gb_emulator / gb_emulator / src / gb_sound_wasapi_renderer.cpp

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/*  Copyright Š 2011 Chris Spencer <spencercw@gmail.com>

    This program is free software: you can redistribute it and/or modify
    it under the terms of the GNU General Public License as published by
    the Free Software Foundation, either version 3 of the License, or
    (at your option) any later version.

    This program is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
    GNU General Public License for more details.

    You should have received a copy of the GNU General Public License
    along with this program.  If not, see <http://www.gnu.org/licenses/>.  */

#include <gb_emulator/gb_sound_wasapi_renderer.h>

#include <algorithm>
#include <stdexcept>
#include <string>

#include <boost/lexical_cast.hpp>

#include <avrt.h>

using ATL::CComPtr;
using boost::circular_buffer;
using boost::lexical_cast;
using boost::shared_ptr;
using std::min;
using std::runtime_error;
using std::string;

GbSoundWasapiRenderer::GbSoundWasapiRenderer():
refCount_(1),
renderFormat_(RENDER_INT16),
samplesPerPeriod_(0),
bufferSize_(0),
switchingStream_(false),
callbackFrequency_(0),
callbackSampleRate_(0),
callbackCountdown_(0)
{
	// Load avrt.dll so we can use MMCSS. We load this dynamically because it is an optional feature
	// and the library is not necessarily available.
	avrt_.reset(LoadLibraryW(L"avrt.dll"), FreeLibrary);
	if (avrt_)
	{
		AvSetMmThreadCharacteristicsW_ = reinterpret_cast<AvSetMmThreadCharacteristicsW_t>(
			GetProcAddress(avrt_.get(), "AvSetMmThreadCharacteristicsW"));
		AvRevertMmThreadCharacteristics_ = reinterpret_cast<AvRevertMmThreadCharacteristics_t>(
			GetProcAddress(avrt_.get(), "AvRevertMmThreadCharacteristics"));

		// In the unlikely event we only get one of the symbols, make sure we NULL out all the rest
		if (AvSetMmThreadCharacteristicsW_ == NULL || AvRevertMmThreadCharacteristics_ == NULL)
		{
			AvSetMmThreadCharacteristicsW_ = NULL;
			AvRevertMmThreadCharacteristics_ = NULL;
		}
	}
}

void GbSoundWasapiRenderer::initialise(CComPtr<IMMDevice> device,
	boost::function<void (unsigned)> sampleRateCallback)
{
	device_ = device;
	sampleRateCallback_ = sampleRateCallback;

	// Create the audio events
	audioSamplesReadyEvent_.reset(CreateEvent(NULL, FALSE, FALSE, NULL), CloseHandle);
	streamSwitchEvent_.reset(CreateEvent(NULL, FALSE, FALSE, NULL), CloseHandle);
	streamSwitchCompleteEvent_.reset(CreateEvent(NULL, TRUE, TRUE, NULL), CloseHandle);

	if (!audioSamplesReadyEvent_ || !streamSwitchEvent_ || !streamSwitchCompleteEvent_)
	{
		throw runtime_error("failed to create audio events: " +
			lexical_cast<string>(GetLastError()));
	}

	// Get an IAudioClient for the audio endpoint
	HRESULT hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
		reinterpret_cast<void **>(&audioClient_));
	if (FAILED(hr))
	{
		throw runtime_error("failed to activate audio client: " + lexical_cast<string>(hr));
	}

	hr = devices_.CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_INPROC_SERVER);
	if (FAILED(hr))
	{
		throw runtime_error("failed to get audio device enumerator: " + lexical_cast<string>(hr));
	}

	// Get the format to use to render samples
	loadFormat();

	// Initialise the audio client
	initialiseAudioClient();

	// Initialise stream switching
	hr = audioClient_->GetService(IID_PPV_ARGS(&audioSessionControl_));
	if (FAILED(hr))
	{
		throw runtime_error("failed to get audio session control: " + lexical_cast<string>(hr));
	}

	// Register for session and endpoint change notifications
	hr = audioSessionControl_->RegisterAudioSessionNotification(this);
	if (FAILED(hr))
	{
		throw runtime_error("failed to register for stream switch notifications: " +
			lexical_cast<string>(hr));
	}

	hr = devices_->RegisterEndpointNotificationCallback(this);
	if (FAILED(hr))
	{
		throw runtime_error("failed to register for stream switch notifications: " +
			lexical_cast<string>(hr));
	}

	// Enable MMCSS on this thread
	if (AvSetMmThreadCharacteristicsW_)
	{
		DWORD taskIndex = 0;
		mmcssHandle_.reset(AvSetMmThreadCharacteristicsW_(L"Playback", &taskIndex),
			AvRevertMmThreadCharacteristics_);
		if (!mmcssHandle_)
		{
			throw runtime_error("failed to enable MMCSS: " + lexical_cast<string>(GetLastError()));
		}
	}

	// Begin playback
	hr = audioClient_->Start();
	if (FAILED(hr))
	{
		throw runtime_error("failed to start audio client: " + lexical_cast<string>(hr));
	}
}

void GbSoundWasapiRenderer::pollEvents()
{
	HANDLE waitArray[] = {
		streamSwitchEvent_.get(),
		audioSamplesReadyEvent_.get() };

	DWORD result = WaitForMultipleObjects(2, waitArray, FALSE, 0);
	switch (result)
	{
	case WAIT_OBJECT_0 + 0:  // streamSwitchEvent_
		handleStreamSwitch();
		break;

	case WAIT_OBJECT_0 + 1:  // audioSamplesReadyEvent_
		sendSamples();
		break;
	}
}

void GbSoundWasapiRenderer::setTimerCallback(double frequency, boost::function<void ()> callback)
{
	callbackFrequency_ = frequency;
	callbackSampleRate_ = mixFormat_->nSamplesPerSec * (1 / frequency);
	callbackCountdown_ = callbackSampleRate_;
	timerCallback_ = callback;
}

void GbSoundWasapiRenderer::pushSample(double left, double right)
{
	switch (renderFormat_)
	{
	case RENDER_FLOAT:
		switch (mixFormat_->nChannels)
		{
		case 1:
			left += right;
			left /= 2;
			soundBufFloat_.push_back(static_cast<float>(left));
			break;

		case 2:
			soundBufFloat_.push_back(static_cast<float>(left));
			soundBufFloat_.push_back(static_cast<float>(right));
			break;

		default:
			soundBufFloat_.push_back(static_cast<float>(left));
			soundBufFloat_.push_back(static_cast<float>(right));
			
			for (int16_t i = 2; i != mixFormat_->nChannels; ++i)
			{
				soundBufFloat_.push_back(0);
			}
			break;
		}
		break;

	case RENDER_INT16:
		left  *= 0x8000;
		right *= 0x8000;

		switch (mixFormat_->nChannels)
		{
		case 1:
			left += right;
			left /= 2;
			soundBufInt16_.push_back(static_cast<int16_t>(left));
			break;

		case 2:
			soundBufInt16_.push_back(static_cast<int16_t>(left));
			soundBufInt16_.push_back(static_cast<int16_t>(right));
			break;

		default:
			soundBufInt16_.push_back(static_cast<int16_t>(left));
			soundBufInt16_.push_back(static_cast<int16_t>(right));
			
			for (int16_t i = 2; i != mixFormat_->nChannels; ++i)
			{
				soundBufInt16_.push_back(0);
			}
			break;
		}
		break;

	default:
		assert(!"unknown render format");
	}
}

void GbSoundWasapiRenderer::loadFormat()
{
	// Load the mix format
	{
		WAVEFORMATEX *tmp;
		HRESULT hr = audioClient_->GetMixFormat(&tmp);
		if (FAILED(hr))
		{
			throw runtime_error("failed to get mix format on the audio client: " +
				lexical_cast<string>(hr));
		}
		mixFormat_.reset(tmp, CoTaskMemFree);
	}

	WAVEFORMATEXTENSIBLE *mixFormatEx = reinterpret_cast<WAVEFORMATEXTENSIBLE *>(mixFormat_.get());
	if (mixFormat_->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
		(mixFormat_->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
		mixFormatEx->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))
	{
		renderFormat_ = RENDER_FLOAT;
	}
	else if (mixFormat_->wFormatTag == WAVE_FORMAT_PCM ||
		(mixFormat_->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
		mixFormatEx->SubFormat == KSDATAFORMAT_SUBTYPE_PCM))
	{
		if (mixFormat_->wBitsPerSample == 16)
		{
			renderFormat_ = RENDER_INT16;
		}
		else
		{
			throw runtime_error("unknown audio render type");
		}
	}
	else
	{
		throw runtime_error("unknown audio render type");
	}

	// Handle any change to the sample rate
	callbackSampleRate_ = mixFormat_->nSamplesPerSec * (1 / callbackFrequency_);
	sampleRateCallback_(mixFormat_->nSamplesPerSec);
}

void GbSoundWasapiRenderer::initialiseAudioClient()
{
	HRESULT hr = audioClient_->Initialize(AUDCLNT_SHAREMODE_SHARED,
		AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST, 0, 0, mixFormat_.get(),
		NULL);
	if (FAILED(hr))
	{
		throw runtime_error("failed to initialise audio client: " + lexical_cast<string>(hr));
	}

	hr = audioClient_->GetBufferSize(&bufferSize_);
	if (FAILED(hr))
	{
		throw runtime_error("failed to get audio client buffer size: " + lexical_cast<string>(hr));
	}

	hr = audioClient_->SetEventHandle(audioSamplesReadyEvent_.get());
	if (FAILED(hr))
	{
		throw runtime_error("failed to set audio event: " + lexical_cast<string>(hr));
	}

	hr = audioClient_->GetService(IID_PPV_ARGS(&renderClient_));
	if (FAILED(hr))
	{
		throw runtime_error("failed to get render client: " + lexical_cast<string>(hr));
	}

	// Allocate the intermediate buffer. We want space for 3 periods' worth of data. Initially
	// fill it half full with silence to avoid glitches when we start playback.
	switch (renderFormat_)
	{
	case RENDER_FLOAT:
		soundBufFloat_.set_capacity(4 * bufferSize_ * mixFormat_->nChannels);
		soundBufFloat_.clear();
		soundBufInt16_.resize(0);
		break;

	case RENDER_INT16:
		soundBufInt16_.set_capacity(4 * bufferSize_ * mixFormat_->nChannels);
		soundBufInt16_.clear();
		soundBufFloat_.resize(0);
		break;

	default:
		assert(!"unknown render format");
	}
}

void GbSoundWasapiRenderer::handleStreamSwitch()
{
	// When a stream switch is triggered several things happen:
	// 1. Stop the current renderer.
	// 2. Release resources associated with the renderer.
	// 3. Wait for the default device to change, or abort if this doesn't happen.
	// 4. Get the new default audio endpoint.
	// 5. Get a new audio client for the endpoint.
	// 6. Get the new mix format and adjust for any changes.
	// 7. Reinitialise the audio client.
	// 8. Reregister the events

	assert(switchingStream_);

	// 1. Stop rendering
	HRESULT hr = audioClient_->Stop();
	if (FAILED(hr))
	{
		throw runtime_error("failed to stop audio client during stream switch: " +
			lexical_cast<string>(hr));
	}

	// 2. Release resources
	hr = audioSessionControl_->UnregisterAudioSessionNotification(this);
	if (FAILED(hr))
	{
		throw runtime_error("failed to unregister from stream switch notifications during stream "
			"switch: " + lexical_cast<string>(hr));
	}

	audioSessionControl_.Release();
	renderClient_.Release();
	audioClient_.Release();
	device_.Release();

	// 3. Wait for the default device to change
	DWORD result = WaitForSingleObject(streamSwitchCompleteEvent_.get(), 500);
	if (result == WAIT_TIMEOUT)
	{
		throw runtime_error("stream switch timeout");
	}

	// 4. Get the new default audio endpoint
	hr = devices_->GetDefaultAudioEndpoint(eRender, eConsole, &device_);
	if (FAILED(hr))
	{
		throw runtime_error("failed to get new default device during stream switch: " +
			lexical_cast<string>(hr));
	}

	// 5. Get a new audio client for the endpoint
	hr = device_->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
		reinterpret_cast<void **>(&audioClient_));
	if (FAILED(hr))
	{
		throw runtime_error("failed to activate audio client on the new endpoint: " +
			lexical_cast<string>(hr));
	}

	// 6. Get the new mix format
	loadFormat();

	// 7. Reinitialise the audio client
	initialiseAudioClient();

	// 8. Reregister for stream switching events
	hr = audioClient_->GetService(IID_PPV_ARGS(&audioSessionControl_));
	if (FAILED(hr))
	{
		throw runtime_error("failed to get audio session control on new client: " +
			lexical_cast<string>(hr));
	}

	hr = audioSessionControl_->RegisterAudioSessionNotification(this);
	if (FAILED(hr))
	{
		throw runtime_error("failed to register for stream switch notifications on new client: " +
			lexical_cast<string>(hr));
	}

	// Finished; resume playback
	ResetEvent(streamSwitchCompleteEvent_.get());
	hr = audioClient_->Start();
	if (FAILED(hr))
	{
		throw runtime_error("failed to start the new audio client: " + lexical_cast<string>(hr));
	}

	switchingStream_ = false;
}

void GbSoundWasapiRenderer::sendSamples()
{
	// Determine how much space is left in the buffer
	uint32_t padding;
	HRESULT hr = audioClient_->GetCurrentPadding(&padding);
	if (FAILED(hr))
	{
		throw runtime_error("failed to get the audio padding size: " + lexical_cast<string>(hr));
	}
	uint32_t framesAvailable = bufferSize_ - padding;
	if (!framesAvailable)
	{
		return;
	}

	// Call the timer callback
	if (timerCallback_)
	{
		callbackCountdown_ -= framesAvailable;
		while (callbackCountdown_ <= 0)
		{
			timerCallback_();
			callbackCountdown_ += callbackSampleRate_;
		}
	}

	// If we have one period's worth of data copy it into the buffer, otherwise skip this pass
	size_t size;
	const uint8_t *range1, *range2;
	size_t range1Size, range2Size;

	switch (renderFormat_)
	{
	case RENDER_FLOAT:
		{
			size = soundBufFloat_.size();

			circular_buffer<float>::array_range array1 = soundBufFloat_.array_one();
			circular_buffer<float>::array_range array2 = soundBufFloat_.array_two();

			range1 = reinterpret_cast<uint8_t *>(array1.first);
			range2 = reinterpret_cast<uint8_t *>(array2.first);

			range1Size = array1.second * sizeof(float);
			range2Size = array2.second * sizeof(float);
		}
		break;

	case RENDER_INT16:
		{
			size = soundBufInt16_.size();

			circular_buffer<int16_t>::array_range array1 = soundBufInt16_.array_one();
			circular_buffer<int16_t>::array_range array2 = soundBufInt16_.array_two();

			range1 = reinterpret_cast<uint8_t *>(array1.first);
			range2 = reinterpret_cast<uint8_t *>(array2.first);

			range1Size = array1.second * sizeof(int16_t);
			range2Size = array2.second * sizeof(int16_t);
		}
		break;

	default:
		assert(!"unknown render format");
		return;
	}

	assert(!(size % mixFormat_->nChannels));
	uint32_t frames = static_cast<uint32_t>(size / mixFormat_->nChannels);
	if (!frames)
	{
		return;
	}
	assert(range1Size);

	// Get the data buffer to fill
	frames = (min)(frames, framesAvailable);
	uint8_t *data;
	hr = renderClient_->GetBuffer(frames, &data);
	if (FAILED(hr))
	{
		throw runtime_error("failed to get audio data buffer: " + lexical_cast<string>(hr));
	}

	// Copy the sound data into the buffer
	size_t remaining = frames * mixFormat_->nBlockAlign;
	size_t copySize = (min)(remaining, range1Size);
	memcpy(data, range1, copySize);
	remaining -= copySize;

	if (remaining && range2Size)
	{
		data += copySize;
		copySize = (min)(remaining, range2Size);
		memcpy(data, range2, copySize);
		remaining -= copySize;
	}

	assert(!remaining);
	switch (renderFormat_)
	{
	case RENDER_FLOAT:
		soundBufFloat_.erase_begin(frames * mixFormat_->nChannels);
		break;

	case RENDER_INT16:
		soundBufInt16_.erase_begin(frames * mixFormat_->nChannels);
		break;

	default:
		assert(!"unknown render format");
	}

	// Release the buffer
	hr = renderClient_->ReleaseBuffer(frames, 0);
	if (FAILED(hr))
	{
		throw runtime_error("failed to release audio data buffer: " + lexical_cast<string>(hr));
	}
}

////////////////////////////////////////////////////////////////////////////////////////////////////
// IAudioSessionEvents

HRESULT GbSoundWasapiRenderer::OnChannelVolumeChanged(DWORD /*ChannelCount*/,
	float * /*NewChannelVolumes*/, DWORD /*ChangedChannel*/, LPCGUID /*EventContext*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnDisplayNameChanged(const wchar_t * /*NewDisplayName*/,
	const GUID * /*EventContext*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnGroupingParamChanged(const GUID * /*NewGroupingParam*/,
	const GUID * /*EventContext*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnIconPathChanged(const wchar_t * /*NewIconPath*/,
	const GUID * /*EventContext*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnSessionDisconnected(AudioSessionDisconnectReason DisconnectReason)
{
	switch (DisconnectReason)
	{
	case DisconnectReasonDeviceRemoval:
		// The device we are rendering to has been removed so set the stream switch event. The
		// stream switch complete event will be set when OnDefaultDeviceChanged is called at which
		// point rendering will continue.
		switchingStream_ = true;
		SetEvent(streamSwitchEvent_.get());
		break;

	case DisconnectReasonFormatChanged:
		// The format of the device has changed so set the stream switch event to restart rendering
		// with the new format. Also set the stream switch complete event because we're not
		// expecting to see a default device change as well.
		switchingStream_ = true;
		SetEvent(streamSwitchEvent_.get());
		SetEvent(streamSwitchCompleteEvent_.get());
	}

	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnSimpleVolumeChanged(float /*NewVolume*/, BOOL /*NewMute*/,
	const GUID * /*EventContext*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnStateChanged(AudioSessionState /*NewState*/)
{
	return S_OK;
}

////////////////////////////////////////////////////////////////////////////////////////////////////
// IMMNotificationClient

HRESULT GbSoundWasapiRenderer::OnDefaultDeviceChanged(EDataFlow flow, ERole role,
	const wchar_t * /*pwstrDefaultDevice*/)
{
	if (flow == eRender && role == eConsole)
	{
		// The default device for our role has changed so initiate a stream switch
		if (!switchingStream_)
		{
			switchingStream_ = true;
			SetEvent(streamSwitchEvent_.get());
		}

		// Set the stream switch complete event to indicate it is ok to re-initialise the audio
		// renderer
		SetEvent(streamSwitchCompleteEvent_.get());
	}

	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnDeviceAdded(const wchar_t * /*pwstrDeviceId*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnDeviceRemoved(const wchar_t * /*pwstrDeviceId*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnDeviceStateChanged(const wchar_t * /*pwstrDeviceId*/,
	DWORD /*dwNewState*/)
{
	return S_OK;
}

HRESULT GbSoundWasapiRenderer::OnPropertyValueChanged(const wchar_t * /*pwstrDeviceId*/,
	const PROPERTYKEY /*key*/)
{
	return S_OK;
}

////////////////////////////////////////////////////////////////////////////////////////////////////
// IUnknown

ULONG GbSoundWasapiRenderer::AddRef()
{
	return InterlockedIncrement(&refCount_);
}

HRESULT GbSoundWasapiRenderer::QueryInterface(REFIID iid, void **pvObject)
{
	if (!pvObject)
	{
		return E_POINTER;
	}
	*pvObject = NULL;

	if (iid == IID_IUnknown)
	{
		*pvObject = static_cast<IUnknown *>(static_cast<IAudioSessionEvents *>(this));
	}
	else if (iid == __uuidof(IAudioSessionEvents))
	{
		*pvObject = static_cast<IAudioSessionEvents *>(this);
	}
	else if (iid == __uuidof(IMMNotificationClient))
	{
		*pvObject = static_cast<IMMNotificationClient *>(this);
	}
	else
	{
		return E_NOINTERFACE;
	}

	AddRef();
	return S_OK;
}

ULONG GbSoundWasapiRenderer::Release()
{
	ULONG result = InterlockedDecrement(&refCount_);
	if (!result)
	{
		delete this;
	}
	return result;
}
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