MyAudioDeviceModule.h

#pragma once

#include "api/task_queue/default_task_queue_factory.h"
#include "modules/audio_device/include/audio_device_default.h"
#include "rtc_base/event.h"

class MyProducerAudioDeviceModule : public webrtc::webrtc_impl::AudioDeviceModuleDefault<webrtc::AudioDeviceModule> {
public:
	MyProducerAudioDeviceModule() : audio_callback_(nullptr), rendering_(false), capturing_(false) {}

	~MyProducerAudioDeviceModule() override
	{
		StopPlayout();
		StopRecording();
	}

	int32_t Init() override { return 0; }

	int32_t RegisterAudioCallback(webrtc::AudioTransport *callback) override
	{
		webrtc::MutexLock lock(&lock_);
		RTC_DCHECK(callback || audio_callback_);
		audio_callback_ = callback;
		return 0;
	}

	int32_t StartPlayout() override
	{
		webrtc::MutexLock lock(&lock_);
		rendering_ = true;
		return 0;
	}

	int32_t StopPlayout() override
	{
		webrtc::MutexLock lock(&lock_);
		rendering_ = false;
		return 0;
	}

	int32_t StartRecording() override
	{
		webrtc::MutexLock lock(&lock_);
		capturing_ = true;
		return 0;
	}

	int32_t StopRecording() override
	{
		webrtc::MutexLock lock(&lock_);
		capturing_ = false;
		return 0;
	}

	bool Playing() const override
	{
		webrtc::MutexLock lock(&lock_);
		return rendering_;
	}

	bool Recording() const override
	{
		webrtc::MutexLock lock(&lock_);
		return capturing_;
	}

public:
	void PlayData(const void *audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samples_per_sec,
		      const uint32_t total_delay_ms, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t &newMicLevel)
	{
		if (audio_callback_ == nullptr)
			return;

		audio_callback_->RecordedDataIsAvailable(audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec, total_delay_ms, clockDrift,
							 currentMicLevel, keyPressed, newMicLevel);
	}

private:
	mutable webrtc::Mutex lock_;

	rtc::Event done_rendering_;
	rtc::Event done_capturing_;

	bool rendering_ RTC_GUARDED_BY(lock_);
	bool capturing_ RTC_GUARDED_BY(lock_);

	webrtc::AudioTransport *audio_callback_ RTC_GUARDED_BY(lock_) = nullptr;
};