Import Cai1Hsu/re3 @ miami (reVC / GTA:VC decompilation)HEADmiami

Snapshot import (no upstream history) into git.ancap.in.ua/claude, per @lzcnt.
Source: https://github.com/Cai1Hsu/re3 branch miami.

8 files changed, 2468 insertions, 0 deletions

diff --git a/src/audio/oal/aldlist.cpp b/src/audio/oal/aldlist.cpp
new file mode 100644
index 0000000..881418c
--- /dev/null
+++ b/src/audio/oal/aldlist.cpp
@@ -0,0 +1,305 @@
+/*
+ * Copyright (c) 2006, Creative Labs Inc.
+ * All rights reserved.
+ * 
+ * Redistribution and use in source and binary forms, with or without modification, are permitted provided
+ * that the following conditions are met:
+ * 
+ *     * Redistributions of source code must retain the above copyright notice, this list of conditions and
+ * 	     the following disclaimer.
+ *     * Redistributions in binary form must reproduce the above copyright notice, this list of conditions
+ * 	     and the following disclaimer in the documentation and/or other materials provided with the distribution.
+ *     * Neither the name of Creative Labs Inc. nor the names of its contributors may be used to endorse or
+ * 	     promote products derived from this software without specific prior written permission.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
+ * PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED
+ * TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ * POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "aldlist.h"
+
+#ifndef _WIN32
+#define _stricmp strcasecmp
+#define _strnicmp strncasecmp
+#define _strdup strdup
+#endif
+
+#ifdef AUDIO_OAL
+/* 
+ * Init call
+ */
+ALDeviceList::ALDeviceList()
+{
+	char *devices;
+	int index;
+	const char *defaultDeviceName;
+	const char *actualDeviceName;
+
+	// DeviceInfo vector stores, for each enumerated device, it's device name, selection status, spec version #, and extension support
+	nNumOfDevices = 0;
+
+	defaultDeviceIndex = 0;
+
+	if (alcIsExtensionPresent(NULL, "ALC_ENUMERATION_EXT")) {
+		devices = (char *)alcGetString(NULL, ALC_DEVICE_SPECIFIER);
+		defaultDeviceName = (char *)alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
+		
+		index = 0;
+		// go through device list (each device terminated with a single NULL, list terminated with double NULL)
+		while (*devices != '\0') {
+			if (strcmp(defaultDeviceName, devices) == 0) {
+				defaultDeviceIndex = index;
+			}
+			ALCdevice *device = alcOpenDevice(devices);
+			if (device) {
+				ALCcontext *context = alcCreateContext(device, NULL);
+				if (context) {
+					alcMakeContextCurrent(context);
+					// if new actual device name isn't already in the list, then add it...
+					actualDeviceName = alcGetString(device, ALC_DEVICE_SPECIFIER);
+					bool bNewName = true;
+					for (unsigned int i = 0; i < GetNumDevices(); i++) {
+						if (strcmp(GetDeviceName(i), actualDeviceName) == 0) {
+							bNewName = false;
+						}
+					}
+					if ((bNewName) && (actualDeviceName != NULL) && (strlen(actualDeviceName) > 0)) {
+						ALDEVICEINFO ALDeviceInfo;
+						ALDeviceInfo.bSelected = true;
+						ALDeviceInfo.strDeviceName = _strdup(actualDeviceName);
+						alcGetIntegerv(device, ALC_MAJOR_VERSION, sizeof(int), &ALDeviceInfo.iMajorVersion);
+						alcGetIntegerv(device, ALC_MINOR_VERSION, sizeof(int), &ALDeviceInfo.iMinorVersion);
+
+						// Check for ALC Extensions
+						if (alcIsExtensionPresent(device, "ALC_EXT_CAPTURE") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EXT_CAPTURE;
+						if (alcIsExtensionPresent(device, "ALC_EXT_EFX") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EXT_EFX;
+
+						// Check for AL Extensions
+						if (alIsExtensionPresent("AL_EXT_OFFSET") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EXT_OFFSET;
+
+						if (alIsExtensionPresent("AL_EXT_LINEAR_DISTANCE") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EXT_LINEAR_DISTANCE;
+						if (alIsExtensionPresent("AL_EXT_EXPONENT_DISTANCE") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EXT_EXPONENT_DISTANCE;
+						
+						if (alIsExtensionPresent("EAX2.0") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EAX2;
+						if (alIsExtensionPresent("EAX3.0") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EAX3;
+						if (alIsExtensionPresent("EAX4.0") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EAX4;
+						if (alIsExtensionPresent("EAX5.0") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EAX5;
+
+						if (alIsExtensionPresent("EAX-RAM") == AL_TRUE)
+							ALDeviceInfo.Extensions |= ADEXT_EAX_RAM;
+
+						// Get Source Count
+						ALDeviceInfo.uiSourceCount = GetMaxNumSources();
+
+						aDeviceInfo[nNumOfDevices++] = ALDeviceInfo;
+					}
+					alcMakeContextCurrent(NULL);
+					alcDestroyContext(context);
+				}
+				alcCloseDevice(device);
+			}
+			devices += strlen(devices) + 1;
+			index += 1;
+		}
+	}
+
+	ResetFilters();
+}
+
+/* 
+ * Exit call
+ */
+ALDeviceList::~ALDeviceList()
+{
+}
+
+/*
+ * Returns the number of devices in the complete device list
+ */
+unsigned int ALDeviceList::GetNumDevices()
+{
+	return nNumOfDevices;
+}
+
+/* 
+ * Returns the device name at an index in the complete device list
+ */
+const char * ALDeviceList::GetDeviceName(unsigned int index)
+{
+	if (index < GetNumDevices())
+		return aDeviceInfo[index].strDeviceName;
+	else
+		return NULL;
+}
+
+/*
+ * Returns the major and minor version numbers for a device at a specified index in the complete list
+ */
+void ALDeviceList::GetDeviceVersion(unsigned int index, int *major, int *minor)
+{
+	if (index < GetNumDevices()) {
+		if (major)
+			*major = aDeviceInfo[index].iMajorVersion;
+		if (minor)
+			*minor = aDeviceInfo[index].iMinorVersion;
+	}
+	return;
+}
+
+/*
+ * Returns the maximum number of Sources that can be generate on the given device
+ */
+unsigned int ALDeviceList::GetMaxNumSources(unsigned int index)
+{
+	if (index < GetNumDevices())
+		return aDeviceInfo[index].uiSourceCount;
+	else
+		return 0;
+}
+
+/*
+ * Checks if the extension is supported on the given device
+ */
+bool ALDeviceList::IsExtensionSupported(int index, unsigned short ext)
+{
+	return !!(aDeviceInfo[index].Extensions & ext);
+}
+
+/*
+ * returns the index of the default device in the complete device list
+ */
+int ALDeviceList::GetDefaultDevice()
+{
+	return defaultDeviceIndex;
+}
+
+/* 
+ * Deselects devices which don't have the specified minimum version
+ */
+void ALDeviceList::FilterDevicesMinVer(int major, int minor)
+{
+	int dMajor, dMinor;
+	for (unsigned int i = 0; i < nNumOfDevices; i++) {
+		GetDeviceVersion(i, &dMajor, &dMinor);
+		if ((dMajor < major) || ((dMajor == major) && (dMinor < minor))) {
+			aDeviceInfo[i].bSelected = false;
+		}
+	}
+}
+
+/* 
+ * Deselects devices which don't have the specified maximum version
+ */
+void ALDeviceList::FilterDevicesMaxVer(int major, int minor)
+{
+	int dMajor, dMinor;
+	for (unsigned int i = 0; i < nNumOfDevices; i++) {
+		GetDeviceVersion(i, &dMajor, &dMinor);
+		if ((dMajor > major) || ((dMajor == major) && (dMinor > minor))) {
+			aDeviceInfo[i].bSelected = false;
+		}
+	}
+}
+
+/*
+ * Deselects device which don't support the given extension name
+ */
+void
+ALDeviceList::FilterDevicesExtension(unsigned short ext)
+{
+	for (unsigned int i = 0; i < nNumOfDevices; i++) {
+		if (!IsExtensionSupported(i, ext))
+			aDeviceInfo[i].bSelected = false;
+	}
+}
+
+/*
+ * Resets all filtering, such that all devices are in the list
+ */
+void ALDeviceList::ResetFilters()
+{
+	for (unsigned int i = 0; i < GetNumDevices(); i++) {
+		aDeviceInfo[i].bSelected = true;
+	}
+	filterIndex = 0;
+}
+
+/*
+ * Gets index of first filtered device
+ */
+int ALDeviceList::GetFirstFilteredDevice()
+{
+	unsigned int i;
+
+	for (i = 0; i < GetNumDevices(); i++) {
+		if (aDeviceInfo[i].bSelected == true) {
+			break;
+		}
+	}
+	filterIndex = i + 1;
+	return i;
+}
+
+/*
+ * Gets index of next filtered device
+ */
+int ALDeviceList::GetNextFilteredDevice()
+{
+	unsigned int i;
+
+	for (i = filterIndex; i < GetNumDevices(); i++) {
+		if (aDeviceInfo[i].bSelected == true) {
+			break;
+		}
+	}
+	filterIndex = i + 1;
+	return i;
+}
+
+/*
+ * Internal function to detemine max number of Sources that can be generated
+ */
+unsigned int ALDeviceList::GetMaxNumSources()
+{
+	ALuint uiSources[256];
+	unsigned int iSourceCount = 0;
+
+	// Clear AL Error Code
+	alGetError();
+
+	// Generate up to 256 Sources, checking for any errors
+	for (iSourceCount = 0; iSourceCount < 256; iSourceCount++)
+	{
+		alGenSources(1, &uiSources[iSourceCount]);
+		if (alGetError() != AL_NO_ERROR)
+			break;
+	}
+
+	// Release the Sources
+	alDeleteSources(iSourceCount, uiSources);
+	if (alGetError() != AL_NO_ERROR)
+	{
+		for (unsigned int i = 0; i < 256; i++)
+		{
+			alDeleteSources(1, &uiSources[i]);
+		}
+	}
+
+	return iSourceCount;
+}
+#endif
diff --git a/src/audio/oal/aldlist.h b/src/audio/oal/aldlist.h
new file mode 100644
index 0000000..417bd31
--- /dev/null
+++ b/src/audio/oal/aldlist.h
@@ -0,0 +1,69 @@
+#ifndef ALDEVICELIST_H
+#define ALDEVICELIST_H
+
+#include "oal_utils.h"
+
+#ifdef AUDIO_OAL
+#pragma warning(disable: 4786)  //disable warning "identifier was truncated to '255' characters in the browser information"
+
+enum
+{
+	ADEXT_EXT_CAPTURE = (1 << 0),
+	ADEXT_EXT_EFX = (1 << 1),
+	ADEXT_EXT_OFFSET = (1 << 2),
+	ADEXT_EXT_LINEAR_DISTANCE = (1 << 3),
+	ADEXT_EXT_EXPONENT_DISTANCE = (1 << 4),
+	ADEXT_EAX2 = (1 << 5),
+	ADEXT_EAX3 = (1 << 6),
+	ADEXT_EAX4 = (1 << 7),
+	ADEXT_EAX5 = (1 << 8),
+	ADEXT_EAX_RAM = (1 << 9),
+};
+
+struct ALDEVICEINFO {
+	const char		*strDeviceName;
+	int				iMajorVersion;
+	int				iMinorVersion;
+	unsigned int	uiSourceCount;
+	unsigned short  Extensions;
+	bool			bSelected;
+
+	ALDEVICEINFO() : iMajorVersion(0), iMinorVersion(0), uiSourceCount(0), bSelected(false)
+	{
+		strDeviceName = NULL;
+		Extensions = 0;
+	}
+};
+
+typedef ALDEVICEINFO *LPALDEVICEINFO;
+
+class ALDeviceList
+{
+private:
+	ALDEVICEINFO aDeviceInfo[64];
+	unsigned int nNumOfDevices;
+	int defaultDeviceIndex;
+	int filterIndex;
+
+public:
+	ALDeviceList ();
+	~ALDeviceList ();
+	unsigned int GetNumDevices();
+	const char *GetDeviceName(unsigned int index);
+	void GetDeviceVersion(unsigned int index, int *major, int *minor);
+	unsigned int GetMaxNumSources(unsigned int index);
+	bool IsExtensionSupported(int index, unsigned short ext);
+	int GetDefaultDevice();
+	void FilterDevicesMinVer(int major, int minor);
+	void FilterDevicesMaxVer(int major, int minor);
+	void FilterDevicesExtension(unsigned short ext);
+	void ResetFilters();
+	int GetFirstFilteredDevice();
+	int GetNextFilteredDevice();
+
+private:
+	unsigned int GetMaxNumSources();
+};
+#endif
+
+#endif // ALDEVICELIST_H
diff --git a/src/audio/oal/channel.cpp b/src/audio/oal/channel.cpp
new file mode 100644
index 0000000..d1fd0ae
--- /dev/null
+++ b/src/audio/oal/channel.cpp
@@ -0,0 +1,278 @@
+#include "common.h"
+
+#ifdef AUDIO_OAL
+#include "channel.h"
+#include "sampman.h"
+
+#ifndef _WIN32
+#include <float.h>
+#endif
+
+extern bool IsFXSupported();
+
+ALuint alSources[MAXCHANNELS+MAX2DCHANNELS];
+ALuint alFilters[MAXCHANNELS+MAX2DCHANNELS];
+ALuint alBuffers[MAXCHANNELS+MAX2DCHANNELS];
+bool bChannelsCreated = false;
+
+int32 CChannel::channelsThatNeedService = 0;
+
+void
+CChannel::InitChannels()
+{
+	alGenSources(MAXCHANNELS+MAX2DCHANNELS, alSources);
+	alGenBuffers(MAXCHANNELS+MAX2DCHANNELS, alBuffers);
+	if (IsFXSupported())
+		alGenFilters(MAXCHANNELS + MAX2DCHANNELS, alFilters);
+	bChannelsCreated = true;
+}
+
+void
+CChannel::DestroyChannels()
+{
+	if (bChannelsCreated) 
+	{
+		alDeleteSources(MAXCHANNELS + MAX2DCHANNELS, alSources);
+		memset(alSources, 0, sizeof(alSources));
+		alDeleteBuffers(MAXCHANNELS + MAX2DCHANNELS, alBuffers);
+		memset(alBuffers, 0, sizeof(alBuffers));
+		if (IsFXSupported())
+		{
+			alDeleteFilters(MAXCHANNELS + MAX2DCHANNELS, alFilters);
+			memset(alFilters, 0, sizeof(alFilters));
+		}
+		bChannelsCreated = false;
+	}
+}
+
+
+CChannel::CChannel()
+{
+	Data = nil;
+	DataSize = 0;
+	SetDefault();
+}
+
+void CChannel::SetDefault()
+{
+	Pitch = 1.0f;
+	Gain = 1.0f;
+	Mix = 0.0f;
+		
+	Position[0] = 0.0f; Position[1] = 0.0f; Position[2] = 0.0f;
+	Distances[0] = 0.0f; Distances[1] = FLT_MAX;
+
+	LoopCount = 1;
+	LastProcessedOffset = UINT32_MAX;
+	LoopPoints[0] = 0; LoopPoints[1] = -1;
+	
+	Frequency = MAX_FREQ;
+}
+
+void CChannel::Reset()
+{
+	// Here is safe because ctor don't call this
+	if (LoopCount > 1)
+		channelsThatNeedService--;
+
+	ClearBuffer();
+	SetDefault();
+}
+
+void CChannel::Init(uint32 _id, bool Is2D)
+{
+	id = _id;
+	if ( HasSource() )
+	{
+		alSourcei(alSources[id], AL_SOURCE_RELATIVE, AL_TRUE);
+		if ( IsFXSupported() )
+			alSource3i(alSources[id], AL_AUXILIARY_SEND_FILTER, AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL);
+		
+		if ( Is2D )
+		{
+			alSource3f(alSources[id], AL_POSITION, 0.0f, 0.0f, 0.0f);
+			alSourcef(alSources[id], AL_GAIN, 1.0f);
+		}
+	}
+}
+
+void CChannel::Term()
+{
+	Stop();
+	if ( HasSource() )
+	{
+		if ( IsFXSupported() )
+		{
+			alSource3i(alSources[id], AL_AUXILIARY_SEND_FILTER, AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL);
+		}
+	}
+}
+
+void CChannel::Start()
+{
+	if ( !HasSource() ) return;
+	if ( !Data ) return;
+
+	alBufferData(alBuffers[id], AL_FORMAT_MONO16, Data, DataSize, Frequency);
+	if ( LoopPoints[0] != 0 && LoopPoints[0] != -1 )
+		alBufferiv(alBuffers[id], AL_LOOP_POINTS_SOFT, LoopPoints);
+	alSourcei(alSources[id], AL_BUFFER, alBuffers[id]);
+	alSourcePlay(alSources[id]);
+}
+
+void CChannel::Stop()
+{
+	if ( HasSource() )
+		alSourceStop(alSources[id]);
+	
+	Reset();
+}
+
+bool CChannel::HasSource()
+{
+	return alSources[id] != AL_NONE;
+}
+	
+bool CChannel::IsUsed()
+{
+	if ( HasSource() )
+	{
+		ALint sourceState;
+		alGetSourcei(alSources[id], AL_SOURCE_STATE, &sourceState);
+		return sourceState == AL_PLAYING;
+	}
+	return false;
+}
+
+void CChannel::SetPitch(float pitch)
+{
+	if ( !HasSource() ) return;
+	alSourcef(alSources[id], AL_PITCH, pitch);
+}
+
+void CChannel::SetGain(float gain)
+{
+	if ( !HasSource() ) return;
+	alSourcef(alSources[id], AL_GAIN, gain);
+}
+	
+void CChannel::SetVolume(int32 vol)
+{
+	SetGain(ALfloat(vol) / MAX_VOLUME);
+}
+
+void CChannel::SetSampleData(void *_data, size_t _DataSize, int32 freq)
+{
+	Data = _data;
+	DataSize = _DataSize;
+	Frequency = freq;
+}
+	
+void CChannel::SetCurrentFreq(uint32 freq)
+{
+	SetPitch(ALfloat(freq) / Frequency);
+}
+
+void CChannel::SetLoopCount(int32 count)
+{
+	if ( !HasSource() ) return;
+
+	// 0: loop indefinitely, 1: play one time, 2: play two times etc...
+	// only > 1 needs manual processing
+
+	if (LoopCount > 1 && count < 2)
+		channelsThatNeedService--;
+	else if (LoopCount < 2 && count > 1)
+		channelsThatNeedService++;
+
+	alSourcei(alSources[id], AL_LOOPING, count == 1 ? AL_FALSE : AL_TRUE);
+	LoopCount = count;
+}
+
+bool CChannel::Update()
+{
+	if (!HasSource()) return false;
+	if (LoopCount < 2) return false;
+
+	ALint state;
+	alGetSourcei(alSources[id], AL_SOURCE_STATE, &state);
+	if (state == AL_STOPPED) {
+		debug("Looping channels(%d in this case) shouldn't report AL_STOPPED, but nvm\n", id);
+		SetLoopCount(1);
+		return true;
+	}
+
+	assert(channelsThatNeedService > 0 && "Ref counting is broken");
+
+	ALint offset;
+	alGetSourcei(alSources[id], AL_SAMPLE_OFFSET, &offset);
+
+	// Rewound
+	if (offset < LastProcessedOffset) {
+		LoopCount--;
+		if (LoopCount == 1) {
+			// Playing last tune...
+			channelsThatNeedService--;
+			alSourcei(alSources[id], AL_LOOPING, AL_FALSE);
+		}
+	}
+	LastProcessedOffset = offset;
+	return true;
+}
+
+void CChannel::SetLoopPoints(ALint start, ALint end)
+{
+	LoopPoints[0] = start;
+	LoopPoints[1] = end;
+}
+	
+void CChannel::SetPosition(float x, float y, float z)
+{
+	if ( !HasSource() ) return;
+	alSource3f(alSources[id], AL_POSITION, x, y, z);
+}
+	
+void CChannel::SetDistances(float max, float min)
+{
+	if ( !HasSource() ) return;
+	alSourcef   (alSources[id], AL_MAX_DISTANCE,       max);
+	alSourcef   (alSources[id], AL_REFERENCE_DISTANCE, min);
+	alSourcef   (alSources[id], AL_MAX_GAIN, 1.0f);
+	alSourcef   (alSources[id], AL_ROLLOFF_FACTOR, 1.0f);
+}
+	
+void CChannel::SetPan(int32 pan)
+{
+	SetPosition((pan-63)/64.0f, 0.0f, Sqrt(1.0f-SQR((pan-63)/64.0f)));
+}
+
+void CChannel::ClearBuffer()
+{
+	if ( !HasSource() ) return;
+	alSourcei(alSources[id], AL_LOOPING, AL_FALSE);
+	alSourcei(alSources[id], AL_BUFFER, AL_NONE);
+	Data = nil;
+	DataSize = 0;
+}
+
+void CChannel::SetReverbMix(ALuint slot, float mix)
+{
+	if ( !IsFXSupported() ) return;
+	if ( !HasSource() ) return;
+	if ( alFilters[id] == AL_FILTER_NULL ) return;
+	
+	Mix = mix;
+	EAX3_SetReverbMix(alFilters[id], mix);
+	alSource3i(alSources[id], AL_AUXILIARY_SEND_FILTER, slot, 0, alFilters[id]);
+}
+
+void CChannel::UpdateReverb(ALuint slot)
+{
+	if ( !IsFXSupported() ) return;
+	if ( !HasSource() ) return;
+	if ( alFilters[id] == AL_FILTER_NULL ) return;
+	EAX3_SetReverbMix(alFilters[id], Mix);
+	alSource3i(alSources[id], AL_AUXILIARY_SEND_FILTER, slot, 0, alFilters[id]);
+}
+
+#endif
diff --git a/src/audio/oal/channel.h b/src/audio/oal/channel.h
new file mode 100644
index 0000000..b081be2
--- /dev/null
+++ b/src/audio/oal/channel.h
@@ -0,0 +1,54 @@
+#pragma once
+
+#ifdef AUDIO_OAL
+#include "oal/oal_utils.h"
+#include <AL/al.h>
+#include <AL/alext.h>
+#include <AL/efx.h>
+
+
+class CChannel
+{
+	uint32 id;
+	float  Pitch, Gain;
+	float  Mix;
+	void  *Data;
+	size_t DataSize;
+	int32  Frequency;
+	float  Position[3];
+	float  Distances[2];
+	int32  LoopCount;
+	ALint  LoopPoints[2];
+	ALint  LastProcessedOffset;
+public:
+	static int32 channelsThatNeedService;
+
+	static void InitChannels();
+	static void DestroyChannels();
+
+	CChannel();
+	void SetDefault();
+	void Reset();
+	void Init(uint32 _id, bool Is2D = false);
+	void Term();
+	void Start();
+	void Stop();
+	bool HasSource();
+	bool IsUsed();
+	void SetPitch(float pitch);
+	void SetGain(float gain);
+	void SetVolume(int32 vol);
+	void SetSampleData(void *_data, size_t _DataSize, int32 freq);
+	void SetCurrentFreq(uint32 freq);
+	void SetLoopCount(int32 count);
+	void SetLoopPoints(ALint start, ALint end);
+	void SetPosition(float x, float y, float z);
+	void SetDistances(float max, float min);
+	void SetPan(int32 pan);
+	void ClearBuffer();
+	void SetReverbMix(ALuint slot, float mix);
+	void UpdateReverb(ALuint slot);
+	bool Update();
+};
+
+#endif
\ No newline at end of file
diff --git a/src/audio/oal/oal_utils.cpp b/src/audio/oal/oal_utils.cpp
new file mode 100644
index 0000000..e4cb0b7
--- /dev/null
+++ b/src/audio/oal/oal_utils.cpp
@@ -0,0 +1,181 @@
+#include "common.h"
+#include "oal_utils.h"
+
+#ifdef AUDIO_OAL
+
+/*
+ * When linking to a static openal-soft library,
+ * the extension function inside the openal library conflict with the variables here.
+ * Therefore declare these re3 owned symbols in a private namespace.
+ */
+
+namespace re3_openal {
+
+LPALGENEFFECTS alGenEffects;
+LPALDELETEEFFECTS alDeleteEffects;
+LPALISEFFECT alIsEffect;
+LPALEFFECTI alEffecti;
+LPALEFFECTIV alEffectiv;
+LPALEFFECTF alEffectf;
+LPALEFFECTFV alEffectfv;
+LPALGETEFFECTI alGetEffecti;
+LPALGETEFFECTIV alGetEffectiv;
+LPALGETEFFECTF alGetEffectf;
+LPALGETEFFECTFV alGetEffectfv;
+LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
+LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
+LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
+LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
+LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
+LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
+LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
+LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
+LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
+LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
+LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
+LPALGENFILTERS alGenFilters;
+LPALDELETEFILTERS alDeleteFilters;
+LPALISFILTER alIsFilter;
+LPALFILTERI alFilteri;
+LPALFILTERIV alFilteriv;
+LPALFILTERF alFilterf;
+LPALFILTERFV alFilterfv;
+LPALGETFILTERI alGetFilteri;
+LPALGETFILTERIV alGetFilteriv;
+LPALGETFILTERF alGetFilterf;
+LPALGETFILTERFV alGetFilterfv;
+
+}
+
+using namespace re3_openal;
+
+void EFXInit()
+{
+	/* Define a macro to help load the function pointers. */
+#define LOAD_PROC(T, x)  ((x) = (T)alGetProcAddress(#x))
+	LOAD_PROC(LPALGENEFFECTS, alGenEffects);
+	LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
+	LOAD_PROC(LPALISEFFECT, alIsEffect);
+	LOAD_PROC(LPALEFFECTI, alEffecti);
+	LOAD_PROC(LPALEFFECTIV, alEffectiv);
+	LOAD_PROC(LPALEFFECTF, alEffectf);
+	LOAD_PROC(LPALEFFECTFV, alEffectfv);
+	LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
+	LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
+	LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
+	LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
+	
+	LOAD_PROC(LPALGENFILTERS, alGenFilters);
+	LOAD_PROC(LPALDELETEFILTERS, alDeleteFilters);
+	LOAD_PROC(LPALISFILTER, alIsFilter);
+	LOAD_PROC(LPALFILTERI, alFilteri);
+	LOAD_PROC(LPALFILTERIV, alFilteriv);
+	LOAD_PROC(LPALFILTERF, alFilterf);
+	LOAD_PROC(LPALFILTERFV, alFilterfv);
+	LOAD_PROC(LPALGETFILTERI, alGetFilteri);
+	LOAD_PROC(LPALGETFILTERIV, alGetFilteriv);
+	LOAD_PROC(LPALGETFILTERF, alGetFilterf);
+	LOAD_PROC(LPALGETFILTERFV, alGetFilterfv);
+	
+	LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
+	LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
+	LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
+	LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
+	LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
+	LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
+	LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
+	LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
+	LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
+	LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
+	LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
+#undef LOAD_PROC
+}
+
+void SetEffectsLevel(ALuint uiFilter, float level)
+{
+	alFilteri(uiFilter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
+	alFilterf(uiFilter, AL_LOWPASS_GAIN, 1.0f);
+	alFilterf(uiFilter, AL_LOWPASS_GAINHF, level);
+}
+
+static inline float gain_to_mB(float gain)
+{
+    return (gain > 1e-5f) ? (float)(log10f(gain) * 2000.0f) : -10000l;
+}
+
+static inline float mB_to_gain(float millibels)
+{
+    return (millibels > -10000.0f) ? powf(10.0f, millibels/2000.0f) : 0.0f;
+}
+
+static inline float clampF(float val, float minval, float maxval)
+{
+    if(val >= maxval) return maxval;
+    if(val <= minval) return minval;
+    return val;
+}
+
+void EAX3_Set(ALuint effect, const EAXLISTENERPROPERTIES *props)
+{
+	alEffecti (effect, AL_EFFECT_TYPE,                     AL_EFFECT_EAXREVERB);
+	alEffectf (effect, AL_EAXREVERB_DENSITY,               clampF(powf(props->flEnvironmentSize, 3.0f) / 16.0f, 0.0f, 1.0f));
+	alEffectf (effect, AL_EAXREVERB_DIFFUSION,             props->flEnvironmentDiffusion);
+	alEffectf (effect, AL_EAXREVERB_GAIN,                  mB_to_gain((float)props->lRoom));
+	alEffectf (effect, AL_EAXREVERB_GAINHF,                mB_to_gain((float)props->lRoomHF));
+	alEffectf (effect, AL_EAXREVERB_GAINLF,                mB_to_gain((float)props->lRoomLF));
+	alEffectf (effect, AL_EAXREVERB_DECAY_TIME,            props->flDecayTime);
+	alEffectf (effect, AL_EAXREVERB_DECAY_HFRATIO,         props->flDecayHFRatio);
+	alEffectf (effect, AL_EAXREVERB_DECAY_LFRATIO,         props->flDecayLFRatio);
+	alEffectf (effect, AL_EAXREVERB_REFLECTIONS_GAIN,      clampF(mB_to_gain((float)props->lReflections), AL_EAXREVERB_MIN_REFLECTIONS_GAIN, AL_EAXREVERB_MAX_REFLECTIONS_GAIN));
+	alEffectf (effect, AL_EAXREVERB_REFLECTIONS_DELAY,     props->flReflectionsDelay);
+	alEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN,       &props->vReflectionsPan.x);
+	alEffectf (effect, AL_EAXREVERB_LATE_REVERB_GAIN,      clampF(mB_to_gain((float)props->lReverb), AL_EAXREVERB_MIN_LATE_REVERB_GAIN, AL_EAXREVERB_MAX_LATE_REVERB_GAIN));
+	alEffectf (effect, AL_EAXREVERB_LATE_REVERB_DELAY,     props->flReverbDelay);
+	alEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN,       &props->vReverbPan.x);
+	alEffectf (effect, AL_EAXREVERB_ECHO_TIME,             props->flEchoTime);
+	alEffectf (effect, AL_EAXREVERB_ECHO_DEPTH,            props->flEchoDepth);
+	alEffectf (effect, AL_EAXREVERB_MODULATION_TIME,       props->flModulationTime);
+	alEffectf (effect, AL_EAXREVERB_MODULATION_DEPTH,      props->flModulationDepth);
+	alEffectf (effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, clampF(mB_to_gain(props->flAirAbsorptionHF), AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF, AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF));
+	alEffectf (effect, AL_EAXREVERB_HFREFERENCE,           props->flHFReference);
+	alEffectf (effect, AL_EAXREVERB_LFREFERENCE,           props->flLFReference);
+	alEffectf (effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR,   props->flRoomRolloffFactor);
+	alEffecti (effect, AL_EAXREVERB_DECAY_HFLIMIT,         (props->ulFlags&EAXLISTENERFLAGS_DECAYHFLIMIT) ? AL_TRUE : AL_FALSE);
+}
+
+void EFX_Set(ALuint effect, const EAXLISTENERPROPERTIES *props)
+{
+	alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
+	
+	alEffectf(effect, AL_REVERB_DENSITY,               clampF(powf(props->flEnvironmentSize, 3.0f) / 16.0f, 0.0f, 1.0f));
+	alEffectf(effect, AL_REVERB_DIFFUSION,             props->flEnvironmentDiffusion);
+	alEffectf(effect, AL_REVERB_GAIN,                  mB_to_gain((float)props->lRoom));
+	alEffectf(effect, AL_REVERB_GAINHF,                mB_to_gain((float)props->lRoomHF));
+	alEffectf(effect, AL_REVERB_DECAY_TIME,            props->flDecayTime);
+	alEffectf(effect, AL_REVERB_DECAY_HFRATIO,         props->flDecayHFRatio);
+	alEffectf(effect, AL_REVERB_REFLECTIONS_GAIN,      clampF(mB_to_gain((float)props->lReflections), AL_EAXREVERB_MIN_REFLECTIONS_GAIN, AL_EAXREVERB_MAX_REFLECTIONS_GAIN));
+	alEffectf(effect, AL_REVERB_REFLECTIONS_DELAY,     props->flReflectionsDelay);
+	alEffectf(effect, AL_REVERB_LATE_REVERB_GAIN,      clampF(mB_to_gain((float)props->lReverb), AL_EAXREVERB_MIN_LATE_REVERB_GAIN, AL_EAXREVERB_MAX_LATE_REVERB_GAIN));
+	alEffectf(effect, AL_REVERB_LATE_REVERB_DELAY,     props->flReverbDelay);
+	alEffectf(effect, AL_REVERB_AIR_ABSORPTION_GAINHF, clampF(mB_to_gain(props->flAirAbsorptionHF), AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF, AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF));
+	alEffectf(effect, AL_REVERB_ROOM_ROLLOFF_FACTOR,   props->flRoomRolloffFactor);
+	alEffecti(effect, AL_REVERB_DECAY_HFLIMIT,         (props->ulFlags&EAXLISTENERFLAGS_DECAYHFLIMIT) ? AL_TRUE : AL_FALSE);
+}
+
+void EAX3_SetReverbMix(ALuint filter, float mix)
+{
+	//long vol=(long)linear_to_dB(mix);
+	//DSPROPERTY_EAXBUFFER_ROOMHF,
+	//DSPROPERTY_EAXBUFFER_ROOM,
+	//DSPROPERTY_EAXBUFFER_REVERBMIX,
+	
+	long mbvol = gain_to_mB(mix);
+	float mb   = mbvol;
+	float mbhf = mbvol;
+	
+	alFilteri(filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
+	alFilterf(filter, AL_LOWPASS_GAIN,   mB_to_gain(Min(mb, 0.0f)));
+	alFilterf(filter, AL_LOWPASS_GAINHF, mB_to_gain(mbhf));
+}
+
+#endif
\ No newline at end of file
diff --git a/src/audio/oal/oal_utils.h b/src/audio/oal/oal_utils.h
new file mode 100644
index 0000000..f0fa090
--- /dev/null
+++ b/src/audio/oal/oal_utils.h
@@ -0,0 +1,54 @@
+#pragma once
+
+#ifdef AUDIO_OAL
+#include "eax.h"
+#include "AL/efx.h"
+
+
+void EFXInit();
+void EAX3_Set(ALuint effect, const EAXLISTENERPROPERTIES *props);
+void EFX_Set(ALuint effect, const EAXLISTENERPROPERTIES *props);
+void EAX3_SetReverbMix(ALuint filter, float mix);
+void SetEffectsLevel(ALuint uiFilter, float level);
+
+namespace re3_openal {
+
+extern LPALGENEFFECTS alGenEffects;
+extern LPALDELETEEFFECTS alDeleteEffects;
+extern LPALISEFFECT alIsEffect;
+extern LPALEFFECTI alEffecti;
+extern LPALEFFECTIV alEffectiv;
+extern LPALEFFECTF alEffectf;
+extern LPALEFFECTFV alEffectfv;
+extern LPALGETEFFECTI alGetEffecti;
+extern LPALGETEFFECTIV alGetEffectiv;
+extern LPALGETEFFECTF alGetEffectf;
+extern LPALGETEFFECTFV alGetEffectfv;
+extern LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
+extern LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
+extern LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
+extern LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
+extern LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
+extern LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
+extern LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
+extern LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
+extern LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
+extern LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
+extern LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
+extern LPALGENFILTERS alGenFilters;
+extern LPALDELETEFILTERS alDeleteFilters;
+extern LPALISFILTER alIsFilter;
+extern LPALFILTERI alFilteri;
+extern LPALFILTERIV alFilteriv;
+extern LPALFILTERF alFilterf;
+extern LPALFILTERFV alFilterfv;
+extern LPALGETFILTERI alGetFilteri;
+extern LPALGETFILTERIV alGetFilteriv;
+extern LPALGETFILTERF alGetFilterf;
+extern LPALGETFILTERFV alGetFilterfv;
+
+}
+
+using namespace re3_openal;
+
+#endif
diff --git a/src/audio/oal/stream.cpp b/src/audio/oal/stream.cpp
new file mode 100644
index 0000000..ed73e94
--- /dev/null
+++ b/src/audio/oal/stream.cpp
@@ -0,0 +1,1413 @@
+#include "common.h"
+
+#ifdef AUDIO_OAL
+#include "stream.h"
+#include "sampman.h"
+
+#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
+
+#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;
+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)
+	{
+		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
+	}
+
+	~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);
+	}
+	
+	~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
+// fuzzy seek eliminates stutter when playing ADF but spams errors a lot (and breaks radio sometimes)
+//#define MP3_USE_FUZZY_SEEK
+
+class CMP3File : public IDecoder
+{
+protected:
+	mpg123_handle *m_pMH;
+	bool m_bOpened;
+	uint32 m_nRate;
+	uint32 m_nChannels;
+	
+	CMP3File() :
+		m_pMH(nil),
+		m_bOpened(false),
+		m_nRate(0),
+		m_nChannels(0) {}
+public:
+	CMP3File(const char *path) :
+		m_pMH(nil),
+		m_bOpened(false),
+		m_nRate(0),
+		m_nChannels(0)
+	{
+		m_pMH = mpg123_new(nil, nil);
+		if ( m_pMH )
+		{
+#ifdef MP3_USE_FUZZY_SEEK
+			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_FUZZY | MPG123_SEEKBUFFER | MPG123_GAPLESS | MPG123_QUIET, 0.0);
+#else
+			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_SEEKBUFFER | MPG123_GAPLESS, 0.0);
+#endif
+			long rate = 0;
+			int channels = 0;
+			int encoding = 0;
+			
+			m_bOpened = mpg123_open(m_pMH, path) == 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);
+			}
+		}
+	}
+	
+	~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() ) return 0;
+		return mpg123_length(m_pMH);
+	}
+	
+	uint32 GetSampleRate()
+	{
+		return m_nRate;
+	}
+	
+	uint32 GetChannels()
+	{
+		return m_nChannels;
+	}
+	
+	void Seek(uint32 milliseconds)
+	{
+		if ( !IsOpened() ) return;
+		mpg123_seek(m_pMH, ms2samples(milliseconds), SEEK_SET);
+	}
+	
+	uint32 Tell()
+	{
+		if ( !IsOpened() ) return 0;
+		return samples2ms(mpg123_tell(m_pMH));
+	}
+	
+	uint32 Decode(void *buffer)
+	{
+		if ( !IsOpened() ) 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)
+		{
+#ifdef MP3_USE_FUZZY_SEEK
+			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_FUZZY | MPG123_SEEKBUFFER | MPG123_GAPLESS | MPG123_QUIET, 0.0);
+#else
+			mpg123_param(m_pMH, MPG123_FLAGS, MPG123_SEEKBUFFER | MPG123_GAPLESS, 0.0);
+#endif
+			long rate = 0;
+			int channels = 0;
+			int encoding = 0;
+
+			FILE* f = fopen(path, "rb");
+
+			m_bOpened = 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);
+			}
+		}
+	}
+};
+
+#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];
+	}
+
+	~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;
+		}
+	}
+	
+	~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
+
+void CStream::Initialise()
+{
+#ifdef AUDIO_OAL_USE_MPG123
+	mpg123_init();
+#endif
+}
+
+void CStream::Terminate()
+{
+#ifdef AUDIO_OAL_USE_MPG123
+	mpg123_exit();
+#endif
+}
+
+CStream::CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS], uint32 overrideSampleRate) :
+	m_pAlSources(sources),
+	m_alBuffers(buffers),
+	m_pBuffer(nil),
+	m_bPaused(false),
+	m_bActive(false),
+	m_pSoundFile(nil),
+	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 ( IsOpened() )
+	{
+		m_pBuffer            = malloc(m_pSoundFile->GetBufferSize());
+		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);
+		
+		return;
+	}
+}
+
+CStream::~CStream()
+{
+	Delete();
+}
+
+void CStream::Delete()
+{
+	Stop();
+	ClearBuffers();
+	
+	if ( m_pSoundFile )
+	{
+		delete m_pSoundFile;
+		m_pSoundFile = nil;
+	}
+	
+	if ( m_pBuffer )
+	{
+		free(m_pBuffer);
+		m_pBuffer = nil;
+	}
+}
+
+bool CStream::HasSource()
+{
+	return (m_pAlSources[0] != AL_NONE) && (m_pAlSources[1] != AL_NONE);
+}
+
+bool CStream::IsOpened()
+{
+	return m_pSoundFile && m_pSoundFile->IsOpened();
+}
+
+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;
+	}
+	
+	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;
+	m_pSoundFile->Seek(nPos);
+	ClearBuffers();
+}
+
+uint32 CStream::GetPosMS()
+{
+	if ( !HasSource() ) return 0;
+	if ( !IsOpened() ) return 0;
+	
+	ALint offset;
+	//alGetSourcei(m_alSource, AL_SAMPLE_OFFSET, &offset);
+	alGetSourcei(m_pAlSources[0], AL_BYTE_OFFSET, &offset);
+
+	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)
+{
+	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;
+	
+	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;
+}
+
+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)
+{
+	if ( IsOpened() )
+	{
+		alSourcei(m_pAlSources[0], AL_LOOPING, AL_FALSE);
+		alSourcei(m_pAlSources[1], AL_LOOPING, AL_FALSE);
+		if (!imSureQueueIsEmpty) {
+			SetPlay(false);
+			ClearBuffers();
+		}
+		m_pSoundFile->Seek(0);
+		//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;
+	if ( FillBuffers() != 0 )
+		SetPlay(true);
+}
+
+void CStream::Stop()
+{
+	if ( !HasSource() ) return;
+	SetPlay(false);
+}
+
+void CStream::Update()
+{
+	if ( !IsOpened() )
+		return;
+	
+	if ( !HasSource() )
+		return;
+	
+	if ( m_bReset )
+		return;
+	
+	if ( !m_bPaused )
+	{
+		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
+
+		bool buffersRefilled = false;
+
+		// 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)
+		{
+			Setup(true);
+			buffersRefilled = FillBuffers() != 0;
+			if (m_nLoopCount != 0)
+				m_nLoopCount--;
+		}
+		else
+		{
+			while( buffersProcessed[0]-- )
+			{
+				ALuint buffer[2];
+				
+				alSourceUnqueueBuffers(m_pAlSources[0], 1, &buffer[0]);
+				alSourceUnqueueBuffers(m_pAlSources[1], 1, &buffer[1]);
+				
+				if (m_bActive && FillBuffer(buffer))
+				{
+					buffersRefilled = true;
+					alSourceQueueBuffers(m_pAlSources[0], 1, &buffer[0]);
+					alSourceQueueBuffers(m_pAlSources[1], 1, &buffer[1]);
+				}
+			}
+		}
+
+		// Two reasons: 1-Source may be starved to audio and stopped itself, 2- We're already waiting it to starve and die for looping track!
+		if (m_bActive && (buffersRefilled || (totalBuffers[1] - buffersProcessed[1] != 0)))
+			SetPlay(true);
+	}
+}
+
+void CStream::ProviderInit()
+{
+	if ( m_bReset )
+	{
+		if ( Setup(true) )
+		{
+			SetPan(m_nPan);
+			SetVolume(m_nVolume);
+			SetLoopCount(m_nLoopCount);
+			SetPosMS(m_nPosBeforeReset);
+			if (m_bActive)
+				FillBuffers();
+			SetPlay(m_bActive);
+			if ( m_bPaused )
+				Pause();
+		}
+	
+		m_bReset = false;
+	}
+}
+
+void CStream::ProviderTerm()
+{
+	m_bReset = true;
+	m_nPosBeforeReset = GetPosMS();
+	
+	ClearBuffers();
+}
+	
+#endif
diff --git a/src/audio/oal/stream.h b/src/audio/oal/stream.h
new file mode 100644
index 0000000..9a2a2fb
--- /dev/null
+++ b/src/audio/oal/stream.h
@@ -0,0 +1,114 @@
+#pragma once
+
+#ifdef AUDIO_OAL
+#include <AL/al.h>
+
+#define NUM_STREAMBUFFERS 8
+
+class IDecoder
+{
+public:
+	virtual ~IDecoder() { }
+	
+	virtual bool   IsOpened() = 0;
+	
+	virtual uint32 GetSampleSize() = 0;
+	virtual uint32 GetSampleCount() = 0;
+	virtual uint32 GetSampleRate() = 0;
+	virtual uint32 GetChannels() = 0;
+	
+	uint32 GetAvgSamplesPerSec()
+	{
+		return GetChannels() * GetSampleRate();
+	}
+	
+	uint32 ms2samples(uint32 ms)
+	{
+		return float(ms) / 1000.0f * float(GetSampleRate());
+	}
+	
+	uint32 samples2ms(uint32 sm)
+	{
+		return float(sm) * 1000.0f / float(GetSampleRate());
+	}
+	
+	uint32 GetBufferSamples()
+	{
+		//return (GetAvgSamplesPerSec() >> 2) - (GetSampleCount() % GetChannels());
+		return (GetAvgSamplesPerSec() / 4); // 250ms
+	}
+	
+	uint32 GetBufferSize()
+	{
+		return GetBufferSamples() * GetSampleSize();
+	}
+	
+	virtual void   Seek(uint32 milliseconds) = 0;
+	virtual uint32 Tell() = 0;
+	
+	uint32 GetLength()
+	{
+		return float(GetSampleCount()) * 1000.0f / float(GetSampleRate());
+	}
+	
+	virtual uint32 Decode(void *buffer) = 0;
+};
+
+class CStream
+{
+	char     m_aFilename[128];
+	ALuint  *m_pAlSources;
+	ALuint (&m_alBuffers)[NUM_STREAMBUFFERS];
+	
+	bool     m_bPaused;
+	bool     m_bActive;
+	
+	void    *m_pBuffer;
+	
+	bool     m_bReset;
+	uint32   m_nVolume;
+	uint8    m_nPan;
+	uint32   m_nPosBeforeReset;
+	int32   m_nLoopCount;
+	
+	IDecoder *m_pSoundFile;
+	
+	bool HasSource();
+	void SetPosition(int i, float x, float y, float z);
+	void SetPitch(float pitch);
+	void SetGain(float gain);
+	void   Pause();
+	void   SetPlay(bool state);
+	
+	bool   FillBuffer(ALuint *alBuffer);
+	int32  FillBuffers();
+	void   ClearBuffers();
+public:
+	static void Initialise();
+	static void Terminate();
+	
+	CStream(char *filename, ALuint *sources, ALuint (&buffers)[NUM_STREAMBUFFERS], uint32 overrideSampleRate = 32000);
+	~CStream();
+	void   Delete();
+	
+	bool   IsOpened();
+	bool   IsPlaying();
+	void   SetPause (bool bPause);
+	void   SetVolume(uint32 nVol);
+	void   SetPan   (uint8 nPan);
+	void   SetPosMS (uint32 nPos); 
+	uint32 GetPosMS();
+	uint32 GetLengthMS();
+	
+	bool Setup(bool imSureQueueIsEmpty = false);
+	void Start();
+	void Stop();
+	void Update(void);
+	void SetLoopCount(int32);
+
+	
+	void ProviderInit();
+	void ProviderTerm();
+};
+
+#endif