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/*
* 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
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