/* * 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 MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_ #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_ #include "absl/types/optional.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/include/audio_device_defines.h" #include "rtc_base/ref_count.h" namespace webrtc { class AudioDeviceModuleForTest; class AudioDeviceModule : public rtc::RefCountInterface { public: enum AudioLayer { kPlatformDefaultAudio = 0, kWindowsCoreAudio, kWindowsCoreAudio2, kLinuxAlsaAudio, kLinuxPulseAudio, kAndroidJavaAudio, kAndroidOpenSLESAudio, kAndroidJavaInputAndOpenSLESOutputAudio, kAndroidAAudioAudio, kAndroidJavaInputAndAAudioOutputAudio, kDummyAudio, }; enum WindowsDeviceType { kDefaultCommunicationDevice = -1, kDefaultDevice = -2 }; 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 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 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 absl::optional GetStats() const { return absl::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 // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_