tinysound.h

/*
	tinysound.h - v1.08

	To create implementation (the function definitions)
		#define TINYSOUND_IMPLEMENTATION
	in *one* C/CPP file (translation unit) that includes this file

	Summary:
	tinysound is a C API for loading, playing, looping, panning and fading mono
	and stero sounds. This means tinysound imparts no external DLLs or large
	libraries that adversely effect shipping size. tinysound can also run on
	Windows XP since DirectSound ships with all recent versions of Windows.
	tinysound implements a custom SSE2 mixer. tinysound uses CoreAudio for Apple
	machines (like OSX and iOS). SDL is used for all other platforms. Define
	TS_FORCE_SDL before placaing the TINYSOUND_IMPLEMENTATION in order to force
	the use of SDL.

	Revision history:
		1.0  (06/04/2016) initial release
		1.01 (06/06/2016) load WAV from memory
		                  separate portable and OS-specific code in tsMix
		                  fixed bug causing audio glitches when sounds ended
		                  added stb_vorbis loaders + demo example
		1.02 (06/08/2016) error checking + strings in vorbis loaders
		                  SSE2 implementation of mixer
		                  fix typos on docs/comments
		                  corrected volume bug introduced in 1.01
		1.03 (07/05/2016) size calculation helper (to know size of sound in
		                  bytes on the heap) tsSoundSize
		1.04 (12/06/2016) merged in Aaron Balint's contributions
		                  SFFT and pitch functions from Stephan M. Bernsee
		                  tsMix can run on its own thread with tsSpawnMixThread
		                  updated documentation, typo fixes
		                  fixed typo in malloc16 that caused heap corruption
		1.05 (12/08/2016) tsStopAllSounds, suggested by Aaron Balint
		1.06 (02/17/2017) port to CoreAudio for Apple machines
		1.07 (06/18/2017) SIMD the pitch shift code; swapped out old Bernsee
		                  code for a new re-write, updated docs as necessary,
		                  support for compiling as .c and .cpp on Windows,
		                  port for SDL (for Linux, or any other platform).
		                  Special thanks to DeXP (Dmitry Hrabrov) for 90% of
		                  the work on the SDL port!
		1.08 (09/06/2017) SDL_RWops support by RobLoach
*/

/*
	Contributors:
		Aaron Balint      1.04 - real time pitch
		                  1.04 - separate thread for tsMix
		                  1.04 - bugfix, removed extra free16 call for second channel
		DeXP              1.07 - initial work on SDL port
		RobLoach          1.08 - SDL_RWops support
*/

/*
	DOCUMENTATION (very quick intro):
	1. create context
	2. load sounds from disk into memory
	3. play sounds
	4. free context

	1. tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, N );
	2. tsPlaySoundDef def = tsMakeDef( &tsLoadWAV( "path_to_file/filename.wav" ) );
	3. tsPlaySound( ctx, def );
	4. tsShutdownContext( ctx );

	DOCUMENTATION (longer introduction):
	tinysound consists of tsLoadedSounds, tsPlayingSounds and the tsContext.
	The tsContext encapsulates an OS sound API, as well as buffers + settings.
	tsLoadedSound holds raw samples of a sound. tsPlayingSound is an instance
	of a tsLoadedSound that represents a sound that can be played through the
	tsContext.

	There are two main versions of the API, the low-level and the high-level
	API. The low-level API does not manage any memory for tsPlayingSounds. The
	high level api holds a memory pool of playing sounds.

	To actually mix sounds together and send audio to the sound card, be sure
	to call either tsMix periodically or call tsSpawnMixThread once.

	High-level API:
		First create a context and pass in non-zero to the final parameter. This
		final parameter controls how large of a memory pool to use for tsPlayingSounds.
		Here's an example where N is the size of the internal pool:

		tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, N );

		We create tsPlayingSounds indirectly with tsPlayDef structs. tsPlayDef is a
		POD struct so feel free to make them straight on the stack. The tsPlayDef
		sets up initialization parameters. Here's an example to load a wav and
		play it:

		tsLoadedSound loaded = tsLoadWAV( "path_to_file/filename.wav" );
		tsPlaySoundDef def = tsMakeDef( &loaded );
		tsPlayingSound* sound = tsPlaySound( ctx, def );

		The same def can be used to play as many sounds as desired (even simultaneously)
		as long as the context playing sound pool is large enough.

	Low-level API:
		First create a context and pass 0 in the final parameter (0 here means
		the context will *not* allocate a tsPlayingSound memory pool):

		tsContext* ctx = tsMakeContext( hwnd, frequency, latency, seconds, 0 );

		parameters:
			hwnd           --  HWND, handle to window (on OSX just pass in 0)
			frequency      --  int, represents Hz frequency rate in which samples are played
			latency        --  int, estimated latency in Hz from PlaySound call to speaker output
			seconds        --  int, number of second of samples internal buffers can hold
			0 (last param) --  int, number of elements in tsPlayingSound pool

		We create a tsPlayingSound like so:
		tsLoadedSound loaded = tsLoadWAV( "path_to_file/filename.wav" );
		tsPlayingSound playing_sound = tsMakePlayingSound( &loaded );

		Then to play the sound we do:
		tsInsertSound( ctx, &playing_sound );

		The above tsInsertSound function call will place playing_sound into
		a singly-linked list inside the context. The context will remove
		the sound from its internal list when it finishes playing.

	WARNING: The high-level API cannot be mixed with the low-level API. If you
	try then the internal code will assert and crash. Pick one and stick with it.
	Usually he high-level API will be used, but if someone is *really* picky about
	their memory usage, or wants more control, the low-level API can be used.

	Here is the Low-Level API:
		tsPlayingSound tsMakePlayingSound( tsLoadedSound* loaded );
		int tsInsertSound( tsContext* ctx, tsPlayingSound* sound );

	Here is the High-Level API:
		tsPlayingSound* tsPlaySound( tsContext* ctx, tsPlaySoundDef def );
		tsPlaySoundDef tsMakeDef( tsLoadedSound* sound );
		void tsStopAllSounds( tsContext( ctx );

	Be sure to link against dsound.dll (or dsound.lib) on Windows.

	Read the rest of the header for specific details on all available functions
	and struct types.
*/

/*
	Known Limitations:

	* PCM mono/stereo format is the only formats the LoadWAV function supports. I don't
		guarantee it will work for all kinds of wav files, but it certainly does for the common
		kind (and can be changed fairly easily if someone wanted to extend it).
	* Only supports 16 bits per sample.
	* Mixer does not do any fancy clipping. The algorithm is to convert all 16 bit samples
		to float, mix all samples, and write back to audio API as 16 bit integers. In
		practice this works very well and clipping is not often a big problem.
	* I'm not super familiar with good ways to avoid the DirectSound play cursor from going
		past the write cursor. To mitigate this pass in a larger number to tsMakeContext's 4th
		parameter (buffer scale in seconds).
	* Pitch shifting code is pretty darn expensive. This is due to the use of a Fast Fourier Transform
		routine. The pitch shifting itself is written in rather efficient SIMD using SSE2 intrinsics,
		but the FFT routine is very basic. FFT is a big bottleneck for pitch shifting. There is a
		TODO optimization listed in this file for the FFT routine, but it's fairly low priority;
		optimizing FFT routines is difficult and requires a lot of specialized knowledge.
*/

/*
	FAQ
	Q : Why DirectSound instead of (insert API here) on Windows?
	A : Casey Muratori documented DS on Handmade Hero, other APIs do not have such good docs. DS has
	shipped on Windows XP all the way through Windows 10 -- using this header effectively intro-
	duces zero dependencies for the foreseeable future. The DS API itself is sane enough to quickly
	implement needed features, and users won't hear the difference between various APIs. Latency is
	not that great with DS but it is shippable. Additionally, many other APIs will in the end speak
	to Windows through the DS API.

	Q : Why not include Linux support?
	A : There have been a couple requests for ALSA support on Linux. For now the only option is to use
	SDL backend, which can indirectly support ALSA. SDL is used only in a very low-level manner;
	to get sound samples to the sound card via callback, so there shouldn't be much in the way of
	considering SDL a good option for "name your flavor" of Linux backend.

	Q : I would like to use my own memory management, how can I achieve this?
	A : This header makes a couple uses of malloc/free, and malloc16/free16. Simply find these bits
	and replace them with your own memory allocation routines. They can be wrapped up into a macro,
	or call your own functions directly -- it's up to you. Generally these functions allocate fairly
	large chunks of memory, and not very often (if at all), with one exception: tsSetPitch is a very
	expensive routine and requires frequent dynamic memory management.
*/

/*
	Some past discussion threads:
	https://www.reddit.com/r/gamedev/comments/6i39j2/tinysound_the_cutest_library_to_get_audio_into/
	https://www.reddit.com/r/gamedev/comments/4ml6l9/tinysound_singlefile_c_audio_library/
	https://forums.tigsource.com/index.php?topic=58706.0
*/

#if !defined( TINYSOUND_H )

#define TS_WINDOWS	1
#define TS_MAC		2
#define TS_UNIX		3
#define TS_SDL		4

#if defined( _WIN32 )

	#define TS_PLATFORM TS_WINDOWS

	#if !defined _CRT_SECURE_NO_WARNINGS
	#define _CRT_SECURE_NO_WARNINGS
	#endif

#elif defined( __APPLE__ )

	#define TS_PLATFORM TS_MAC

#else

	#define TS_PLATFORM TS_SDL

	// please note TS_UNIX is not directly support
	// instead, unix-style OSes are encouraged to use SDL
	// see: https://www.libsdl.org/

#endif

// Use TS_FORCE_SDL to override the above macros and use
// the SDL port.
#ifdef TINYSOUND_FORCE_SDL

	#undef TS_PLATFORM
	#define TS_PLATFORM TS_SDL

#endif

#if TS_PLATFORM == TS_SDL

	#include <SDL2/SDL.h>
	
#endif

#include <stdint.h>

// read this in the event of tsLoadWAV/tsLoadOGG errors
// also read this in the event of certain errors from tsMakeContext
extern const char* g_tsErrorReason;

// stores a loaded sound in memory
typedef struct
{
	int sample_rate;
	int sample_count;
	int channel_count;
	void* channels[ 2 ];
} tsLoadedSound;

struct tsPitchData;
typedef struct tsPitchData tsPitchData;

// represents an instance of a tsLoadedSound, can be played through the tsContext
typedef struct tsPlayingSound
{
	int active;
	int paused;
	int looped;
	float volume0;
	float volume1;
	float pan0;
	float pan1;
	float pitch;
	tsPitchData* pitch_filter[ 2 ];
	int sample_index;
	tsLoadedSound* loaded_sound;
	struct tsPlayingSound* next;
} tsPlayingSound;

// holds audio API info and other info
struct tsContext;
typedef struct tsContext tsContext;

// The returned struct will contain a null pointer in tsLoadedSound::channel[ 0 ]
// in the case of errors. Read g_tsErrorReason string for details on what happened.
// Calls tsReadMemWAV internally.
tsLoadedSound tsLoadWAV( const char* path );

// Reads a WAV file from memory. Still allocates memory for the tsLoadedSound since
// WAV format will interlace stereo, and we need separate data streams to do SIMD
// properly.
void tsReadMemWAV( const void* memory, tsLoadedSound* sound );

// If stb_vorbis was included *before* tinysound go ahead and create
// some functions for dealing with OGG files.
#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H

	void tsReadMemOGG( const void* memory, int length, tsLoadedSound* sound );
	tsLoadedSound tsLoadOGG( const char* path );

#endif

// Uses free16 (aligned free, implemented later in this file) to free up both of
// the channels stored within sound
void tsFreeSound( tsLoadedSound* sound );

// Returns the size, in bytes, of all heap-allocated memory for this particular
// loaded sound
int tsSoundSize( tsLoadedSound* sound );

// playing_pool_count -- 0 to setup low-level API, non-zero to size the internal
// memory pool for tsPlayingSound instances
tsContext* tsMakeContext( void* hwnd, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count );
void tsShutdownContext( tsContext* ctx );

// Call tsSpawnMixThread once to setup a separate thread for the context to run
// upon. The separate thread will continually call tsMix and perform mixing
// operations.
void tsSpawnMixThread( tsContext* ctx );

// Use tsThreadSleepDelay to specify a custom sleep delay time.
// A sleep will occur after each call to tsMix. By default YieldProcessor
// is used, and no sleep occurs. Use a sleep delay to conserve CPU bandwidth.
// A recommended sleep time is a little less than 1/2 your predicted 1/FPS.
// 60 fps is 16 ms, so about 1-5 should work well in most cases.
void tsThreadSleepDelay( tsContext* ctx, int milliseconds );

// Call this manually, once per game tick recommended, if you haven't ever
// called tsSpawnMixThread. Otherwise the thread will call tsMix itself.
// num_samples_to_write is not used on Windows. On Mac it is used to push
// samples into a circular buffer while CoreAudio simultaneously pulls samples
// off of the buffer. num_samples_to_write should be computed each update tick
// as delta_time * play_frequency_in_Hz + 1.
void tsMix( tsContext* ctx );

// All of the functions in this next section should only be called if tsIsActive
// returns true. Calling them otherwise probably won't do anything bad, but it
// won't do anything at all. If a sound is active it resides in the context's
// internal list of playing sounds.
int tsIsActive( tsPlayingSound* sound );

// Flags sound for removal. Upon next tsMix call will remove sound from playing
// list. If high-level API used sound is placed onto the internal free list.
void tsStopSound( tsPlayingSound* sound );

void tsLoopSound( tsPlayingSound* sound, int zero_for_no_loop );
void tsPauseSound( tsPlayingSound* sound, int one_for_paused );

// lerp from 0 to 1, 0 full left, 1 full right
void tsSetPan( tsPlayingSound* sound, float pan );

// explicitly set volume of each channel. Can be used as panning (but it's
// recommended to use the tsSetPan function for panning).
void tsSetVolume( tsPlayingSound* sound, float volume_left, float volume_right );

// Change pitch (not duration) of sound. pitch = 0.5f for one octave lower, pitch = 2.0f for one octave higher.
// pitch at 1.0f applies no change. pitch settings farther away from 1.0f create more distortion and lower
// the output sample quality. pitch can be adjusted in real-time for doppler effects and the like. Going beyond
// 0.5f and 2.0f may require some tweaking the pitch shifting parameters, and is not recommended.

// Additional important information about performance: This function
// is quite expensive -- you have been warned! Try it out and be aware of how much CPU consumption it uses.
// To avoid destroying the originally loaded sound samples, tsSetPitch will do a one-time allocation to copy
// sound samples into a new buffer. The new buffer contains the pitch adjusted samples, and these will be played
// through tsMix. This lets the pitch be modulated at run-time, but requires dynamically allocated memory. The
// memory is freed once the sound finishes playing. If a one-time pitch adjustment is desired, for performance
// reasons please consider doing an off-line pitch adjustment manually as a pre-processing step for your sounds.
// Also, consider changing malloc16 and free16 to match your custom memory allocation needs. Try adjusting
// TS_PITCH_QUALITY (must be a power of two) and see how this affects your performance.
void tsSetPitch( tsPlayingSound* sound, float pitch );

// Delays sound before actually playing it. Requires context to be passed in
// since there's a conversion from seconds to samples per second.
// If one were so inclined another version could be implemented like:
// void tsSetDelay( tsPlayingSound* sound, float delay, int samples_per_second )
void tsSetDelay( tsContext* ctx, tsPlayingSound* sound, float delay_in_seconds );

// Portable sleep function
void tsSleep( int milliseconds );

// LOW-LEVEL API
tsPlayingSound tsMakePlayingSound( tsLoadedSound* loaded );
int tsInsertSound( tsContext* ctx, tsPlayingSound* sound ); // returns 1 if sound was successfully inserted, 0 otherwise

// HIGH-LEVEL API
typedef struct
{
	int paused;
	int looped;
	float volume_left;
	float volume_right;
	float pan;
	float pitch;
	float delay;
	tsLoadedSound* loaded;
} tsPlaySoundDef;

tsPlayingSound* tsPlaySound( tsContext* ctx, tsPlaySoundDef def );
tsPlaySoundDef tsMakeDef( tsLoadedSound* sound );
void tsStopAllSounds( tsContext* ctx );

// SDL_RWops specific functions
#if TS_PLATFORM == TS_SDL

	// Provides the ability to use tsLoadWAV with an SDL_RWops object.
	tsLoadedSound tsLoadWAVRW( SDL_RWops* context );

	#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H

		// Provides the ability to use tsLoadOGG with an SDL_RWops object.
		tsLoadedSound tsLoadOGGRW( SDL_RWops* rw );

	#endif

#endif

#define TINYSOUND_H
#endif

#ifdef TINYSOUND_IMPLEMENTATION
#undef TINYSOUND_IMPLEMENTATION

// Change the allocator as necessary
#if !defined( TS_ALLOC )
	#include <stdlib.h>	// malloc, free
	#define TS_ALLOC(size) malloc(size)
	#define TS_FREE(mem) free(mem)
#endif

#include <stdio.h>	// fopen, fclose
#include <string.h>	// memcmp, memset, memcpy
#include <xmmintrin.h>
#include <emmintrin.h>

#if TS_PLATFORM == TS_WINDOWS

	#ifndef _WINDOWS_
		#include <Windows.h>
	#endif

	#ifndef _WAVEFORMATEX_
		#include <mmreg.h>
	#endif

	#include <dsound.h>
	#undef PlaySound

	#if defined( _MSC_VER )
		#pragma comment( lib, "dsound.lib" )
	#endif

#elif TS_PLATFORM == TS_MAC

	#include <CoreAudio/CoreAudio.h>
	#include <AudioUnit/AudioUnit.h>
	#include <pthread.h>
	#include <mach/mach_time.h>

#else

#endif

#define TS_CHECK( X, Y ) do { if ( !(X) ) { g_tsErrorReason = Y; goto ts_err; } } while ( 0 )
#if TS_PLATFORM == TS_MAC && defined( __clang__ )
	#define TS_ASSERT_INTERNAL __builtin_trap( )
#else
	#define TS_ASSERT_INTERNAL *(int*)0 = 0
#endif
#define TS_ASSERT( X ) do { if ( !(X) ) TS_ASSERT_INTERNAL; } while ( 0 )
#define TS_ALIGN( X, Y ) ((((size_t)X) + ((Y) - 1)) & ~((Y) - 1))
#define TS_TRUNC( X, Y ) ((size_t)(X) & ~((Y) - 1))

const char* g_tsErrorReason;

static void* tsReadFileToMemory( const char* path, int* size )
{
	void* data = 0;
	FILE* fp = fopen( path, "rb" );
	int sizeNum = 0;

	if ( fp )
	{
		fseek( fp, 0, SEEK_END );
		sizeNum = (int)ftell( fp );
		fseek( fp, 0, SEEK_SET );
		data = TS_ALLOC( sizeNum );
		fread( data, sizeNum, 1, fp );
		fclose( fp );
	}

	if ( size ) *size = sizeNum;
	return data;
}

static int tsFourCC( const char* CC, void* memory )
{
	if ( !memcmp( CC, memory, 4 ) ) return 1;
	return 0;
}

static char* tsNext( char* data )
{
	uint32_t size = *(uint32_t*)(data + 4);
	size = (size + 1) & ~1;
	return data + 8 + size;
}

static void* malloc16( size_t size )
{
	void* p = TS_ALLOC( size + 16 );
	if ( !p ) return 0;
	unsigned char offset = (size_t)p & 15;
	p = (void*)TS_ALIGN( p + 1, 16 );
	*((char*)p - 1) = 16 - offset;
	TS_ASSERT( !((size_t)p & 15) );
	return p;
}

static void free16( void* p )
{
	if ( !p ) return;
	TS_FREE( (char*)p - (size_t)*((char*)p - 1) );
}

static void tsLastElement( __m128* a, int i, int j, int16_t* samples, int offset )
{
	switch ( offset )
	{
	case 1:
		a[ i ] = _mm_set_ps( samples[ j ], 0.0f, 0.0f, 0.0f );
		break;

	case 2:
		a[ i ] = _mm_set_ps( samples[ j ], samples[ j + 1 ], 0.0f, 0.0f );
		break;

	case 3:
		a[ i ] = _mm_set_ps( samples[ j ], samples[ j + 1 ], samples[ j + 2 ], 0.0f );
		break;

	case 0:
		a[ i ] = _mm_set_ps( samples[ j ], samples[ j + 1 ], samples[ j + 2 ], samples[ j + 3 ] );
		break;
	}
}

void tsReadMemWAV( const void* memory, tsLoadedSound* sound )
{
	#pragma pack( push, 1 )
	typedef struct
	{
		uint16_t wFormatTag;
		uint16_t nChannels;
		uint32_t nSamplesPerSec;
		uint32_t nAvgBytesPerSec;
		uint16_t nBlockAlign;
		uint16_t wBitsPerSample;
		uint16_t cbSize;
		uint16_t wValidBitsPerSample;
		uint32_t dwChannelMask;
		uint8_t SubFormat[ 18 ];
	} Fmt;
	#pragma pack( pop )

	char* data = (char*)memory;
	TS_CHECK( data, "Unable to read input file (file doesn't exist, or could not allocate heap memory." );
	TS_CHECK( tsFourCC( "RIFF", data ), "Incorrect file header; is this a WAV file?" );
	TS_CHECK( tsFourCC( "WAVE", data + 8 ), "Incorrect file header; is this a WAV file?" );

	data += 12;

	TS_CHECK( tsFourCC( "fmt ", data ), "fmt chunk not found." );
	Fmt fmt;
	fmt = *(Fmt*)(data + 8);
	TS_CHECK( fmt.wFormatTag == 1, "Only PCM WAV files are supported." );
	TS_CHECK( fmt.nChannels == 1 || fmt.nChannels == 2, "Only mono or stereo supported (too many channels detected)." );
	TS_CHECK( fmt.wBitsPerSample == 16, "Only 16 bits per sample supported." );
	TS_CHECK( fmt.nBlockAlign == fmt.nChannels * 2, "implementation error" );

	sound->sample_rate = (int)fmt.nSamplesPerSec;

	data = tsNext( data ); // skip fmt chunk
	if ( tsFourCC( "fact", data ) ) data = tsNext( data ); // skip possible fact chunk
	
	TS_CHECK( tsFourCC( "data", data ), "data chunk not found." );
	int sample_size = *((uint32_t*)(data + 4));
	int sample_count = sample_size / (fmt.nChannels * sizeof( uint16_t ));
	sound->sample_count = sample_count;
	sound->channel_count = fmt.nChannels;

	int wide_count = (int)TS_ALIGN( sample_count, 4 );
	wide_count /= 4;
	int wide_offset = sample_count & 3;
	int16_t* samples = (int16_t*)(data + 8);
	float* sample = (float*)alloca( sizeof( float ) * 4 + 16 );
	sample = (float*)TS_ALIGN( sample, 16 );

	switch ( sound->channel_count )
	{
	case 1:
	{
		sound->channels[ 0 ] = malloc16( wide_count * sizeof( __m128 ) );
		sound->channels[ 1 ] = 0;
		__m128* a = (__m128*)sound->channels[ 0 ];

		for ( int i = 0, j = 0; i < wide_count - 1; ++i, j += 4 )
		{
			sample[ 0 ] = (float)samples[ j ];
			sample[ 1 ] = (float)samples[ j + 1 ];
			sample[ 2 ] = (float)samples[ j + 2 ];
			sample[ 3 ] = (float)samples[ j + 3 ];
			a[ i ] = _mm_load_ps( sample );
		}

		tsLastElement( a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset );
	}	break;

	case 2:
	{
		__m128* a = (__m128*)malloc16( wide_count * sizeof( __m128 ) * 2 );
		__m128* b = a + wide_count;

		for ( int i = 0, j = 0; i < wide_count - 1; ++i, j += 8 )
		{
			sample[ 0 ] = (float)samples[ j ];
			sample[ 1 ] = (float)samples[ j + 2 ];
			sample[ 2 ] = (float)samples[ j + 4 ];
			sample[ 3 ] = (float)samples[ j + 6 ];
			a[ i ] = _mm_load_ps( sample );

			sample[ 0 ] = (float)samples[ j + 1 ];
			sample[ 1 ] = (float)samples[ j + 3 ];
			sample[ 2 ] = (float)samples[ j + 5 ];
			sample[ 3 ] = (float)samples[ j + 7 ];
			b[ i ] = _mm_load_ps( sample );
		}

		tsLastElement( a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset );
		tsLastElement( b, wide_count - 1, (wide_count - 1) * 4 + 4, samples, wide_offset );
		sound->channels[ 0 ] = a;
		sound->channels[ 1 ] = b;
	}	break;

	default:
		TS_CHECK( 0, "unsupported channel count (only support mono and stereo)." );
	}

	return;

ts_err:
	memset( &sound, 0, sizeof( sound ) );
}

tsLoadedSound tsLoadWAV( const char* path )
{
	tsLoadedSound sound = { 0 };
	char* wav = (char*)tsReadFileToMemory( path, 0 );
	tsReadMemWAV( wav, &sound );
	TS_FREE( wav );
	return sound;
}

#if TS_PLATFORM == TS_SDL

	// Load an SDL_RWops object's data into memory.
	// Ripped straight from: https://wiki.libsdl.org/SDL_RWread
	static void* tsReadRWToMemory( SDL_RWops* rw, int* size )
	{
		Sint64 res_size = SDL_RWsize( rw );
		char* data = (char*)TS_ALLOC( (size_t)(res_size + 1) );

		Sint64 nb_read_total = 0, nb_read = 1;
		char* buf = data;
		while ( nb_read_total < res_size && nb_read != 0 )
		{
			nb_read = SDL_RWread( rw, buf, 1, (size_t)(res_size - nb_read_total) );
			nb_read_total += nb_read;
			buf += nb_read;
		}

		SDL_RWclose(rw);

		if ( nb_read_total != res_size )
		{
			TS_FREE( data );
			return NULL;
		}

		if ( size ) *size = (int)res_size;
		return data;
	}

	tsLoadedSound tsLoadWAVRW( SDL_RWops* context )
	{
		tsLoadedSound sound = { 0 };
		char* wav = (char*)tsReadRWToMemory( context, 0 );
		tsReadMemWAV( wav, &sound );
		TS_FREE( wav );
		return sound;
	}

#endif

// If stb_vorbis was included *before* tinysound go ahead and create
// some functions for dealing with OGG files.
#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H

	void tsReadMemOGG( const void* memory, int length, tsLoadedSound* sound )
	{
		int16_t* samples = 0;
		int channel_count;
		int sample_count = stb_vorbis_decode_memory( (const unsigned char*)memory, length, &channel_count, &sound->sample_rate, &samples );

		TS_CHECK( sample_count > 0, "stb_vorbis_decode_memory failed. Make sure your file exists and is a valid OGG file." );

		int wide_count = (int)TS_ALIGN( sample_count, 4 ) / 4;
		int wide_offset = sample_count & 3;
		float* sample = (float*)alloca( sizeof( float ) * 4 + 16 );
		sample = (float*)TS_ALIGN( sample, 16 );
		__m128* a;
		__m128* b;

		switch ( channel_count )
		{
		case 1:
		{
			a = (__m128*)malloc16( wide_count * sizeof( __m128 ) );
			b = 0;

			for ( int i = 0, j = 0; i < wide_count - 1; ++i, j += 4 )
			{
				sample[ 0 ] = (float)samples[ j ];
				sample[ 1 ] = (float)samples[ j + 1 ];
				sample[ 2 ] = (float)samples[ j + 2 ];
				sample[ 3 ] = (float)samples[ j + 3 ];
				a[ i ] = _mm_load_ps( sample );
			}

			tsLastElement( a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset );
		}	break;

		case 2:
			a = (__m128*)malloc16( wide_count * sizeof( __m128 ) * 2 );
			b = a + wide_count;

			for ( int i = 0, j = 0; i < wide_count - 1; ++i, j += 8 )
			{
				sample[ 0 ] = (float)samples[ j ];
				sample[ 1 ] = (float)samples[ j + 2 ];
				sample[ 2 ] = (float)samples[ j + 4 ];
				sample[ 3 ] = (float)samples[ j + 6 ];
				a[ i ] = _mm_load_ps( sample );

				sample[ 0 ] = (float)samples[ j + 1 ];
				sample[ 1 ] = (float)samples[ j + 3 ];
				sample[ 2 ] = (float)samples[ j + 5 ];
				sample[ 3 ] = (float)samples[ j + 7 ];
				b[ i ] = _mm_load_ps( sample );
			}

			tsLastElement( a, wide_count - 1, (wide_count - 1) * 4, samples, wide_offset );
			tsLastElement( b, wide_count - 1, (wide_count - 1) * 4 + 4, samples, wide_offset );
			break;

		default:
			TS_CHECK( 0, "Unsupported channel count." );
		}

		sound->sample_count = sample_count;
		sound->channel_count = channel_count;
		sound->channels[ 0 ] = a;
		sound->channels[ 1 ] = b;
		TS_FREE( samples );
		return;

	ts_err:
		TS_FREE( samples );
		memset( sound, 0, sizeof( tsLoadedSound ) );
	}

	tsLoadedSound tsLoadOGG( const char* path )
	{
		int length;
		void* memory = tsReadFileToMemory( path, &length );
		tsLoadedSound sound;
		tsReadMemOGG( memory, length, &sound );
		TS_FREE( memory );

		return sound;
	}

	#if TS_PLATFORM == TS_SDL

		tsLoadedSound tsLoadOGGRW( SDL_RWops* rw )
		{
			int length;
			void* memory = tsReadRWToMemory( rw, &length );
			tsLoadedSound sound;
			tsReadMemOGG( memory, length, &sound );
			TS_FREE( memory );

			return sound;
		}
		
	#endif

#endif

void tsFreeSound( tsLoadedSound* sound )
{
	free16( sound->channels[ 0 ] );
	memset( sound, 0, sizeof( tsLoadedSound ) );
}

int tsSoundSize( tsLoadedSound* sound )
{
	return sound->sample_count * sound->channel_count * sizeof( uint16_t );
}

tsPlayingSound tsMakePlayingSound( tsLoadedSound* loaded )
{
	tsPlayingSound playing;
	playing.active = 0;
	playing.paused = 0;
	playing.looped = 0;
	playing.volume0 = 1.0f;
	playing.volume1 = 1.0f;
	playing.pan0 = 0.5f;
	playing.pan1 = 0.5f;
	playing.pitch = 1.0f;
	playing.pitch_filter[ 0 ] = 0;
	playing.pitch_filter[ 1 ] = 0;
	playing.sample_index = 0;
	playing.loaded_sound = loaded;
	playing.next = 0;
	return playing;
}

int tsIsActive( tsPlayingSound* sound )
{
	return sound->active;
}

void tsStopSound( tsPlayingSound* sound )
{
	sound->active = 0;
}

void tsLoopSound( tsPlayingSound* sound, int zero_for_no_loop )
{
	sound->looped = zero_for_no_loop;
}

void tsPauseSound( tsPlayingSound* sound, int one_for_paused )
{
	sound->paused = one_for_paused;
}

void tsSetPan( tsPlayingSound* sound, float pan )
{
	if ( pan > 1.0f ) pan = 1.0f;
	else if ( pan < 0.0f ) pan = 0.0f;
	float left = 1.0f - pan;
	float right = pan;
	sound->pan0 = left;
	sound->pan1 = right;
}

void tsSetPitch( tsPlayingSound* sound, float pitch )
{
	sound->pitch = pitch;
}

void tsSetVolume( tsPlayingSound* sound, float volume_left, float volume_right )
{
	if ( volume_left < 0.0f ) volume_left = 0.0f;
	if ( volume_right < 0.0f ) volume_right = 0.0f;
	sound->volume0 = volume_left;
	sound->volume1 = volume_right;
}

static void tsRemoveFilter( tsPlayingSound* playing );

#if TS_PLATFORM == TS_WINDOWS

    void tsSleep( int milliseconds )
    {
        Sleep( milliseconds );
    }

	struct tsContext
	{
		unsigned latency_samples;
		unsigned running_index;
		int Hz;
		int bps;
		int buffer_size;
		int wide_count;
		tsPlayingSound* playing;
		__m128* floatA;
		__m128* floatB;
		__m128i* samples;
		tsPlayingSound* playing_pool;
		tsPlayingSound* playing_free;

		// platform specific stuff
		LPDIRECTSOUND dsound;
		LPDIRECTSOUNDBUFFER buffer;
		LPDIRECTSOUNDBUFFER primary;

		// data for tsMix thread, enable these with tsSpawnMixThread
		CRITICAL_SECTION critical_section;
		int separate_thread;
		int running;
		int sleep_milliseconds;
	};

	static void tsReleaseContext( tsContext* ctx )
	{
		if ( ctx->separate_thread )	DeleteCriticalSection( &ctx->critical_section );
#ifdef __cplusplus
		ctx->buffer->Release( );
		ctx->primary->Release( );
		ctx->dsound->Release( );
#else
		ctx->buffer->lpVtbl->Release( ctx->buffer );
		ctx->primary->lpVtbl->Release( ctx->primary );
		ctx->dsound->lpVtbl->Release( ctx->dsound );
#endif
		tsPlayingSound* playing = ctx->playing;
		while ( playing )
		{
			tsRemoveFilter( playing );
			playing = playing->next;
		}
		TS_FREE( ctx );
	}

	static DWORD WINAPI tsCtxThread( LPVOID lpParameter )
	{
		tsContext* ctx = (tsContext*)lpParameter;

		while ( ctx->running )
		{
			tsMix( ctx );
			if ( ctx->sleep_milliseconds ) tsSleep( ctx->sleep_milliseconds );
			else YieldProcessor( );
		}

		ctx->separate_thread = 0;
		return 0;
	}

	static void tsLock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) EnterCriticalSection( &ctx->critical_section );
	}

	static void tsUnlock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) LeaveCriticalSection( &ctx->critical_section );
	}

	tsContext* tsMakeContext( void* hwnd, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count )
	{
		int bps = sizeof( INT16 ) * 2;
		int buffer_size = play_frequency_in_Hz * bps * num_buffered_seconds;
		tsContext* ctx = 0;
		WAVEFORMATEX format = { 0 };
		DSBUFFERDESC bufdesc = { 0 };
		LPDIRECTSOUND dsound;

		TS_CHECK( hwnd, "Invalid hwnd passed to tsMakeContext." );

		HRESULT res = DirectSoundCreate( 0, &dsound, 0 );
		TS_CHECK( res == DS_OK, "DirectSoundCreate failed" );
#ifdef __cplusplus
		dsound->SetCooperativeLevel( (HWND)hwnd, DSSCL_PRIORITY );
#else
		dsound->lpVtbl->SetCooperativeLevel( dsound, (HWND)hwnd, DSSCL_PRIORITY );
#endif
		bufdesc.dwSize = sizeof( bufdesc );
		bufdesc.dwFlags = DSBCAPS_PRIMARYBUFFER;

		LPDIRECTSOUNDBUFFER primary_buffer;
#ifdef __cplusplus
		res = dsound->CreateSoundBuffer( &bufdesc, &primary_buffer, 0 );
#else
		res = dsound->lpVtbl->CreateSoundBuffer( dsound, &bufdesc, &primary_buffer, 0 );
#endif
		TS_CHECK( res == DS_OK, "Failed to create primary sound buffer" );

		format.wFormatTag = WAVE_FORMAT_PCM;
		format.nChannels = 2;
		format.nSamplesPerSec = play_frequency_in_Hz;
		format.wBitsPerSample = 16;
		format.nBlockAlign = (format.nChannels * format.wBitsPerSample) / 8;
		format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
		format.cbSize = 0;
#ifdef __cplusplus
		res = primary_buffer->SetFormat( &format );
#else
		res = primary_buffer->lpVtbl->SetFormat( primary_buffer, &format );
#endif
		TS_CHECK( res == DS_OK, "Failed to set format on primary buffer" );

		LPDIRECTSOUNDBUFFER secondary_buffer;
		bufdesc.dwSize = sizeof( bufdesc );
		bufdesc.dwFlags = 0;
		bufdesc.dwBufferBytes = buffer_size;
		bufdesc.lpwfxFormat = &format;
#ifdef __cplusplus
		res = dsound->CreateSoundBuffer( &bufdesc, &secondary_buffer, 0 );
#else
		res = dsound->lpVtbl->CreateSoundBuffer( dsound, &bufdesc, &secondary_buffer, 0 );
#endif
		TS_CHECK( res == DS_OK, "Failed to set format on secondary buffer" );

		int sample_count = play_frequency_in_Hz * num_buffered_seconds;
		int wide_count = (int)TS_ALIGN( sample_count, 4 );
		int pool_size = playing_pool_count * sizeof( tsPlayingSound );
		int mix_buffers_size = sizeof( __m128 ) * wide_count * 2;
		int sample_buffer_size = sizeof( __m128i ) * wide_count;
		ctx = (tsContext*)TS_ALLOC( sizeof( tsContext ) + mix_buffers_size + sample_buffer_size + 16 + pool_size );
		ctx->latency_samples = (unsigned)TS_ALIGN( play_frequency_in_Hz / latency_factor_in_Hz, 4 );
		ctx->running_index = 0;
		ctx->Hz = play_frequency_in_Hz;
		ctx->bps = bps;
		ctx->buffer_size = buffer_size;
		ctx->wide_count = wide_count;
		ctx->dsound = dsound;
		ctx->buffer = secondary_buffer;
		ctx->primary = primary_buffer;
		ctx->playing = 0;
		ctx->floatA = (__m128*)(ctx + 1);
		ctx->floatA = (__m128*)TS_ALIGN( ctx->floatA, 16 );
		TS_ASSERT( !((size_t)ctx->floatA & 15) );
		ctx->floatB = ctx->floatA + wide_count;
		ctx->samples = (__m128i*)ctx->floatB + wide_count;
		ctx->running = 1;
		ctx->separate_thread = 0;
		ctx->sleep_milliseconds = 0;

		if ( playing_pool_count )
		{
			ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
			for ( int i = 0; i < playing_pool_count - 1; ++i )
				ctx->playing_pool[ i ].next = ctx->playing_pool + i + 1;
			ctx->playing_pool[ playing_pool_count - 1 ].next = 0;
			ctx->playing_free = ctx->playing_pool;
		}

		else
		{
			ctx->playing_pool = 0;
			ctx->playing_free = 0;
		}

		return ctx;

	ts_err:
		TS_FREE( ctx );
		return 0;
	}

	void tsSpawnMixThread( tsContext* ctx )
	{
		if ( ctx->separate_thread ) return;
		InitializeCriticalSectionAndSpinCount( &ctx->critical_section, 0x00000400 );
		ctx->separate_thread = 1;
		CreateThread( 0, 0, tsCtxThread, ctx, 0, 0 );
	}

#elif TS_PLATFORM == TS_MAC

	void tsSleep( int milliseconds )
	{
		usleep( milliseconds * 1000 );
	}

	struct tsContext
	{
		unsigned latency_samples;
		unsigned index0; // read
		unsigned index1; // write
		int Hz;
		int bps;
		int wide_count;
		int sample_count;
		tsPlayingSound* playing;
		__m128* floatA;
		__m128* floatB;
		__m128i* samples;
		tsPlayingSound* playing_pool;
		tsPlayingSound* playing_free;

		// platform specific stuff
		AudioComponentInstance inst;

		// data for tsMix thread, enable these with tsSpawnMixThread
		pthread_t thread;
		pthread_mutex_t mutex;
		int separate_thread;
		int running;
		int sleep_milliseconds;
	};

	static void tsReleaseContext( tsContext* ctx )
	{
		if ( ctx->separate_thread )	pthread_mutex_destroy( &ctx->mutex );
		AudioOutputUnitStop( ctx->inst );
		AudioUnitUninitialize( ctx->inst );
		AudioComponentInstanceDispose( ctx->inst );
		tsPlayingSound* playing = ctx->playing;
		while ( playing )
		{
			tsRemoveFilter( playing );
			playing = playing->next;
		}
		TS_FREE( ctx );
	}

	static void* tsCtxThread( void* udata )
	{
		tsContext* ctx = (tsContext*)udata;

		while ( ctx->running )
		{
			tsMix( ctx );
			if ( ctx->sleep_milliseconds ) tsSleep( ctx->sleep_milliseconds );
			else pthread_yield_np( );
		}

		ctx->separate_thread = 0;
		pthread_exit( 0 );
		return 0;
	}

	static void tsLock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) pthread_mutex_lock( &ctx->mutex );
	}

	static void tsUnlock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) pthread_mutex_unlock( &ctx->mutex );
	}

	static OSStatus tsMemcpyToCA( void* udata, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData );

	tsContext* tsMakeContext( void* unused, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count )
	{
		int bps = sizeof( uint16_t ) * 2;

		AudioComponentDescription comp_desc = { 0 };
		comp_desc.componentType = kAudioUnitType_Output;
		comp_desc.componentSubType = kAudioUnitSubType_DefaultOutput;
		comp_desc.componentFlags = 0;
		comp_desc.componentFlagsMask = 0;
		comp_desc.componentManufacturer = kAudioUnitManufacturer_Apple;

		AudioComponent comp = AudioComponentFindNext( NULL, &comp_desc );
		if ( !comp )
		{
			g_tsErrorReason = "Failed to create output unit from AudioComponentFindNext.";
			return 0;
		}

		AudioStreamBasicDescription stream_desc = { 0 };
		stream_desc.mSampleRate = (double)play_frequency_in_Hz;
		stream_desc.mFormatID = kAudioFormatLinearPCM;
		stream_desc.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
		stream_desc.mFramesPerPacket = 1;
		stream_desc.mChannelsPerFrame = 2;
		stream_desc.mBitsPerChannel = sizeof( uint16_t ) * 8;
		stream_desc.mBytesPerPacket = bps;
		stream_desc.mBytesPerFrame = bps;
		stream_desc.mReserved = 0;

		AudioComponentInstance inst;
		OSStatus ret;
		AURenderCallbackStruct input;

		ret = AudioComponentInstanceNew( comp, &inst );

		int sample_count = play_frequency_in_Hz * num_buffered_seconds;
		int latency_count = (unsigned)TS_ALIGN( play_frequency_in_Hz / latency_factor_in_Hz, 4 );
		TS_ASSERT( sample_count > latency_count );
		int wide_count = (int)TS_ALIGN( sample_count, 4 ) / 4;
		int pool_size = playing_pool_count * sizeof( tsPlayingSound );
		int mix_buffers_size = sizeof( __m128 ) * wide_count * 2;
		int sample_buffer_size = sizeof( __m128i ) * wide_count;
		tsContext* ctx = (tsContext*)TS_ALLOC( sizeof( tsContext ) + mix_buffers_size + sample_buffer_size + 16 + pool_size );
		TS_CHECK( ret == noErr, "AudioComponentInstanceNew failed" );
		ctx->latency_samples = latency_count;
		ctx->index0 = 0;
		ctx->index1 = 0;
		ctx->Hz = play_frequency_in_Hz;
		ctx->bps = bps;
		ctx->wide_count = wide_count;
		ctx->sample_count = wide_count * 4;
		ctx->inst = inst;
		ctx->playing = 0;
		ctx->floatA = (__m128*)(ctx + 1);
		ctx->floatA = (__m128*)TS_ALIGN( ctx->floatA, 16 );
		TS_ASSERT( !((size_t)ctx->floatA & 15) );
		ctx->floatB = ctx->floatA + wide_count;
		ctx->samples = (__m128i*)ctx->floatB + wide_count;
		ctx->running = 1;
		ctx->separate_thread = 0;
		ctx->sleep_milliseconds = 0;

		ret = AudioUnitSetProperty( inst, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &stream_desc, sizeof( stream_desc ) );
		TS_CHECK( ret == noErr, "Failed to set stream forat" );

		input.inputProc = tsMemcpyToCA;
		input.inputProcRefCon = ctx;
		ret = AudioUnitSetProperty( inst, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof( input ) );
		TS_CHECK( ret == noErr, "AudioUnitSetProperty failed" );

		ret = AudioUnitInitialize( inst );
		TS_CHECK( ret == noErr, "Couldn't initialize output unit" );

		ret = AudioOutputUnitStart( inst );
		TS_CHECK( ret == noErr, "Couldn't start output unit" );

		if ( playing_pool_count )
		{
			ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
			for ( int i = 0; i < playing_pool_count - 1; ++i )
				ctx->playing_pool[ i ].next = ctx->playing_pool + i + 1;
			ctx->playing_pool[ playing_pool_count - 1 ].next = 0;
			ctx->playing_free = ctx->playing_pool;
		}

		else
		{
			ctx->playing_pool = 0;
			ctx->playing_free = 0;
		}

		return ctx;

	ts_err:
		TS_FREE( ctx );
		return 0;
	}

	void tsSpawnMixThread( tsContext* ctx )
	{
		if ( ctx->separate_thread ) return;
		pthread_mutex_init( &ctx->mutex, 0 );
		ctx->separate_thread = 1;
		pthread_create( &ctx->thread, 0, tsCtxThread, ctx );
	}

#else

	void tsSleep( int milliseconds )
	{
		SDL_Delay( milliseconds );
	}

	struct tsContext
	{
		unsigned latency_samples;
		unsigned index0; // read
		unsigned index1; // write
		unsigned running_index;
		int Hz;
		int bps;
		int buffer_size;
		int wide_count;
		int sample_count;
		tsPlayingSound* playing;
		__m128* floatA;
		__m128* floatB;
		__m128i* samples;
		tsPlayingSound* playing_pool;
		tsPlayingSound* playing_free;

		// data for tsMix thread, enable these with tsSpawnMixThread
		SDL_Thread* thread;
		SDL_mutex* mutex;
		int separate_thread;
		int running;
		int sleep_milliseconds;
	};

	static void tsReleaseContext( tsContext* ctx )
	{
		if ( ctx->separate_thread )	SDL_DestroyMutex( ctx->mutex );
		tsPlayingSound* playing = ctx->playing;
		while ( playing )
		{
			tsRemoveFilter( playing );
			playing = playing->next;
		}
		SDL_CloseAudio( );
		TS_FREE( ctx );
	}

	int tsCtxThread( void* udata )
	{
		tsContext* ctx = (tsContext*)udata;

		while ( ctx->running )
		{
			tsMix( ctx );
			if ( ctx->sleep_milliseconds ) tsSleep( ctx->sleep_milliseconds );
			else tsSleep( 1 );
		}

		ctx->separate_thread = 0;
		return 0;
	}

	static void tsLock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) SDL_LockMutex( ctx->mutex );
	}

	static void tsUnlock( tsContext* ctx )
	{
		if ( ctx->separate_thread ) SDL_UnlockMutex( ctx->mutex );
	}

	static void tsSDL_AudioCallback( void* udata, Uint8* stream, int len );

	tsContext* tsMakeContext( void* unused, unsigned play_frequency_in_Hz, int latency_factor_in_Hz, int num_buffered_seconds, int playing_pool_count )
	{
		(void)unused;
		int bps = sizeof( uint16_t ) * 2;
		int sample_count = play_frequency_in_Hz * num_buffered_seconds;
		int latency_count = (unsigned)TS_ALIGN( play_frequency_in_Hz / latency_factor_in_Hz, 4 );
		TS_ASSERT( sample_count > latency_count );
		int wide_count = (int)TS_ALIGN( sample_count, 4 ) / 4;
		int pool_size = playing_pool_count * sizeof( tsPlayingSound );
		int mix_buffers_size = sizeof( __m128 ) * wide_count * 2;
		int sample_buffer_size = sizeof( __m128i ) * wide_count;
		tsContext* ctx = 0;
		SDL_AudioSpec wanted;
		int ret = SDL_Init( SDL_INIT_AUDIO );
		TS_CHECK( ret >= 0, "Can't init SDL audio" );

		ctx = (tsContext*)TS_ALLOC( sizeof( tsContext ) + mix_buffers_size + sample_buffer_size + 16 + pool_size );
		TS_CHECK( ctx != NULL, "Can't create audio context" );
		ctx->latency_samples = latency_count;
		ctx->index0 = 0;
		ctx->index1 = 0;
		ctx->Hz = play_frequency_in_Hz;
		ctx->bps = bps;
		ctx->wide_count = wide_count;
		ctx->sample_count = wide_count * 4;
		ctx->playing = 0;
		ctx->floatA = (__m128*)(ctx + 1);
		ctx->floatA = (__m128*)TS_ALIGN( ctx->floatA, 16 );
		TS_ASSERT( !((size_t)ctx->floatA & 15) );
		ctx->floatB = ctx->floatA + wide_count;
		ctx->samples = (__m128i*)ctx->floatB + wide_count;
		ctx->running = 1;
		ctx->separate_thread = 0;
		ctx->sleep_milliseconds = 0;

		SDL_memset( &wanted, 0, sizeof( wanted ) );
		wanted.freq = play_frequency_in_Hz;
		wanted.format = AUDIO_S16SYS;
		wanted.channels = 2; /* 1 = mono, 2 = stereo */
		wanted.samples = 1024;
		wanted.callback = tsSDL_AudioCallback;
		wanted.userdata = ctx;
		ret = SDL_OpenAudio( &wanted, NULL );
		TS_CHECK( ret >= 0, "Can't open SDL audio" );
		SDL_PauseAudio( 0 );

		if ( playing_pool_count )
		{
			ctx->playing_pool = (tsPlayingSound*)(ctx->samples + wide_count);
			for ( int i = 0; i < playing_pool_count - 1; ++i )
				ctx->playing_pool[ i ].next = ctx->playing_pool + i + 1;
			ctx->playing_pool[ playing_pool_count - 1 ].next = 0;
			ctx->playing_free = ctx->playing_pool;
		}

		else
		{
			ctx->playing_pool = 0;
			ctx->playing_free = 0;
		}

		return ctx;

	ts_err:
		if ( ctx ) TS_FREE( ctx );
		return 0;
	}

	void tsSpawnMixThread( tsContext* ctx )
	{
		if ( ctx->separate_thread ) return;
		ctx->mutex = SDL_CreateMutex( );
		ctx->separate_thread = 1;
		ctx->thread = SDL_CreateThread( &tsCtxThread, "TinySoundThread", ctx );
	}

#endif

#if TS_PLATFORM == TS_SDL || TS_PLATFORM == TS_MAC

	static int tsSamplesWritten( tsContext* ctx )
	{
		int index0 = ctx->index0;
		int index1 = ctx->index1;
		if ( index0 <= index1 ) return index1 - index0;
		else return ctx->sample_count - index0 + index1;
	}

	static int tsSamplesUnwritten( tsContext* ctx )
	{
		int index0 = ctx->index0;
		int index1 = ctx->index1;
		if ( index0 <= index1 ) return ctx->sample_count - index1 + index0;
		else return index0 - index1;
	}

	static int tsSamplesToMix( tsContext* ctx )
	{
		int lat = ctx->latency_samples;
		int written = tsSamplesWritten( ctx );
		int dif = lat - written;
		TS_ASSERT( dif >= 0 );
		if ( dif )
		{
			int unwritten = tsSamplesUnwritten( ctx );
			return dif < unwritten ? dif : unwritten;
		}
		return 0;
	}

#define TS_SAMPLES_TO_BYTES( interleaved_sample_count ) ((interleaved_sample_count) * ctx->bps)
#define TS_BYTES_TO_SAMPLES( byte_count ) ((byte_count) / ctx->bps)

	static void tsPushBytes( tsContext* ctx, void* data, int size )
	{
		int index0 = ctx->index0;
		int index1 = ctx->index1;
		int samples = TS_BYTES_TO_SAMPLES( size );
		int sample_count = ctx->sample_count;

		int unwritten = tsSamplesUnwritten( ctx );
		if ( unwritten < samples ) samples = unwritten;
		int can_overflow = index0 <= index1;
		int would_overflow = index1 + samples > sample_count;

		if ( can_overflow && would_overflow )
		{
			int first_size = TS_SAMPLES_TO_BYTES( sample_count - index1 );
			int second_size = size - first_size;
			memcpy( (char*)ctx->samples + TS_SAMPLES_TO_BYTES( index1 ), data, first_size );
			memcpy( ctx->samples, (char*)data + first_size, second_size );
			ctx->index1 = TS_BYTES_TO_SAMPLES( second_size );
		}

		else
		{
			memcpy( (char*)ctx->samples + TS_SAMPLES_TO_BYTES( index1 ), data, size );
			ctx->index1 += TS_BYTES_TO_SAMPLES( size );
		}
	}

	static int tsPullBytes( tsContext* ctx, void* dst, int size )
	{
		int index0 = ctx->index0;
		int index1 = ctx->index1;
		int allowed_size = TS_SAMPLES_TO_BYTES( tsSamplesWritten( ctx ) );
		int zeros = 0;

		if ( allowed_size < size )
		{
			zeros = size - allowed_size;
			size = allowed_size;
		}

		if ( index1 >= index0 )
		{
			memcpy( dst, ((char*)ctx->samples) + TS_SAMPLES_TO_BYTES( index0 ), size );
			ctx->index0 += TS_BYTES_TO_SAMPLES( size );
		}

		else
		{
			int first_size = TS_SAMPLES_TO_BYTES( ctx->sample_count ) - TS_SAMPLES_TO_BYTES( index0 );
			if ( first_size > size ) first_size = size;
			int second_size = size - first_size;
			memcpy( dst, ((char*)ctx->samples) + TS_SAMPLES_TO_BYTES( index0 ), first_size );
			memcpy( ((char*)dst) + first_size, ctx->samples, second_size );
			if ( second_size ) ctx->index0 = TS_BYTES_TO_SAMPLES( second_size );
			else ctx->index0 += TS_BYTES_TO_SAMPLES( first_size );
		}

		return zeros;
	}

#endif

void tsShutdownContext( tsContext* ctx )
{
	if ( ctx->separate_thread )
	{
		tsLock( ctx );
		ctx->running = 0;
		tsUnlock( ctx );
	}

	while ( ctx->separate_thread ) tsSleep( 1 );
	tsReleaseContext( ctx );
}

void tsThreadSleepDelay( tsContext* ctx, int milliseconds )
{
	ctx->sleep_milliseconds = milliseconds;
}

int tsInsertSound( tsContext* ctx, tsPlayingSound* sound )
{
	// Cannot use tsPlayingSound if tsMakeContext was passed non-zero for playing_pool_count
	// since non-zero playing_pool_count means the context is doing some memory-management
	// for a playing sound pool. InsertSound assumes the pool does not exist, and is apart
	// of the lower-level API (see top of this header for documentation details).
	TS_ASSERT( ctx->playing_pool == 0 );

	if ( sound->active ) return 0;
	tsLock( ctx );
	sound->next = ctx->playing;
	ctx->playing = sound;
	sound->active = 1;
	tsUnlock( ctx );
	return 1;
}

// NOTE: does not allow delay_in_seconds to be negative (clamps at 0)
void tsSetDelay( tsContext* ctx, tsPlayingSound* sound, float delay_in_seconds )
{
	if ( delay_in_seconds < 0.0f ) delay_in_seconds = 0.0f;
	sound->sample_index = (int)(delay_in_seconds * (float)ctx->Hz);
	sound->sample_index = -(int)TS_ALIGN( sound->sample_index, 4 );
}

tsPlaySoundDef tsMakeDef( tsLoadedSound* sound )
{
	tsPlaySoundDef def;
	def.paused = 0;
	def.looped = 0;
	def.volume_left = 1.0f;
	def.volume_right = 1.0f;
	def.pan = 0.5f;
	def.pitch = 1.0f;
	def.delay = 0.0f;
	def.loaded = sound;
	return def;
}

tsPlayingSound* tsPlaySound( tsContext* ctx, tsPlaySoundDef def )
{
	tsLock( ctx );

	tsPlayingSound* playing = ctx->playing_free;
	if ( !playing ) return 0;
	ctx->playing_free = playing->next;
	*playing = tsMakePlayingSound( def.loaded );
	playing->active = 1;
	playing->paused = def.paused;
	playing->looped = def.looped;
	tsSetVolume( playing, def.volume_left, def.volume_right );
	tsSetPan( playing, def.pan );
	tsSetPitch( playing, def.pitch );
	tsSetDelay( ctx, playing, def.delay );
	playing->next = ctx->playing;
	ctx->playing = playing;

	tsUnlock( ctx );

	return playing;
}


void tsStopAllSounds( tsContext* ctx )
{
	// This is apart of the high level API, not the low level API.
	// If using the low level API you must write your own function to
	// stop playing all sounds.
	TS_ASSERT( ctx->playing_pool != 0 );

	tsLock( ctx );
	tsPlayingSound* sound = ctx->playing;
	while ( sound )
	{
		// let tsMix() remove the sound
		sound->active = 0;
		sound = sound->next;
	}
	tsUnlock( ctx );
}

#if TS_PLATFORM == TS_WINDOWS

	static void tsPosition( tsContext* ctx, int* byte_to_lock, int* bytes_to_write )
	{
		// compute bytes to be written to direct sound
		DWORD play_cursor;
		DWORD write_cursor;
#ifdef __cplusplus
		HRESULT hr = ctx->buffer->GetCurrentPosition( &play_cursor, &write_cursor );
#else
		HRESULT hr = ctx->buffer->lpVtbl->GetCurrentPosition( ctx->buffer, &play_cursor, &write_cursor );
#endif
		TS_ASSERT( hr == DS_OK );

		DWORD lock = (ctx->running_index * ctx->bps) % ctx->buffer_size;
		DWORD target_cursor = (write_cursor + ctx->latency_samples * ctx->bps) % ctx->buffer_size;
		target_cursor = (DWORD)TS_ALIGN( target_cursor, 16 );
		DWORD write;

		if ( lock > target_cursor )
		{
			write = (ctx->buffer_size - lock) + target_cursor;
		}

		else
		{
			write = target_cursor - lock;
		}

		*byte_to_lock = lock;
		*bytes_to_write = write;
	}

	static void tsMemcpyToDS( tsContext* ctx, int16_t* samples, int byte_to_lock, int bytes_to_write )
	{
		// copy mixer buffers to direct sound
		void* region1;
		DWORD size1;
		void* region2;
		DWORD size2;
#ifdef __cplusplus
		HRESULT hr = ctx->buffer->Lock( byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0 );

		if ( hr == DSERR_BUFFERLOST )
		{
			ctx->buffer->Restore( );
			hr = ctx->buffer->Lock( byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0 );
		}
#else
		HRESULT hr = ctx->buffer->lpVtbl->Lock( ctx->buffer, byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0 );

		if ( hr == DSERR_BUFFERLOST )
		{
			ctx->buffer->lpVtbl->Restore( ctx->buffer );
			hr = ctx->buffer->lpVtbl->Lock( ctx->buffer, byte_to_lock, bytes_to_write, &region1, &size1, &region2, &size2, 0 );
		}
#endif

		if ( !SUCCEEDED( hr ) )
			return;

		unsigned running_index = ctx->running_index;
		INT16* sample1 = (INT16*)region1;
		DWORD sample1_count = size1 / ctx->bps;
		memcpy( sample1, samples, sample1_count * sizeof( INT16 ) * 2 );
		samples += sample1_count * 2;
		running_index += sample1_count;

		INT16* sample2 = (INT16*)region2;
		DWORD sample2_count = size2 / ctx->bps;
		memcpy( sample2, samples, sample2_count * sizeof( INT16 ) * 2 );
		samples += sample2_count * 2;
		running_index += sample2_count;
		
#ifdef __cplusplus
		ctx->buffer->Unlock( region1, size1, region2, size2 );
#else
		ctx->buffer->lpVtbl->Unlock( ctx->buffer, region1, size1, region2, size2 );
#endif
		ctx->running_index = running_index;

		// meager hack to fill out sound buffer before playing
		static int first;
		if ( !first )
		{
#ifdef __cplusplus
			ctx->buffer->Play( 0, 0, DSBPLAY_LOOPING );
#else
			ctx->buffer->lpVtbl->Play( ctx->buffer, 0, 0, DSBPLAY_LOOPING );
#endif
			first = 1;
		}
	}

#elif TS_PLATFORM == TS_MAC

	static OSStatus tsMemcpyToCA( void* udata, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData )
	{
		tsContext* ctx = (tsContext*)udata;
		int bps = ctx->bps;
		int samples_requested_to_consume = inNumberFrames;
		AudioBuffer* buffer = ioData->mBuffers;

		TS_ASSERT( ioData->mNumberBuffers == 1 );
		TS_ASSERT( buffer->mNumberChannels == 2 );
		int byte_size = buffer->mDataByteSize;
		TS_ASSERT( byte_size == samples_requested_to_consume * bps );

		int zero_bytes = tsPullBytes( ctx, buffer->mData, byte_size );
		memset( ((char*)buffer->mData) + (byte_size - zero_bytes), 0, zero_bytes );

		return noErr;
	}

#elif TS_PLATFORM == TS_SDL

	static void tsSDL_AudioCallback( void* udata, Uint8* stream, int len )
	{
		tsContext* ctx = (tsContext*)udata;
		int zero_bytes = tsPullBytes( ctx, stream, len );
		memset( stream + (len - zero_bytes), 0, zero_bytes );
	}

#endif

static void tsPitchShift( float pitchShift, int num_samples_to_process, float sampleRate, float* indata, tsPitchData** pitch_filter );

// Pitch processing tunables
#define TS_MAX_FRAME_LENGTH 4096
#define TS_PITCH_FRAME_SIZE 512
#define TS_PITCH_QUALITY 8

// interals
#define TS_STEPSIZE (TS_PITCH_FRAME_SIZE / TS_PITCH_QUALITY)
#define TS_OVERLAP (TS_PITCH_FRAME_SIZE - TS_STEPSIZE)
#define TS_EXPECTED_FREQUENCY (2.0f * 3.14159265359f * (float)TS_STEPSIZE / (float)TS_PITCH_FRAME_SIZE)

// TODO:
// Use a memory pool for these things. For now they are just malloc16'd/free16'd
// Not high priority to use a pool, since pitch shifting is already really expensive,
// and cost of malloc is dwarfed. But would be a nice-to-have for potential memory
// fragmentation issues.
typedef struct tsPitchData
{
	float pitch_shifted_output_samples[ TS_MAX_FRAME_LENGTH ];
	float in_FIFO[ TS_STEPSIZE + TS_PITCH_FRAME_SIZE ];
	float out_FIFO[ TS_STEPSIZE + TS_PITCH_FRAME_SIZE ];
	float fft_data[ 2 * TS_PITCH_FRAME_SIZE ];
	float previous_phase[ TS_PITCH_FRAME_SIZE / 2 + 4 ];
	float sum_phase[ TS_PITCH_FRAME_SIZE / 2 + 4 ];
	float window_accumulator[ TS_STEPSIZE + TS_PITCH_FRAME_SIZE ];
	float freq[ TS_PITCH_FRAME_SIZE ];
	float mag[ TS_PITCH_FRAME_SIZE ];
	float pitch_shift_workspace[ TS_PITCH_FRAME_SIZE ];
	int index;
} tsPitchData;

static void tsRemoveFilter( tsPlayingSound* playing )
{
	for ( int i = 0; i < 2; i++ )
	{
		if ( playing->pitch_filter[ i ] )
		{
			free16( playing->pitch_filter[ i ] );
			playing->pitch_filter[ i ] = 0;
		}
	}
}

void tsMix( tsContext* ctx )
{
	tsLock( ctx );

#if TS_PLATFORM == TS_WINDOWS

	int byte_to_lock;
	int bytes_to_write;
	tsPosition( ctx, &byte_to_lock, &bytes_to_write );

	if ( !bytes_to_write ) goto unlock;
	int samples_to_write = bytes_to_write / ctx->bps;

#elif TS_PLATFORM == TS_MAC || TS_PLATFORM == TS_SDL

	int samples_to_write = tsSamplesToMix( ctx );
	if ( !samples_to_write ) goto unlock;
	int bytes_to_write = samples_to_write * ctx->bps;

#else
#endif

	// clear mixer buffers
	int wide_count = samples_to_write / 4;
	TS_ASSERT( !(samples_to_write & 3) );

	__m128* floatA = ctx->floatA;
	__m128* floatB = ctx->floatB;
	__m128 zero = _mm_set1_ps( 0.0f );

	for ( int i = 0; i < wide_count; ++i )
	{
		floatA[ i ] = zero;
		floatB[ i ] = zero;
	}

	// mix all playing sounds into the mixer buffers
	tsPlayingSound** ptr = &ctx->playing;
	while ( *ptr )
	{
		tsPlayingSound* playing = *ptr;

		// immediately remove any inactive elements
		if ( !playing->active || !ctx->running )
			goto remove;

		tsLoadedSound* loaded = playing->loaded_sound;
		__m128* cA = (__m128*)loaded->channels[ 0 ];
		__m128* cB = (__m128*)loaded->channels[ 1 ];

		// Attempted to play a sound with no audio.
		// Make sure the audio file was loaded properly. Check for
		// error messages in g_tsErrorReason.
		TS_ASSERT( cA );

		int mix_count = samples_to_write;
		int offset = playing->sample_index;
		int remaining = loaded->sample_count - offset;
		if ( remaining < mix_count ) mix_count = remaining;
		TS_ASSERT( remaining > 0 );

		float vA0 = playing->volume0 * playing->pan0;
		float vB0 = playing->volume1 * playing->pan1;
		__m128 vA = _mm_set1_ps( vA0 );
		__m128 vB = _mm_set1_ps( vB0 );

		// skip sound if it's delay is longer than mix_count and
		// handle various delay cases
		int delay_offset = 0;
		if ( offset < 0 )
		{
			int samples_till_positive = -offset;
			int mix_leftover = mix_count - samples_till_positive;

			if ( mix_leftover <= 0 )
			{
				playing->sample_index += mix_count;
				goto get_next_playing_sound;
			}

			else
			{
				offset = 0;
				delay_offset = samples_till_positive;
				mix_count = mix_leftover;
			}
		}
		TS_ASSERT( !(delay_offset & 3) );

		// skip all paused sounds
		if ( playing->paused )
			goto get_next_playing_sound;

		// SIMD offets
		int mix_wide = (int)TS_ALIGN( mix_count, 4 ) / 4;
		int offset_wide = (int)TS_TRUNC( offset, 4 ) / 4;
		int delay_wide = (int)TS_ALIGN( delay_offset, 4 ) / 4;

		// use tsPitchShift to on-the-fly pitch shift some samples
		// only call this function if the user set a custom pitch value
		float pitch = playing->pitch;
		if ( pitch != 1.0f )
		{
			int sample_count = (mix_wide - 2 * delay_wide) * 4;
			int falling_behind = sample_count > TS_MAX_FRAME_LENGTH;

			// TS_MAX_FRAME_LENGTH represents max samples we can pitch shift in one go. In the event
			// that this process takes longer than the time required to play the actual sound, just
			// fall back to the original sound (non-pitch shifted). This will sound very ugly. To
			// prevent falling behind, make sure not to pitch shift too many sounds at once. Try tweaking
			// TS_PITCH_QUALITY to make it lower (must be a power of 2).
			if ( !falling_behind )
			{
				tsPitchShift( pitch, sample_count, (float)ctx->Hz, (float*)(cA + delay_wide + offset_wide), playing->pitch_filter );
				cA = (__m128 *)playing->pitch_filter[ 0 ]->pitch_shifted_output_samples;

				if ( loaded->channel_count == 2 )
				{
					tsPitchShift( pitch, sample_count, (float)ctx->Hz, (float*)(cB + delay_wide + offset_wide), playing->pitch_filter + 1 );
					cB = (__m128 *)playing->pitch_filter[ 1 ]->pitch_shifted_output_samples;
				}

				offset_wide = -delay_wide;
			}
		}

		// apply volume, load samples into float buffers
		switch ( loaded->channel_count )
		{
		case 1:
			for ( int i = delay_wide; i < mix_wide - delay_wide; ++i )
			{
				__m128 A = cA[ i + offset_wide ];
				__m128 B = _mm_mul_ps( A, vB );
				A = _mm_mul_ps( A, vA );
				floatA[ i ] = _mm_add_ps( floatA[ i ], A );
				floatB[ i ] = _mm_add_ps( floatB[ i ], B );
			}
			break;

		case 2:
		{
			for ( int i = delay_wide; i < mix_wide - delay_wide; ++i )
			{
				__m128 A = cA[ i + offset_wide ];
				__m128 B = cB[ i + offset_wide ];

				A = _mm_mul_ps( A, vA );
				B = _mm_mul_ps( B, vB );
				floatA[ i ] = _mm_add_ps( floatA[ i ], A );
				floatB[ i ] = _mm_add_ps( floatB[ i ], B );
			}
		}	break;
		}

		// playing list logic
		playing->sample_index += mix_count;
		if ( playing->sample_index == loaded->sample_count )
		{
			if ( playing->looped )
			{
				playing->sample_index = 0;
				goto get_next_playing_sound;
			}

			remove:
			playing->sample_index = 0;
			*ptr = (*ptr)->next;
			playing->next = 0;
			playing->active = 0;

			tsRemoveFilter( playing );

			// if using high-level API manage the tsPlayingSound memory ourselves
			if ( ctx->playing_pool )
			{
				playing->next = ctx->playing_free;
				ctx->playing_free = playing;
			}

			// we already incremented next pointer, so don't do it again
			continue;
		}

	get_next_playing_sound:
		if ( *ptr ) ptr = &(*ptr)->next;
		else break;
	}

	// load all floats into 16 bit packed interleaved samples
#if TS_PLATFORM == TS_WINDOWS

	__m128i* samples = ctx->samples;
	for ( int i = 0; i < wide_count; ++i )
	{
		__m128i a = _mm_cvtps_epi32( floatA[ i ] );
		__m128i b = _mm_cvtps_epi32( floatB[ i ] );
		__m128i a0b0a1b1 = _mm_unpacklo_epi32( a, b );
		__m128i a2b2a3b3 = _mm_unpackhi_epi32( a, b );
		samples[ i ] = _mm_packs_epi32( a0b0a1b1, a2b2a3b3 );
	}
	tsMemcpyToDS( ctx, (int16_t*)samples, byte_to_lock, bytes_to_write );

#elif TS_PLATFORM == TS_MAC || TS_PLATFORM == TS_SDL

	// Since the ctx->samples array is already in use as a ring buffer
	// reusing floatA to store output is a good way to temporarly store
	// the final samples. Then a single ring buffer push can be used
	// afterwards. Pretty hacky, but whatever :)
	__m128i* samples = (__m128i*)floatA;
	for ( int i = 0; i < wide_count; ++i )
	{
		__m128i a = _mm_cvtps_epi32( floatA[ i ] );
		__m128i b = _mm_cvtps_epi32( floatB[ i ] );
		__m128i a0b0a1b1 = _mm_unpacklo_epi32( a, b );
		__m128i a2b2a3b3 = _mm_unpackhi_epi32( a, b );
		samples[ i ] = _mm_packs_epi32( a0b0a1b1, a2b2a3b3 );
	}
	tsPushBytes( ctx, samples, bytes_to_write );

#else
#endif

unlock:
	tsUnlock( ctx );
}

// TODO:
// Try this optimization out (2N POINT REAL FFT USING AN N POINT COMPLEX FFT)
// http://www.fftguru.com/fftguru.com.tutorial2.pdf

#include <math.h>

static uint32_t tsRev32( uint32_t x )
{
	uint32_t a = ((x & 0xAAAAAAAA) >> 1) | ((x & 0x55555555) << 1);
	uint32_t b = ((a & 0xCCCCCCCC) >> 2) | ((a & 0x33333333) << 2);
	uint32_t c = ((b & 0xF0F0F0F0) >> 4) | ((b & 0x0F0F0F0F) << 4);
	uint32_t d = ((c & 0xFF00FF00) >> 8) | ((c & 0x00FF00FF) << 8);
	return (d >> 16) | (d << 16);
}

static uint32_t tsPopCount( uint32_t x )
{
	uint32_t a = x - ((x >> 1) & 0x55555555);
	uint32_t b = (((a >> 2) & 0x33333333) + (a & 0x33333333));
	uint32_t c = (((b >> 4) + b) & 0x0F0F0F0F);
	uint32_t d = c + (c >> 8);
	uint32_t e = d + (d >> 16);
	uint32_t f = e & 0x0000003F;
	return f;
}

static uint32_t tsLog2( uint32_t x )
{
	uint32_t a = x | ( x >> 1 );
	uint32_t b = a | ( a >> 2 );
	uint32_t c = b | ( b >> 4 );
	uint32_t d = c | ( c >> 8 );
	uint32_t e = d | ( d >> 16 );
	uint32_t f = e >> 1;
	return tsPopCount( f );
}

// x contains real inputs
// y contains imaginary inputs
// count must be a power of 2
// sign must be 1.0 (forward transform) or -1.0f (inverse transform)
static void tsFFT( float* x, float* y, int count, float sign )
{
	int exponent = (int)tsLog2( (uint32_t)count );

	// bit reversal stage
	// swap all elements with their bit reversed index within the
	// lowest level of the Cooley-Tukey recursion tree
	for ( int i = 1; i < count - 1; i++ )
	{
		uint32_t j = tsRev32( (uint32_t)i );
		j >>= (32 - exponent);
		if ( i < (int)j )
		{
			float tx = x[ i ];
			float ty = y[ i ];
			x[ i ] = x[ j ];
			y[ i ] = y[ j ];
			x[ j ] = tx;
			y[ j ] = ty;
		}
	}

	// for each recursive iteration
	for ( int iter = 0, L = 1; iter < exponent; ++iter )
	{
		int Ls = L;
		L <<= 1;
		float ur = 1.0f; // cos( pi / 2 )
		float ui = 0;    // sin( pi / 2 )
		float arg = 3.14159265359f / (float)Ls;
		float wr = cosf( arg );
		float wi = -sign * sinf( arg );

		// rows in DFT submatrix
		for ( int j = 0; j < Ls; ++j )
		{
			// do butterflies upon DFT row elements
			for ( int i = j; i < count; i += L )
			{
				int index = i + Ls;
				float x_index = x[ index ];
				float y_index = y[ index ];
				float x_i = x[ i ];
				float y_i = y[ i ];

				float tr = ur * x_index - ui * y_index;
				float ti = ur * y_index + ui * x_index;
				float x_low = x_i - tr;
				float x_high = x_i + tr;
				float y_low = y_i - ti;
				float y_high = y_i + ti;

				x[ index ] = x_low; 
				y[ index ] = y_low;
				x[ i ] = x_high;
				y[ i ] = y_high;
			}

			// Rotate u1 and u2 via Givens rotations (2d planar rotation).
			// This keeps cos/sin calls in the outermost loop.
			// Floating point error is scaled proportionally to Ls.
			float t = ur * wr - ui * wi;
			ui = ur * wi + ui * wr;
			ur = t;
		}
	}

	// scale factor for forward transform
	if ( sign > 0 )
	{
		float inv_count = 1.0f / (float)count;
		for ( int i = 0; i < count; i++ )
		{
			x[ i ] *= inv_count;
			y[ i ] *= inv_count;
		}
	}
}

#ifdef _MSC_VER

	#define TS_ALIGN16_0 __declspec( align( 16 ) )
	#define TS_ALIGN16_1

#else

	#define TS_ALIGN16_0
	#define TS_ALIGN16_1 __attribute__((aligned(16)))

#endif

// SSE2 trig funcs from https://github.com/to-miz/sse_mathfun_extension/
#define _PS_CONST( Name, Val ) \
	TS_ALIGN16_0 float _ps_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }

#define _PS_CONST_TYPE( Name, Type, Val ) \
	TS_ALIGN16_0 Type _ps_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }

#define _PI32_CONST( Name, Val ) \
	TS_ALIGN16_0 int _pi32_##Name[ 4 ] TS_ALIGN16_1 = { Val, Val, Val, Val }

_PS_CONST_TYPE( sign_mask, int, (int)0x80000000 );
_PS_CONST_TYPE( inv_sign_mask, int, (int)~0x80000000 );

_PS_CONST( atanrange_hi, 2.414213562373095f );
_PS_CONST( atanrange_lo, 0.4142135623730950f );
_PS_CONST( cephes_PIO2F, 1.5707963267948966192f );
_PS_CONST( cephes_PIO4F, 0.7853981633974483096f );
_PS_CONST( 1 , 1.0f );
_PS_CONST( 0p5, 0.5f );
_PS_CONST( 0, 0 );
_PS_CONST( sincof_p0, -1.9515295891E-4f );
_PS_CONST( sincof_p1,  8.3321608736E-3f );
_PS_CONST( sincof_p2, -1.6666654611E-1f );
_PS_CONST( atancof_p0, 8.05374449538e-2f );
_PS_CONST( atancof_p1, 1.38776856032E-1f );
_PS_CONST( atancof_p2, 1.99777106478E-1f );
_PS_CONST( atancof_p3, 3.33329491539E-1f );
_PS_CONST( cephes_PIF, 3.141592653589793238f );
_PS_CONST( cephes_2PIF, 2.0f * 3.141592653589793238f );
_PS_CONST( cephes_FOPI, 1.27323954473516f ); // 4 / M_PI
_PS_CONST( minus_cephes_DP1, -0.78515625f );
_PS_CONST( minus_cephes_DP2, -2.4187564849853515625e-4f );
_PS_CONST( minus_cephes_DP3, -3.77489497744594108e-8f );
_PS_CONST( coscof_p0,  2.443315711809948E-005f );
_PS_CONST( coscof_p1, -1.388731625493765E-003f );
_PS_CONST( coscof_p2,  4.166664568298827E-002f );
_PS_CONST( frame_size, (float)TS_PITCH_FRAME_SIZE );

_PI32_CONST( 1, 1 );
_PI32_CONST( inv1, ~1 );
_PI32_CONST( 2, 2 );
_PI32_CONST( 4, 4 );

#if 0 /* temporary comment it out, remove "unused functions" warning */
static __m128 _mm_atan_ps( __m128 x )
{
	__m128 sign_bit, y;

	sign_bit = x;
	/* take the absolute value */
	x = _mm_and_ps( x, *(__m128*)_ps_inv_sign_mask );
	/* extract the sign bit (upper one) */
	sign_bit = _mm_and_ps( sign_bit, *(__m128*)_ps_sign_mask );

/* range reduction, init x and y depending on range */
	/* x > 2.414213562373095 */
	__m128 cmp0 = _mm_cmpgt_ps( x, *(__m128*)_ps_atanrange_hi );
	/* x > 0.4142135623730950 */
	__m128 cmp1 = _mm_cmpgt_ps( x, *(__m128*)_ps_atanrange_lo );

	/* x > 0.4142135623730950 && !( x > 2.414213562373095 ) */
	__m128 cmp2 = _mm_andnot_ps( cmp0, cmp1 );

	/* -( 1.0/x ) */
	__m128 y0 = _mm_and_ps( cmp0, *(__m128*)_ps_cephes_PIO2F );
	__m128 x0 = _mm_div_ps( *(__m128*)_ps_1, x );
	x0 = _mm_xor_ps( x0, *(__m128*)_ps_sign_mask );

	__m128 y1 = _mm_and_ps( cmp2, *(__m128*)_ps_cephes_PIO4F );
	/* (x-1.0)/(x+1.0) */
	__m128 x1_o = _mm_sub_ps( x, *(__m128*)_ps_1 );
	__m128 x1_u = _mm_add_ps( x, *(__m128*)_ps_1 );
	__m128 x1 = _mm_div_ps( x1_o, x1_u );

	__m128 x2 = _mm_and_ps( cmp2, x1 );
	x0 = _mm_and_ps( cmp0, x0 );
	x2 = _mm_or_ps( x2, x0 );
	cmp1 = _mm_or_ps( cmp0, cmp2 );
	x2 = _mm_and_ps( cmp1, x2 );
	x = _mm_andnot_ps( cmp1, x );
	x = _mm_or_ps( x2, x );

	y = _mm_or_ps( y0, y1 );

	__m128 zz = _mm_mul_ps( x, x );
	__m128 acc = *(__m128*)_ps_atancof_p0;
	acc = _mm_mul_ps( acc, zz );
	acc = _mm_sub_ps( acc, *(__m128*)_ps_atancof_p1 );
	acc = _mm_mul_ps( acc, zz );
	acc = _mm_add_ps( acc, *(__m128*)_ps_atancof_p2 );
	acc = _mm_mul_ps( acc, zz );
	acc = _mm_sub_ps( acc, *(__m128*)_ps_atancof_p3 );
	acc = _mm_mul_ps( acc, zz );
	acc = _mm_mul_ps( acc, x );
	acc = _mm_add_ps( acc, x );
	y = _mm_add_ps( y, acc );

	/* update the sign */
	y = _mm_xor_ps( y, sign_bit );

	return y;
}

static __m128 _mm_atan2_ps( __m128 y, __m128 x )
{
	__m128 x_eq_0 = _mm_cmpeq_ps( x, *(__m128*)_ps_0 );
	__m128 x_gt_0 = _mm_cmpgt_ps( x, *(__m128*)_ps_0 );
	__m128 x_le_0 = _mm_cmple_ps( x, *(__m128*)_ps_0 );
	__m128 y_eq_0 = _mm_cmpeq_ps( y, *(__m128*)_ps_0 );
	__m128 x_lt_0 = _mm_cmplt_ps( x, *(__m128*)_ps_0 );
	__m128 y_lt_0 = _mm_cmplt_ps( y, *(__m128*)_ps_0 );

	__m128 zero_mask = _mm_and_ps( x_eq_0, y_eq_0 );
	__m128 zero_mask_other_case = _mm_and_ps( y_eq_0, x_gt_0 );
	zero_mask = _mm_or_ps( zero_mask, zero_mask_other_case );

	__m128 pio2_mask = _mm_andnot_ps( y_eq_0, x_eq_0 );
	__m128 pio2_mask_sign = _mm_and_ps( y_lt_0, *(__m128*)_ps_sign_mask );
	__m128 pio2_result = *(__m128*)_ps_cephes_PIO2F;
	pio2_result = _mm_xor_ps( pio2_result, pio2_mask_sign );
	pio2_result = _mm_and_ps( pio2_mask, pio2_result );

	__m128 pi_mask = _mm_and_ps( y_eq_0, x_le_0 );
	__m128 pi = *(__m128*)_ps_cephes_PIF;
	__m128 pi_result = _mm_and_ps( pi_mask, pi );

	__m128 swap_sign_mask_offset = _mm_and_ps( x_lt_0, y_lt_0 );
	swap_sign_mask_offset = _mm_and_ps( swap_sign_mask_offset, *(__m128*)_ps_sign_mask );

	__m128 offset0 = _mm_setzero_ps();
	__m128 offset1 = *(__m128*)_ps_cephes_PIF;
	offset1 = _mm_xor_ps( offset1, swap_sign_mask_offset );

	__m128 offset = _mm_andnot_ps( x_lt_0, offset0 );
	offset = _mm_and_ps( x_lt_0, offset1 );

	__m128 arg = _mm_div_ps( y, x );
	__m128 atan_result = _mm_atan_ps( arg );
	atan_result = _mm_add_ps( atan_result, offset );

	/* select between zero_result, pio2_result and atan_result */

	__m128 result = _mm_andnot_ps( zero_mask, pio2_result );
	atan_result = _mm_andnot_ps( pio2_mask, atan_result );
	atan_result = _mm_andnot_ps( pio2_mask, atan_result);
	result = _mm_or_ps( result, atan_result );
	result = _mm_or_ps( result, pi_result );

	return result;
}
#endif

static void _mm_sincos_ps( __m128 x, __m128 *s, __m128 *c )
{
	__m128 xmm1, xmm2, xmm3 = _mm_setzero_ps(), sign_bit_sin, y;
	__m128i emm0, emm2, emm4;
	sign_bit_sin = x;
	/* take the absolute value */
	x = _mm_and_ps(x, *(__m128*)_ps_inv_sign_mask);
	/* extract the sign bit (upper one) */
	sign_bit_sin = _mm_and_ps(sign_bit_sin, *(__m128*)_ps_sign_mask);
  
	/* scale by 4/Pi */
	y = _mm_mul_ps(x, *(__m128*)_ps_cephes_FOPI);

	/* store the integer part of y in emm2 */
	emm2 = _mm_cvttps_epi32(y);

	/* j=(j+1) & (~1) (see the cephes sources) */
	emm2 = _mm_add_epi32(emm2, *(__m128i*)_pi32_1);
	emm2 = _mm_and_si128(emm2, *(__m128i*)_pi32_inv1);
	y = _mm_cvtepi32_ps(emm2);

	emm4 = emm2;

	/* get the swap sign flag for the sine */
	emm0 = _mm_and_si128(emm2, *(__m128i*)_pi32_4);
	emm0 = _mm_slli_epi32(emm0, 29);
	__m128 swap_sign_bit_sin = _mm_castsi128_ps(emm0);

	/* get the polynom selection mask for the sine*/
	emm2 = _mm_and_si128(emm2, *(__m128i*)_pi32_2);
	emm2 = _mm_cmpeq_epi32(emm2, _mm_setzero_si128());
	__m128 poly_mask = _mm_castsi128_ps(emm2);

	/* The magic pass: "Extended precision modular arithmetic" 
		x = ((x - y * DP1) - y * DP2) - y * DP3; */
	xmm1 = *(__m128*)_ps_minus_cephes_DP1;
	xmm2 = *(__m128*)_ps_minus_cephes_DP2;
	xmm3 = *(__m128*)_ps_minus_cephes_DP3;
	xmm1 = _mm_mul_ps(y, xmm1);
	xmm2 = _mm_mul_ps(y, xmm2);
	xmm3 = _mm_mul_ps(y, xmm3);
	x = _mm_add_ps(x, xmm1);
	x = _mm_add_ps(x, xmm2);
	x = _mm_add_ps(x, xmm3);

	emm4 = _mm_sub_epi32(emm4, *(__m128i*)_pi32_2);
	emm4 = _mm_andnot_si128(emm4, *(__m128i*)_pi32_4);
	emm4 = _mm_slli_epi32(emm4, 29);
	__m128 sign_bit_cos = _mm_castsi128_ps(emm4);

	sign_bit_sin = _mm_xor_ps(sign_bit_sin, swap_sign_bit_sin);

  
	/* Evaluate the first polynom  (0 <= x <= Pi/4) */
	__m128 z = _mm_mul_ps(x,x);
	y = *(__m128*)_ps_coscof_p0;

	y = _mm_mul_ps(y, z);
	y = _mm_add_ps(y, *(__m128*)_ps_coscof_p1);
	y = _mm_mul_ps(y, z);
	y = _mm_add_ps(y, *(__m128*)_ps_coscof_p2);
	y = _mm_mul_ps(y, z);
	y = _mm_mul_ps(y, z);
	__m128 tmp = _mm_mul_ps(z, *(__m128*)_ps_0p5);
	y = _mm_sub_ps(y, tmp);
	y = _mm_add_ps(y, *(__m128*)_ps_1);
  
	/* Evaluate the second polynom  (Pi/4 <= x <= 0) */

	__m128 y2 = *(__m128*)_ps_sincof_p0;
	y2 = _mm_mul_ps(y2, z);
	y2 = _mm_add_ps(y2, *(__m128*)_ps_sincof_p1);
	y2 = _mm_mul_ps(y2, z);
	y2 = _mm_add_ps(y2, *(__m128*)_ps_sincof_p2);
	y2 = _mm_mul_ps(y2, z);
	y2 = _mm_mul_ps(y2, x);
	y2 = _mm_add_ps(y2, x);

	/* select the correct result from the two polynoms */  
	xmm3 = poly_mask;
	__m128 ysin2 = _mm_and_ps(xmm3, y2);
	__m128 ysin1 = _mm_andnot_ps(xmm3, y);
	y2 = _mm_sub_ps(y2,ysin2);
	y = _mm_sub_ps(y, ysin1);

	xmm1 = _mm_add_ps(ysin1,ysin2);
	xmm2 = _mm_add_ps(y,y2);
 
	/* update the sign */
	*s = _mm_xor_ps(xmm1, sign_bit_sin);
	*c = _mm_xor_ps(xmm2, sign_bit_cos);
}

static __m128i select_si( __m128i a, __m128i b, __m128i mask )
{
	return _mm_xor_si128( a, _mm_and_si128( mask, _mm_xor_si128( b, a ) ) );
}

#define tsVonHann( i ) (-0.5f * cosf( 2.0f * 3.14159265359f * (float)(i) / (float)TS_PITCH_FRAME_SIZE ) + 0.5f)

static __m128 tsVonHann4( int i )
{
	__m128 k4 = _mm_set_ps( (float)(i * 4 + 3), (float)(i * 4 + 2), (float)(i * 4 + 1), (float)(i * 4) );
	k4 = _mm_mul_ps( *(__m128*)_ps_cephes_2PIF, k4 );
	k4 = _mm_div_ps( k4, *(__m128*)_ps_frame_size );

	// Seems like _mm_cos_ps and _mm_sincos_ps was causing some audio popping...
	// I'm not really skilled enough to fix it, but feel free to try: http://gruntthepeon.free.fr/ssemath/sse_mathfun.h
	// My guess is some large negative or positive values were causing some
	// precision trouble. In this case manually calling 4 cosines is not
	// really a big deal, since this function is not a bottleneck.

#if 0
	__m128 c = _mm_cos_ps( k4 );
#elif 0
	__m128 s, c;
	_mm_sincos_ps( k4, &s, &c );
#else
	__m128 c = k4;
	float* cf = (float*)&c;
	cf[ 0 ] = cosf( cf[ 0 ] );
	cf[ 1 ] = cosf( cf[ 1 ] );
	cf[ 2 ] = cosf( cf[ 2 ] );
	cf[ 3 ] = cosf( cf[ 3 ] );
#endif

	__m128 von_hann = _mm_add_ps( _mm_mul_ps( _mm_set_ps1( -0.5f ), c ), _mm_set_ps1( 0.5f ) );
	return von_hann;
}

float smbAtan2f( float x, float y )
{
	float signx = x > 0 ? 1.0f : -1.0f;
	if (x == 0) return 0;
	if (y == 0) return signx * 3.14159265f / 2.0f;
	return atan2f( x, y );
}

// Analysis and synthesis steps learned from Bernsee's wonderful blog post:
// http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
static void tsPitchShift( float pitchShift, int num_samples_to_process, float sampleRate, float* indata, tsPitchData** pitch_filter )
{
	TS_ASSERT( num_samples_to_process <= TS_MAX_FRAME_LENGTH );

	// make sure compiler didn't do anything weird with the member
	// offsets of tsPitchData. All arrays must be 16 byte aligned
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->pitch_shifted_output_samples) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->fft_data) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->previous_phase) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->sum_phase) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->window_accumulator) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->freq) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->mag) & 15) );
	TS_ASSERT( !((size_t)&(((tsPitchData*)0)->pitch_shift_workspace) & 15) );

	tsPitchData* pf;

	if ( *pitch_filter == NULL )
	{
		pf = (tsPitchData*)malloc16( sizeof( tsPitchData ) );
		memset( pf, 0, sizeof( tsPitchData ) );
		*pitch_filter = pf;
	}
	else
	{
		pf = *pitch_filter;
	}

	float freqPerBin = sampleRate / (float)TS_PITCH_FRAME_SIZE;
	__m128 freq_per_bin = _mm_set_ps1( sampleRate / (float)TS_PITCH_FRAME_SIZE );
	__m128 pi = *(__m128*)_ps_cephes_PIF;
	__m128 two_pi = *(__m128*)_ps_cephes_2PIF;
	__m128 pitch_quality = _mm_set_ps1( (float)TS_PITCH_QUALITY );
	float* out_samples = pf->pitch_shifted_output_samples;
	if ( pf->index == 0 ) pf->index = TS_OVERLAP;

	while ( num_samples_to_process )
	{
		int copy_count = TS_PITCH_FRAME_SIZE - pf->index;
		if ( num_samples_to_process < copy_count ) copy_count = num_samples_to_process;

		memcpy( pf->in_FIFO + pf->index, indata, sizeof( float ) * copy_count );
		memcpy( out_samples, pf->out_FIFO + pf->index - TS_OVERLAP, sizeof( float ) * copy_count );

		int start_index = pf->index;
		int offset = start_index & 3;
		start_index += 4 - offset;

		for ( int i = 0; i < offset; ++i )
			pf->in_FIFO[ pf->index + i ] /= 32768.0f;

		int extra = copy_count & 3;
		copy_count = copy_count / 4 - extra;
		__m128* in_FIFO = (__m128*)(pf->in_FIFO + pf->index + offset);
		TS_ASSERT( !((size_t)in_FIFO & 15) );
		__m128 int16_max = _mm_set_ps1( 32768.0f );

		for ( int i = 0; i < copy_count; ++i )
		{
			__m128 val = in_FIFO[ i ];
			__m128 div = _mm_div_ps( val, int16_max );
			in_FIFO[ i ] = div;
		}

		for ( int i = 0, copy_count4 = copy_count * 4; i < extra; ++i )
		{
			int index = copy_count4 + i;
			pf->in_FIFO[ pf->index + index ] /= 32768.0f;
		}

		TS_ASSERT( !((size_t)out_samples & 15) );
		__m128* out_samples4 = (__m128*)out_samples;
		for ( int i = 0; i < copy_count; ++i )
		{
			__m128 val = out_samples4[ i ];
			__m128 mul = _mm_mul_ps( val, int16_max );
			out_samples4[ i ] = mul;
		}

		for ( int i = 0, copy_count4 = copy_count * 4; i < extra; ++i )
		{
			int index = copy_count4 + i;
			out_samples[ index ] *= 32768.0f;
		}

		copy_count = copy_count * 4 + extra;
		num_samples_to_process -= copy_count;
		pf->index += copy_count;
		indata += copy_count;
		out_samples += copy_count;

		if ( pf->index >= TS_PITCH_FRAME_SIZE )
		{
			pf->index = TS_OVERLAP;
			{
				__m128* fft_data = (__m128*)pf->fft_data;
				__m128* in_FIFO = (__m128*)pf->in_FIFO;

				for ( int k = 0; k < TS_PITCH_FRAME_SIZE / 4; k++ )
				{
					__m128 von_hann = tsVonHann4( k );
					__m128 sample = in_FIFO[ k ];
					__m128 windowed_sample = _mm_mul_ps( sample, von_hann );
					fft_data[ k ] = windowed_sample;
				}
			}

			memset( pf->fft_data + TS_PITCH_FRAME_SIZE, 0, TS_PITCH_FRAME_SIZE * sizeof( float ) );
			tsFFT( pf->fft_data, pf->fft_data + TS_PITCH_FRAME_SIZE, TS_PITCH_FRAME_SIZE, 1.0f );

			{
				__m128* fft_data = (__m128*)pf->fft_data;
				__m128* previous_phase = (__m128*)pf->previous_phase;
				__m128* magnitudes = (__m128*)pf->mag;
				__m128* frequencies = (__m128*)pf->freq;
				int simd_count = (TS_PITCH_FRAME_SIZE / 2) / 4;

				for ( int k = 0; k <= simd_count; k++ )
				{
					__m128 real = fft_data[ k ];
					__m128 imag = fft_data[ (TS_PITCH_FRAME_SIZE / 4) + k ];
					__m128 overlap_phase = _mm_set_ps( (float)(k * 4 + 3) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 2) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 1) * TS_EXPECTED_FREQUENCY, (float)(k * 4) * TS_EXPECTED_FREQUENCY );
					__m128 k4 = _mm_set_ps( (float)(k * 4 + 3), (float)(k * 4 + 2), (float)(k * 4 + 1), (float)(k * 4) );

					__m128 mag = _mm_mul_ps( _mm_set_ps1( 2.0f ), _mm_sqrt_ps( _mm_add_ps( _mm_mul_ps( real, real ), _mm_mul_ps( imag, imag ) ) ) );
#if 0
					__m128 phase = _mm_atan2_ps( imag, real );
#else
					__m128 phase; // = _mm_atan2_ps( imag, real );
					float *phasef = (float*)&phase;
					float *realf = (float*)&real;
					float *imagf = (float*)&imag;
					for ( int i=0; i<4; i++ ) phasef[ i ] = smbAtan2f( imagf[ i ], realf[ i ] );
#endif
					__m128 phase_dif = _mm_sub_ps( phase, previous_phase[ k ] );

					previous_phase[ k ] = phase;
					phase_dif = _mm_sub_ps( phase_dif, overlap_phase );

					// map delta phase into +/- pi interval
					__m128i qpd = _mm_cvttps_epi32( _mm_div_ps( phase_dif, pi ) );
					__m128i zero = _mm_setzero_si128( );
					__m128i ltzero_mask = _mm_cmplt_epi32( qpd, zero );
					__m128i ones_bit = _mm_and_si128( qpd, _mm_set1_epi32( 1 ) );
					__m128i neg_qpd = _mm_sub_epi32( qpd, ones_bit );
					__m128i pos_qpd = _mm_add_epi32( qpd, ones_bit );
					qpd = select_si( pos_qpd, neg_qpd, ltzero_mask );
					__m128 pi_range_offset = _mm_mul_ps( pi, _mm_cvtepi32_ps( qpd ) );
					phase_dif = _mm_sub_ps( phase_dif, pi_range_offset );

					__m128 deviation = _mm_div_ps( _mm_mul_ps( _mm_set_ps1( (float)TS_PITCH_QUALITY ), phase_dif ), two_pi );
					__m128 true_freq_estimated = _mm_add_ps( _mm_mul_ps( k4, freq_per_bin ), _mm_mul_ps( deviation, freq_per_bin ) );

					magnitudes[ k ] = mag;
					frequencies[ k ] = true_freq_estimated;
				}
			}

			// actual pitch shifting work
			// shift frequencies into workspace
			memset( pf->pitch_shift_workspace, 0, (TS_PITCH_FRAME_SIZE / 2) * sizeof( float ) );
			for ( int k = 0; k <= TS_PITCH_FRAME_SIZE / 2; k++ )
			{
				int index = (int)(k * pitchShift);
				if ( index <= TS_PITCH_FRAME_SIZE / 2 )
					pf->pitch_shift_workspace[ index ] = pf->freq[ k ] * pitchShift;
			}

			// swap buffers around to reuse old pf->preq buffer as the new workspace
			float* frequencies = pf->pitch_shift_workspace;
			float* pitch_shift_workspace = pf->freq;
			float* magnitudes = pf->mag;

			// shift magnitudes into workspace
			memset( pitch_shift_workspace, 0, TS_PITCH_FRAME_SIZE * sizeof( float ) );
			for ( int k = 0; k <= TS_PITCH_FRAME_SIZE / 2; k++ )
			{
				int index = (int)(k * pitchShift);
				if ( index <= TS_PITCH_FRAME_SIZE / 2 )
					pitch_shift_workspace[ index ] += magnitudes[ k ];
			}

			// track where the shifted magnitudes are
			magnitudes = pitch_shift_workspace;
			
			{
				__m128* magnitudes4 = (__m128*)magnitudes;
				__m128* frequencies4 = (__m128*)frequencies;
				__m128* fft_data = (__m128*)pf->fft_data;
				__m128* sum_phase = (__m128*)pf->sum_phase;
				int simd_count = (TS_PITCH_FRAME_SIZE / 2) / 4;

				for ( int k = 0; k <= simd_count; k++ )
				{
					__m128 mag = magnitudes4[ k ];
					__m128 freq = frequencies4[ k ];
					__m128 freq_per_bin_k = _mm_set_ps( (float)(k * 4 + 3) * freqPerBin, (float)(k * 4 + 2) * freqPerBin, (float)(k * 4 + 1) * freqPerBin, (float)(k * 4) * freqPerBin );

					freq = _mm_sub_ps( freq, freq_per_bin_k );
					freq = _mm_div_ps( freq, freq_per_bin );

					freq = _mm_mul_ps( two_pi, freq );
					freq = _mm_div_ps( freq, pitch_quality );

					__m128 overlap_phase = _mm_set_ps( (float)(k * 4 + 3) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 2) * TS_EXPECTED_FREQUENCY, (float)(k * 4 + 1) * TS_EXPECTED_FREQUENCY, (float)(k * 4) * TS_EXPECTED_FREQUENCY );
					freq = _mm_add_ps( freq, overlap_phase );

					__m128 phase = sum_phase[ k ];
					phase = _mm_add_ps( phase, freq );
					sum_phase[ k ] = phase;

					__m128 c, s;
					_mm_sincos_ps( phase, &s, &c );
					__m128 real = _mm_mul_ps( mag, c );
					__m128 imag = _mm_mul_ps( mag, s );

					fft_data[ k ] = real;
					fft_data[ (TS_PITCH_FRAME_SIZE / 4) + k ] = imag;
				}
			}

			for ( int k = TS_PITCH_FRAME_SIZE + 2; k < 2 * TS_PITCH_FRAME_SIZE - 2; ++k )
				pf->fft_data[ k ] = 0;

			tsFFT( pf->fft_data, pf->fft_data + TS_PITCH_FRAME_SIZE, TS_PITCH_FRAME_SIZE, -1 );

			{
				__m128* fft_data = (__m128*)pf->fft_data;
				__m128* window_accumulator = (__m128*)pf->window_accumulator;

				for ( int k = 0; k < TS_PITCH_FRAME_SIZE / 4; ++k )
				{
					__m128 von_hann = tsVonHann4( k );
					__m128 fft_data_segment = fft_data[ k ];
					__m128 accumulator_segment = window_accumulator[ k ];
					__m128 divisor = _mm_div_ps( pitch_quality, _mm_set_ps1( 8.0f ) );
					fft_data_segment = _mm_mul_ps( von_hann, fft_data_segment );
					fft_data_segment = _mm_div_ps( fft_data_segment, divisor );
					accumulator_segment = _mm_add_ps( accumulator_segment, fft_data_segment );
					window_accumulator[ k ] = accumulator_segment;
				}
			}

			memcpy( pf->out_FIFO, pf->window_accumulator, TS_STEPSIZE * sizeof( float ) );
			memmove( pf->window_accumulator, pf->window_accumulator + TS_STEPSIZE, TS_PITCH_FRAME_SIZE * sizeof( float ) );
			memmove( pf->in_FIFO, pf->in_FIFO + TS_STEPSIZE, TS_OVERLAP * sizeof( float ) );
		}
	}
}

#endif // TINYSOUND_IMPLEMENTATION

/*
	zlib license:

	Copyright (c) 2017 Randy Gaul http://www.randygaul.net

	This software is provided 'as-is', without any express or implied warranty.
	In no event will the authors be held liable for any damages arising from
	the use of this software.

	Permission is granted to anyone to use this software for any purpose,
	including commercial applications, and to alter it and redistribute it
	freely, subject to the following restrictions:
	  1. The origin of this software must not be misrepresented; you must not
	     claim that you wrote the original software. If you use this software
	     in a product, an acknowledgment in the product documentation would be
	     appreciated but is not required.
	  2. Altered source versions must be plainly marked as such, and must not
	     be misrepresented as being the original software.
	  3. This notice may not be removed or altered from any source distribution.
*/