audio_device.h - mozsearch

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/*

 *  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.

*/

#ifndef API_AUDIO_AUDIO_DEVICE_H_

#define API_AUDIO_AUDIO_DEVICE_H_

#include <optional>

#include "api/audio/audio_device_defines.h"

#include "api/ref_count.h"

#include "api/scoped_refptr.h"

#include "api/task_queue/task_queue_factory.h"

namespace webrtc {

class AudioDeviceModuleForTest;

class AudioDeviceModule : public webrtc::RefCountInterface {

 public:

  enum AudioLayer {

    kPlatformDefaultAudio = 0,

    kWindowsCoreAudio,

    kWindowsCoreAudio2,

    kLinuxAlsaAudio,

    kLinuxPulseAudio,

    kAndroidJavaAudio,

    kAndroidOpenSLESAudio,

    kAndroidJavaInputAndOpenSLESOutputAudio,

    kAndroidAAudioAudio,

    kAndroidJavaInputAndAAudioOutputAudio,

    kDummyAudio,

};

  enum WindowsDeviceType {

    kDefaultCommunicationDevice = -1,

    kDefaultDevice = -2

};

// Only supported on iOS.

#if defined(WEBRTC_IOS)

  enum MutedSpeechEvent { kMutedSpeechStarted, kMutedSpeechEnded };

  typedef void (^MutedSpeechEventHandler)(MutedSpeechEvent event);

#endif  // WEBRTC_IOS

  struct Stats {

    // The fields below correspond to similarly-named fields in the WebRTC stats

    // spec. https://w3c.github.io/webrtc-stats/#playoutstats-dict*

    double synthesized_samples_duration_s = 0;

    uint64_t synthesized_samples_events = 0;

    double total_samples_duration_s = 0;

    double total_playout_delay_s = 0;

    uint64_t total_samples_count = 0;

};

 public:

  // Creates a default ADM for usage in production code.

  static rtc::scoped_refptr<AudioDeviceModule> Create(

      AudioLayer audio_layer,

      TaskQueueFactory* task_queue_factory);

  // Creates an ADM with support for extra test methods. Don't use this factory

  // in production code.

  static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(

      AudioLayer audio_layer,

      TaskQueueFactory* task_queue_factory);

  // Retrieve the currently utilized audio layer

  virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;

  // Full-duplex transportation of PCM audio

  virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;

  // Main initialization and termination

  virtual int32_t Init() = 0;

  virtual int32_t Terminate() = 0;

  virtual bool Initialized() const = 0;

  // Device enumeration

  virtual int16_t PlayoutDevices() = 0;

  virtual int16_t RecordingDevices() = 0;

  virtual int32_t PlayoutDeviceName(uint16_t index,

                                    char name[kAdmMaxDeviceNameSize],

                                    char guid[kAdmMaxGuidSize]) = 0;

  virtual int32_t RecordingDeviceName(uint16_t index,

                                      char name[kAdmMaxDeviceNameSize],

                                      char guid[kAdmMaxGuidSize]) = 0;

  // Device selection

  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;

  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;

  virtual int32_t SetRecordingDevice(uint16_t index) = 0;

  virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;

  // Audio transport initialization

  virtual int32_t PlayoutIsAvailable(bool* available) = 0;

  virtual int32_t InitPlayout() = 0;

  virtual bool PlayoutIsInitialized() const = 0;

  virtual int32_t RecordingIsAvailable(bool* available) = 0;

  virtual int32_t InitRecording() = 0;

  virtual bool RecordingIsInitialized() const = 0;

  // Audio transport control

  virtual int32_t StartPlayout() = 0;

  virtual int32_t StopPlayout() = 0;

  virtual bool Playing() const = 0;

  virtual int32_t StartRecording() = 0;

  virtual int32_t StopRecording() = 0;

  virtual bool Recording() const = 0;

  // Audio mixer initialization

  virtual int32_t InitSpeaker() = 0;

  virtual bool SpeakerIsInitialized() const = 0;

  virtual int32_t InitMicrophone() = 0;

  virtual bool MicrophoneIsInitialized() const = 0;

  // Speaker volume controls

  virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;

  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;

  virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;

  virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;

  virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;

  // Microphone volume controls

  virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;

  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;

  virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;

  virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;

  virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;

  // Speaker mute control

  virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;

  virtual int32_t SetSpeakerMute(bool enable) = 0;

  virtual int32_t SpeakerMute(bool* enabled) const = 0;

  // Microphone mute control

  virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;

  virtual int32_t SetMicrophoneMute(bool enable) = 0;

  virtual int32_t MicrophoneMute(bool* enabled) const = 0;

  // Stereo support

  virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;

  virtual int32_t SetStereoPlayout(bool enable) = 0;

  virtual int32_t StereoPlayout(bool* enabled) const = 0;

  virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;

  virtual int32_t SetStereoRecording(bool enable) = 0;

  virtual int32_t StereoRecording(bool* enabled) const = 0;

  // Playout delay

  virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;

  // Only supported on Android.

  virtual bool BuiltInAECIsAvailable() const = 0;

  virtual bool BuiltInAGCIsAvailable() const = 0;

  virtual bool BuiltInNSIsAvailable() const = 0;

  // Enables the built-in audio effects. Only supported on Android.

  virtual int32_t EnableBuiltInAEC(bool enable) = 0;

  virtual int32_t EnableBuiltInAGC(bool enable) = 0;

  virtual int32_t EnableBuiltInNS(bool enable) = 0;

  // Play underrun count. Only supported on Android.

  // TODO(alexnarest): Make it abstract after upstream projects support it.

  virtual int32_t GetPlayoutUnderrunCount() const { return -1; }

  // Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will

  // not be present in the stats.

  virtual std::optional<Stats> GetStats() const { return std::nullopt; }

// Only supported on iOS.

#if defined(WEBRTC_IOS)

  virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;

  virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;

#endif  // WEBRTC_IOS

 protected:

  ~AudioDeviceModule() override {}

};

// Extends the default ADM interface with some extra test methods.

// Intended for usage in tests only and requires a unique factory method.

class AudioDeviceModuleForTest : public AudioDeviceModule {

 public:

  // Triggers internal restart sequences of audio streaming. Can be used by

  // tests to emulate events corresponding to e.g. removal of an active audio

  // device or other actions which causes the stream to be disconnected.

  virtual int RestartPlayoutInternally() = 0;

  virtual int RestartRecordingInternally() = 0;

  virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;

  virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;

};

}  // namespace webrtc

#endif  // API_AUDIO_AUDIO_DEVICE_H_