| /* |
| * 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_ |
