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eb8c4ca608
This removes some code in the AudioDeviceWindowsCore::CoreAudioIsSupported function that was checking that every audio input and output device was functional. There are legitimate cases where some, or all, audio devices may not be accessible, and that was causing CoreAudioIsSupported to return false. If CoreAudioIsSupported returns false, a subsequent RTC_CHECK call fails, which causes the entire app to exit. After this change, the CoreAudioIsSupported() function simply checks if the Core Audio APIs are supported and no longer tries to do extra stuff unrelated to checking if the APIs are supported. Note that Core Audio is actually supported in all versions of Windows after Windows XP. There were log messages in the code saying that if CoreAudioIsSupported() returns false, WebRTC will use the Wave Audio APIs instead. But this is no longer the case. The Wave Audio APIs would only be needed for Windows XP, and this code appears to have already been removed from WebRTC. It is tempting to simply make CoreAudioIsSupported() do a "return true;" but for now I only removed the part of the logging messages that mentioned the Wave Audio APIs. I understand that there is a new Audio Device Module (ADM) called WindowsCoreAudio2, which is now recommended for use by apps. Apps are supposed to instantiate WindowsCoreAudio2 and pass it in to WebRTC. When the app supplies its own ADM, CoreAudioIsSupported() does not get invoked, which avoids the bug. To help make it clearer that using WindowsCoreAudio2 is an acceptable solution, I am removing a comment that says that kWindowsCoreAudio2 is "experimental". Bug: webrtc:11081 Change-Id: I7ed1684a276799f4c83006b45629e48814f0b18b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161463 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30025}
179 lines
6.7 KiB
C++
179 lines
6.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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#include "api/scoped_refptr.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "rtc_base/ref_count.h"
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namespace webrtc {
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class AudioDeviceModuleForTest;
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class AudioDeviceModule : public rtc::RefCountInterface {
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public:
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enum AudioLayer {
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kPlatformDefaultAudio = 0,
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kWindowsCoreAudio,
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kWindowsCoreAudio2,
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kLinuxAlsaAudio,
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kLinuxPulseAudio,
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kAndroidJavaAudio,
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kAndroidOpenSLESAudio,
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kAndroidJavaInputAndOpenSLESOutputAudio,
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kAndroidAAudioAudio,
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kAndroidJavaInputAndAAudioOutputAudio,
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kDummyAudio,
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};
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enum WindowsDeviceType {
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kDefaultCommunicationDevice = -1,
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kDefaultDevice = -2
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};
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public:
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// Creates a default ADM for usage in production code.
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static rtc::scoped_refptr<AudioDeviceModule> Create(
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AudioLayer audio_layer,
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TaskQueueFactory* task_queue_factory);
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// Creates an ADM with support for extra test methods. Don't use this factory
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// in production code.
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static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
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AudioLayer audio_layer,
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TaskQueueFactory* task_queue_factory);
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// Retrieve the currently utilized audio layer
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virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
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// Full-duplex transportation of PCM audio
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virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
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// Main initialization and termination
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virtual int32_t Init() = 0;
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virtual int32_t Terminate() = 0;
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virtual bool Initialized() const = 0;
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// Device enumeration
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virtual int16_t PlayoutDevices() = 0;
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virtual int16_t RecordingDevices() = 0;
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virtual int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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virtual int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) = 0;
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// Device selection
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virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
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virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
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virtual int32_t SetRecordingDevice(uint16_t index) = 0;
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virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
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// Audio transport initialization
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virtual int32_t PlayoutIsAvailable(bool* available) = 0;
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virtual int32_t InitPlayout() = 0;
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virtual bool PlayoutIsInitialized() const = 0;
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virtual int32_t RecordingIsAvailable(bool* available) = 0;
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virtual int32_t InitRecording() = 0;
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virtual bool RecordingIsInitialized() const = 0;
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// Audio transport control
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virtual int32_t StartPlayout() = 0;
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virtual int32_t StopPlayout() = 0;
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virtual bool Playing() const = 0;
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virtual int32_t StartRecording() = 0;
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virtual int32_t StopRecording() = 0;
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virtual bool Recording() const = 0;
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// Audio mixer initialization
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virtual int32_t InitSpeaker() = 0;
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virtual bool SpeakerIsInitialized() const = 0;
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virtual int32_t InitMicrophone() = 0;
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virtual bool MicrophoneIsInitialized() const = 0;
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// Speaker volume controls
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virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
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virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
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// Microphone volume controls
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virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
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virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
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virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
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virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
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// Speaker mute control
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virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetSpeakerMute(bool enable) = 0;
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virtual int32_t SpeakerMute(bool* enabled) const = 0;
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// Microphone mute control
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virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
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virtual int32_t SetMicrophoneMute(bool enable) = 0;
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virtual int32_t MicrophoneMute(bool* enabled) const = 0;
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// Stereo support
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virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoPlayout(bool enable) = 0;
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virtual int32_t StereoPlayout(bool* enabled) const = 0;
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virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
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virtual int32_t SetStereoRecording(bool enable) = 0;
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virtual int32_t StereoRecording(bool* enabled) const = 0;
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// Playout delay
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virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
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// Only supported on Android.
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virtual bool BuiltInAECIsAvailable() const = 0;
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virtual bool BuiltInAGCIsAvailable() const = 0;
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virtual bool BuiltInNSIsAvailable() const = 0;
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// Enables the built-in audio effects. Only supported on Android.
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virtual int32_t EnableBuiltInAEC(bool enable) = 0;
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virtual int32_t EnableBuiltInAGC(bool enable) = 0;
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virtual int32_t EnableBuiltInNS(bool enable) = 0;
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// Play underrun count. Only supported on Android.
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// TODO(alexnarest): Make it abstract after upstream projects support it.
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virtual int32_t GetPlayoutUnderrunCount() const { return -1; }
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// Only supported on iOS.
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#if defined(WEBRTC_IOS)
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virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
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virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
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#endif // WEBRTC_IOS
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protected:
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~AudioDeviceModule() override {}
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};
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// Extends the default ADM interface with some extra test methods.
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// Intended for usage in tests only and requires a unique factory method.
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class AudioDeviceModuleForTest : public AudioDeviceModule {
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public:
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// Triggers internal restart sequences of audio streaming. Can be used by
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// tests to emulate events corresponding to e.g. removal of an active audio
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// device or other actions which causes the stream to be disconnected.
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virtual int RestartPlayoutInternally() = 0;
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virtual int RestartRecordingInternally() = 0;
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virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
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virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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