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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 <cstdint>
#include <optional>
#include "api/audio/audio_device_defines.h"
#include "api/ref_count.h"
namespace webrtc {
class AudioDeviceModuleForTest;
class AudioDeviceModule : public 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:
// 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_
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