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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm | 194 |
1 files changed, 194 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm new file mode 100644 index 0000000000..156d6326a4 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm @@ -0,0 +1,194 @@ +/* + * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#import <AudioUnit/AudioUnit.h> +#import <Foundation/Foundation.h> + +#import "objc_audio_device.h" +#import "objc_audio_device_delegate.h" + +#include "api/make_ref_counted.h" +#include "api/ref_counted_base.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/thread.h" + +namespace { + +constexpr double kPreferredInputSampleRate = 48000.0; +constexpr double kPreferredOutputSampleRate = 48000.0; + +// WebRTC processes audio in chunks of 10ms. Preferring 20ms audio chunks +// is a compromize between performance and power consumption. +constexpr NSTimeInterval kPeferredInputIOBufferDuration = 0.02; +constexpr NSTimeInterval kPeferredOutputIOBufferDuration = 0.02; + +class AudioDeviceDelegateImpl final : public rtc::RefCountedNonVirtual<AudioDeviceDelegateImpl> { + public: + AudioDeviceDelegateImpl( + rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule> audio_device_module, + rtc::Thread* thread) + : audio_device_module_(audio_device_module), thread_(thread) { + RTC_DCHECK(audio_device_module_); + RTC_DCHECK(thread_); + } + + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module() const { + return audio_device_module_.get(); + } + + rtc::Thread* thread() const { return thread_; } + + void reset_audio_device_module() { audio_device_module_ = nullptr; } + + private: + rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule> audio_device_module_; + rtc::Thread* thread_; +}; + +} // namespace + +@implementation ObjCAudioDeviceDelegate { + rtc::scoped_refptr<AudioDeviceDelegateImpl> impl_; +} + +@synthesize getPlayoutData = getPlayoutData_; + +@synthesize deliverRecordedData = deliverRecordedData_; + +@synthesize preferredInputSampleRate = preferredInputSampleRate_; + +@synthesize preferredInputIOBufferDuration = preferredInputIOBufferDuration_; + +@synthesize preferredOutputSampleRate = preferredOutputSampleRate_; + +@synthesize preferredOutputIOBufferDuration = preferredOutputIOBufferDuration_; + +- (instancetype)initWithAudioDeviceModule: + (rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule>)audioDeviceModule + audioDeviceThread:(rtc::Thread*)thread { + RTC_DCHECK_RUN_ON(thread); + if (self = [super init]) { + impl_ = rtc::make_ref_counted<AudioDeviceDelegateImpl>(audioDeviceModule, thread); + preferredInputSampleRate_ = kPreferredInputSampleRate; + preferredInputIOBufferDuration_ = kPeferredInputIOBufferDuration; + preferredOutputSampleRate_ = kPreferredOutputSampleRate; + preferredOutputIOBufferDuration_ = kPeferredOutputIOBufferDuration; + + rtc::scoped_refptr<AudioDeviceDelegateImpl> playout_delegate = impl_; + getPlayoutData_ = ^OSStatus(AudioUnitRenderActionFlags* _Nonnull actionFlags, + const AudioTimeStamp* _Nonnull timestamp, + NSInteger inputBusNumber, + UInt32 frameCount, + AudioBufferList* _Nonnull outputData) { + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device = + playout_delegate->audio_device_module(); + if (audio_device) { + return audio_device->OnGetPlayoutData( + actionFlags, timestamp, inputBusNumber, frameCount, outputData); + } else { + *actionFlags |= kAudioUnitRenderAction_OutputIsSilence; + RTC_LOG(LS_VERBOSE) << "No alive audio device"; + return noErr; + } + }; + + rtc::scoped_refptr<AudioDeviceDelegateImpl> record_delegate = impl_; + deliverRecordedData_ = + ^OSStatus(AudioUnitRenderActionFlags* _Nonnull actionFlags, + const AudioTimeStamp* _Nonnull timestamp, + NSInteger inputBusNumber, + UInt32 frameCount, + const AudioBufferList* _Nullable inputData, + void* renderContext, + RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) _Nullable renderBlock) { + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device = + record_delegate->audio_device_module(); + if (audio_device) { + return audio_device->OnDeliverRecordedData(actionFlags, + timestamp, + inputBusNumber, + frameCount, + inputData, + renderContext, + renderBlock); + } else { + RTC_LOG(LS_VERBOSE) << "No alive audio device"; + return noErr; + } + }; + } + return self; +} + +- (void)notifyAudioInputParametersChange { + RTC_DCHECK_RUN_ON(impl_->thread()); + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module(); + if (audio_device_module) { + audio_device_module->HandleAudioInputParametersChange(); + } +} + +- (void)notifyAudioOutputParametersChange { + RTC_DCHECK_RUN_ON(impl_->thread()); + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module(); + if (audio_device_module) { + audio_device_module->HandleAudioOutputParametersChange(); + } +} + +- (void)notifyAudioInputInterrupted { + RTC_DCHECK_RUN_ON(impl_->thread()); + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module(); + if (audio_device_module) { + audio_device_module->HandleAudioInputInterrupted(); + } +} + +- (void)notifyAudioOutputInterrupted { + RTC_DCHECK_RUN_ON(impl_->thread()); + webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module(); + if (audio_device_module) { + audio_device_module->HandleAudioOutputInterrupted(); + } +} + +- (void)dispatchAsync:(dispatch_block_t)block { + rtc::Thread* thread = impl_->thread(); + RTC_DCHECK(thread); + thread->PostTask([block] { + @autoreleasepool { + block(); + } + }); +} + +- (void)dispatchSync:(dispatch_block_t)block { + rtc::Thread* thread = impl_->thread(); + RTC_DCHECK(thread); + if (thread->IsCurrent()) { + @autoreleasepool { + block(); + } + } else { + thread->BlockingCall([block] { + @autoreleasepool { + block(); + } + }); + } +} + +- (void)resetAudioDeviceModule { + RTC_DCHECK_RUN_ON(impl_->thread()); + impl_->reset_audio_device_module(); +} + +@end |