Diff
Modified: trunk/LayoutTests/ChangeLog (294101 => 294102)
--- trunk/LayoutTests/ChangeLog 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/LayoutTests/ChangeLog 2022-05-12 13:56:19 UTC (rev 294102)
@@ -1,3 +1,13 @@
+2022-05-12 Youenn Fablet <you...@apple.com>
+
+ Add a better mock for audio units used by CoreAudioSharedUnit
+ https://bugs.webkit.org/show_bug.cgi?id=240231
+
+ Reviewed by Eric Carlson.
+
+ * fast/mediastream/audio-unit-reconfigure-expected.txt: Added.
+ * fast/mediastream/audio-unit-reconfigure.html: Added.
+
2022-05-11 John Cunningham <johncunning...@apple.com>
[GPU Process] webrtc/vp8-then-h264-gpu-process-crash.html flaky fails after turning DOM in GPUP on by default
Added: trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure-expected.txt (0 => 294102)
--- trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure-expected.txt (rev 0)
+++ trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure-expected.txt 2022-05-12 13:56:19 UTC (rev 294102)
@@ -0,0 +1,5 @@
+
+
+PASS Reconfigure audio sample rate
+PASS Reconfigure audio unit when being in render-only mode
+
Added: trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure.html (0 => 294102)
--- trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure.html (rev 0)
+++ trunk/LayoutTests/fast/mediastream/audio-unit-reconfigure.html 2022-05-12 13:56:19 UTC (rev 294102)
@@ -0,0 +1,70 @@
+<!DOCTYPE html>
+<html>
+<head>
+ <meta charset="utf-8">
+ <script src=""
+ <script src=""
+ </head>
+<body>
+<video id="video1" autoplay playsInline controls></video>
+<video id="video2" autoplay playsInline controls></video>
+<script>
+promise_test(async (test) => {
+ const context = new AudioContext();
+ const oscillator = context.createOscillator();
+ const streamDestination = context.createMediaStreamDestination();
+ oscillator.connect(streamDestination);
+ video1.srcObject = streamDestination.stream;
+ // video1 will be played through VPIO will fix the rate to a single value.
+ await video1.play();
+ oscillator.start();
+
+ const stream = await navigator.mediaDevices.getUserMedia({ audio: { sampleRate: 48000 }});
+ stream.getAudioTracks()[0]._onended_ = () => assert_not_reached("should not end");
+ await new Promise(resolve => setTimeout(resolve, 100));
+ stream.getAudioTracks()[0].stop();
+
+ const stream2 = await navigator.mediaDevices.getUserMedia({ audio: { sampleRate: 44100 }});
+ stream2.getAudioTracks()[0]._onended_ = () => assert_not_reached("should not end");
+ await new Promise(resolve => setTimeout(resolve, 100));
+}, "Reconfigure audio sample rate");
+
+promise_test(async (test) => {
+ const context = new AudioContext();
+ const oscillator = context.createOscillator();
+ const streamDestination = context.createMediaStreamDestination();
+ oscillator.connect(streamDestination);
+ video1.srcObject = streamDestination.stream;
+ await video1.play();
+ oscillator.start();
+
+ const defaultAudioConstraints = {
+ sampleRate: 48000,
+ sampleSize: 16,
+ channelCount: 1,
+ echoCancellation: true,
+ noiseSuppression: true
+ };
+ const musicModeAudioConstraints = {
+ sampleRate: 48000,
+ sampleSize: 16,
+ channelCount: 2,
+ echoCancellation: false,
+ noiseSuppression: false
+ };
+
+ const stream1 = await navigator.mediaDevices.getUserMedia({ audio: defaultAudioConstraints });
+ video2.srcObject = stream1;
+ await video2.play();
+ await new Promise(resolve => setTimeout(resolve, 1000));
+
+ // By stopping the capture audio track, the VPIO unit will go in render only mode as it needs to play video 1.
+ stream1.getAudioTracks()[0].stop();
+
+ const stream2 = await navigator.mediaDevices.getUserMedia({ audio: musicModeAudioConstraints });
+ stream2.getAudioTracks()[0]._onended_ = () => log.innerHTML += 'track ended';
+ await new Promise(resolve => setTimeout(resolve, 1000));
+}, "Reconfigure audio unit when being in render-only mode");
+</script>
+</body>
+</html>
Modified: trunk/Source/WebCore/ChangeLog (294101 => 294102)
--- trunk/Source/WebCore/ChangeLog 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/ChangeLog 2022-05-12 13:56:19 UTC (rev 294102)
@@ -1,3 +1,29 @@
+2022-05-12 Youenn Fablet <you...@apple.com>
+
+ Add a better mock for audio units used by CoreAudioSharedUnit
+ https://bugs.webkit.org/show_bug.cgi?id=240231
+
+ Reviewed by Eric Carlson.
+
+ Move CoreAudioSharedUnit to its own file, CoreAudioSharedUnit.h/CoreAudioSharedUnit.cpp.
+ Introduce an internal unit to abstract out CoreAudioSharedUnit from actual CoreAudio units.
+ Implement a CoreAudio based internal unit in CoreAudioSharedUnit.cpp.
+ Implement a Mock based internal unit in MockAudioSharedUnit.h/MockAudioSharedUnit.mm.
+ Add inputput/ouput sample rate validation (debug assert and release error returned).
+ Add a test that uses this validation.
+
+ Test: fast/mediastream/audio-unit-reconfigure.html
+
+ * SourcesCocoa.txt:
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/mediastream/mac/BaseAudioSharedUnit.cpp:
+ * platform/mediastream/mac/BaseAudioSharedUnit.h:
+ * platform/mediastream/mac/CoreAudioCaptureSource.cpp:
+ * platform/mediastream/mac/CoreAudioSharedUnit.cpp: Added.
+ * platform/mediastream/mac/CoreAudioSharedUnit.h: Added.
+ * platform/mediastream/mac/MockAudioSharedUnit.h:
+ * platform/mediastream/mac/MockAudioSharedUnit.mm:
+
2022-05-12 Oriol Brufau <obru...@igalia.com>
[cssom] Serialize computed '-webkit-text-combine: none'
Modified: trunk/Source/WebCore/SourcesCocoa.txt (294101 => 294102)
--- trunk/Source/WebCore/SourcesCocoa.txt 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/SourcesCocoa.txt 2022-05-12 13:56:19 UTC (rev 294102)
@@ -565,6 +565,7 @@
platform/mediastream/mac/CoreAudioCaptureDevice.cpp
platform/mediastream/mac/CoreAudioCaptureDeviceManager.cpp
platform/mediastream/mac/CoreAudioCaptureSource.cpp
+platform/mediastream/mac/CoreAudioSharedUnit.cpp
platform/mediastream/mac/DisplayCaptureManagerCocoa.cpp
platform/mediastream/mac/MediaStreamTrackAudioSourceProviderCocoa.cpp
platform/mediastream/mac/MockAudioSharedUnit.mm
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (294101 => 294102)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2022-05-12 13:56:19 UTC (rev 294102)
@@ -8567,6 +8567,8 @@
410A8EF724F8F3F6004F9070 /* TextEncoderStreamEncoder.idl */ = {isa = PBXFileReference; lastKnownFileType = text; path = TextEncoderStreamEncoder.idl; sourceTree = "<group>"; };
410A8EF824F8F47A004F9070 /* TextEncoderStreamEncoder.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = TextEncoderStreamEncoder.cpp; sourceTree = "<group>"; };
410A8EF924F8F47B004F9070 /* TextEncoderStreamEncoder.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = TextEncoderStreamEncoder.h; sourceTree = "<group>"; };
+ 410AD2DE2822BBF6009C93C8 /* CoreAudioSharedUnit.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CoreAudioSharedUnit.h; sourceTree = "<group>"; };
+ 410AD2DF2822BBF6009C93C8 /* CoreAudioSharedUnit.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = CoreAudioSharedUnit.cpp; sourceTree = "<group>"; };
410B7E711045FAB000D8224F /* JSMessageEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSMessageEventCustom.cpp; sourceTree = "<group>"; };
410BA1312570FE57002E2F8A /* LibWebRTCRtpTransformableFrame.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = LibWebRTCRtpTransformableFrame.cpp; path = libwebrtc/LibWebRTCRtpTransformableFrame.cpp; sourceTree = "<group>"; };
410BCF5326F0CD8B0040B124 /* RTCLocalSessionDescriptionInit.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCLocalSessionDescriptionInit.h; sourceTree = "<group>"; };
@@ -18813,6 +18815,8 @@
3F8020341E9E381D00DEC61D /* CoreAudioCaptureDeviceManager.h */,
3F3BB5821E709EE400C701F2 /* CoreAudioCaptureSource.cpp */,
3F3BB5831E709EE400C701F2 /* CoreAudioCaptureSource.h */,
+ 410AD2DF2822BBF6009C93C8 /* CoreAudioSharedUnit.cpp */,
+ 410AD2DE2822BBF6009C93C8 /* CoreAudioSharedUnit.h */,
07A6D8481FEB700B006441DE /* DisplayCaptureManagerCocoa.cpp */,
07A6D8491FEB700C006441DE /* DisplayCaptureManagerCocoa.h */,
4174E9202535DCDD00FE4202 /* MediaStreamTrackAudioSourceProviderCocoa.cpp */,
Modified: trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.cpp (294101 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.cpp 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.cpp 2022-05-12 13:56:19 UTC (rev 294102)
@@ -113,11 +113,6 @@
return 0;
}
-void BaseAudioSharedUnit::resetSampleRate()
-{
- m_sampleRate = AudioSession::sharedSession().sampleRate();
-}
-
void BaseAudioSharedUnit::prepareForNewCapture()
{
m_volume = 1;
Modified: trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.h (294101 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.h 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/platform/mediastream/mac/BaseAudioSharedUnit.h 2022-05-12 13:56:19 UTC (rev 294102)
@@ -83,6 +83,8 @@
void whenAudioCaptureUnitIsNotRunning(Function<void()>&&);
bool isRenderingAudio() const { return m_isRenderingAudio; }
+ const String& persistentIDForTesting() const { return m_capturingDevice ? m_capturingDevice->first : emptyString(); }
+
protected:
void forEachClient(const Function<void(CoreAudioCaptureSource&)>&) const;
bool hasClients() const { return !m_clients.isEmpty(); }
@@ -92,7 +94,7 @@
virtual OSStatus startInternal() = 0;
virtual void stopInternal() = 0;
virtual OSStatus reconfigureAudioUnit() = 0;
- virtual void resetSampleRate();
+ virtual void resetSampleRate() = 0;
virtual void captureDeviceChanged() = 0;
void setSuspended(bool value) { m_suspended = value; }
Modified: trunk/Source/WebCore/platform/mediastream/mac/CoreAudioCaptureSource.cpp (294101 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/CoreAudioCaptureSource.cpp 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/platform/mediastream/mac/CoreAudioCaptureSource.cpp 2022-05-12 13:56:19 UTC (rev 294102)
@@ -31,9 +31,9 @@
#include "AudioSampleBufferList.h"
#include "AudioSampleDataSource.h"
#include "AudioSession.h"
-#include "BaseAudioSharedUnit.h"
#include "CoreAudioCaptureDevice.h"
#include "CoreAudioCaptureDeviceManager.h"
+#include "CoreAudioSharedUnit.h"
#include "Logging.h"
#include "PlatformMediaSessionManager.h"
#include "Timer.h"
@@ -51,9 +51,6 @@
#include <wtf/NeverDestroyed.h>
#include <wtf/Scope.h>
-#include <pal/cf/AudioToolboxSoftLink.h>
-#include <pal/cf/CoreMediaSoftLink.h>
-
#if PLATFORM(IOS_FAMILY)
#include "AVAudioSessionCaptureDevice.h"
#include "AVAudioSessionCaptureDeviceManager.h"
@@ -60,6 +57,9 @@
#include "CoreAudioCaptureSourceIOS.h"
#endif
+#include <pal/cf/AudioToolboxSoftLink.h>
+#include <pal/cf/CoreMediaSoftLink.h>
+
namespace WebCore {
#if PLATFORM(MAC)
@@ -70,614 +70,6 @@
}
#endif
-const UInt32 outputBus = 0;
-const UInt32 inputBus = 1;
-
-class CoreAudioSharedUnit final : public BaseAudioSharedUnit {
-public:
- static CoreAudioSharedUnit& unit();
- static BaseAudioSharedUnit& singleton() { return unit(); }
- CoreAudioSharedUnit();
-
- void registerSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer&);
- void unregisterSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer&);
- bool isRunning() const { return m_ioUnitStarted; }
-
-private:
- static size_t preferredIOBufferSize();
-
- CapabilityValueOrRange sampleRateCapacities() const final { return CapabilityValueOrRange(8000, 96000); }
- const CAAudioStreamDescription& microphoneFormat() const { return m_microphoneProcFormat; }
-
- bool hasAudioUnit() const final { return m_ioUnit; }
- void captureDeviceChanged() final;
- OSStatus reconfigureAudioUnit() final;
-
- OSStatus setupAudioUnit();
- void cleanupAudioUnit() final;
-
- OSStatus startInternal() final;
- void stopInternal() final;
- bool isProducingData() const final { return m_ioUnitStarted; }
- void isProducingMicrophoneSamplesChanged() final;
- void validateOutputDevice(uint32_t deviceID) final;
- int actualSampleRate() const final;
-
- OSStatus configureSpeakerProc(int sampleRate);
- OSStatus configureMicrophoneProc(int sampleRate);
- OSStatus defaultOutputDevice(uint32_t*);
- OSStatus defaultInputDevice(uint32_t*);
-
- static OSStatus microphoneCallback(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*);
- OSStatus processMicrophoneSamples(AudioUnitRenderActionFlags&, const AudioTimeStamp&, UInt32, UInt32, AudioBufferList*);
-
- static OSStatus speakerCallback(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*);
- OSStatus provideSpeakerData(AudioUnitRenderActionFlags&, const AudioTimeStamp&, UInt32, UInt32, AudioBufferList&);
-
- void unduck();
-
- void verifyIsCapturing();
-
- Seconds verifyCaptureInterval() { return isProducingMicrophoneSamples() ? 10_s : 2_s; }
-
- AudioUnit m_ioUnit { nullptr };
-
- // Only read/modified from the IO thread.
- Vector<Ref<AudioSampleDataSource>> m_activeSources;
-
- CAAudioStreamDescription m_microphoneProcFormat;
- RefPtr<AudioSampleBufferList> m_microphoneSampleBuffer;
- uint64_t m_latestMicTimeStamp { 0 };
-
- CAAudioStreamDescription m_speakerProcFormat;
-
- double m_DTSConversionRatio { 0 };
-
- bool m_ioUnitInitialized { false };
- bool m_ioUnitStarted { false };
-
- mutable std::unique_ptr<RealtimeMediaSourceCapabilities> m_capabilities;
- mutable std::optional<RealtimeMediaSourceSettings> m_currentSettings;
-
-#if !LOG_DISABLED
- void checkTimestamps(const AudioTimeStamp&, uint64_t, double);
-
- String m_ioUnitName;
-#endif
-
- uint64_t m_microphoneProcsCalled { 0 };
- uint64_t m_microphoneProcsCalledLastTime { 0 };
- Timer m_verifyCapturingTimer;
-
- bool m_isReconfiguring { false };
- mutable Lock m_speakerSamplesProducerLock;
- CoreAudioSpeakerSamplesProducer* m_speakerSamplesProducer WTF_GUARDED_BY_LOCK(m_speakerSamplesProducerLock) { nullptr };
-};
-
-CoreAudioSharedUnit& CoreAudioSharedUnit::unit()
-{
- static NeverDestroyed<CoreAudioSharedUnit> singleton;
- return singleton;
-}
-
-CoreAudioSharedUnit::CoreAudioSharedUnit()
- : m_verifyCapturingTimer(*this, &CoreAudioSharedUnit::verifyIsCapturing)
-{
-}
-
-void CoreAudioSharedUnit::captureDeviceChanged()
-{
-#if PLATFORM(MAC)
- reconfigureAudioUnit();
-#else
- AVAudioSessionCaptureDeviceManager::singleton().setPreferredAudioSessionDeviceUID(persistentID());
-#endif
-}
-
-size_t CoreAudioSharedUnit::preferredIOBufferSize()
-{
- return AudioSession::sharedSession().bufferSize();
-}
-
-OSStatus CoreAudioSharedUnit::setupAudioUnit()
-{
- if (m_ioUnit)
- return 0;
-
- ASSERT(hasClients());
-
- mach_timebase_info_data_t timebaseInfo;
- mach_timebase_info(&timebaseInfo);
- m_DTSConversionRatio = 1e-9 * static_cast<double>(timebaseInfo.numer) / static_cast<double>(timebaseInfo.denom);
-
- AudioComponentDescription ioUnitDescription = { kAudioUnitType_Output, kAudioUnitSubType_VoiceProcessingIO, kAudioUnitManufacturer_Apple, 0, 0 };
- AudioComponent ioComponent = PAL::AudioComponentFindNext(nullptr, &ioUnitDescription);
- ASSERT(ioComponent);
- if (!ioComponent) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to find vpio unit component", this);
- return -1;
- }
-
-#if !LOG_DISABLED
- CFStringRef name = nullptr;
- PAL::AudioComponentCopyName(ioComponent, &name);
- if (name) {
- m_ioUnitName = name;
- CFRelease(name);
- RELEASE_LOG(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) created \"%s\" component", this, m_ioUnitName.utf8().data());
- }
-#endif
-
- auto err = PAL::AudioComponentInstanceNew(ioComponent, &m_ioUnit);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to open vpio unit, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- if (!enableEchoCancellation()) {
- uint32_t param = 0;
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAUVoiceIOProperty_VoiceProcessingEnableAGC, kAudioUnitScope_Global, inputBus, ¶m, sizeof(param));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio automatic gain control, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- param = 1;
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAUVoiceIOProperty_BypassVoiceProcessing, kAudioUnitScope_Global, inputBus, ¶m, sizeof(param));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit echo cancellation, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- }
-
-#if PLATFORM(IOS_FAMILY)
- uint32_t param = 1;
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, inputBus, ¶m, sizeof(param));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to enable vpio unit input, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-#else
- auto deviceID = captureDeviceID();
- if (!deviceID) {
- err = defaultInputDevice(&deviceID);
- if (err)
- return err;
- }
-
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, inputBus, &deviceID, sizeof(deviceID));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit capture device ID %d, error %d (%.4s)", this, (int)deviceID, (int)err, (char*)&err);
- return err;
- }
-
- uint32_t defaultOutputDeviceID;
- err = defaultOutputDevice(&defaultOutputDeviceID);
- if (!err) {
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, outputBus, &defaultOutputDeviceID, sizeof(defaultOutputDeviceID));
- RELEASE_LOG_ERROR_IF(err, WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit output device ID %d, error %d (%.4s)", this, (int)defaultOutputDeviceID, (int)err, (char*)&err);
- }
- setOutputDeviceID(!err ? defaultOutputDeviceID : 0);
-#endif
-
- // FIXME: Add support for different speaker/microphone sample rates.
- int actualSampleRate = this->actualSampleRate();
- err = configureMicrophoneProc(actualSampleRate);
- if (err)
- return err;
-
- err = configureSpeakerProc(actualSampleRate);
- if (err)
- return err;
-
- err = PAL::AudioUnitInitialize(m_ioUnit);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) AudioUnitInitialize() failed, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- m_ioUnitInitialized = true;
-
- unduck();
-
- return err;
-}
-
-void CoreAudioSharedUnit::unduck()
-{
- uint32_t outputDevice;
- if (!defaultOutputDevice(&outputDevice))
- AudioDeviceDuck(outputDevice, 1.0, nullptr, 0);
-}
-
-int CoreAudioSharedUnit::actualSampleRate() const
-{
- Locker locker { m_speakerSamplesProducerLock };
- return m_speakerSamplesProducer ? m_speakerSamplesProducer->format().streamDescription().mSampleRate : sampleRate();
-}
-
-OSStatus CoreAudioSharedUnit::configureMicrophoneProc(int sampleRate)
-{
- ASSERT(isMainThread());
-
- AURenderCallbackStruct callback = { microphoneCallback, this };
- auto err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, inputBus, &callback, sizeof(callback));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to set vpio unit mic proc, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- AudioStreamBasicDescription microphoneProcFormat = { };
-
- UInt32 size = sizeof(microphoneProcFormat);
- err = PAL::AudioUnitGetProperty(m_ioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, inputBus, µphoneProcFormat, &size);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to get output stream format, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- microphoneProcFormat.mSampleRate = sampleRate;
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, inputBus, µphoneProcFormat, size);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to set output stream format, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- m_microphoneSampleBuffer = AudioSampleBufferList::create(microphoneProcFormat, preferredIOBufferSize() * 2);
- m_microphoneProcFormat = microphoneProcFormat;
-
- return err;
-}
-
-OSStatus CoreAudioSharedUnit::configureSpeakerProc(int sampleRate)
-{
- ASSERT(isMainThread());
-
- AURenderCallbackStruct callback = { speakerCallback, this };
- auto err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, outputBus, &callback, sizeof(callback));
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to set vpio unit speaker proc, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- AudioStreamBasicDescription speakerProcFormat;
- UInt32 size = sizeof(speakerProcFormat);
- {
- Locker locker { m_speakerSamplesProducerLock };
- if (m_speakerSamplesProducer) {
- speakerProcFormat = m_speakerSamplesProducer->format().streamDescription();
- ASSERT(speakerProcFormat.mSampleRate == sampleRate);
- } else {
- err = PAL::AudioUnitGetProperty(m_ioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, outputBus, &speakerProcFormat, &size);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to get input stream format, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- }
- }
- speakerProcFormat.mSampleRate = sampleRate;
-
- err = PAL::AudioUnitSetProperty(m_ioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, outputBus, &speakerProcFormat, size);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to get input stream format, error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
-
- m_speakerProcFormat = speakerProcFormat;
-
- return err;
-}
-
-#if !LOG_DISABLED
-void CoreAudioSharedUnit::checkTimestamps(const AudioTimeStamp& timeStamp, uint64_t sampleTime, double hostTime)
-{
- if (!timeStamp.mSampleTime || sampleTime == m_latestMicTimeStamp || !hostTime)
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::checkTimestamps: unusual timestamps, sample time = %lld, previous sample time = %lld, hostTime %f", sampleTime, m_latestMicTimeStamp, hostTime);
-}
-#endif
-
-OSStatus CoreAudioSharedUnit::provideSpeakerData(AudioUnitRenderActionFlags& flags, const AudioTimeStamp& timeStamp, UInt32 /*inBusNumber*/, UInt32 inNumberFrames, AudioBufferList& ioData)
-{
- if (m_isReconfiguring || !m_speakerSamplesProducerLock.tryLock()) {
- AudioSampleBufferList::zeroABL(ioData, static_cast<size_t>(inNumberFrames * m_speakerProcFormat.bytesPerFrame()));
- flags = kAudioUnitRenderAction_OutputIsSilence;
- return noErr;
- }
-
- Locker locker { AdoptLock, m_speakerSamplesProducerLock };
- if (!m_speakerSamplesProducer) {
- AudioSampleBufferList::zeroABL(ioData, static_cast<size_t>(inNumberFrames * m_speakerProcFormat.bytesPerFrame()));
- flags = kAudioUnitRenderAction_OutputIsSilence;
- return noErr;
- }
- return m_speakerSamplesProducer->produceSpeakerSamples(inNumberFrames, ioData, timeStamp.mSampleTime, timeStamp.mHostTime, flags);
-}
-
-OSStatus CoreAudioSharedUnit::speakerCallback(void *inRefCon, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData)
-{
- ASSERT(ioActionFlags);
- ASSERT(inTimeStamp);
- auto dataSource = static_cast<CoreAudioSharedUnit*>(inRefCon);
- return dataSource->provideSpeakerData(*ioActionFlags, *inTimeStamp, inBusNumber, inNumberFrames, *ioData);
-}
-
-OSStatus CoreAudioSharedUnit::processMicrophoneSamples(AudioUnitRenderActionFlags& ioActionFlags, const AudioTimeStamp& timeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* /*ioData*/)
-{
- ++m_microphoneProcsCalled;
-
- if (m_isReconfiguring)
- return false;
-
- // Pull through the vpio unit to our mic buffer.
- m_microphoneSampleBuffer->reset();
- AudioBufferList& bufferList = m_microphoneSampleBuffer->bufferList();
- auto err = AudioUnitRender(m_ioUnit, &ioActionFlags, &timeStamp, inBusNumber, inNumberFrames, &bufferList);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::processMicrophoneSamples(%p) AudioUnitRender failed with error %d (%.4s), bufferList size %d, inNumberFrames %d ", this, (int)err, (char*)&err, (int)bufferList.mBuffers[0].mDataByteSize, (int)inNumberFrames);
- if (err == kAudio_ParamError) {
- // Our buffer might be too small, the preferred buffer size or sample rate might have changed.
- callOnMainThread([] {
- CoreAudioSharedUnit::singleton().reconfigure();
- });
- }
- // We return early so that if this error happens, we do not increment m_microphoneProcsCalled and fail the capture once timer kicks in.
- return err;
- }
-
- if (!isProducingMicrophoneSamples())
- return noErr;
-
- double adjustedHostTime = m_DTSConversionRatio * timeStamp.mHostTime;
- uint64_t sampleTime = timeStamp.mSampleTime;
-#if !LOG_DISABLED
- checkTimestamps(timeStamp, sampleTime, adjustedHostTime);
-#endif
- m_latestMicTimeStamp = sampleTime;
- m_microphoneSampleBuffer->setTimes(adjustedHostTime, sampleTime);
-
- if (volume() != 1.0)
- m_microphoneSampleBuffer->applyGain(volume());
-
- audioSamplesAvailable(MediaTime(sampleTime, m_microphoneProcFormat.sampleRate()), m_microphoneSampleBuffer->bufferList(), m_microphoneProcFormat, inNumberFrames);
- return noErr;
-}
-
-OSStatus CoreAudioSharedUnit::microphoneCallback(void *inRefCon, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData)
-{
- ASSERT(ioActionFlags);
- ASSERT(inTimeStamp);
- CoreAudioSharedUnit* dataSource = static_cast<CoreAudioSharedUnit*>(inRefCon);
- return dataSource->processMicrophoneSamples(*ioActionFlags, *inTimeStamp, inBusNumber, inNumberFrames, ioData);
-}
-
-void CoreAudioSharedUnit::cleanupAudioUnit()
-{
- if (m_ioUnitInitialized) {
- ASSERT(m_ioUnit);
- auto err = AudioUnitUninitialize(m_ioUnit);
- if (err)
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::cleanupAudioUnit(%p) AudioUnitUninitialize failed with error %d (%.4s)", this, (int)err, (char*)&err);
- m_ioUnitInitialized = false;
- }
-
- if (m_ioUnit) {
- PAL::AudioComponentInstanceDispose(m_ioUnit);
- m_ioUnit = nullptr;
- }
-
- m_microphoneSampleBuffer = nullptr;
-#if !LOG_DISABLED
- m_ioUnitName = emptyString();
-#endif
-}
-
-OSStatus CoreAudioSharedUnit::reconfigureAudioUnit()
-{
- ASSERT(isMainThread());
- OSStatus err;
- if (!hasAudioUnit())
- return 0;
-
- m_isReconfiguring = true;
- auto scope = makeScopeExit([this] { m_isReconfiguring = false; });
-
- if (m_ioUnitStarted) {
- err = PAL::AudioOutputUnitStop(m_ioUnit);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::reconfigureAudioUnit(%p) AudioOutputUnitStop failed with error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- }
-
- cleanupAudioUnit();
- err = setupAudioUnit();
- if (err)
- return err;
-
- if (m_ioUnitStarted) {
- err = PAL::AudioOutputUnitStart(m_ioUnit);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::reconfigureAudioUnit(%p) AudioOutputUnitStart failed with error %d (%.4s)", this, (int)err, (char*)&err);
- return err;
- }
- }
- return err;
-}
-
-OSStatus CoreAudioSharedUnit::startInternal()
-{
- ASSERT(isMainThread());
-
- setIsProducingMicrophoneSamples(true);
-
- OSStatus err;
- if (!m_ioUnit) {
- err = setupAudioUnit();
- if (err) {
- cleanupAudioUnit();
- ASSERT(!m_ioUnit);
- return err;
- }
- ASSERT(m_ioUnit);
- }
-
- unduck();
-
- {
- Locker locker { m_speakerSamplesProducerLock };
- if (m_speakerSamplesProducer)
- m_speakerSamplesProducer->captureUnitIsStarting();
- }
-
- err = PAL::AudioOutputUnitStart(m_ioUnit);
- if (err) {
- {
- Locker locker { m_speakerSamplesProducerLock };
- if (m_speakerSamplesProducer)
- m_speakerSamplesProducer->captureUnitHasStopped();
- }
-
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::start(%p) AudioOutputUnitStart failed with error %d (%.4s)", this, (int)err, (char*)&err);
- cleanupAudioUnit();
- ASSERT(!m_ioUnit);
- return err;
- }
-
- m_ioUnitStarted = true;
-
- m_verifyCapturingTimer.startRepeating(verifyCaptureInterval());
- m_microphoneProcsCalled = 0;
- m_microphoneProcsCalledLastTime = 0;
-
- return 0;
-}
-
-void CoreAudioSharedUnit::isProducingMicrophoneSamplesChanged()
-{
- if (!isProducingData())
- return;
- m_verifyCapturingTimer.startRepeating(verifyCaptureInterval());
-}
-
-void CoreAudioSharedUnit::validateOutputDevice(uint32_t currentOutputDeviceID)
-{
-#if PLATFORM(MAC)
- uint32_t currentDefaultOutputDeviceID = 0;
- if (auto err = defaultOutputDevice(¤tDefaultOutputDeviceID))
- return;
-
- if (!currentDefaultOutputDeviceID || currentOutputDeviceID == currentDefaultOutputDeviceID)
- return;
-
- reconfigure();
-#else
- UNUSED_PARAM(currentOutputDeviceID);
-#endif
-}
-
-void CoreAudioSharedUnit::verifyIsCapturing()
-{
- if (m_microphoneProcsCalledLastTime != m_microphoneProcsCalled) {
- m_microphoneProcsCalledLastTime = m_microphoneProcsCalled;
- return;
- }
-
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::verifyIsCapturing - no audio received in %d seconds, failing", static_cast<int>(m_verifyCapturingTimer.repeatInterval().value()));
- captureFailed();
-}
-
-void CoreAudioSharedUnit::stopInternal()
-{
- ASSERT(isMainThread());
-
- m_verifyCapturingTimer.stop();
-
- if (!m_ioUnit || !m_ioUnitStarted)
- return;
-
- auto err = PAL::AudioOutputUnitStop(m_ioUnit);
- if (err) {
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::stop(%p) AudioOutputUnitStop failed with error %d (%.4s)", this, (int)err, (char*)&err);
- return;
- }
- {
- Locker locker { m_speakerSamplesProducerLock };
- if (m_speakerSamplesProducer)
- m_speakerSamplesProducer->captureUnitHasStopped();
- }
-
- m_ioUnitStarted = false;
-}
-
-OSStatus CoreAudioSharedUnit::defaultInputDevice(uint32_t* deviceID)
-{
- ASSERT(m_ioUnit);
-
- UInt32 propertySize = sizeof(*deviceID);
- auto err = PAL::AudioUnitGetProperty(m_ioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, inputBus, deviceID, &propertySize);
- if (err)
- RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::defaultInputDevice(%p) unable to get default input device ID, error %d (%.4s)", this, (int)err, (char*)&err);
-
- return err;
-}
-
-OSStatus CoreAudioSharedUnit::defaultOutputDevice(uint32_t* deviceID)
-{
- OSErr err = -1;
-#if PLATFORM(MAC)
- AudioObjectPropertyAddress address = {
- kAudioHardwarePropertyDefaultOutputDevice,
- kAudioObjectPropertyScopeGlobal,
-#if HAVE(AUDIO_OBJECT_PROPERTY_ELEMENT_MAIN)
- kAudioObjectPropertyElementMain
-#else
- ALLOW_DEPRECATED_DECLARATIONS_BEGIN
- kAudioObjectPropertyElementMaster
- ALLOW_DEPRECATED_DECLARATIONS_END
-#endif
- };
-
- if (AudioObjectHasProperty(kAudioObjectSystemObject, &address)) {
- UInt32 propertySize = sizeof(AudioDeviceID);
- err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &address, 0, nullptr, &propertySize, deviceID);
- }
-#else
- UNUSED_PARAM(deviceID);
-#endif
- return err;
-}
-
-void CoreAudioSharedUnit::registerSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer& producer)
-{
- ASSERT(isMainThread());
-
- setIsRenderingAudio(true);
-
- CoreAudioSpeakerSamplesProducer* oldProducer;
- {
- Locker locker { m_speakerSamplesProducerLock };
- oldProducer = m_speakerSamplesProducer;
- m_speakerSamplesProducer = &producer;
- }
- if (oldProducer && oldProducer != &producer)
- oldProducer->captureUnitHasStopped();
-
- if (hasAudioUnit() && producer.format() != m_speakerProcFormat)
- reconfigure();
-}
-
-void CoreAudioSharedUnit::unregisterSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer& producer)
-{
- ASSERT(isMainThread());
-
- {
- Locker locker { m_speakerSamplesProducerLock };
- if (m_speakerSamplesProducer != &producer)
- return;
- m_speakerSamplesProducer = nullptr;
- }
-
- setIsRenderingAudio(false);
-}
-
static CaptureSourceOrError initializeCoreAudioCaptureSource(Ref<CoreAudioCaptureSource>&& source, const MediaConstraints* constraints)
{
if (constraints) {
Copied: trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.cpp (from rev 294101, trunk/Source/WebCore/platform/mediastream/mac/CoreAudioCaptureSource.cpp) (0 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.cpp (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.cpp 2022-05-12 13:56:19 UTC (rev 294102)
@@ -0,0 +1,652 @@
+/*
+ * Copyright (C) 2017-2022 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+ * OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "CoreAudioSharedUnit.h"
+
+#if ENABLE(MEDIA_STREAM)
+
+#include "AudioSampleBufferList.h"
+#include "AudioSession.h"
+#include "CoreAudioCaptureSource.h"
+#include "Logging.h"
+#include <AudioToolbox/AudioConverter.h>
+#include <AudioUnit/AudioUnit.h>
+#include <CoreMedia/CMSync.h>
+#include <mach/mach_time.h>
+#include <pal/avfoundation/MediaTimeAVFoundation.h>
+#include <pal/spi/cf/CoreAudioSPI.h>
+#include <sys/time.h>
+#include <wtf/Algorithms.h>
+#include <wtf/Lock.h>
+#include <wtf/MainThread.h>
+#include <wtf/NeverDestroyed.h>
+#include <wtf/Scope.h>
+
+#if PLATFORM(IOS_FAMILY)
+#include "AVAudioSessionCaptureDeviceManager.h"
+#endif
+
+#include <pal/cf/AudioToolboxSoftLink.h>
+#include <pal/cf/CoreMediaSoftLink.h>
+
+namespace WebCore {
+
+const UInt32 outputBus = 0;
+const UInt32 inputBus = 1;
+
+class CoreAudioSharedInternalUnit : public CoreAudioSharedUnit::InternalUnit {
+ WTF_MAKE_FAST_ALLOCATED;
+public:
+ static Expected<UniqueRef<InternalUnit>, OSStatus> create();
+ explicit CoreAudioSharedInternalUnit(AudioUnit);
+ ~CoreAudioSharedInternalUnit() final;
+
+private:
+ OSStatus initialize() final;
+ OSStatus uninitialize() final;
+ OSStatus start() final;
+ OSStatus stop() final;
+ OSStatus set(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, const void*, UInt32) final;
+ OSStatus get(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, void*, UInt32*) final;
+ OSStatus render(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*) final;
+ OSStatus defaultInputDevice(uint32_t*) final;
+ OSStatus defaultOutputDevice(uint32_t*) final;
+
+ AudioUnit m_ioUnit { nullptr };
+};
+
+Expected<UniqueRef<CoreAudioSharedUnit::InternalUnit>, OSStatus> CoreAudioSharedInternalUnit::create()
+{
+ AudioComponentDescription ioUnitDescription = { kAudioUnitType_Output, kAudioUnitSubType_VoiceProcessingIO, kAudioUnitManufacturer_Apple, 0, 0 };
+ AudioComponent ioComponent = PAL::AudioComponentFindNext(nullptr, &ioUnitDescription);
+ ASSERT(ioComponent);
+ if (!ioComponent) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedInternalUnit unable to find vpio unit component");
+ return makeUnexpected(-1);
+ }
+
+#if !LOG_DISABLED
+ CFStringRef name = nullptr;
+ PAL::AudioComponentCopyName(ioComponent, &name);
+ if (name) {
+ String ioUnitName = name;
+ CFRelease(name);
+ RELEASE_LOG(WebRTC, "CoreAudioSharedInternalUnit created \"%{private}s\" component", ioUnitName.utf8().data());
+ }
+#endif
+
+ AudioUnit ioUnit;
+ auto err = PAL::AudioComponentInstanceNew(ioComponent, &ioUnit);
+ if (err) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedInternalUnit unable to open vpio unit, error %d (%.4s)", (int)err, (char*)&err);
+ return makeUnexpected(err);
+ }
+
+ RELEASE_LOG(WebRTC, "Successfully created a CoreAudioSharedInternalUnit");
+
+ UniqueRef<CoreAudioSharedUnit::InternalUnit> result = makeUniqueRef<CoreAudioSharedInternalUnit>(ioUnit);
+ return result;
+}
+
+CoreAudioSharedInternalUnit::CoreAudioSharedInternalUnit(AudioUnit ioUnit)
+ : m_ioUnit(ioUnit)
+{
+}
+
+CoreAudioSharedInternalUnit::~CoreAudioSharedInternalUnit()
+{
+ PAL::AudioComponentInstanceDispose(m_ioUnit);
+}
+
+OSStatus CoreAudioSharedInternalUnit::initialize()
+{
+ return PAL::AudioUnitInitialize(m_ioUnit);
+}
+
+OSStatus CoreAudioSharedInternalUnit::uninitialize()
+{
+ return AudioUnitUninitialize(m_ioUnit);
+}
+
+OSStatus CoreAudioSharedInternalUnit::start()
+{
+ return PAL::AudioOutputUnitStart(m_ioUnit);
+}
+
+OSStatus CoreAudioSharedInternalUnit::stop()
+{
+ return PAL::AudioOutputUnitStop(m_ioUnit);
+}
+
+OSStatus CoreAudioSharedInternalUnit::set(AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element, const void* value, UInt32 size)
+{
+ return PAL::AudioUnitSetProperty(m_ioUnit, propertyID, scope, element, &value, size);
+}
+
+OSStatus CoreAudioSharedInternalUnit::get(AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element, void* value, UInt32* size)
+{
+ return PAL::AudioUnitGetProperty(m_ioUnit, propertyID, scope, element, value, size);
+}
+
+OSStatus CoreAudioSharedInternalUnit::render(AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* list)
+{
+ return AudioUnitRender(m_ioUnit, ioActionFlags, inTimeStamp, inOutputBusNumber, inNumberFrames, list);
+}
+
+OSStatus CoreAudioSharedInternalUnit::defaultInputDevice(uint32_t* deviceID)
+{
+#if PLATFORM(MAC)
+ UInt32 propertySize = sizeof(*deviceID);
+ auto err = get(kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, inputBus, deviceID, &propertySize);
+ RELEASE_LOG_ERROR_IF(err, WebRTC, "CoreAudioSharedInternalUnit unable to get default input device ID, error %d (%.4s)", (int)err, (char*)&err);
+ return err;
+#else
+ UNUSED_PARAM(deviceID);
+ return -1;
+#endif
+}
+
+OSStatus CoreAudioSharedInternalUnit::defaultOutputDevice(uint32_t* deviceID)
+{
+#if PLATFORM(MAC)
+ AudioObjectPropertyAddress address = {
+ kAudioHardwarePropertyDefaultOutputDevice,
+ kAudioObjectPropertyScopeGlobal,
+#if HAVE(AUDIO_OBJECT_PROPERTY_ELEMENT_MAIN)
+ kAudioObjectPropertyElementMain
+#else
+ ALLOW_DEPRECATED_DECLARATIONS_BEGIN
+ kAudioObjectPropertyElementMaster
+ ALLOW_DEPRECATED_DECLARATIONS_END
+#endif
+ };
+
+ if (AudioObjectHasProperty(kAudioObjectSystemObject, &address)) {
+ UInt32 propertySize = sizeof(AudioDeviceID);
+ return AudioObjectGetPropertyData(kAudioObjectSystemObject, &address, 0, nullptr, &propertySize, deviceID);
+ }
+#else
+ UNUSED_PARAM(deviceID);
+#endif
+ return -1;
+}
+
+CoreAudioSharedUnit& CoreAudioSharedUnit::unit()
+{
+ static NeverDestroyed<CoreAudioSharedUnit> singleton;
+ return singleton;
+}
+
+CoreAudioSharedUnit::CoreAudioSharedUnit()
+ : m_sampleRateCapabilities(8000, 96000)
+ , m_verifyCapturingTimer(*this, &CoreAudioSharedUnit::verifyIsCapturing)
+{
+}
+
+void CoreAudioSharedUnit::resetSampleRate()
+{
+ setSampleRate(m_getSampleRateCallback ? m_getSampleRateCallback() : AudioSession::sharedSession().sampleRate());
+}
+
+void CoreAudioSharedUnit::captureDeviceChanged()
+{
+#if PLATFORM(MAC)
+ reconfigureAudioUnit();
+#else
+ AVAudioSessionCaptureDeviceManager::singleton().setPreferredAudioSessionDeviceUID(persistentID());
+#endif
+}
+
+size_t CoreAudioSharedUnit::preferredIOBufferSize()
+{
+ return AudioSession::sharedSession().bufferSize();
+}
+
+OSStatus CoreAudioSharedUnit::setupAudioUnit()
+{
+ if (m_ioUnit)
+ return 0;
+
+ ASSERT(hasClients());
+
+ mach_timebase_info_data_t timebaseInfo;
+ mach_timebase_info(&timebaseInfo);
+ m_DTSConversionRatio = 1e-9 * static_cast<double>(timebaseInfo.numer) / static_cast<double>(timebaseInfo.denom);
+
+ auto result = m_creationCallback ? m_creationCallback() : CoreAudioSharedInternalUnit::create();
+ if (!result.has_value())
+ return result.error();
+
+ m_ioUnit = WTFMove(result.value()).moveToUniquePtr();
+ if (!enableEchoCancellation()) {
+ uint32_t param = 0;
+ if (auto err = m_ioUnit->set(kAUVoiceIOProperty_VoiceProcessingEnableAGC, kAudioUnitScope_Global, inputBus, ¶m, sizeof(param))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio automatic gain control, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ param = 1;
+ if (auto err = m_ioUnit->set(kAUVoiceIOProperty_BypassVoiceProcessing, kAudioUnitScope_Global, inputBus, ¶m, sizeof(param))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit echo cancellation, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ }
+
+#if PLATFORM(IOS_FAMILY)
+ uint32_t param = 1;
+ if (auto err = m_ioUnit->set(kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, inputBus, ¶m, sizeof(param))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to enable vpio unit input, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+#else
+ auto deviceID = captureDeviceID();
+ // FIXME: We probably want to make default input/output devices.
+ if (!deviceID) {
+ if (auto err = m_ioUnit->defaultInputDevice(&deviceID))
+ return err;
+ }
+
+ if (auto err = m_ioUnit->set(kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, inputBus, &deviceID, sizeof(deviceID))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit capture device ID %d, error %d (%.4s)", this, (int)deviceID, (int)err, (char*)&err);
+ return err;
+ }
+
+ uint32_t defaultOutputDeviceID;
+ auto err = m_ioUnit->defaultOutputDevice(&defaultOutputDeviceID);
+ if (!err) {
+ err = m_ioUnit->set(kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, outputBus, &defaultOutputDeviceID, sizeof(defaultOutputDeviceID));
+ RELEASE_LOG_ERROR_IF(err, WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) unable to set vpio unit output device ID %d, error %d (%.4s)", this, (int)defaultOutputDeviceID, (int)err, (char*)&err);
+ }
+ setOutputDeviceID(!err ? defaultOutputDeviceID : 0);
+#endif
+
+ // FIXME: Add support for different speaker/microphone sample rates.
+ int actualSampleRate = this->actualSampleRate();
+ if (auto err = configureMicrophoneProc(actualSampleRate))
+ return err;
+
+ if (auto err = configureSpeakerProc(actualSampleRate))
+ return err;
+
+ if (auto err = m_ioUnit->initialize()) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::setupAudioUnit(%p) AudioUnitInitialize() failed, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ m_ioUnitInitialized = true;
+
+ unduck();
+
+ return 0;
+}
+
+void CoreAudioSharedUnit::unduck()
+{
+ uint32_t outputDevice;
+ if (m_ioUnit && !m_ioUnit->defaultOutputDevice(&outputDevice))
+ AudioDeviceDuck(outputDevice, 1.0, nullptr, 0);
+}
+
+int CoreAudioSharedUnit::actualSampleRate() const
+{
+ Locker locker { m_speakerSamplesProducerLock };
+ return m_speakerSamplesProducer ? m_speakerSamplesProducer->format().streamDescription().mSampleRate : sampleRate();
+}
+
+OSStatus CoreAudioSharedUnit::configureMicrophoneProc(int sampleRate)
+{
+ ASSERT(isMainThread());
+
+ AURenderCallbackStruct callback = { microphoneCallback, this };
+ if (auto err = m_ioUnit->set(kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, inputBus, &callback, sizeof(callback))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to set vpio unit mic proc, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+
+ AudioStreamBasicDescription microphoneProcFormat = { };
+
+ UInt32 size = sizeof(microphoneProcFormat);
+ if (auto err = m_ioUnit->get(kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, inputBus, µphoneProcFormat, &size)) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to get output stream format, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+
+ microphoneProcFormat.mSampleRate = sampleRate;
+ if (auto err = m_ioUnit->set(kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, inputBus, µphoneProcFormat, size)) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureMicrophoneProc(%p) unable to set output stream format, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+
+ m_microphoneSampleBuffer = AudioSampleBufferList::create(microphoneProcFormat, preferredIOBufferSize() * 2);
+ m_microphoneProcFormat = microphoneProcFormat;
+
+ return noErr;
+}
+
+OSStatus CoreAudioSharedUnit::configureSpeakerProc(int sampleRate)
+{
+ ASSERT(isMainThread());
+
+ AURenderCallbackStruct callback = { speakerCallback, this };
+ if (auto err = m_ioUnit->set(kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, outputBus, &callback, sizeof(callback))) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to set vpio unit speaker proc, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+
+ AudioStreamBasicDescription speakerProcFormat;
+ UInt32 size = sizeof(speakerProcFormat);
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ if (m_speakerSamplesProducer) {
+ speakerProcFormat = m_speakerSamplesProducer->format().streamDescription();
+ ASSERT(speakerProcFormat.mSampleRate == sampleRate);
+ } else {
+ if (auto err = m_ioUnit->get(kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, outputBus, &speakerProcFormat, &size)) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to get input stream format, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ }
+ }
+ speakerProcFormat.mSampleRate = sampleRate;
+
+ if (auto err = m_ioUnit->set(kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, outputBus, &speakerProcFormat, size)) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::configureSpeakerProc(%p) unable to get input stream format, error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+
+ m_speakerProcFormat = speakerProcFormat;
+
+ return noErr;
+}
+
+#if !LOG_DISABLED
+void CoreAudioSharedUnit::checkTimestamps(const AudioTimeStamp& timeStamp, uint64_t sampleTime, double hostTime)
+{
+ if (!timeStamp.mSampleTime || sampleTime == m_latestMicTimeStamp || !hostTime)
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::checkTimestamps: unusual timestamps, sample time = %lld, previous sample time = %lld, hostTime %f", sampleTime, m_latestMicTimeStamp, hostTime);
+}
+#endif
+
+OSStatus CoreAudioSharedUnit::provideSpeakerData(AudioUnitRenderActionFlags& flags, const AudioTimeStamp& timeStamp, UInt32 /*inBusNumber*/, UInt32 inNumberFrames, AudioBufferList& ioData)
+{
+ if (m_isReconfiguring || !m_speakerSamplesProducerLock.tryLock()) {
+ AudioSampleBufferList::zeroABL(ioData, static_cast<size_t>(inNumberFrames * m_speakerProcFormat.bytesPerFrame()));
+ flags = kAudioUnitRenderAction_OutputIsSilence;
+ return noErr;
+ }
+
+ Locker locker { AdoptLock, m_speakerSamplesProducerLock };
+ if (!m_speakerSamplesProducer) {
+ AudioSampleBufferList::zeroABL(ioData, static_cast<size_t>(inNumberFrames * m_speakerProcFormat.bytesPerFrame()));
+ flags = kAudioUnitRenderAction_OutputIsSilence;
+ return noErr;
+ }
+ return m_speakerSamplesProducer->produceSpeakerSamples(inNumberFrames, ioData, timeStamp.mSampleTime, timeStamp.mHostTime, flags);
+}
+
+OSStatus CoreAudioSharedUnit::speakerCallback(void *inRefCon, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData)
+{
+ ASSERT(ioActionFlags);
+ ASSERT(inTimeStamp);
+ auto dataSource = static_cast<CoreAudioSharedUnit*>(inRefCon);
+ return dataSource->provideSpeakerData(*ioActionFlags, *inTimeStamp, inBusNumber, inNumberFrames, *ioData);
+}
+
+OSStatus CoreAudioSharedUnit::processMicrophoneSamples(AudioUnitRenderActionFlags& ioActionFlags, const AudioTimeStamp& timeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* /*ioData*/)
+{
+ ++m_microphoneProcsCalled;
+
+ if (m_isReconfiguring)
+ return false;
+
+ // Pull through the vpio unit to our mic buffer.
+ m_microphoneSampleBuffer->reset();
+ AudioBufferList& bufferList = m_microphoneSampleBuffer->bufferList();
+ if (auto err = m_ioUnit->render(&ioActionFlags, &timeStamp, inBusNumber, inNumberFrames, &bufferList)) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::processMicrophoneSamples(%p) AudioUnitRender failed with error %d (%.4s), bufferList size %d, inNumberFrames %d ", this, (int)err, (char*)&err, (int)bufferList.mBuffers[0].mDataByteSize, (int)inNumberFrames);
+ if (err == kAudio_ParamError) {
+ // Our buffer might be too small, the preferred buffer size or sample rate might have changed.
+ callOnMainThread([] {
+ CoreAudioSharedUnit::singleton().reconfigure();
+ });
+ }
+ // We return early so that if this error happens, we do not increment m_microphoneProcsCalled and fail the capture once timer kicks in.
+ return err;
+ }
+
+ if (!isProducingMicrophoneSamples())
+ return noErr;
+
+ double adjustedHostTime = m_DTSConversionRatio * timeStamp.mHostTime;
+ uint64_t sampleTime = timeStamp.mSampleTime;
+#if !LOG_DISABLED
+ checkTimestamps(timeStamp, sampleTime, adjustedHostTime);
+#endif
+ m_latestMicTimeStamp = sampleTime;
+ m_microphoneSampleBuffer->setTimes(adjustedHostTime, sampleTime);
+
+ if (volume() != 1.0)
+ m_microphoneSampleBuffer->applyGain(volume());
+
+ audioSamplesAvailable(MediaTime(sampleTime, m_microphoneProcFormat.sampleRate()), m_microphoneSampleBuffer->bufferList(), m_microphoneProcFormat, inNumberFrames);
+ return noErr;
+}
+
+OSStatus CoreAudioSharedUnit::microphoneCallback(void *inRefCon, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData)
+{
+ ASSERT(ioActionFlags);
+ ASSERT(inTimeStamp);
+ CoreAudioSharedUnit* dataSource = static_cast<CoreAudioSharedUnit*>(inRefCon);
+ return dataSource->processMicrophoneSamples(*ioActionFlags, *inTimeStamp, inBusNumber, inNumberFrames, ioData);
+}
+
+void CoreAudioSharedUnit::cleanupAudioUnit()
+{
+ if (m_ioUnitInitialized) {
+ ASSERT(m_ioUnit);
+ if (auto err = m_ioUnit->uninitialize())
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::cleanupAudioUnit(%p) AudioUnitUninitialize failed with error %d (%.4s)", this, (int)err, (char*)&err);
+ m_ioUnitInitialized = false;
+ }
+
+ m_ioUnit = nullptr;
+
+ m_microphoneSampleBuffer = nullptr;
+}
+
+void CoreAudioSharedUnit::delaySamples(Seconds seconds)
+{
+ if (m_ioUnit)
+ m_ioUnit->delaySamples(seconds);
+}
+
+OSStatus CoreAudioSharedUnit::reconfigureAudioUnit()
+{
+ ASSERT(isMainThread());
+ if (!hasAudioUnit())
+ return 0;
+
+ m_isReconfiguring = true;
+ auto scope = makeScopeExit([this] { m_isReconfiguring = false; });
+
+ if (m_ioUnitStarted) {
+ if (auto err = m_ioUnit->stop()) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::reconfigureAudioUnit(%p) AudioOutputUnitStop failed with error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ }
+
+ cleanupAudioUnit();
+ if (auto err = setupAudioUnit())
+ return err;
+
+ if (m_ioUnitStarted) {
+ if (auto err = m_ioUnit->start()) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::reconfigureAudioUnit(%p) AudioOutputUnitStart failed with error %d (%.4s)", this, (int)err, (char*)&err);
+ return err;
+ }
+ }
+ return noErr;
+}
+
+OSStatus CoreAudioSharedUnit::startInternal()
+{
+ ASSERT(isMainThread());
+
+ setIsProducingMicrophoneSamples(true);
+
+ if (!m_ioUnit) {
+ if (auto err = setupAudioUnit()) {
+ cleanupAudioUnit();
+ ASSERT(!m_ioUnit);
+ return err;
+ }
+ ASSERT(m_ioUnit);
+ }
+
+ unduck();
+
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ if (m_speakerSamplesProducer)
+ m_speakerSamplesProducer->captureUnitIsStarting();
+ }
+
+ if (auto err = m_ioUnit->start()) {
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ if (m_speakerSamplesProducer)
+ m_speakerSamplesProducer->captureUnitHasStopped();
+ }
+
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::start(%p) AudioOutputUnitStart failed with error %d (%.4s)", this, (int)err, (char*)&err);
+ cleanupAudioUnit();
+ ASSERT(!m_ioUnit);
+ return err;
+ }
+
+ m_ioUnitStarted = true;
+
+ m_verifyCapturingTimer.startRepeating(verifyCaptureInterval());
+ m_microphoneProcsCalled = 0;
+ m_microphoneProcsCalledLastTime = 0;
+
+ return noErr;
+}
+
+void CoreAudioSharedUnit::isProducingMicrophoneSamplesChanged()
+{
+ if (!isProducingData())
+ return;
+ m_verifyCapturingTimer.startRepeating(verifyCaptureInterval());
+}
+
+void CoreAudioSharedUnit::validateOutputDevice(uint32_t currentOutputDeviceID)
+{
+#if PLATFORM(MAC)
+ if (!m_ioUnit)
+ return;
+
+ uint32_t currentDefaultOutputDeviceID = 0;
+ if (auto err = m_ioUnit->defaultOutputDevice(¤tDefaultOutputDeviceID))
+ return;
+
+ if (!currentDefaultOutputDeviceID || currentOutputDeviceID == currentDefaultOutputDeviceID)
+ return;
+
+ reconfigure();
+#else
+ UNUSED_PARAM(currentOutputDeviceID);
+#endif
+}
+
+void CoreAudioSharedUnit::verifyIsCapturing()
+{
+ if (m_microphoneProcsCalledLastTime != m_microphoneProcsCalled) {
+ m_microphoneProcsCalledLastTime = m_microphoneProcsCalled;
+ return;
+ }
+
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::verifyIsCapturing - no audio received in %d seconds, failing", static_cast<int>(m_verifyCapturingTimer.repeatInterval().value()));
+ captureFailed();
+}
+
+void CoreAudioSharedUnit::stopInternal()
+{
+ ASSERT(isMainThread());
+
+ m_verifyCapturingTimer.stop();
+
+ if (!m_ioUnit || !m_ioUnitStarted)
+ return;
+
+ if (auto err = m_ioUnit->stop()) {
+ RELEASE_LOG_ERROR(WebRTC, "CoreAudioSharedUnit::stop(%p) AudioOutputUnitStop failed with error %d (%.4s)", this, (int)err, (char*)&err);
+ return;
+ }
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ if (m_speakerSamplesProducer)
+ m_speakerSamplesProducer->captureUnitHasStopped();
+ }
+
+ m_ioUnitStarted = false;
+}
+
+void CoreAudioSharedUnit::registerSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer& producer)
+{
+ ASSERT(isMainThread());
+
+ setIsRenderingAudio(true);
+
+ CoreAudioSpeakerSamplesProducer* oldProducer;
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ oldProducer = m_speakerSamplesProducer;
+ m_speakerSamplesProducer = &producer;
+ }
+ if (oldProducer && oldProducer != &producer)
+ oldProducer->captureUnitHasStopped();
+
+ if (hasAudioUnit() && producer.format() != m_speakerProcFormat)
+ reconfigure();
+}
+
+void CoreAudioSharedUnit::unregisterSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer& producer)
+{
+ ASSERT(isMainThread());
+
+ {
+ Locker locker { m_speakerSamplesProducerLock };
+ if (m_speakerSamplesProducer != &producer)
+ return;
+ m_speakerSamplesProducer = nullptr;
+ }
+
+ setIsRenderingAudio(false);
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(MEDIA_STREAM)
Added: trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.h (0 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/mac/CoreAudioSharedUnit.h 2022-05-12 13:56:19 UTC (rev 294102)
@@ -0,0 +1,155 @@
+/*
+ * Copyright (C) 2017-2022 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+ * OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if ENABLE(MEDIA_STREAM)
+
+#include "AudioSampleDataSource.h"
+#include "BaseAudioSharedUnit.h"
+#include "CAAudioStreamDescription.h"
+#include "Timer.h"
+
+typedef UInt32 AudioUnitPropertyID;
+typedef UInt32 AudioUnitScope;
+typedef UInt32 AudioUnitElement;
+typedef UInt32 AudioUnitRenderActionFlags;
+
+struct AudioBufferList;
+struct AudioTimeStamp;
+
+namespace WebCore {
+
+class CoreAudioSpeakerSamplesProducer;
+
+class CoreAudioSharedUnit final : public BaseAudioSharedUnit {
+public:
+ class InternalUnit {
+ public:
+ virtual ~InternalUnit() = default;
+ virtual OSStatus initialize() = 0;
+ virtual OSStatus uninitialize() = 0;
+ virtual OSStatus start() = 0;
+ virtual OSStatus stop() = 0;
+ virtual OSStatus set(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, const void*, UInt32) = 0;
+ virtual OSStatus get(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, void*, UInt32*) = 0;
+ virtual OSStatus render(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*) = 0;
+ virtual OSStatus defaultInputDevice(uint32_t*) = 0;
+ virtual OSStatus defaultOutputDevice(uint32_t*) = 0;
+ virtual void delaySamples(Seconds) { }
+ };
+
+ static CoreAudioSharedUnit& unit();
+ static BaseAudioSharedUnit& singleton() { return unit(); }
+ CoreAudioSharedUnit();
+
+ using CreationCallback = Function<Expected<UniqueRef<InternalUnit>, OSStatus>()>;
+ void setInternalUnitCreationCallback(CreationCallback&& callback) { m_creationCallback = WTFMove(callback); }
+ using GetSampleRateCallback = Function<int()>;
+ void setInternalUnitGetSampleRateCallback(GetSampleRateCallback&& callback) { m_getSampleRateCallback = WTFMove(callback); }
+
+ void registerSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer&);
+ void unregisterSpeakerSamplesProducer(CoreAudioSpeakerSamplesProducer&);
+ bool isRunning() const { return m_ioUnitStarted; }
+ void setSampleRateRange(CapabilityValueOrRange range) { m_sampleRateCapabilities = range; }
+
+private:
+ static size_t preferredIOBufferSize();
+
+ CapabilityValueOrRange sampleRateCapacities() const final { return m_sampleRateCapabilities; }
+ const CAAudioStreamDescription& microphoneFormat() const { return m_microphoneProcFormat; }
+
+ bool hasAudioUnit() const final { return !!m_ioUnit; }
+ void captureDeviceChanged() final;
+ OSStatus reconfigureAudioUnit() final;
+ void delaySamples(Seconds) final;
+
+ OSStatus setupAudioUnit();
+ void cleanupAudioUnit() final;
+
+ OSStatus startInternal() final;
+ void stopInternal() final;
+ bool isProducingData() const final { return m_ioUnitStarted; }
+ void isProducingMicrophoneSamplesChanged() final;
+ void validateOutputDevice(uint32_t deviceID) final;
+ int actualSampleRate() const final;
+ void resetSampleRate();
+
+ OSStatus configureSpeakerProc(int sampleRate);
+ OSStatus configureMicrophoneProc(int sampleRate);
+
+ static OSStatus microphoneCallback(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*);
+ OSStatus processMicrophoneSamples(AudioUnitRenderActionFlags&, const AudioTimeStamp&, UInt32, UInt32, AudioBufferList*);
+
+ static OSStatus speakerCallback(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*);
+ OSStatus provideSpeakerData(AudioUnitRenderActionFlags&, const AudioTimeStamp&, UInt32, UInt32, AudioBufferList&);
+
+ void unduck();
+
+ void verifyIsCapturing();
+
+ Seconds verifyCaptureInterval() { return isProducingMicrophoneSamples() ? 10_s : 2_s; }
+
+ CreationCallback m_creationCallback;
+ GetSampleRateCallback m_getSampleRateCallback;
+ std::unique_ptr<InternalUnit> m_ioUnit;
+
+ // Only read/modified from the IO thread.
+ Vector<Ref<AudioSampleDataSource>> m_activeSources;
+
+ CAAudioStreamDescription m_microphoneProcFormat;
+ RefPtr<AudioSampleBufferList> m_microphoneSampleBuffer;
+ uint64_t m_latestMicTimeStamp { 0 };
+
+ CAAudioStreamDescription m_speakerProcFormat;
+
+ double m_DTSConversionRatio { 0 };
+
+ bool m_ioUnitInitialized { false };
+ bool m_ioUnitStarted { false };
+
+ mutable std::unique_ptr<RealtimeMediaSourceCapabilities> m_capabilities;
+ mutable std::optional<RealtimeMediaSourceSettings> m_currentSettings;
+
+#if !LOG_DISABLED
+ void checkTimestamps(const AudioTimeStamp&, uint64_t, double);
+
+ String m_ioUnitName;
+#endif
+
+ CapabilityValueOrRange m_sampleRateCapabilities;
+
+ uint64_t m_microphoneProcsCalled { 0 };
+ uint64_t m_microphoneProcsCalledLastTime { 0 };
+ Timer m_verifyCapturingTimer;
+
+ bool m_isReconfiguring { false };
+ mutable Lock m_speakerSamplesProducerLock;
+ CoreAudioSpeakerSamplesProducer* m_speakerSamplesProducer WTF_GUARDED_BY_LOCK(m_speakerSamplesProducerLock) { nullptr };
+};
+
+} // namespace WebCore
+
+#endif // ENABLE(MEDIA_STREAM)
Modified: trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.h (294101 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.h 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.h 2022-05-12 13:56:19 UTC (rev 294102)
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2016-2019 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2022 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -27,73 +27,16 @@
#if ENABLE(MEDIA_STREAM)
-#include "BaseAudioSharedUnit.h"
-#include <CoreAudio/CoreAudioTypes.h>
-#include <wtf/RunLoop.h>
-#include <wtf/Vector.h>
-#include <wtf/WorkQueue.h>
+#include "CoreAudioSharedUnit.h"
-OBJC_CLASS AVAudioPCMBuffer;
-typedef struct OpaqueCMClock* CMClockRef;
-typedef const struct opaqueCMFormatDescription* CMFormatDescriptionRef;
-
namespace WebCore {
-class WebAudioBufferList;
-class WebAudioSourceProviderCocoa;
+namespace MockAudioSharedUnit {
-class MockAudioSharedUnit final : public BaseAudioSharedUnit {
-public:
- WEBCORE_EXPORT static MockAudioSharedUnit& singleton();
- MockAudioSharedUnit();
+CoreAudioSharedUnit& singleton();
- void setDeviceID(const String& deviceID) { setCaptureDevice(String { deviceID }, 0); }
+}
-private:
- bool hasAudioUnit() const final;
- void captureDeviceChanged() final;
- OSStatus reconfigureAudioUnit() final;
- void resetSampleRate() final;
-
- void cleanupAudioUnit() final;
- OSStatus startInternal() final;
- void stopInternal() final;
- bool isProducingData() const final;
-
- void delaySamples(Seconds) final;
-
- void start();
- CapabilityValueOrRange sampleRateCapacities() const final { return CapabilityValueOrRange(44100, 48000); }
-
- void tick();
-
- void render(MonotonicTime);
- void emitSampleBuffers(uint32_t frameCount);
- void reconfigure();
-
- static Seconds renderInterval() { return 20_ms; }
-
- std::unique_ptr<WebAudioBufferList> m_audioBufferList;
-
- uint32_t m_maximiumFrameCount;
- uint64_t m_samplesEmitted { 0 };
- uint64_t m_samplesRendered { 0 };
-
- RetainPtr<CMFormatDescriptionRef> m_formatDescription;
- AudioStreamBasicDescription m_streamFormat;
-
- Vector<float> m_bipBopBuffer;
- bool m_hasAudioUnit { false };
- bool m_isProducingData { false };
-
- RunLoop::Timer<MockAudioSharedUnit> m_timer;
- MonotonicTime m_lastRenderTime { MonotonicTime::nan() };
- MonotonicTime m_delayUntil;
-
- Ref<WorkQueue> m_workQueue;
- unsigned m_channelCount { 2 };
-};
-
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM)
Modified: trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.mm (294101 => 294102)
--- trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.mm 2022-05-12 13:49:11 UTC (rev 294101)
+++ trunk/Source/WebCore/platform/mediastream/mac/MockAudioSharedUnit.mm 2022-05-12 13:56:19 UTC (rev 294102)
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2016-2019 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2022 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -43,6 +43,9 @@
#import <AVFoundation/AVAudioBuffer.h>
#import <AudioToolbox/AudioConverter.h>
#import <CoreAudio/CoreAudioTypes.h>
+#include <wtf/RunLoop.h>
+#include <wtf/Vector.h>
+#include <wtf/WorkQueue.h>
#import <pal/cf/AudioToolboxSoftLink.h>
#import <pal/cf/CoreMediaSoftLink.h>
@@ -90,67 +93,111 @@
if (!device)
return { "No mock microphone device"_s };
- MockAudioSharedUnit::singleton().setDeviceID(deviceID);
+ MockAudioSharedUnit::singleton().setCaptureDevice(String { deviceID }, 0);
return CoreAudioCaptureSource::createForTesting(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalt), constraints, MockAudioSharedUnit::singleton(), pageIdentifier);
}
-MockAudioSharedUnit& MockAudioSharedUnit::singleton()
-{
- static NeverDestroyed<MockAudioSharedUnit> singleton;
- return singleton;
-}
+class MockAudioSharedInternalUnit : public CoreAudioSharedUnit::InternalUnit {
+ WTF_MAKE_FAST_ALLOCATED;
+public:
+ MockAudioSharedInternalUnit();
+ ~MockAudioSharedInternalUnit();
-MockAudioSharedUnit::MockAudioSharedUnit()
- : m_timer(RunLoop::current(), this, &MockAudioSharedUnit::start)
- , m_workQueue(WorkQueue::create("MockAudioSharedUnit Capture Queue", WorkQueue::QOS::UserInteractive))
-{
-}
+private:
+ OSStatus initialize() final;
+ OSStatus uninitialize() final { return 0; }
+ OSStatus start() final;
+ OSStatus stop() final;
+ OSStatus set(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, const void*, UInt32) final;
+ OSStatus get(AudioUnitPropertyID, AudioUnitScope, AudioUnitElement, void*, UInt32*) final;
+ OSStatus render(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32, AudioBufferList*) final;
+ OSStatus defaultInputDevice(uint32_t*) final;
+ OSStatus defaultOutputDevice(uint32_t*) final;
+ void delaySamples(Seconds) final;
-void MockAudioSharedUnit::resetSampleRate()
+ int sampleRate() const { return m_streamFormat.mSampleRate; }
+ void tick();
+
+ void generateSampleBuffers(MonotonicTime);
+ void emitSampleBuffers(uint32_t frameCount);
+ void reconfigure();
+
+ static Seconds renderInterval() { return 20_ms; }
+
+ std::unique_ptr<WebAudioBufferList> m_audioBufferList;
+
+ uint32_t m_maximiumFrameCount;
+ uint64_t m_samplesEmitted { 0 };
+ uint64_t m_samplesRendered { 0 };
+
+ RetainPtr<CMFormatDescriptionRef> m_formatDescription;
+ AudioStreamBasicDescription m_outputStreamFormat;
+ AudioStreamBasicDescription m_streamFormat;
+
+ Vector<float> m_bipBopBuffer;
+ bool m_hasAudioUnit { false };
+ bool m_isProducingData { false };
+ bool m_enableEchoCancellation { true };
+ RunLoop::Timer<MockAudioSharedInternalUnit> m_timer;
+ MonotonicTime m_lastRenderTime { MonotonicTime::nan() };
+ MonotonicTime m_delayUntil;
+
+ Ref<WorkQueue> m_workQueue;
+ unsigned m_channelCount { 2 };
+
+ AURenderCallbackStruct m_microphoneCallback;
+ AURenderCallbackStruct m_speakerCallback;
+};
+
+CoreAudioSharedUnit& MockAudioSharedUnit::singleton()
{
- if (auto device = MockRealtimeMediaSourceCenter::mockDeviceWithPersistentID(persistentID()))
- setSampleRate(std::get<MockMicrophoneProperties>(device->properties).defaultSampleRate);
+ static NeverDestroyed<CoreAudioSharedUnit> unit;
+ static std::once_flag onceFlag;
+ std::call_once(onceFlag, [&] () {
+ unit->setSampleRateRange(CapabilityValueOrRange(44100, 48000));
+ unit->setInternalUnitCreationCallback([] {
+ UniqueRef<CoreAudioSharedUnit::InternalUnit> result = makeUniqueRef<MockAudioSharedInternalUnit>();
+ return result;
+ });
+ unit->setInternalUnitGetSampleRateCallback([] { return 44100; });
+ });
+ return unit;
}
-bool MockAudioSharedUnit::hasAudioUnit() const
+static AudioStreamBasicDescription createAudioFormat(Float64 sampleRate, UInt32 channelCount)
{
- return m_hasAudioUnit;
+ AudioStreamBasicDescription format;
+ const int bytesPerFloat = sizeof(Float32);
+ const int bitsPerByte = 8;
+ const bool isFloat = true;
+ const bool isBigEndian = false;
+ const bool isNonInterleaved = true;
+ FillOutASBDForLPCM(format, sampleRate, channelCount, bitsPerByte * bytesPerFloat, bitsPerByte * bytesPerFloat, isFloat, isBigEndian, isNonInterleaved);
+ return format;
}
-void MockAudioSharedUnit::captureDeviceChanged()
+MockAudioSharedInternalUnit::MockAudioSharedInternalUnit()
+ : m_timer(RunLoop::current(), [this] { this->start(); })
+ , m_workQueue(WorkQueue::create("MockAudioSharedInternalUnit Capture Queue", WorkQueue::QOS::UserInteractive))
{
- reconfigureAudioUnit();
+ m_streamFormat = m_outputStreamFormat = createAudioFormat(44100, 2);
}
-OSStatus MockAudioSharedUnit::reconfigureAudioUnit()
+MockAudioSharedInternalUnit::~MockAudioSharedInternalUnit()
{
- if (!hasAudioUnit())
- return 0;
-
- m_lastRenderTime = MonotonicTime::nan();
- m_workQueue->dispatch([this] {
- reconfigure();
- callOnMainThread([this] {
- startInternal();
- });
- });
- return 0;
+ ASSERT(!m_isProducingData);
}
-void MockAudioSharedUnit::cleanupAudioUnit()
+OSStatus MockAudioSharedInternalUnit::initialize()
{
- m_hasAudioUnit = false;
- m_isProducingData = false;
- m_lastRenderTime = MonotonicTime::nan();
-}
+ ASSERT(m_outputStreamFormat.mSampleRate == m_streamFormat.mSampleRate);
+ if (m_outputStreamFormat.mSampleRate != m_streamFormat.mSampleRate)
+ return -1;
-OSStatus MockAudioSharedUnit::startInternal()
-{
- start();
return 0;
}
-void MockAudioSharedUnit::start()
+OSStatus MockAudioSharedInternalUnit::start()
{
if (!m_hasAudioUnit)
m_hasAudioUnit = true;
@@ -158,30 +205,29 @@
m_lastRenderTime = MonotonicTime::now();
m_isProducingData = true;
m_workQueue->dispatch([this, renderTime = m_lastRenderTime] {
- render(renderTime);
+ generateSampleBuffers(renderTime);
});
+ return 0;
}
-void MockAudioSharedUnit::stopInternal()
+OSStatus MockAudioSharedInternalUnit::stop()
{
m_isProducingData = false;
- if (!m_hasAudioUnit)
- return;
- m_lastRenderTime = MonotonicTime::nan();
-}
+ if (m_hasAudioUnit)
+ m_lastRenderTime = MonotonicTime::nan();
-bool MockAudioSharedUnit::isProducingData() const
-{
- return m_isProducingData;
+ m_workQueue->dispatchSync([] { });
+
+ return 0;
}
-void MockAudioSharedUnit::delaySamples(Seconds delta)
+void MockAudioSharedInternalUnit::delaySamples(Seconds delta)
{
- stopInternal();
+ stop();
m_timer.startOneShot(delta);
}
-void MockAudioSharedUnit::reconfigure()
+void MockAudioSharedInternalUnit::reconfigure()
{
ASSERT(!isMainThread());
@@ -191,14 +237,6 @@
m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * rate * 2);
ASSERT(m_maximiumFrameCount);
- const int bytesPerFloat = sizeof(Float32);
- const int bitsPerByte = 8;
- const int channelCount = m_channelCount;
- const bool isFloat = true;
- const bool isBigEndian = false;
- const bool isNonInterleaved = true;
- FillOutASBDForLPCM(m_streamFormat, rate, channelCount, bitsPerByte * bytesPerFloat, bitsPerByte * bytesPerFloat, isFloat, isBigEndian, isNonInterleaved);
-
m_audioBufferList = makeUnique<WebAudioBufferList>(m_streamFormat, m_maximiumFrameCount);
CMFormatDescriptionRef formatDescription;
@@ -215,25 +253,38 @@
addHum(BipBopVolume, BipFrequency, rate, 0, m_bipBopBuffer.data() + bipStart, bipBopSampleCount);
addHum(BipBopVolume, BopFrequency, rate, 0, m_bipBopBuffer.data() + bopStart, bipBopSampleCount);
- if (!enableEchoCancellation())
+ if (!m_enableEchoCancellation)
addHum(NoiseVolume, NoiseFrequency, rate, 0, m_bipBopBuffer.data(), sampleCount);
}
-void MockAudioSharedUnit::emitSampleBuffers(uint32_t frameCount)
+void MockAudioSharedInternalUnit::emitSampleBuffers(uint32_t frameCount)
{
ASSERT(!isMainThread());
ASSERT(m_formatDescription);
CMTime startTime = PAL::CMTimeMake(m_samplesEmitted, sampleRate());
+ auto sampleTime = PAL::CMTimeGetSeconds(startTime);
m_samplesEmitted += frameCount;
- audioSamplesAvailable(PAL::toMediaTime(startTime), *m_audioBufferList, CAAudioStreamDescription(m_streamFormat), frameCount);
+ auto* bufferList = m_audioBufferList->list();
+ AudioUnitRenderActionFlags ioActionFlags = 0;
+
+ AudioTimeStamp timeStamp;
+ memset(&timeStamp, 0, sizeof(AudioTimeStamp));
+ timeStamp.mSampleTime = sampleTime;
+ timeStamp.mHostTime = static_cast<UInt64>(sampleTime);
+ if (m_microphoneCallback.inputProc)
+ m_microphoneCallback.inputProc(m_microphoneCallback.inputProcRefCon, &ioActionFlags, &timeStamp, 1, frameCount, bufferList);
+
+ ioActionFlags = 0;
+ if (m_speakerCallback.inputProc)
+ m_speakerCallback.inputProc(m_speakerCallback.inputProcRefCon, &ioActionFlags, &timeStamp, 1, frameCount, bufferList);
}
-void MockAudioSharedUnit::render(MonotonicTime renderTime)
+void MockAudioSharedInternalUnit::generateSampleBuffers(MonotonicTime renderTime)
{
ASSERT(!isMainThread());
- if (!isProducingData())
+ if (!m_isProducingData)
return;
auto delta = renderInterval();
@@ -245,7 +296,7 @@
nextRenderDelay = 0_s;
}
m_workQueue->dispatchAfter(nextRenderDelay, [this, nextRenderTime] {
- render(nextRenderTime);
+ generateSampleBuffers(nextRenderTime);
});
if (!m_audioBufferList || !m_bipBopBuffer.size())
@@ -270,6 +321,80 @@
}
}
+OSStatus MockAudioSharedInternalUnit::render(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32, UInt32 frameCount, AudioBufferList* buffer)
+{
+ auto* sourceBuffer = m_audioBufferList->list();
+ if (buffer->mNumberBuffers > sourceBuffer->mNumberBuffers)
+ return kAudio_ParamError;
+
+ for (uint32_t i = 0; i < buffer->mNumberBuffers; i++) {
+ auto* source = static_cast<uint8_t*>(sourceBuffer->mBuffers[i].mData);
+ auto* destination = static_cast<uint8_t*>(buffer->mBuffers[i].mData);
+ memcpy(destination, source, frameCount * m_streamFormat.mBytesPerPacket);
+ }
+
+ return 0;
+}
+
+OSStatus MockAudioSharedInternalUnit::set(AudioUnitPropertyID property, AudioUnitScope scope, AudioUnitElement, const void* value, UInt32)
+{
+ if (property == kAudioUnitProperty_StreamFormat) {
+ auto& typedValue = *static_cast<const AudioStreamBasicDescription*>(value);
+ if (scope == kAudioUnitScope_Input)
+ m_streamFormat = typedValue;
+ else
+ m_outputStreamFormat = typedValue;
+ return 0;
+ }
+ if (property == kAUVoiceIOProperty_VoiceProcessingEnableAGC) {
+ m_enableEchoCancellation = !!*static_cast<const uint32_t*>(value);
+ return 0;
+ }
+ if (property == kAudioOutputUnitProperty_SetInputCallback) {
+ m_microphoneCallback = *static_cast<const AURenderCallbackStruct*>(value);
+ return 0;
+ }
+ if (property == kAudioUnitProperty_SetRenderCallback) {
+ m_speakerCallback = *static_cast<const AURenderCallbackStruct*>(value);
+ return 0;
+ }
+ if (property == kAudioOutputUnitProperty_CurrentDevice) {
+ ASSERT(!*static_cast<const uint32_t*>(value));
+ if (auto device = MockRealtimeMediaSourceCenter::mockDeviceWithPersistentID(MockAudioSharedUnit::singleton().persistentIDForTesting()))
+ m_streamFormat.mSampleRate = m_outputStreamFormat.mSampleRate = std::get<MockMicrophoneProperties>(device->properties).defaultSampleRate;
+ return 0;
+ }
+
+ return 0;
+}
+
+OSStatus MockAudioSharedInternalUnit::get(AudioUnitPropertyID property, AudioUnitScope scope, AudioUnitElement, void* value, UInt32* valueSize)
+{
+ if (property == kAudioUnitProperty_StreamFormat) {
+ auto& typedValue = *static_cast<AudioStreamBasicDescription*>(value);
+ if (scope == kAudioUnitScope_Input)
+ typedValue = m_streamFormat;
+ else
+ typedValue = m_outputStreamFormat;
+ *valueSize = sizeof(AudioStreamBasicDescription);
+ return 0;
+ }
+
+ return 0;
+}
+
+OSStatus MockAudioSharedInternalUnit::defaultInputDevice(uint32_t* device)
+{
+ *device = 0;
+ return 0;
+}
+
+OSStatus MockAudioSharedInternalUnit::defaultOutputDevice(uint32_t* device)
+{
+ *device = 0;
+ return 0;
+}
+
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM)
