == 0 will be ignored
    // and left with default config.
    // TODO(niklase) Remove default argument as soon as libJingle is updated!
    virtual int SetTypingDetectionParameters(int timeWindow,
                                             int costPerTyping,
                                             int reportingThreshold,
                                             int penaltyDecay,
                                             int typeEventDelay = 0) = 0;

    // Swaps the capture-side left and right audio channels when enabled. It
    // only has an effect when using a stereo send codec. The setting is
    // persistent; it will be applied whenever a stereo send codec is enabled.
    //
    // The swap is applied only to the captured audio, and not mixed files. The
    // swap will appear in file recordings and when accessing audio through the
    // external media interface.
    virtual void EnableStereoChannelSwapping(bool enable) = 0;
    virtual bool IsStereoChannelSwappingEnabled() = 0;

protected:
    VoEAudioProcessing() {}
    virtual ~VoEAudioProcessing() {}
};

}  // namespace webrtc

#endif  // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H