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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_device/include/audio_device.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_device/include/audio_device.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_device/include/audio_device.h | 194 |
1 files changed, 194 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device.h b/third_party/libwebrtc/modules/audio_device/include/audio_device.h new file mode 100644 index 0000000000..936ee6cb04 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_device/include/audio_device.h @@ -0,0 +1,194 @@ +/* + * 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<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 absl::optional<Stats> 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_ |