// Copyright 2008 Dolphin Emulator Project
// Licensed under GPLv2+
// Refer to the license.txt file included.

#include <thread>

#include "AudioCommon/aldlist.h"
#include "AudioCommon/DPL2Decoder.h"
#include "AudioCommon/OpenALStream.h"
#include "Common/Thread.h"
#include "Core/ConfigManager.h"

#if defined HAVE_OPENAL && HAVE_OPENAL

#ifdef _WIN32
#pragma comment(lib, "openal32.lib")
#endif

static soundtouch::SoundTouch soundTouch;

//
// AyuanX: Spec says OpenAL1.1 is thread safe already
//
bool OpenALStream::Start()
{
	m_run_thread.store(true);
	bool bReturn = false;

	ALDeviceList pDeviceList;
	if (pDeviceList.GetNumDevices())
	{
		char *defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());

		WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);

		ALCdevice *pDevice = alcOpenDevice(defDevName);
		if (pDevice)
		{
			ALCcontext *pContext = alcCreateContext(pDevice, nullptr);
			if (pContext)
			{
				// Used to determine an appropriate period size (2x period = total buffer size)
				//ALCint refresh;
				//alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
				//period_size_in_millisec = 1000 / refresh;

				alcMakeContextCurrent(pContext);
				thread = std::thread(&OpenALStream::SoundLoop, this);
				bReturn = true;
			}
			else
			{
				alcCloseDevice(pDevice);
				PanicAlertT("OpenAL: can't create context for device %s", defDevName);
			}
		}
		else
		{
			PanicAlertT("OpenAL: can't open device %s", defDevName);
		}
	}
	else
	{
		PanicAlertT("OpenAL: can't find sound devices");
	}

	// Initialize DPL2 parameters
	DPL2Reset();

	soundTouch.clear();
	return bReturn;
}

void OpenALStream::Stop()
{
	m_run_thread.store(false);
	// kick the thread if it's waiting
	soundSyncEvent.Set();

	soundTouch.clear();

	thread.join();

	alSourceStop(uiSource);
	alSourcei(uiSource, AL_BUFFER, 0);

	// Clean up buffers and sources
	alDeleteSources(1, &uiSource);
	uiSource = 0;
	alDeleteBuffers(numBuffers, uiBuffers);

	ALCcontext *pContext = alcGetCurrentContext();
	ALCdevice *pDevice = alcGetContextsDevice(pContext);

	alcMakeContextCurrent(nullptr);
	alcDestroyContext(pContext);
	alcCloseDevice(pDevice);
}

void OpenALStream::SetVolume(int volume)
{
	fVolume = (float)volume / 100.0f;

	if (uiSource)
		alSourcef(uiSource, AL_GAIN, fVolume);
}

void OpenALStream::Update()
{
	soundSyncEvent.Set();
}

void OpenALStream::Clear(bool mute)
{
	m_muted = mute;

	if (m_muted)
	{
		soundTouch.clear();
		alSourceStop(uiSource);
	}
	else
	{
		alSourcePlay(uiSource);
	}
}

void OpenALStream::SoundLoop()
{
	Common::SetCurrentThreadName("Audio thread - openal");

	bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
#if defined(__APPLE__)
	bool float32_capable = false;
	const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
	// OS X does not have the alext AL_FORMAT_51CHN32 yet.
	surround_capable = false;
	const ALenum AL_FORMAT_51CHN32 = 0;
#else
	bool float32_capable = true;
#endif

	u32 ulFrequency = m_mixer->GetSampleRate();
	numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers

	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
	uiSource = 0;

	// Generate some AL Buffers for streaming
	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
	// Generate a Source to playback the Buffers
	alGenSources(1, &uiSource);

	// Short Silence
	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
	for (int i = 0; i < numBuffers; i++)
	{
		if (surround_capable)
			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
		else
			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT, ulFrequency);
	}
	alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
	alSourcePlay(uiSource);

	// Set the default sound volume as saved in the config file.
	alSourcef(uiSource, AL_GAIN, fVolume);

	// TODO: Error handling
	//ALenum err = alGetError();

	ALint iBuffersFilled = 0;
	ALint iBuffersProcessed = 0;
	ALint iState = 0;
	ALuint uiBufferTemp[OAL_MAX_BUFFERS] = { 0 };

	soundTouch.setChannels(2);
	soundTouch.setSampleRate(ulFrequency);
	soundTouch.setTempo(1.0);
	soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
	soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
	soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
	soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
	soundTouch.setSetting(SETTING_OVERLAP_MS, 12);

	while (m_run_thread.load())
	{
		// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
		const u32 stereo_16_bit_size = 4;
		const u32 dma_length = 32;
		const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
		u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
		u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();

		unsigned int numSamples = (unsigned int)num_samples_to_render;
		unsigned int minSamples = surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)

		numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
		numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);

		// Convert the samples from short to float
		float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
		for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
			dest[i] = (float)realtimeBuffer[i] / (1 << 15);

		soundTouch.putSamples(dest, numSamples);

		if (iBuffersProcessed == iBuffersFilled)
		{
			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
			iBuffersFilled = 0;
		}

		if (iBuffersProcessed)
		{
			float rate = m_mixer->GetCurrentSpeed();
			if (rate <= 0)
			{
				Core::RequestRefreshInfo();
				rate = m_mixer->GetCurrentSpeed();
			}

			// Place a lower limit of 10% speed.  When a game boots up, there will be
			// many silence samples.  These do not need to be timestretched.
			if (rate > 0.10)
			{
				soundTouch.setTempo(rate);
				if (rate > 10)
				{
					soundTouch.clear();
				}
			}

			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);

			if (nSamples <= minSamples)
				continue;

			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
			if (iBuffersFilled == 0)
			{
				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
				ALenum err = alGetError();
				if (err != 0)
				{
					ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
				}
			}

			if (surround_capable)
			{
				float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
				DPL2Decode(sampleBuffer, nSamples, dpl2);
				// zero-out the subwoofer channel - DPL2Decode generates a pretty
				// good 5.0 but not a good 5.1 output.  Sadly there is not a 5.0
				// AL_FORMAT_50CHN32 to make this super-explicit.
				// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
				for (u32 i = 0; i < nSamples; ++i)
				{
					dpl2[i*SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
				}
				alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
				ALenum err = alGetError();
				if (err == AL_INVALID_ENUM)
				{
					// 5.1 is not supported by the host, fallback to stereo
					WARN_LOG(AUDIO, "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
					surround_capable = false;
				}
				else if (err != 0)
				{
					ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
				}
			}

			else
			{
				if (float32_capable)
				{
					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * FRAME_STEREO_FLOAT, ulFrequency);
					ALenum err = alGetError();
					if (err == AL_INVALID_ENUM)
					{
						float32_capable = false;
					}
					else if (err != 0)
					{
						ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
					}
				}

				else
				{
					// Convert the samples from float to short
					short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
					for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
						stereo[i] = (short)((float)sampleBuffer[i] * (1 << 16));

					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * FRAME_STEREO_SHORT, ulFrequency);
				}
			}

			alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
			ALenum err = alGetError();
			if (err != 0)
			{
				ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
			}
			iBuffersFilled++;

			if (iBuffersFilled == numBuffers)
			{
				alSourcePlay(uiSource);
				err = alGetError();
				if (err != 0)
				{
					ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
				}
			}

			alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
			if (iState != AL_PLAYING)
			{
				// Buffer underrun occurred, resume playback
				alSourcePlay(uiSource);
				err = alGetError();
				if (err != 0)
				{
					ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
				}
			}
		}
		else
		{
			soundSyncEvent.Wait();
		}
	}
}

#endif //HAVE_OPENAL