path: root/src/audio/oal/stream.cpp

#include "common.h"

#ifdef AUDIO_OAL

#if defined _MSC_VER && !defined CMAKE_NO_AUTOLINK
#ifdef AUDIO_OAL_USE_SNDFILE
#pragma comment( lib, "libsndfile-1.lib" )
#endif
#ifdef AUDIO_OAL_USE_MPG123
#pragma comment( lib, "libmpg123-0.lib" )
#endif
#endif
#ifdef AUDIO_OAL_USE_SNDFILE
#include <sndfile.h>
#endif
#ifdef AUDIO_OAL_USE_MPG123
#include <mpg123.h>
#endif
#ifdef AUDIO_OAL_USE_OPUS
#include <opusfile.h>
#endif

#include <queue>
#include <utility>

#ifdef MULTITHREADED_AUDIO
#include <iostream>
#include <thread>
#include <mutex>
#include <condition_variable>
#include "MusicManager.h"
#include "stream.h"

std::thread	gAudioThread;
std::mutex gAudioThreadQueueMutex;
std::condition_variable gAudioThreadCv;
bool gAudioThreadTerm = false;
std::queue<CStream*> gStreamsToProcess; // values are not unique, we will handle that ourself
std::queue<std::pair<IDecoder*, void*>> gStreamsToClose;
#else
#include "stream.h"
#endif

#include "sampman.h"

#ifndef _WIN32
#include "crossplatform.h"
#endif

/*
As we ran onto an issue of having different volume levels for mono streams
and stereo streams we are now handling all the stereo panning ourselves.
Each stream now has two sources - one panned to the left and one to the right,
and uses two separate buffers to store data for each individual channel.
For that we also have to reshuffle all decoded PCM stereo data from LRLRLRLR to
LLLLRRRR (handled by CSortStereoBuffer).
*/

class CSortStereoBuffer
{
	uint16* PcmBuf;
	size_t BufSize;
//#ifdef MULTITHREADED_AUDIO
//	std::mutex Mutex;
//#endif

public:
	CSortStereoBuffer() : PcmBuf(nil), BufSize(0) {}
	~CSortStereoBuffer()
	{
		if (PcmBuf)
			free(PcmBuf);
	}

	uint16* GetBuffer(size_t size)
	{
		if (size == 0) return nil;
		if (!PcmBuf)
		{
			BufSize = size;
			PcmBuf = (uint16*)malloc(BufSize);
		}
		else if (BufSize < size)
		{
			BufSize = size;
			PcmBuf = (uint16*)realloc(PcmBuf, size);
		}
		return PcmBuf;
	}

	void SortStereo(void* buf, size_t size)
	{
//#ifdef MULTITHREADED_AUDIO
//		std::lock_guard<std::mutex> lock(Mutex);
//#endif
		uint16* InBuf = (uint16*)buf;
		uint16* OutBuf = GetBuffer(size);

		if (!OutBuf) return;

		size_t rightStart = size / 4;
		for (size_t i = 0; i < size / 4; i++)
		{
			OutBuf[i] = InBuf[i*2];
			OutBuf[i+rightStart] = InBuf[i*2+1];
		}

		memcpy(InBuf, OutBuf, size);
	}

};

CSortStereoBuffer SortStereoBuffer;

class CImaADPCMDecoder
{
	const uint16 StepTable[89] = {
		7, 8, 9, 10, 11, 12, 13, 14,
		16, 17, 19, 21, 23, 25, 28, 31,
		34, 37, 41, 45, 50, 55, 60, 66,
		73, 80, 88, 97, 107, 118, 130, 143,
		157, 173, 190, 209, 230, 253, 279, 307,
		337, 371, 408, 449, 494, 544, 598, 658,
		724, 796, 876, 963, 1060, 1166, 1282, 1411,
		1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
		3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
		7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
		15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
		32767
	};

	int16 Sample, StepIndex;

public:
	CImaADPCMDecoder()
	{
		Init(0, 0);
	}

	void Init(int16 _Sample, int16 _StepIndex)
	{
		Sample = _Sample;
		StepIndex = _StepIndex;
	}

	void Decode(uint8 *inbuf, int16 *_outbuf, size_t size)
	{
		int16* outbuf = _outbuf;
		for (size_t i = 0; i < size; i++)
		{
			*(outbuf++) = DecodeSample(inbuf[i] & 0xF);
			*(outbuf++) = DecodeSample(inbuf[i] >> 4);
		}
	}

	int16 DecodeSample(uint8 adpcm)
	{
		uint16 step = StepTable[StepIndex];

		if (adpcm & 4)
			StepIndex += ((adpcm & 3) + 1) * 2;
		else
			StepIndex--;

		StepIndex = clamp(StepIndex, 0, 88);

		int delta = step >> 3;
		if (adpcm & 1) delta += step >> 2;
		if (adpcm & 2) delta += step >> 1;
		if (adpcm & 4) delta += step;
		if (adpcm & 8) delta = -delta;

		int newSample = Sample + delta;
		Sample = clamp(newSample, -32768, 32767);
		return Sample;
	}
};

class CWavFile : public IDecoder
{
	enum
	{
		WAVEFMT_PCM = 1,
		WAVEFMT_IMA_ADPCM = 0x11,
		WAVEFMT_XBOX_ADPCM = 0x69,
	};

	struct tDataHeader
	{
		uint32 ID;
		uint32 Size;
	};

	struct tFormatHeader
	{
		uint16 AudioFormat;
		uint16 NumChannels;
		uint32 SampleRate;
		uint32 ByteRate;
		uint16 BlockAlign;
		uint16 BitsPerSample;
		uint16 extra[2]; // adpcm only

		tFormatHeader() { memset(this, 0, sizeof(*this)); }
	};

	FILE *m_pFile;
	bool  m_bIsOpen;

	tFormatHeader m_FormatHeader;

	uint32 m_DataStartOffset; // TODO: 64 bit?
	uint32 m_nSampleCount;
	uint32 m_nSamplesPerBlock;

	// ADPCM things
	uint8            *m_pAdpcmBuffer;
	int16           **m_ppPcmBuffers;
	CImaADPCMDecoder *m_pAdpcmDecoders;

	void Close()
	{
		if (m_pFile) {
			fclose(m_pFile);
			m_pFile = nil;
		}
		delete[] m_pAdpcmBuffer;
		delete[] m_ppPcmBuffers;
		delete[] m_pAdpcmDecoders;
	}

	uint32 GetCurrentSample() const
	{
		// TODO: 64 bit?
		uint32 FilePos = ftell(m_pFile);
		if (FilePos <= m_DataStartOffset)
			return 0;
		return (FilePos - m_DataStartOffset) / m_FormatHeader.BlockAlign * m_nSamplesPerBlock;
	}

public:
	CWavFile(const char* path) : m_bIsOpen(false), m_DataStartOffset(0), m_nSampleCount(0), m_nSamplesPerBlock(0), m_pAdpcmBuffer(nil), m_ppPcmBuffers(nil), m_pAdpcmDecoders(nil)
	{
		m_pFile = fopen(path, "rb");
		if (!m_pFile) return;

#define CLOSE_ON_ERROR(op)\
			if (op) { \
				Close(); \
				return; \
			}

		tDataHeader DataHeader;

		CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0);
		CLOSE_ON_ERROR(DataHeader.ID != 'FFIR');

		// TODO? validate filesizes

		int WAVE;
		CLOSE_ON_ERROR(fread(&WAVE, 4, 1, m_pFile) == 0);
		CLOSE_ON_ERROR(WAVE != 'EVAW')
		CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0);
		CLOSE_ON_ERROR(DataHeader.ID != ' tmf');

		CLOSE_ON_ERROR(fread(&m_FormatHeader, Min(DataHeader.Size, sizeof(tFormatHeader)), 1, m_pFile) == 0);
		CLOSE_ON_ERROR(DataHeader.Size > sizeof(tFormatHeader));

		switch (m_FormatHeader.AudioFormat)
		{
		case WAVEFMT_XBOX_ADPCM:
			m_FormatHeader.AudioFormat = WAVEFMT_IMA_ADPCM;
		case WAVEFMT_IMA_ADPCM:
			m_nSamplesPerBlock = (m_FormatHeader.BlockAlign / m_FormatHeader.NumChannels - 4) * 2 + 1;
			m_pAdpcmBuffer = new uint8[m_FormatHeader.BlockAlign];
			m_ppPcmBuffers = new int16*[m_FormatHeader.NumChannels];
			m_pAdpcmDecoders = new CImaADPCMDecoder[m_FormatHeader.NumChannels];
			break;
		case WAVEFMT_PCM:
			m_nSamplesPerBlock = 1;
			if (m_FormatHeader.BitsPerSample != 16)
			{
				debug("Unsupported PCM (%d bits), only signed 16-bit is supported (%s)\n", m_FormatHeader.BitsPerSample, path);
				Close();
				return;
			}
			break;
		default:
			debug("Unsupported wav format 0x%x (%s)\n", m_FormatHeader.AudioFormat, path);
			Close();
			return;
		}

		while (true) {
			CLOSE_ON_ERROR(fread(&DataHeader, sizeof(DataHeader), 1, m_pFile) == 0);
			if (DataHeader.ID == 'atad')
				break;
			fseek(m_pFile, DataHeader.Size, SEEK_CUR);
			// TODO? validate data size
			// maybe check if there no extreme custom headers that might break this
		}
		
		m_DataStartOffset = ftell(m_pFile);
		m_nSampleCount = DataHeader.Size / m_FormatHeader.BlockAlign * m_nSamplesPerBlock;

		m_bIsOpen = true;
#undef CLOSE_ON_ERROR
	}

	void FileOpen()
	{
	}

	~CWavFile()
	{
		Close();
	}

	bool IsOpened()
	{
		return m_bIsOpen;
	}


	uint32 GetSampleSize()
	{
		return sizeof(uint16);
	}

	uint32 GetSampleCount()
	{
		return m_nSampleCount;
	}

	uint32 GetSampleRate()
	{
		return m_FormatHeader.SampleRate;
	}

	uint32 GetChannels()
	{
		return m_FormatHeader.NumChannels;
	}

	void Seek(uint32 milliseconds)
	{
		if (!IsOpened()) return;
		fseek(m_pFile, m_DataStartOffset + ms2samples(milliseconds) / m_nSamplesPerBlock * m_FormatHeader.BlockAlign, SEEK_SET);
	}

	uint32 Tell()
	{
		if (!IsOpened()) return 0;
		return samples2ms(GetCurrentSample());
	}

#define SAMPLES_IN_LINE (8)

	uint32 Decode(void* buffer)
	{
		if (!IsOpened()) return 0;
		
		if (m_FormatHeader.AudioFormat == WAVEFMT_PCM)
		{
			// just read the file and sort the samples
			uint32 size = fread(buffer, 1, GetBufferSize(), m_pFile);
			if (m_FormatHeader.NumChannels == 2)
				SortStereoBuffer.SortStereo(buffer, size);
			return size;
		}
		else if (m_FormatHeader.AudioFormat == WAVEFMT_IMA_ADPCM)
		{
			// trim the buffer size if we're at the end of our file
			uint32 nMaxSamples = GetBufferSamples() / m_FormatHeader.NumChannels;
			uint32 nSamplesLeft = m_nSampleCount - GetCurrentSample();
			nMaxSamples = Min(nMaxSamples, nSamplesLeft);

			// align sample count to our block
			nMaxSamples = nMaxSamples / m_nSamplesPerBlock * m_nSamplesPerBlock;

			// count the size of output buffer
			uint32 OutBufSizePerChannel = nMaxSamples * GetSampleSize();
			uint32 OutBufSize = OutBufSizePerChannel * m_FormatHeader.NumChannels;

			// calculate the pointers to individual channel buffers
			for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
				m_ppPcmBuffers[i] = (int16*)((int8*)buffer + OutBufSizePerChannel * i);

			uint32 samplesRead = 0;
			while (samplesRead < nMaxSamples)
			{
				// read the file
				uint8 *pAdpcmBuf = m_pAdpcmBuffer;
				if (fread(m_pAdpcmBuffer, 1, m_FormatHeader.BlockAlign, m_pFile) == 0)
					return 0;

				// get the first sample in adpcm block and initialise the decoder(s)
				for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
				{
					int16 Sample = *(int16*)pAdpcmBuf;
					pAdpcmBuf += sizeof(int16);
					int16 Step = *(int16*)pAdpcmBuf;
					pAdpcmBuf += sizeof(int16);
					m_pAdpcmDecoders[i].Init(Sample, Step);
					*(m_ppPcmBuffers[i]) = Sample;
					m_ppPcmBuffers[i]++;
				}
				samplesRead++;

				// decode the rest of the block
				for (uint32 s = 1; s < m_nSamplesPerBlock; s += SAMPLES_IN_LINE)
				{
					for (uint32 i = 0; i < m_FormatHeader.NumChannels; i++)
					{
						m_pAdpcmDecoders[i].Decode(pAdpcmBuf, m_ppPcmBuffers[i], SAMPLES_IN_LINE / 2);
						pAdpcmBuf += SAMPLES_IN_LINE / 2;
						m_ppPcmBuffers[i] += SAMPLES_IN_LINE;
					}
					samplesRead += SAMPLES_IN_LINE;
				}
			}
			return OutBufSize;
		}
		return 0;
	}
};

#ifdef AUDIO_OAL_USE_SNDFILE
class CSndFile : public IDecoder
{
	SNDFILE *m_pfSound;
	SF_INFO m_soundInfo;
public:
	CSndFile(const char *path) :
		m_pfSound(nil)
	{
		memset(&m_soundInfo, 0, sizeof(m_soundInfo));
		m_pfSound = sf_open(path, SFM_READ, &m_soundInfo);
	}
	
	void FileOpen()
	{
	}

	~CSndFile()
	{
		if ( m_pfSound )
		{
			sf_close(m_pfSound);
			m_pfSound = nil;
		}
	}
	
	bool IsOpened()
	{
		return m_pfSound != nil;
	}
	
	uint32 GetSampleSize()
	{
		return sizeof(uint16);
	}
	
	uint32 GetSampleCount()
	{
		return m_soundInfo.frames;
	}
	
	uint32 GetSampleRate()
	{
		return m_soundInfo.samplerate;
	}
	
	uint32 GetChannels()
	{
		return m_soundInfo.channels;
	}
	
	void Seek(uint32 milliseconds)
	{
		if ( !IsOpened() ) return;
		sf_seek(m_pfSound, ms2samples(milliseconds), SF_SEEK_SET);
	}
	
	uint32 Tell()
	{
		if ( !IsOpened() ) return 0;
		return samples2ms(sf_seek(m_pfSound, 0, SF_SEEK_CUR));
	}
	
	uint32 Decode(void *buffer)
	{
		if ( !IsOpened() ) return 0;

		size_t size = sf_read_short(m_pfSound, (short*)buffer, GetBufferSamples()) * GetSampleSize();
		if (GetChannels()==2)
			SortStereoBuffer.SortStereo(buffer, size);
		return size;
	}
};
#endif

#ifdef AUDIO_OAL_USE_MPG123

class CMP3File : public IDecoder
{
protected:
	mpg123_handle *m_pMH;
	bool m_bOpened;
	uint32 m_nRate;
	uint32 m_nChannels;
	const char* m_pPath;
	bool m_bFileNotOpenedYet;

	CMP3File() :
		m_pMH(nil),
		m_bOpened(false),
		m_nRate(0),
		m_bFileNotOpenedYet(false),
		m_nChannels(0) {}
public:
	CMP3File(const char *path) :
		m_pMH(nil),
		m_bOpened(false),
		m_nRate(0),
		m_nChannels(0),
		m_pPath(path),
		m_bFileNotOpenedYet(false)
	{
		m_pMH = mpg123_new(nil, nil);
		if ( m_pMH )
		{
			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_SEEKBUFFER | MPG123_GAPLESS, 0.0);

			m_bOpened = true;
			m_bFileNotOpenedYet = true;
			// It's possible to move this to audioFileOpsThread(), but effect isn't noticable + probably not compatible with our current cutscene audio handling
#if 1
			FileOpen();
#endif
		}
	}
	
	void FileOpen()
	{
		if(!m_bFileNotOpenedYet) return;

		long rate = 0;
		int channels = 0;
		int encoding = 0;
		m_bOpened = mpg123_open(m_pMH, m_pPath) == MPG123_OK
		        && mpg123_getformat(m_pMH, &rate, &channels, &encoding) == MPG123_OK;

		m_nRate = rate;
		m_nChannels = channels;

		if(IsOpened()) {
			mpg123_format_none(m_pMH);
			mpg123_format(m_pMH, rate, channels, encoding);
		}
		m_bFileNotOpenedYet = false;
	}

	~CMP3File()
	{
		if ( m_pMH )
		{
			mpg123_close(m_pMH);
			mpg123_delete(m_pMH);
			m_pMH = nil;
		}
	}
	
	bool IsOpened()
	{
		return m_bOpened;
	}
	
	uint32 GetSampleSize()
	{
		return sizeof(uint16);
	}
	
	uint32 GetSampleCount()
	{
		if ( !IsOpened() || m_bFileNotOpenedYet ) return 0;
		return mpg123_length(m_pMH);
	}
	
	uint32 GetSampleRate()
	{
		return m_nRate;
	}
	
	uint32 GetChannels()
	{
		return m_nChannels;
	}
	
	void Seek(uint32 milliseconds)
	{
		if ( !IsOpened() || m_bFileNotOpenedYet ) return;
		mpg123_seek(m_pMH, ms2samples(milliseconds), SEEK_SET);
	}
	
	uint32 Tell()
	{
		if ( !IsOpened() || m_bFileNotOpenedYet ) return 0;
		return samples2ms(mpg123_tell(m_pMH));
	}
	
	uint32 Decode(void *buffer)
	{
		if ( !IsOpened() || m_bFileNotOpenedYet ) return 0;
		
		size_t size;
		int err = mpg123_read(m_pMH, (unsigned char *)buffer, GetBufferSize(), &size);
#if defined(__LP64__) || defined(_WIN64)
		assert("We can't handle audio files more then 2 GB yet :shrug:" && (size < UINT32_MAX));
#endif
		if (err != MPG123_OK && err != MPG123_DONE) return 0;
		if (GetChannels() == 2)
			SortStereoBuffer.SortStereo(buffer, size);
		return (uint32)size;
	}
};

class CADFFile : public CMP3File
{
	static ssize_t r_read(void* fh, void* buf, size_t size)
	{
		size_t bytesRead = fread(buf, 1, size, (FILE*)fh);
		uint8* _buf = (uint8*)buf;
		for (size_t i = 0; i < size; i++)
			_buf[i] ^= 0x22;
		return bytesRead;
	}
	static off_t r_seek(void* fh, off_t pos, int seekType)
	{
		fseek((FILE*)fh, pos, seekType);
		return ftell((FILE*)fh);
	}
	static void r_close(void* fh)
	{
		fclose((FILE*)fh);
	}
public:
	CADFFile(const char* path)
	{
		m_pMH = mpg123_new(nil, nil);
		if (m_pMH)
		{
			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_SEEKBUFFER | MPG123_GAPLESS, 0.0);

			m_bOpened = true;
			m_bFileNotOpenedYet = true;
			m_pPath = path;
			// It's possible to move this to audioFileOpsThread(), but effect isn't noticable + probably not compatible with our current cutscene audio handling
#if 1
			FileOpen();
#endif

		}
	}

	void FileOpen()
	{
		if(!m_bFileNotOpenedYet) return;

		long rate = 0;
		int channels = 0;
		int encoding = 0;

		FILE *f = fopen(m_pPath, "rb");

		m_bOpened = f && mpg123_replace_reader_handle(m_pMH, r_read, r_seek, r_close) == MPG123_OK
			&& mpg123_open_handle(m_pMH, f) == MPG123_OK && mpg123_getformat(m_pMH, &rate, &channels, &encoding) == MPG123_OK;

		m_nRate = rate;
		m_nChannels = channels;

		if(IsOpened()) {
			mpg123_format_none(m_pMH);
			mpg123_format(m_pMH, rate, channels, encoding);
		}

		m_bFileNotOpenedYet = false;
	}
};

#endif
#define VAG_LINE_SIZE (0x10)
#define VAG_SAMPLES_IN_LINE (28)

class CVagDecoder
{
	const double f[5][2] = { { 0.0, 0.0 },
					{  60.0 / 64.0,  0.0 },
					{  115.0 / 64.0, -52.0 / 64.0 },
					{  98.0 / 64.0, -55.0 / 64.0 },
					{  122.0 / 64.0, -60.0 / 64.0 } };

	double s_1;
	double s_2;
public:
	CVagDecoder()
	{
		ResetState();
	}

	void ResetState()
	{
		s_1 = s_2 = 0.0;
	}

	static short quantize(double sample)
	{
		int a = int(sample + 0.5);
		return short(clamp(a, -32768, 32767));
	}

	void Decode(void* _inbuf, int16* _outbuf, size_t size)
	{
		uint8* inbuf = (uint8*)_inbuf;
		int16* outbuf = _outbuf;
		size &= ~(VAG_LINE_SIZE - 1);

		while (size > 0) {
			double samples[VAG_SAMPLES_IN_LINE];

			int predict_nr, shift_factor, flags;
			predict_nr = *(inbuf++);
			shift_factor = predict_nr & 0xf;
			predict_nr >>= 4;
			flags = *(inbuf++);
			if (flags == 7) // TODO: ignore?
				break;
			for (int i = 0; i < VAG_SAMPLES_IN_LINE; i += 2) {
				int d = *(inbuf++);
				int16 s = int16((d & 0xf) << 12);
				samples[i] = (double)(s >> shift_factor);
				s = int16((d & 0xf0) << 8);
				samples[i + 1] = (double)(s >> shift_factor);
			}

			for (int i = 0; i < VAG_SAMPLES_IN_LINE; i++) {
				samples[i] = samples[i] + s_1 * f[predict_nr][0] + s_2 * f[predict_nr][1];
				s_2 = s_1;
				s_1 = samples[i];
				*(outbuf++) = quantize(samples[i] + 0.5);
			}
			size -= VAG_LINE_SIZE;
		}
	}
};

#define VB_BLOCK_SIZE (0x2000)
#define NUM_VAG_LINES_IN_BLOCK (VB_BLOCK_SIZE / VAG_LINE_SIZE)
#define NUM_VAG_SAMPLES_IN_BLOCK (NUM_VAG_LINES_IN_BLOCK * VAG_SAMPLES_IN_LINE)

class CVbFile : public IDecoder
{
	FILE        *m_pFile;
	CVagDecoder *m_pVagDecoders;

	size_t  m_FileSize;
	size_t  m_nNumberOfBlocks;

	uint32  m_nSampleRate;
	uint8   m_nChannels;
	bool    m_bBlockRead;
	uint16  m_LineInBlock;
	size_t  m_CurrentBlock;

	uint8 **m_ppVagBuffers; // buffers that cache actual ADPCM file data
	int16 **m_ppPcmBuffers;

	void ReadBlock(int32 block = -1)
	{
		// just read next block if -1
		if (block != -1)
			fseek(m_pFile, block * m_nChannels * VB_BLOCK_SIZE, SEEK_SET);

		for (int i = 0; i < m_nChannels; i++)
			fread(m_ppVagBuffers[i], VB_BLOCK_SIZE, 1, m_pFile);
		m_bBlockRead = true;
	}

public:
	CVbFile(const char* path, uint32 nSampleRate = 32000, uint8 nChannels = 2) : m_nSampleRate(nSampleRate), m_nChannels(nChannels), m_pVagDecoders(nil), m_ppVagBuffers(nil), m_ppPcmBuffers(nil),
		m_FileSize(0), m_nNumberOfBlocks(0), m_bBlockRead(false), m_LineInBlock(0), m_CurrentBlock(0)
	{
		m_pFile = fopen(path, "rb");
		if (!m_pFile) return;

		fseek(m_pFile, 0, SEEK_END);
		m_FileSize = ftell(m_pFile);
		fseek(m_pFile, 0, SEEK_SET);

		m_nNumberOfBlocks = m_FileSize / (nChannels * VB_BLOCK_SIZE);
		m_pVagDecoders = new CVagDecoder[nChannels];
		m_ppVagBuffers = new uint8*[nChannels];
		m_ppPcmBuffers = new int16*[nChannels];
		for (uint8 i = 0; i < nChannels; i++)
			m_ppVagBuffers[i] = new uint8[VB_BLOCK_SIZE];
	}

	void FileOpen()
	{
	}

	~CVbFile()
	{
		if (m_pFile)
		{
			fclose(m_pFile);

			delete[] m_pVagDecoders;
			for (int i = 0; i < m_nChannels; i++)
				delete[] m_ppVagBuffers[i];
			delete[] m_ppVagBuffers;
			delete[] m_ppPcmBuffers;
		}
	}

	bool IsOpened()
	{
		return m_pFile != nil;
	}

	uint32 GetSampleSize()
	{
		return sizeof(uint16);
	}

	uint32 GetSampleCount()
	{
		if (!IsOpened()) return 0;
		return m_nNumberOfBlocks * NUM_VAG_LINES_IN_BLOCK * VAG_SAMPLES_IN_LINE;
	}

	uint32 GetSampleRate()
	{
		return m_nSampleRate;
	}

	uint32 GetChannels()
	{
		return m_nChannels;
	}

	void Seek(uint32 milliseconds)
	{
		if (!IsOpened()) return;
		uint32 samples = ms2samples(milliseconds);

		// find the block of our sample
		uint32 block = samples / NUM_VAG_SAMPLES_IN_BLOCK;
		if (block > m_nNumberOfBlocks)
		{
			samples = 0;
			block = 0;
		}
		if (block != m_CurrentBlock)
			m_bBlockRead = false;

		// find a line of our sample within our block
		uint32 remainingSamples = samples - block * NUM_VAG_SAMPLES_IN_BLOCK;
		uint32 newLine = remainingSamples / VAG_SAMPLES_IN_LINE / VAG_LINE_SIZE;

		if (m_CurrentBlock != block || m_LineInBlock != newLine)
		{
			m_CurrentBlock = block;
			m_LineInBlock = newLine;
			for (uint32 i = 0; i < GetChannels(); i++)
				m_pVagDecoders[i].ResetState();
		}

	}

	uint32 Tell()
	{
		if (!IsOpened()) return 0;
		uint32 pos = (m_CurrentBlock * NUM_VAG_LINES_IN_BLOCK + m_LineInBlock) * VAG_SAMPLES_IN_LINE;
		return samples2ms(pos);
	}

	uint32 Decode(void* buffer)
	{
		if (!IsOpened()) return 0;

		if (m_CurrentBlock >= m_nNumberOfBlocks) return 0;

		// cache current ADPCM block
		if (!m_bBlockRead)
			ReadBlock(m_CurrentBlock);

		// trim the buffer size if we're at the end of our file
		int numberOfRequiredLines = GetBufferSamples() / m_nChannels / VAG_SAMPLES_IN_LINE;
		int numberOfRemainingLines = (m_nNumberOfBlocks - m_CurrentBlock) * NUM_VAG_LINES_IN_BLOCK - m_LineInBlock;
		int bufSizePerChannel = Min(numberOfRequiredLines, numberOfRemainingLines) * VAG_SAMPLES_IN_LINE * GetSampleSize();

		// calculate the pointers to individual channel buffers
		for (uint32 i = 0; i < m_nChannels; i++)
			m_ppPcmBuffers[i] = (int16*)((int8*)buffer + bufSizePerChannel * i);

		int size = 0;
		while (size < bufSizePerChannel)
		{
			// decode the VAG lines
			for (uint32 i = 0; i < m_nChannels; i++)
			{
				m_pVagDecoders[i].Decode(m_ppVagBuffers[i] + m_LineInBlock * VAG_LINE_SIZE, m_ppPcmBuffers[i], VAG_LINE_SIZE);
				m_ppPcmBuffers[i] += VAG_SAMPLES_IN_LINE;
			}
			size += VAG_SAMPLES_IN_LINE * GetSampleSize();
			m_LineInBlock++;

			// block is over, read the next block
			if (m_LineInBlock >= NUM_VAG_LINES_IN_BLOCK)
			{
				m_CurrentBlock++;
				if (m_CurrentBlock >= m_nNumberOfBlocks) // end of file
					break;
				m_LineInBlock = 0;
				ReadBlock();
			}
		}

		return bufSizePerChannel * m_nChannels;
	}
};
#ifdef AUDIO_OAL_USE_OPUS
class COpusFile : public IDecoder
{
	OggOpusFile *m_FileH;
	bool m_bOpened;
	uint32 m_nRate;
	uint32 m_nChannels;
public:
	COpusFile(const char *path) : m_FileH(nil),
		m_bOpened(false),
		m_nRate(0),
		m_nChannels(0)
	{
		int ret;
		m_FileH = op_open_file(path, &ret);

		if (m_FileH) {
			m_nChannels = op_head(m_FileH, 0)->channel_count;
			m_nRate = 48000;
			const OpusTags *tags = op_tags(m_FileH, 0);
			for (int i = 0; i < tags->comments; i++) {
				if (strncmp(tags->user_comments[i], "SAMPLERATE", sizeof("SAMPLERATE")-1) == 0)
				{
					sscanf(tags->user_comments[i], "SAMPLERATE=%i", &m_nRate);
					break;
				}
			}
			
			m_bOpened = true;
		}
	}

	void FileOpen()
	{
	}
	
	~COpusFile()
	{
		if (m_FileH)
		{
			op_free(m_FileH);
			m_FileH = nil;
		}
	}
	
	bool IsOpened()
	{
		return m_bOpened;
	}
	
	uint32 GetSampleSize()
	{
		return sizeof(uint16);
	}
	
	uint32 GetSampleCount()
	{
		if ( !IsOpened() ) return 0;
		return op_pcm_total(m_FileH, 0);
	}
	
	uint32 GetSampleRate()
	{
		return m_nRate;
	}
	
	uint32 GetChannels()
	{
		return m_nChannels;
	}
	
	void Seek(uint32 milliseconds)
	{
		if ( !IsOpened() ) return;
		op_pcm_seek(m_FileH, ms2samples(milliseconds) / GetChannels());
	}
	
	uint32 Tell()
	{
		if ( !IsOpened() ) return 0;
		return samples2ms(op_pcm_tell(m_FileH) * GetChannels());
	}
	
	uint32 Decode(void *buffer)
	{
		if ( !IsOpened() ) return 0;

		int size = op_read(m_FileH, (opus_int16 *)buffer, GetBufferSamples(), NULL);

		if (size < 0)
			return 0;

		if (GetChannels() == 2)
			SortStereoBuffer.SortStereo(buffer, size * m_nChannels * GetSampleSize());

		return size * m_nChannels * GetSampleSize();
	}
};
#endif


// For multi-thread: Someone always acquire stream's mutex before entering here
void
CStream::BuffersShouldBeFilled()
{
#ifdef MULTITHREADED_AUDIO
	if (MusicManager.m_nMusicMode != MUSICMODE_CUTSCENE) {
		std::queue<std::pair<ALuint, ALuint>> tempQueue;
		for(int i = 0; i < NUM_STREAMBUFFERS / 2; i++) {
			tempQueue.push(std::pair<ALuint, ALuint>(m_alBuffers[i * 2], m_alBuffers[i * 2 + 1]));
		}
		m_fillBuffers.swap(tempQueue);

		FlagAsToBeProcessed();

		m_bActive = true; // to allow Update() to queue the filled buffers & play
		return;
	}
	std::queue<std::pair<ALuint, ALuint>>().swap(m_fillBuffers);
#endif
	if ( FillBuffers() != 0 )
	{
		SetPlay(true);
	}
}

// returns whether it's queued (not on multi-thread)
bool
CStream::BufferShouldBeFilledAndQueued(std::pair<ALuint, ALuint>* bufs)
{
#ifdef MULTITHREADED_AUDIO
	if (MusicManager.m_nMusicMode != MUSICMODE_CUTSCENE)
		m_fillBuffers.push(*bufs);
	else
#endif
	{
		ALuint alBuffers[2] = {(*bufs).first, (*bufs).second}; // left - right
		if (FillBuffer(alBuffers)) {
			alSourceQueueBuffers(m_pAlSources[0], 1, &alBuffers[0]);
			alSourceQueueBuffers(m_pAlSources[1], 1, &alBuffers[1]);
			return true;
		}
	}
	return false;
}

#ifdef MULTITHREADED_AUDIO
void
CStream::FlagAsToBeProcessed(bool close)
{
	if (!close && MusicManager.m_nMusicMode == MUSICMODE_CUTSCENE)
		return;

	gAudioThreadQueueMutex.lock();
	if (close)
		gStreamsToClose.push(std::pair<IDecoder*, void*>(m_pSoundFile ? m_pSoundFile : nil, m_pBuffer ? m_pBuffer : nil));
	else
		gStreamsToProcess.push(this);
	
	gAudioThreadQueueMutex.unlock();

	gAudioThreadCv.notify_one();
}

void audioFileOpsThread()
{
	do
	{
		CStream *stream;
		{
			// Just a semaphore
			std::unique_lock<std::mutex> queueMutex(gAudioThreadQueueMutex);
			gAudioThreadCv.wait(queueMutex, [] { return gStreamsToProcess.size() > 0 || gStreamsToClose.size() > 0 || gAudioThreadTerm; });
			if (gAudioThreadTerm)
				return;

			if (!gStreamsToClose.empty()) {
				auto streamToClose = gStreamsToClose.front();
				gStreamsToClose.pop();
				if (streamToClose.first) { // pSoundFile
					delete streamToClose.first;
				}

				if (streamToClose.second) { // pBuffer
					free(streamToClose.second);
				}
			}

			if (!gStreamsToProcess.empty()) {
				stream = gStreamsToProcess.front();
				gStreamsToProcess.pop();
			} else
				continue;
		}

		std::unique_lock<std::mutex> lock(stream->m_mutex);

		std::pair<ALuint, ALuint> buffers, *lastBufAddr;
		bool insertBufsAfterCheck = false;

		do {
			if (!stream->IsOpened()) {
				break;
			}

			if (stream->m_bReset)
				break;

			// We gave up this idea for now
			/*
			stream->m_pSoundFile->FileOpen();

			// Deffered allocation, do it now
			if (stream->m_pBuffer == nil) {
				stream->m_pBuffer = malloc(stream->m_pSoundFile->GetBufferSize());
				ASSERT(stream->m_pBuffer != nil);
			}
			*/

			if (stream->m_bDoSeek) {
				stream->m_bDoSeek = false;
				int pos = stream->m_SeekPos;
				lock.unlock();
				stream->m_pSoundFile->Seek(pos);
				lock.lock();

				continue; // let's do the checks again, make sure we didn't miss anything while Seeking
			}

			if (insertBufsAfterCheck) {
				stream->m_queueBuffers.push(buffers);
				insertBufsAfterCheck = false;
			}

			if (!stream->m_fillBuffers.empty()) {
				lastBufAddr = &stream->m_fillBuffers.front();
				buffers = *lastBufAddr;
				lock.unlock();

				ALuint alBuffers[2] = {buffers.first, buffers.second}; // left - right
				bool filled = stream->FillBuffer(alBuffers);
				
				lock.lock();
			
				// Make sure queue isn't touched after we released mutex
				if (!stream->m_fillBuffers.empty() && lastBufAddr == &stream->m_fillBuffers.front()) {
					stream->m_fillBuffers.pop();
					if (filled)
						insertBufsAfterCheck = true; // Also make sure stream's properties aren't changed. So make one more pass, and push it to m_queueBuffers only if it pass checks again.
				}
			} else
				break;

		} while (true);

	} while(true);
}
#endif

void CStream::Initialise()
{
#ifdef AUDIO_OAL_USE_MPG123
	mpg123_init();
#endif
#ifdef MULTITHREADED_AUDIO
	gAudioThread = std::thread(audioFileOpsThread);
#endif
}

void CStream::Terminate()
{
#ifdef AUDIO_OAL_USE_MPG123
	mpg123_exit();
#endif
#ifdef MULTITHREADED_AUDIO
	gAudioThreadQueueMutex.lock();
	gAudioThreadTerm = true;
	gAudioThreadQueueMutex.unlock();

	gAudioThreadCv.notify_one();
	gAudioThread.join();
#endif
}

CStream::CStream(ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS]) :
	m_pAlSources(sources),
	m_alBuffers(buffers),
	m_pBuffer(nil),
	m_bPaused(false),
	m_bActive(false),
#ifdef MULTITHREADED_AUDIO
	m_bIExist(false),
	m_bDoSeek(false),
	m_SeekPos(0),
#endif
	m_pSoundFile(nil),
	m_bReset(false),
	m_nVolume(0),
	m_nPan(0),
	m_nPosBeforeReset(0),
	m_nLoopCount(1)
	
{
}

bool CStream::Open(const char* filename, uint32 overrideSampleRate)
{
	if (IsOpened()) return false;

#ifdef MULTITHREADED_AUDIO
	std::unique_lock<std::mutex> lock(m_mutex);

	m_bDoSeek = false;
	m_SeekPos = 0;
#endif

	m_bPaused = false;
	m_bActive = false;
	m_bReset = false;
	m_nVolume = 0;
	m_nPan = 0;
	m_nPosBeforeReset = 0;
	m_nLoopCount = 1;

// Be case-insensitive on linux (from https://github.com/OneSadCookie/fcaseopen/)
#if !defined(_WIN32)
	char *real = casepath(filename);
	if (real) {
		strcpy(m_aFilename, real);
		free(real);
	} else {
#else
	{
#endif
		strcpy(m_aFilename, filename);
	}
		
	DEV("Stream %s\n", m_aFilename);

	if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".wav")], ".wav"))
#ifdef AUDIO_OAL_USE_SNDFILE
		m_pSoundFile = new CSndFile(m_aFilename);
#else
		m_pSoundFile = new CWavFile(m_aFilename);
#endif
#ifdef AUDIO_OAL_USE_MPG123
	else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".mp3")], ".mp3"))
		m_pSoundFile = new CMP3File(m_aFilename);
	else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".adf")], ".adf"))
		m_pSoundFile = new CADFFile(m_aFilename);
#endif
	else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".vb")], ".VB"))
		m_pSoundFile = new CVbFile(m_aFilename, overrideSampleRate);
#ifdef AUDIO_OAL_USE_OPUS
	else if (!strcasecmp(&m_aFilename[strlen(m_aFilename) - strlen(".opus")], ".opus"))
		m_pSoundFile = new COpusFile(m_aFilename);
#endif
	else 
		m_pSoundFile = nil;

	if ( m_pSoundFile && m_pSoundFile->IsOpened() )
	{
		uint32 bufSize = m_pSoundFile->GetBufferSize();
		if(bufSize != 0) { // Otherwise it's deferred
			m_pBuffer = malloc(bufSize);
			ASSERT(m_pBuffer != nil);

			DEV("AvgSamplesPerSec: %d\n", m_pSoundFile->GetAvgSamplesPerSec());
			DEV("SampleCount: %d\n",      m_pSoundFile->GetSampleCount());
			DEV("SampleRate: %d\n",       m_pSoundFile->GetSampleRate());
			DEV("Channels: %d\n",         m_pSoundFile->GetChannels());
			DEV("Buffer Samples: %d\n",   m_pSoundFile->GetBufferSamples());
			DEV("Buffer sec: %f\n",       (float(m_pSoundFile->GetBufferSamples()) / float(m_pSoundFile->GetChannels())/ float(m_pSoundFile->GetSampleRate())));
			DEV("Length MS: %02d:%02d\n", (m_pSoundFile->GetLength() / 1000) / 60, (m_pSoundFile->GetLength() / 1000) % 60);
		}
#ifdef MULTITHREADED_AUDIO
		m_bIExist = true;
#endif
		return true;
	}
	return false;
}

CStream::~CStream()
{
	assert(!IsOpened());
}

void CStream::Close()
{
	if(!IsOpened()) return;

#ifdef MULTITHREADED_AUDIO
	{
		std::lock_guard<std::mutex> lock(m_mutex);

		Stop();
		ClearBuffers();
		m_bIExist = false;
		std::queue<std::pair<ALuint, ALuint>>().swap(m_fillBuffers);
		tsQueue<std::pair<ALuint, ALuint>>().swapNts(m_queueBuffers); // TSness not required, mutex is acquired
	}

	FlagAsToBeProcessed(true);
#else

	Stop();
	ClearBuffers();

	if ( m_pSoundFile )
	{
		delete m_pSoundFile;
		m_pSoundFile = nil;
	}

	if ( m_pBuffer )
	{
		free(m_pBuffer);
		m_pBuffer = nil;
	}
#endif
}

bool CStream::HasSource()
{
	return (m_pAlSources[0] != AL_NONE) && (m_pAlSources[1] != AL_NONE);
}

// m_bIExist only written in main thread, thus mutex is not needed on main thread
bool CStream::IsOpened()
{
#ifdef MULTITHREADED_AUDIO
	return m_bIExist;
#else
	return m_pSoundFile && m_pSoundFile->IsOpened();
#endif
}

bool CStream::IsPlaying()
{
	if ( !HasSource() || !IsOpened() ) return false;
	
	if ( !m_bPaused )
	{
		ALint sourceState[2];
		alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState[0]);
		alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState[1]);
		if (sourceState[0] == AL_PLAYING || sourceState[1] == AL_PLAYING)
			return true;

#ifdef MULTITHREADED_AUDIO
		std::lock_guard<std::mutex> lock(m_mutex);

		// Streams are designed in such a way that m_fillBuffers and m_queueBuffers will be *always* filled if audio is playing, and mutex is acquired
		if (!m_fillBuffers.empty() || !m_queueBuffers.emptyNts())
			return true;
#endif
	}
	
	return false;
}

void CStream::Pause()
{
	if ( !HasSource() ) return;
	ALint sourceState = AL_PAUSED;
	alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
	if (sourceState != AL_PAUSED)
		alSourcePause(m_pAlSources[0]);
	alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
	if (sourceState != AL_PAUSED)
		alSourcePause(m_pAlSources[1]);
}

void CStream::SetPause(bool bPause)
{
	if ( !HasSource() ) return;
	if ( bPause )
	{
		Pause();
		m_bPaused = true;
	}
	else
	{
		if (m_bPaused)
			SetPlay(true);
		m_bPaused = false;
	}
}

void CStream::SetPitch(float pitch)
{
	if ( !HasSource() ) return;
	alSourcef(m_pAlSources[0], AL_PITCH, pitch);
	alSourcef(m_pAlSources[1], AL_PITCH, pitch);
}

void CStream::SetGain(float gain)
{
	if ( !HasSource() ) return;
	alSourcef(m_pAlSources[0], AL_GAIN, gain);
	alSourcef(m_pAlSources[1], AL_GAIN, gain);
}

void CStream::SetPosition(int i, float x, float y, float z)
{
	if ( !HasSource() ) return;
	alSource3f(m_pAlSources[i], AL_POSITION, x, y, z);
}

void CStream::SetVolume(uint32 nVol)
{
	m_nVolume = nVol;
	SetGain(ALfloat(nVol) / MAX_VOLUME);
}

void CStream::SetPan(uint8 nPan)
{
	m_nPan = clamp((int8)nPan - 63, 0, 63);
	SetPosition(0, (m_nPan - 63) / 64.0f, 0.0f, Sqrt(1.0f - SQR((m_nPan - 63) / 64.0f)));

	m_nPan = clamp((int8)nPan + 64, 64, 127);
	SetPosition(1, (m_nPan - 63) / 64.0f, 0.0f, Sqrt(1.0f - SQR((m_nPan - 63) / 64.0f)));

	m_nPan = nPan;
}

// Should only be called if source is stopped
void CStream::SetPosMS(uint32 nPos)
{
	if ( !IsOpened() ) return;
	
#ifdef MULTITHREADED_AUDIO
	std::lock_guard<std::mutex> lock(m_mutex);

	std::queue<std::pair<ALuint, ALuint>>().swap(m_fillBuffers);
	tsQueue<std::pair<ALuint, ALuint>>().swapNts(m_queueBuffers); // TSness not required, second thread always access it when stream mutex acquired

	if (MusicManager.m_nMusicMode != MUSICMODE_CUTSCENE) {
		m_bDoSeek = true;
		m_SeekPos = nPos;
	} else
#endif	
	{
		m_pSoundFile->Seek(nPos);
	}	
	ClearBuffers();
	
	// adding to gStreamsToProcess not needed, someone always calls Start() / BuffersShouldBeFilled() after SetPosMS
}

uint32 CStream::GetPosMS()
{
	if ( !HasSource() ) return 0;
	if ( !IsOpened() ) return 0;
	
	// Deferred init causes division by zero
	if (m_pSoundFile->GetChannels() == 0)
		return 0;

	ALint offset;
	//alGetSourcei(m_alSource, AL_SAMPLE_OFFSET, &offset);
	alGetSourcei(m_pAlSources[0], AL_BYTE_OFFSET, &offset);

	//std::lock_guard<std::mutex> lock(m_mutex);

	return m_pSoundFile->Tell()
		- m_pSoundFile->samples2ms(m_pSoundFile->GetBufferSamples() * (NUM_STREAMBUFFERS/2-1)) / m_pSoundFile->GetChannels()
		+ m_pSoundFile->samples2ms(offset/m_pSoundFile->GetSampleSize()) / m_pSoundFile->GetChannels();
}

uint32 CStream::GetLengthMS()
{
	if ( !IsOpened() ) return 0;
	return m_pSoundFile->GetLength();
}

bool CStream::FillBuffer(ALuint *alBuffer)
{
#ifndef MULTITHREADED_AUDIO
	if ( !HasSource() )
		return false;
	if ( !IsOpened() )
		return false;
	if ( !(alBuffer[0] != AL_NONE && alIsBuffer(alBuffer[0])) )
		return false;
	if ( !(alBuffer[1] != AL_NONE && alIsBuffer(alBuffer[1])) )
		return false;
#endif

	uint32 size = m_pSoundFile->Decode(m_pBuffer);
	if( size == 0 )
		return false;

	uint32 channelSize = size / m_pSoundFile->GetChannels();
	
	alBufferData(alBuffer[0], AL_FORMAT_MONO16, m_pBuffer, channelSize, m_pSoundFile->GetSampleRate());
	// TODO: use just one buffer if we play mono
	if (m_pSoundFile->GetChannels() == 1)
		alBufferData(alBuffer[1], AL_FORMAT_MONO16, m_pBuffer, channelSize, m_pSoundFile->GetSampleRate());
	else
		alBufferData(alBuffer[1], AL_FORMAT_MONO16, (uint8*)m_pBuffer + channelSize, channelSize, m_pSoundFile->GetSampleRate());
	return true;
}

#ifdef MULTITHREADED_AUDIO
bool CStream::QueueBuffers()
{
	bool buffersQueued = false;
	std::pair<ALuint, ALuint> buffers;
	while (m_queueBuffers.peekPop(&buffers)) // beware: m_queueBuffers is tsQueue
	{
		ALuint leftBuf = buffers.first;
		ALuint rightBuf = buffers.second;

		alSourceQueueBuffers(m_pAlSources[0], 1, &leftBuf);
		alSourceQueueBuffers(m_pAlSources[1], 1, &rightBuf);

		buffersQueued = true;
	}
	return buffersQueued;	
}
#endif

// Only used in single-threaded audio or cutscene audio
int32 CStream::FillBuffers()
{
	int32 i = 0;
	for ( i = 0; i < NUM_STREAMBUFFERS/2; i++ )
	{
		if ( !FillBuffer(&m_alBuffers[i*2]) )
			break;
		alSourceQueueBuffers(m_pAlSources[0], 1, &m_alBuffers[i*2]);
		alSourceQueueBuffers(m_pAlSources[1], 1, &m_alBuffers[i*2+1]);
	}
	
	return i;
}

void CStream::ClearBuffers()
{
	if ( !HasSource() ) return;
	
	ALint buffersQueued[2];
	alGetSourcei(m_pAlSources[0], AL_BUFFERS_QUEUED, &buffersQueued[0]);
	alGetSourcei(m_pAlSources[1], AL_BUFFERS_QUEUED, &buffersQueued[1]);

	ALuint value;
	while (buffersQueued[0]--)
		alSourceUnqueueBuffers(m_pAlSources[0], 1, &value);
	while (buffersQueued[1]--)
		alSourceUnqueueBuffers(m_pAlSources[1], 1, &value);
}

bool CStream::Setup(bool imSureQueueIsEmpty, bool lock)
{
	if ( IsOpened() )
	{
#ifdef MULTITHREADED_AUDIO
		if (lock)
			m_mutex.lock();
#endif

		if (!imSureQueueIsEmpty) {
			Stop();
			ClearBuffers();
		}
#ifdef MULTITHREADED_AUDIO
		if (MusicManager.m_nMusicMode == MUSICMODE_CUTSCENE) {
			m_pSoundFile->Seek(0);
		} else {
			m_bDoSeek = true;
			m_SeekPos = 0;
		}

		if (lock)
			m_mutex.unlock();
#else
		m_pSoundFile->Seek(0);
#endif

		//SetPosition(0.0f, 0.0f, 0.0f);
		SetPitch(1.0f);
		//SetPan(m_nPan);
		//SetVolume(100);
	}
	
	return IsOpened();
}

void CStream::SetLoopCount(int32 count)
{
	if ( !HasSource() ) return;

	m_nLoopCount = count;
}

void CStream::SetPlay(bool state)
{
	if ( !HasSource() ) return;
	if ( state )
	{
		ALint sourceState = AL_PLAYING;
		alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
		if (sourceState != AL_PLAYING )
			alSourcePlay(m_pAlSources[0]);

		sourceState = AL_PLAYING;
		alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
		if (sourceState != AL_PLAYING)
			alSourcePlay(m_pAlSources[1]);

		m_bActive = true;
	}
	else
	{
		ALint sourceState = AL_STOPPED;
		alGetSourcei(m_pAlSources[0], AL_SOURCE_STATE, &sourceState);
		if (sourceState != AL_STOPPED )
			alSourceStop(m_pAlSources[0]);

		sourceState = AL_STOPPED;
		alGetSourcei(m_pAlSources[1], AL_SOURCE_STATE, &sourceState);
		if (sourceState != AL_STOPPED)
			alSourceStop(m_pAlSources[1]);

		m_bActive = false;
	}
}

void CStream::Start()
{
	if ( !HasSource() ) return;
	
#ifdef MULTITHREADED_AUDIO
	std::lock_guard<std::mutex> lock(m_mutex);
	tsQueue<std::pair<ALuint, ALuint>>().swapNts(m_queueBuffers); // TSness not required, second thread always access it when stream mutex acquired
#endif
	BuffersShouldBeFilled();
}

void CStream::Stop()
{
	if ( !HasSource() ) return;
	SetPlay(false);
}

void CStream::Update()
{
	if ( !IsOpened() )
		return;
	
	if ( !HasSource() )
		return;
	
	if ( m_bReset )
		return;
	
	if ( !m_bPaused )
	{
		
		bool buffersQueuedAndStarted = false;
		bool buffersQueuedButNotStarted = false;
#ifdef MULTITHREADED_AUDIO
		// Put it in here because we need totalBuffers after queueing to decide when to loop audio
		if (m_bActive)
		{
			buffersQueuedAndStarted = QueueBuffers();
			if(buffersQueuedAndStarted) {
				SetPlay(true);
			}
		}
#endif

		ALint totalBuffers[2] = {0, 0};
		ALint buffersProcessed[2] = {0, 0};

		// Relying a lot on left buffer states in here

		do
		{
			//alSourcef(m_pAlSources[0], AL_ROLLOFF_FACTOR, 0.0f);
			alGetSourcei(m_pAlSources[0], AL_BUFFERS_QUEUED, &totalBuffers[0]);
			alGetSourcei(m_pAlSources[0], AL_BUFFERS_PROCESSED, &buffersProcessed[0]);
			//alSourcef(m_pAlSources[1], AL_ROLLOFF_FACTOR, 0.0f);
			alGetSourcei(m_pAlSources[1], AL_BUFFERS_QUEUED, &totalBuffers[1]);
			alGetSourcei(m_pAlSources[1], AL_BUFFERS_PROCESSED, &buffersProcessed[1]);
		} while (buffersProcessed[0] != buffersProcessed[1]);

		assert(buffersProcessed[0] == buffersProcessed[1]);

		// Correcting OpenAL concepts here:
		// AL_BUFFERS_QUEUED = Number of *all* buffers in queue, including processed, processing and pending
		// AL_BUFFERS_PROCESSED = Index of the buffer being processing right now. Buffers coming after that(have greater index) are pending buffers.
		// which means: totalBuffers[0] - buffersProcessed[0] = pending buffers
		
		// We should wait queue to be cleared to loop track, because position calculation relies on queue.
		if (m_nLoopCount != 1 && m_bActive && totalBuffers[0] == 0)
		{
#ifdef MULTITHREADED_AUDIO
			std::lock_guard<std::mutex> lock(m_mutex);

			if (m_fillBuffers.empty() && m_queueBuffers.emptyNts()) // we already acquired stream mutex, which is enough for second thread. thus Nts variant
#endif
			{
				Setup(true, false);
				BuffersShouldBeFilled(); // will also call SetPlay(true)
				if (m_nLoopCount != 0)
					m_nLoopCount--;
			}
		}
		else
		{
			static std::queue<std::pair<ALuint, ALuint>> tempFillBuffer;

			while ( buffersProcessed[0]-- )
			{
				ALuint buffer[2];
				
				alSourceUnqueueBuffers(m_pAlSources[0], 1, &buffer[0]);
				alSourceUnqueueBuffers(m_pAlSources[1], 1, &buffer[1]);

				if (m_bActive)
				{
					tempFillBuffer.push(std::pair<ALuint, ALuint>(buffer[0], buffer[1]));
				}
			}

			if (m_bActive && buffersProcessed[1])
			{
#ifdef MULTITHREADED_AUDIO
				m_mutex.lock();
#endif
				while (!tempFillBuffer.empty()) {
					auto elem = tempFillBuffer.front();
					tempFillBuffer.pop();
					buffersQueuedButNotStarted = BufferShouldBeFilledAndQueued(&elem);
				}
#ifdef MULTITHREADED_AUDIO
				m_mutex.unlock();
				FlagAsToBeProcessed();
#endif

			}
		}

		// Source may be starved to audio and stopped itself
		if (m_bActive && !buffersQueuedAndStarted && (buffersQueuedButNotStarted || (totalBuffers[1] - buffersProcessed[1] != 0)))
			SetPlay(true);
	}
}

void CStream::ProviderInit()
{
	if ( m_bReset )
	{
		if ( Setup(true, false) ) // lock not needed, thread can't process streams with m_bReset set
		{
			SetPan(m_nPan);
			SetVolume(m_nVolume);
			SetLoopCount(m_nLoopCount);
			SetPosMS(m_nPosBeforeReset);
#ifdef MULTITHREADED_AUDIO
			std::unique_lock<std::mutex> lock(m_mutex);
#endif
			if(m_bActive)
				BuffersShouldBeFilled();

			if (m_bPaused)
				Pause();

			m_bReset = false;

		} else {
#ifdef MULTITHREADED_AUDIO
			std::unique_lock<std::mutex> lock(m_mutex);
#endif
			m_bReset = false;
		}
	}
}

void CStream::ProviderTerm()
{
#ifdef MULTITHREADED_AUDIO
	std::lock_guard<std::mutex> lock(m_mutex);

	// unlike Close() we will reuse this stream, so clearing queues are important.
	std::queue<std::pair<ALuint, ALuint>>().swap(m_fillBuffers);
	tsQueue<std::pair<ALuint, ALuint>>().swapNts(m_queueBuffers); // stream mutex is already acquired, thus Nts variant
#endif
	m_bReset = true;
	m_nPosBeforeReset = GetPosMS();

	Stop();
	ClearBuffers();
}
	
#endif