shithub: ft²

--- a/make-linux-nomidi.sh

+++ b/make-linux-nomidi.sh

@@ -3,7 +3,7 @@

 rm release/other/ft2-clone &> /dev/null

 echo Compiling \(with no MIDI functionality\), please wait patiently...

-gcc -DNDEBUG src/gfxdata/*.c src/mixer/*.c src/*.c -lSDL2 -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone

+gcc -DNDEBUG src/gfxdata/*.c src/mixer/*.c src/modloaders/*.c src/smploaders/*.c src/*.c -lSDL2 -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone

 rm src/gfxdata/*.o src/*.o &> /dev/null

--- a/make-linux.sh

+++ b/make-linux.sh

@@ -3,7 +3,7 @@

 rm release/other/ft2-clone &> /dev/null

 echo Compiling, please wait patiently...

-gcc -DNDEBUG -DHAS_MIDI -D__LINUX_ALSA__ src/rtmidi/*.cpp src/gfxdata/*.c src/mixer/*.c src/*.c -lSDL2 -lpthread -lasound -lstdc++ -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone

+gcc -DNDEBUG -DHAS_MIDI -D__LINUX_ALSA__ src/rtmidi/*.cpp src/gfxdata/*.c src/mixer/*.c src/modloaders/*.c src/smploaders/*.c src/*.c -lSDL2 -lpthread -lasound -lstdc++ -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone

 rm src/rtmidi/*.o src/gfxdata/*.o src/*.o &> /dev/null

--- a/make-macos.sh

+++ b/make-macos.sh

@@ -36,7 +36,7 @@

 function compile() {

     rm $1 &> /dev/null

-    clang $VERBOSE $CFLAGS -F /Library/Frameworks -g0 -DNDEBUG -DHAS_MIDI -D__MACOSX_CORE__ -stdlib=libc++ src/rtmidi/*.cpp src/gfxdata/*.c src/mixer/*.c src/*.c -Winit-self -Wno-deprecated -Wextra -Wunused -mno-ms-bitfields -Wno-missing-field-initializers -Wswitch-default $LDFLAGS -L /Library/Frameworks -framework SDL2 -framework CoreMidi -framework CoreAudio -framework Cocoa -liconv -lpthread -lm -lstdc++ -o $1

+    clang $VERBOSE $CFLAGS -F /Library/Frameworks -g0 -DNDEBUG -DHAS_MIDI -D__MACOSX_CORE__ -stdlib=libc++ src/rtmidi/*.cpp src/gfxdata/*.c src/mixer/*.c src/modloaders/*.c src/smploaders/*.c src/*.c -Winit-self -Wno-deprecated -Wextra -Wunused -mno-ms-bitfields -Wno-missing-field-initializers -Wswitch-default $LDFLAGS -L /Library/Frameworks -framework SDL2 -framework CoreMidi -framework CoreAudio -framework Cocoa -liconv -lpthread -lm -lstdc++ -o $1

     return $?

--- a/src/ft2_audio.c

+++ b/src/ft2_audio.c

@@ -47,8 +47,8 @@

 	voice_t *v = voice;

 	for (int32_t i = 0; i < MAX_VOICES*2; i++, v++)

+		v->dOldHz = 0.0;

 		v->oldDelta = 0;

-		v->oldRevDelta = UINT32_MAX;

@@ -109,7 +109,6 @@

 		return false;

-	calcRevMixDeltaTable();

 	resumeAudio();

 	setWavRenderFrequency(audio.freq);

@@ -140,10 +139,7 @@

 	const bool mustRecalcTables = audio.freq != oldAudioFreq;

 	if (mustRecalcTables)

-	{

 		calcReplayerVars(audio.freq);

-		calcRevMixDeltaTable();

-	}

 void setBackOldAudioFreq(void) // for song-to-WAV rendering

@@ -153,10 +149,7 @@

 	audio.freq = oldAudioFreq;

 	if (mustRecalcTables)

-	{

 		calcReplayerVars(audio.freq);

-		calcRevMixDeltaTable();

-	}

 void P_SetSpeed(uint16_t bpm)

@@ -383,7 +376,7 @@

 		if (status & IS_Period)

-			// use cached values if possible

+			// use cached values when possible

 			if (ch->finalPeriod != ch->oldFinalPeriod)

 				ch->oldFinalPeriod = ch->finalPeriod;

@@ -390,19 +383,22 @@

 				if (ch->finalPeriod == 0) // in FT2, period 0 -> delta 0

+					v->dOldHz = 0.0;

 					v->oldDelta = 0;

-					v->oldRevDelta = UINT32_MAX;

 					v->dSincLUT = gKaiserSinc;

 				else

-					v->oldDelta = getMixerDelta(ch->finalPeriod);

-					v->oldRevDelta = getRevMixerDelta(ch->finalPeriod); // for calc'ing max samples in inner mix loop before sample end/loop end

+					const double dHz = dPeriod2Hz(ch->finalPeriod);

+					const uint64_t delta64 = (int64_t)((dHz * audio.dHz2MixDeltaMul) + 0.5); // Hz -> rounded 32.32 fixed-point

+					v->dOldHz = dHz;

+					v->oldDelta = delta64;

 					// decide which sinc LUT to use according to resampling ratio

-					if (v->oldDelta <= 0x130000000) // 1.1875 (32.32 fixed-point)

+					if (delta64 <= 0x130000000) // 1.1875 (32.32fp)

 						v->dSincLUT = gKaiserSinc;

-					else if (v->oldDelta <= 0x180000000) // 1.5 (32.32 fixed-point)

+					else if (delta64 <= 0x180000000) // 1.5 (32.32fp)

 						v->dSincLUT = gDownSample1;

 					else

 						v->dSincLUT = gDownSample2;

@@ -410,7 +406,7 @@

 			v->delta = v->oldDelta;

-			v->revDelta = v->oldRevDelta;

+			v->dHz = v->dOldHz;

 		if (status & IS_NyTon)

@@ -878,18 +874,24 @@

 	syncedChannel_t *c = chSyncData.channels;

 	stmTyp *s = stm;

+	voice_t *v = voice;

-	for (int32_t i = 0; i < song.antChn; i++, c++, s++)

+	for (int32_t i = 0; i < song.antChn; i++, c++, s++, v++)

-		c->finalPeriod = s->finalPeriod;

-		c->fineTune = s->fineTune;

-		c->relTonNr = s->relTonNr;

+		c->vol = s->finalVol;

+		c->dHz = v->dHz;

 		c->instrNr = s->instrNr;

 		c->sampleNr = s->sampleNr;

-		c->envSustainActive = s->envSustainActive;

 		c->status = s->tmpStatus;

-		c->dFinalVol = s->dFinalVol;

-		c->smpStartPos = s->smpStartPos;

+		c->smpStartPos = (uint16_t)s->smpStartPos;

+		c->pianoNoteNr = 255; // no piano key

+		if (songPlaying && (c->status & IS_Period) && s->envSustainActive)

+		{

+			const int32_t note = getPianoKey(s->finalPeriod, s->fineTune, s->relTonNr);

+			if (note >= 0 && note <= 95)

+				c->pianoNoteNr = (uint8_t)note;

+		}

 	chSyncData.timestamp = audio.tickTime64;

--- a/src/ft2_audio.h

+++ b/src/ft2_audio.h

@@ -34,7 +34,7 @@

 	uint32_t freq, audLatencyPerfValInt, audLatencyPerfValFrac, samplesPerTick, musicTimeSpeedVal;

 	uint64_t tickTime64, tickTime64Frac, tickTimeTab[MAX_BPM+1];

 	double dRampQuickVolMul, dRampTickMul, dRampTickMulTab[MAX_BPM+1];

-	double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned;

+	double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned, dHz2MixDeltaMul;

 	double dAudioLatencyMs, dSamplesPerTick, dTickSampleCounter, dSamplesPerTickTab[MAX_BPM+1];

 	SDL_AudioDeviceID dev;

@@ -47,8 +47,8 @@

 	const int16_t *base16, *revBase16;

 	bool active, backwards, isFadeOutVoice, hasLooped;

 	uint8_t mixFuncOffset, pan, loopType;

-	int32_t pos, end, loopStart, loopLength;

-	uint32_t volRampSamples, revDelta, oldRevDelta;

+	int32_t pos, end, loopStart, loopLength, oldPeriod;

+	uint32_t volRampSamples;

 	uint64_t posFrac, delta, oldDelta;

 	// if (loopEnabled && hasLooped && samplingPos <= loopStart+SINC_LEFT_TAPS) readFixedTapsFromThisPointer();

@@ -56,7 +56,7 @@

 	const int16_t *leftEdgeTaps16;

 	const double *dSincLUT;

-	double dVol, dDestVolL, dDestVolR, dVolL, dVolR, dVolDeltaL, dVolDeltaR;

+	double dOldHz, dHz, dVol, dDestVolL, dDestVolR, dVolL, dVolR, dVolDeltaL, dVolDeltaR;

 } voice_t;

 typedef struct pattSyncData_t

--- a/src/ft2_diskop.c

+++ b/src/ft2_diskop.c

@@ -1187,6 +1187,38 @@

+static bool moduleExtensionAccepted(char *extPtr)

+{

+	int32_t i = 0;

+	while (true)

+	{

+		const char *str = supportedModExtensions[i++];

+		if (!_stricmp(str, "END_OF_LIST"))

+			return false;

+		if (!_stricmp(str, extPtr))

+			break;

+	}

+	return true;

+}

+static bool sampleExtensionAccepted(char *extPtr)

+{

+	int32_t i = 0;

+	while (true)

+	{

+		const char *str = supportedSmpExtensions[i++];

+		if (!_stricmp(str, "END_OF_LIST"))

+			return false;

+		if (!_stricmp(str, extPtr))

+			break;

+	}

+	return true;

+}

 static uint8_t handleEntrySkip(UNICHAR *nameU, bool isDir)

 	// skip if illegal name or filesize >32-bit

@@ -1223,12 +1255,13 @@

 	else if (!FReq_ShowAllFiles)

-		const int32_t extOffset = getExtOffset(name, nameLen);

+		int32_t extOffset = getExtOffset(name, nameLen);

 		if (extOffset == -1)

 			goto skipEntry;

+		extOffset++; // skip '.'

 		const int32_t extLen = (int32_t)strlen(&name[extOffset]);

-		if (extLen < 3 || extLen > 5)

+		if (extLen < 2 || extLen > 6)

 			goto skipEntry; // no possibly known extensions to filter out

 		char *extPtr = &name[extOffset];

@@ -1239,117 +1272,46 @@

 			default:

 			case DISKOP_ITEM_MODULE:

-				if (extLen == 3)

-				{

-					if (_stricmp(".xm", extPtr) && _stricmp(".ft", extPtr))

-						goto skipEntry; // skip, none of the extensions above

-				}

-				else if (extLen == 4)

-				{

-					if (editor.moduleSaveMode == MOD_SAVE_MODE_WAV)

-					{

-						if (_stricmp(".nst", extPtr) && _stricmp(".mod", extPtr) &&

-							_stricmp(".s3m", extPtr) && _stricmp(".stm", extPtr) &&

-							_stricmp(".fst", extPtr) && _stricmp(".wav", extPtr))

-						{

-							goto skipEntry; // skip, none of the extensions above

-						}

-					}

-					else

-					{

-						if (_stricmp(".nst", extPtr) && _stricmp(".mod", extPtr) &&

-							_stricmp(".s3m", extPtr) && _stricmp(".stm", extPtr) &&

-							_stricmp(".fst", extPtr))

-						{

-							goto skipEntry; // skip, none of the extensions above

-						}

-					}

-				}

-				else

-				{

+				if (editor.moduleSaveMode == MOD_SAVE_MODE_WAV && !_stricmp("wav", extPtr))

+					break; // show .wav files when save mode is "WAV"

+				if (!moduleExtensionAccepted(extPtr))

 					goto skipEntry;

-				}

 			break;

 			case DISKOP_ITEM_INSTR:

-				if (extLen == 3)

-				{

-					if (_stricmp(".xi", extPtr))

-						goto skipEntry; // skip, none of the extensions above

-				}

-				else if (extLen == 4)

-				{

-					if (_stricmp(".iff", extPtr) && _stricmp(".raw", extPtr) &&

-						_stricmp(".wav", extPtr) && _stricmp(".snd", extPtr) &&

-						_stricmp(".smp", extPtr) && _stricmp(".sam", extPtr) &&

-						_stricmp(".aif", extPtr) && _stricmp(".pat", extPtr))

-					{

-						goto skipEntry; // skip, none of the extensions above

-					}

-				}

-				else if (extLen == 5)

-				{

-					if (_stricmp(".aiff", extPtr))

-						goto skipEntry; // skip, not the extension above

-				}

-				else

-				{

+				if (!_stricmp("xi", extPtr))

+					break;

+				if (!sampleExtensionAccepted(extPtr))

 					goto skipEntry;

-				}

 			break;

 			case DISKOP_ITEM_SAMPLE:

-				if (extLen == 4)

-				{

-					if (_stricmp(".iff", extPtr) && _stricmp(".raw", extPtr) &&

-						_stricmp(".wav", extPtr) && _stricmp(".snd", extPtr) &&

-						_stricmp(".smp", extPtr) && _stricmp(".sam", extPtr) &&

-						_stricmp(".aif", extPtr))

-					{

-						goto skipEntry; // skip, none of the extensions above

-					}

-				}

-				else if (extLen == 5)

-				{

-					if (_stricmp(".aiff", extPtr))

-						goto skipEntry; // skip, not the extension above

-				}

-				else

-				{

+				if (!sampleExtensionAccepted(extPtr))

 					goto skipEntry;

-				}

 			break;

 			case DISKOP_ITEM_PATTERN:

-				if (extLen == 3)

-				{

-					if (_stricmp(".xp", extPtr))

-						goto skipEntry; // skip, not the extension above

-				}

-				else

-				{

-					goto skipEntry;

-				}

+				if (!_stricmp("xp", extPtr))

+					break;

+				goto skipEntry;

 			break;

 			case DISKOP_ITEM_TRACK:

-				if (extLen == 3)

-				{

-					if (_stricmp(".xt", extPtr))

-						goto skipEntry;  // skip, not the extension above

-				}

-				else

-				{

-					goto skipEntry;

-				}

+				if (!_stricmp("xt", extPtr))

+					break;

+				goto skipEntry;

 			break;

--- a/src/ft2_inst_ed.c

+++ b/src/ft2_inst_ed.c

@@ -18,7 +18,6 @@

 #include "ft2_video.h"

 #include "ft2_sample_loader.h"

 #include "ft2_diskop.h"

-#include "ft2_module_loader.h"

 #include "ft2_tables.h"

 #include "ft2_bmp.h"

 #include "ft2_structs.h"

@@ -1657,12 +1656,8 @@

 			syncedChannel_t *c = chSyncData->channels;

 			for (int32_t i = 0; i < song.antChn; i++, c++)

-				if (c->instrNr == editor.curInstr && c->envSustainActive)

-				{

-					const int32_t note = getPianoKey(c->finalPeriod, c->fineTune, c->relTonNr);

-					if (note >= 0 && note <= 95)

-						newStatus[note] = true;

-				}

+				if (c->instrNr == editor.curInstr && c->pianoNoteNr <= 95)

+					newStatus[c->pianoNoteNr] = true;

 		else // song is not playing (jamming from keyboard/MIDI)

@@ -3412,7 +3407,7 @@

 loadDone:

 	fclose(f);

-	fixSampleName(editor.curInstr);

+	fixInstrAndSampleNames(editor.curInstr);

 	editor.updateCurInstr = true; // setMouseBusy(false) is called in the input/video thread when done

 	if (ih.antSamp > MAX_SMP_PER_INST)

@@ -3426,14 +3421,13 @@

 	(void)ptr;

-static bool fileIsInstr(UNICHAR *filename)

+bool fileIsInstr(UNICHAR *filenameU)

-	char header[22];

-	FILE *f = UNICHAR_FOPEN(filename, "rb");

+	FILE *f = UNICHAR_FOPEN(filenameU, "rb");

 	if (f == NULL)

 		return false;

+	char header[22];

 	fread(header, 1, sizeof (header), f);

 	fclose(f);

--- a/src/ft2_inst_ed.h

+++ b/src/ft2_inst_ed.h

@@ -8,6 +8,8 @@

 void drawC4Rate(void);

+bool fileIsInstr(UNICHAR *filenameU);

 void saveInstr(UNICHAR *filenameU, int16_t nr);

 void loadInstr(UNICHAR *filenameU);

 void copyInstr(void); // dstInstr = srcInstr

--- a/src/ft2_module_loader.c

+++ b/src/ft2_module_loader.c

@@ -10,7 +10,6 @@

 #include <unistd.h>

 #endif

 #include "ft2_header.h"

-#include "ft2_config.h"

 #include "ft2_scopes.h"

 #include "ft2_trim.h"

 #include "ft2_inst_ed.h"

@@ -24,1857 +23,206 @@

 #include "ft2_midi.h"

 #include "ft2_events.h"

 #include "ft2_video.h"

-#include "ft2_tables.h"

 #include "ft2_structs.h"

+#include "ft2_sysreqs.h"

-/* This is a *HUGE* mess!

-** I hope you never have to modify it, and you probably shouldn't either.

-** You will experience a lot of headaches if you dig into this file...

-** If something looks to be wrong, it's probably right!

-**

-** The actual module load routines are ported from FT2 and slightly modified.

-*/

+bool loadDIGI(FILE *f, uint32_t filesize);

+bool loadMOD(FILE *f, uint32_t filesize);

+bool loadS3M(FILE *f, uint32_t filesize);

+bool loadSTK(FILE *f, uint32_t filesize);

+bool loadSTM(FILE *f, uint32_t filesize);

+bool loadXM(FILE *f, uint32_t filesize);

 enum

-	FORMAT_NONE = 0,

-	FORMAT_XM = 1,

-	FORMAT_MOD = 2,

-	FORMAT_S3M = 3,

-	FORMAT_STM = 4

+	FORMAT_UNKNOWN = 0,

+	FORMAT_POSSIBLY_STK = 1,

+	FORMAT_XM = 2,

+	FORMAT_MOD = 3,

+	FORMAT_S3M = 4,

+	FORMAT_STM = 5,

+	FORMAT_DIGI = 6

};

-// .MOD types

-enum

+// file extensions accepted by Disk Op. in module mode

+char *supportedModExtensions[] =

-	FORMAT_MK, // ProTracker or compatible

-	FORMAT_FLT, // StarTrekker (4-channel modules only)

-	FORMAT_FT2, // FT2 (or other trackers, multichannel)

-	FORMAT_STK, // The Ultimate SoundTracker (15 samples)

-	FORMAT_NT, // NoiseTracker

-	FORMAT_HMNT, // His Master's NoiseTracker (special one)

+	"xm", "ft", "nst", "stk", "mod", "s3m", "stm", "fst", "digi",

-	FORMAT_UNKNOWN // may be The Ultimate Soundtracker (set to FORMAT_STK later)

+	"END_OF_LIST" // do NOT move, remove or edit this line!

};

-// DO NOT TOUCH THESE STRUCTS!

+// globals for module loaders

+volatile bool tmpLinearPeriodsFlag;

+int16_t pattLensTmp[MAX_PATTERNS];

+tonTyp *pattTmp[MAX_PATTERNS];

+instrTyp *instrTmp[1+256];

+songTyp songTmp;

+// --------------------------

-#ifdef _MSC_VER

-#pragma pack(push)

-#pragma pack(1)

-#endif

-typedef struct songSTMinstrHeaderTyp_t

-{

-	char name[12];

-	uint8_t nul, insDisk;

-	uint16_t reserved1, len, repS, repE;

-	uint8_t vol, reserved2;

-	uint16_t rate;

-	int32_t reserved3;

-	uint16_t paraLen;

-}

-#ifdef __GNUC__

-__attribute__ ((packed))

-#endif

-songSTMinstrHeaderTyp;

-typedef struct songSTMHeaderTyp_t

-{

-	char name[20], sig[8];

-	uint8_t id1a, typ;

-	uint8_t verMajor, verMinor;

-	uint8_t tempo, ap, vol, reserved[13];

-	songSTMinstrHeaderTyp instr[31];

-	uint8_t songTab[128];

-}

-#ifdef __GNUC__

-__attribute__ ((packed))

-#endif

-songSTMHeaderTyp;

-typedef struct songS3MinstrHeaderTyp_t

-{

-	uint8_t typ;

-	char dosName[12];

-	uint8_t memSegH;

-	uint16_t memSeg;

-	int32_t len, repS, repE;

-	uint8_t vol, dsk, pack, flags;

-	int32_t c2Spd, res1;

-	uint16_t gusPos;

-	uint8_t res2[6];

-	char name[28], id[4];

-}

-#ifdef __GNUC__

-__attribute__ ((packed))

-#endif

-songS3MinstrHeaderTyp;

-typedef struct songS3MHeaderTyp_t

-{

-	char name[28];

-	uint8_t id1a, typ;

-	uint16_t res1;

-	int16_t songTabLen, antInstr, antPatt;

-	uint16_t flags, trackerID, ver;

-	char id[4];

-	uint8_t globalVol, defSpeed, defTempo, masterVol, res2[12], chanType[32];

-}

-#ifdef __GNUC__

-__attribute__ ((packed))

-#endif

-songS3MHeaderTyp;

-#ifdef _MSC_VER

-#pragma pack(pop)

-#endif

-static volatile uint8_t loadedFormat;

-static volatile bool linearFreqTable, musicIsLoading, moduleLoaded, moduleFailedToLoad;

-static uint8_t oldPlayMode, pattBuff[12288], packedPattData[65536];

-static const uint8_t stmEff[16] = { 0, 0, 11, 0, 10, 2, 1, 3, 4, 7, 0, 5 ,6, 0, 0, 0 };

+static volatile bool musicIsLoading, moduleLoaded, moduleFailedToLoad;

 static SDL_Thread *thread;

-// these temporarily read to, then copied to real struct if load was OK (should not need to be volatile'd)

-static int16_t pattLensTmp[MAX_PATTERNS];

-static uint32_t extraSampleLengths[32-MAX_SMP_PER_INST]; // ModPlug Tracker & OpenMPT supports up to 32 samples per instrument in .XM

-static tonTyp *pattTmp[MAX_PATTERNS];

-static instrTyp *instrTmp[1+256];

-static songTyp songTmp;

+static uint8_t oldPlayMode;

 static void setupLoadedModule(void);

 static void freeTmpModule(void);

-static bool loadInstrHeader(FILE *f, uint16_t i, bool externalThreadFlag);

-static bool loadInstrSample(FILE *f, uint16_t i, bool externalThreadFlag);

-void unpackPatt(uint8_t *dst, uint8_t *src, uint16_t len, int32_t antChn);

-static bool tmpPatternEmpty(uint16_t nr);

-static bool loadPatterns(FILE *f, uint16_t antPtn, bool externalThreadFlag);

-void checkSampleRepeat(sampleTyp *s);

-// ft2_replayer.c

-extern const char modSig[32][5];

-static bool allocateTmpInstr(int16_t nr)

+// Crude module detection routine. These aren't always accurate detections!

+static int8_t detectModule(FILE *f)

-	if (instrTmp[nr] != NULL)

-		return false; // already allocated

+	uint8_t D[256], I[4];

-	instrTyp *p = (instrTyp *)calloc(1, sizeof (instrTyp));

-	if (p == NULL)

-		return false;

-	sampleTyp *s = p->samp;

-	for (int32_t i = 0; i < MAX_SMP_PER_INST; i++, s++)

-	{

-		s->pan = 128;

-		s->vol = 64;

-	}

-	instrTmp[nr] = p;

-	return true;

-}

-#define IS_ID(s, b) !strncmp(s, b, 4)

-static uint8_t getModType(uint8_t *numChannels, const char *id)

-{

-	uint8_t modFormat = FORMAT_UNKNOWN;

-	*numChannels = 4;

-	if (IS_ID("M.K.", id) || IS_ID("M!K!", id) || IS_ID("NSMS", id) || IS_ID("LARD", id) || IS_ID("PATT", id))

-	{

-		modFormat = FORMAT_MK; // ProTracker or compatible

-	}

-	else if (id[1] == 'C' && id[2] == 'H' && id[3] == 'N')

-	{

-		modFormat = FORMAT_FT2; // FT2 or generic multichannel

-		*numChannels = id[0] - '0';

-	}

-	else if (id[2] == 'C' && (id[3] == 'H' || id[3] == 'N'))

-	{

-		modFormat = FORMAT_FT2; // FT2 or generic multichannel

-		*numChannels = ((id[0] - '0') * 10) + (id[1] - '0');

-	}

-	else if (IS_ID("FLT4", id))

-	{

-		modFormat = FORMAT_FLT; // StarTrekker (4-channel modules only)

-	}

-	else if (IS_ID("N.T.", id))

-	{

-		modFormat = FORMAT_NT; // NoiseTracker

-	}

-	else if (IS_ID("M&K!", id) || IS_ID("FEST", id))

-	{

-		modFormat = FORMAT_HMNT; // His Master's NoiseTracker

-	}

-	return modFormat;

-}

-static bool loadMusicMOD(FILE *f, uint32_t fileLength, bool externalThreadFlag)

-{

-	char ID[16];

-	uint8_t bytes[4], modFormat, numChannels;

-	int16_t i, j, k, ai;

-	uint16_t a, b, period;

-	tonTyp *ton;

-	sampleTyp *s;

-	songMOD31HeaderTyp h_MOD31;

-	songMOD15HeaderTyp h_MOD15;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

-	bool veryLateSTKVerFlag = false; // "DFJ SoundTracker III" nad later

-	bool lateSTKVerFlag = false; // "TJC SoundTracker II" and later

-	bool mightBeSTK = false;

-	memset(&songTmp, 0, sizeof (songTmp));

-	memset(&h_MOD31, 0, sizeof (songMOD31HeaderTyp));

-	memset(&h_MOD15, 0, sizeof (songMOD15HeaderTyp));

-	// start loading MOD

-	loadedFormat = FORMAT_MOD;

+	fseek(f, 0, SEEK_END);

+	uint32_t fileLength = (uint32_t)ftell(f);

 	rewind(f);

-	fread(ID, 1, 16, f);

-	fseek(f, 0x1D, SEEK_SET);

-	fread(bytes, 1, 1, f);

-	rewind(f);

-	// since .mod is the last format tested, check if the file is an .it module (Impulse Tracker)

-	if (!memcmp(ID, "IMPM", 4) && bytes[0] == 0)

-	{

-		showMsg(0, "System message", "Error: Impulse Tracker modules are not supported!");

-		goto modLoadError;

-	}

-	// check if the file to load is a WAV, if so reject it

-	if (!memcmp(ID, "RIFF", 4) && !memcmp(&ID[8], "WAVEfmt", 7))

-	{

-		showMsg(0, "System message", "Error: Can't load a .wav file as a module!");

-		goto modLoadError;

-	}

-	if (fileLength < 1596 || fileLength > 20842494) // minimum and maximum possible size for a supported .mod

-	{

-		showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-		goto modLoadError;

-	}

-	if (fread(&h_MOD31, 1, sizeof (h_MOD31), f) != sizeof (h_MOD31))

-	{

-		showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-		goto modLoadError;

-	}

-	modFormat = getModType(&numChannels, h_MOD31.sig);

-	if (modFormat == FORMAT_FLT && numChannels == 8)

-	{

-		showMsg(0, "System message", "8-channel Startrekker modules are not supported!");

-		goto modLoadError;

-	}

-	if (modFormat != FORMAT_UNKNOWN)

-	{

-		if (fileLength < sizeof (h_MOD31))

-		{

-			showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-			goto modLoadError;

-		}

-		songTmp.antChn = numChannels;

-		songTmp.len = h_MOD31.len;

-		songTmp.repS = h_MOD31.repS;

-		memcpy(songTmp.songTab, h_MOD31.songTab, 128);

-		ai = 31;

-	}

-	else

-	{

-		mightBeSTK = true;

-		if (fileLength < sizeof (h_MOD15))

-		{

-			showMsg(0, "System message", "Error: This file is not a module!");

-			goto modLoadError;

-		}

-		rewind(f);

-		if (fread(&h_MOD15, 1, sizeof (h_MOD15), f) != sizeof (h_MOD15))

-		{

-			showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-			goto modLoadError;

-		}

-		songTmp.antChn = numChannels;

-		songTmp.len = h_MOD15.len;

-		songTmp.repS = h_MOD15.repS;

-		memcpy(songTmp.songTab, h_MOD15.songTab, 128);

-		ai = 15;

-	}

-	if (modFormat == FORMAT_MK && songTmp.len == 129)

-		songTmp.len = 127; // fixes a specific copy of beatwave.mod (FIXME: Do we want to keep this digsuting hack?)

-	if (songTmp.antChn == 0 || songTmp.len < 1 || songTmp.len > 128 || (mightBeSTK && songTmp.repS > 220))

-	{

-		showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-		goto modLoadError;

-	}

-	for (a = 0; a < ai; a++)

-	{

-		songMODInstrHeaderTyp *smp = &h_MOD31.instr[a];

-		if (modFormat != FORMAT_HMNT) // most of "His Master's Noisetracker" songs have junk sample names, so let's not load them

-		{

-			// trim off spaces at end of name

-			for (i = 21; i >= 0; i--)

-			{

-				if (smp->name[i] == ' ' || smp->name[i] == 0x1A)

-					smp->name[i] = '\0';

-				else

-					break;

-			}

-			memcpy(songTmp.instrName[1+a], smp->name, 22);

-			songTmp.instrName[1+a][22] = '\0';

-		}

-		/* Only late versions of Ultimate SoundTracker could have samples larger than 9999 bytes.

-		** If found, we know for sure that this is a late STK module.

-		*/

-		if (mightBeSTK && 2*SWAP16(smp->len) > 9999)

-			lateSTKVerFlag = true;

-	}

-	songTmp.speed = 125;

-	if (mightBeSTK)

-	{

-		/* If we're still here at this point and the mightBeSTK flag is set,

-		** then it's most likely a proper The Ultimate SoundTracker (STK/UST) module.

-		*/

-		modFormat = FORMAT_STK;

-		if (songTmp.repS == 0)

-			songTmp.repS = 120;

-		// jjk55.mod by Jesper Kyd has a bogus STK tempo value that should be ignored (hackish!)

-		if (!strcmp("jjk55", h_MOD31.name))

-			songTmp.repS = 120;

-		// The "restart pos" field in STK is the inital tempo (must be converted to BPM first)

-		if (songTmp.repS != 120) // 120 is a special case and means 50Hz (125BPM)

-		{

-			if (songTmp.repS > 220)

-				songTmp.repS = 220;

-			// convert UST tempo to BPM

-			uint16_t ciaPeriod = (240 - songTmp.repS) * 122;

-			double dHz = 709379.0 / ciaPeriod;

-			int32_t BPM = (int32_t)((dHz * 2.5) + 0.5);

-			songTmp.speed = (uint16_t)BPM;

-		}

-		songTmp.repS = 0;

-	}

-	else if (songTmp.repS >= songTmp.len)

-	{

-		songTmp.repS = 0;

-	}

-	// trim off spaces at end of name

-	for (i = 19; i >= 0; i--)

-	{

-		if (h_MOD31.name[i] == ' ' || h_MOD31.name[i] == 0x1A)

-			h_MOD31.name[i] = '\0';

-		else

-			break;

-	}

-	memcpy(songTmp.name, h_MOD31.name, 20);

-	songTmp.name[20] = '\0';

-	// count number of patterns

-	b = 0;

-	for (a = 0; a < 128; a++)

-	{

-		if (songTmp.songTab[a] > b)

-			b = songTmp.songTab[a];

-	}

-	b++;

-	for (a = 0; a < b; a++)

-	{

-		pattTmp[a] = (tonTyp *)calloc((MAX_PATT_LEN * TRACK_WIDTH) + 16, 1);

-		if (pattTmp[a] == NULL)

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			goto modLoadError;

-		}

-		pattLensTmp[a] = 64;

-		for (j = 0; j < 64; j++)

-		{

-			for (k = 0; k < songTmp.antChn; k++)

-			{

-				ton = &pattTmp[a][(j * MAX_VOICES) + k];

-				if (fread(bytes, 1, 4, f) != 4)

-				{

-					showMsg(0, "System message", "Error: This file is either not a module, or is not supported.");

-					goto modLoadError;

-				}

-				// period to note

-				period = ((bytes[0] & 0x0F) << 8) | bytes[1];

-				for (i = 0; i < 8*12; i++)

-				{

-					if (period >= amigaPeriod[i])

-					{

-						ton->ton = (uint8_t)i + 1;

-						break;

-					}

-				}

-				ton->instr = (bytes[0] & 0xF0) | (bytes[2] >> 4);

-				ton->effTyp = bytes[2] & 0x0F;

-				ton->eff = bytes[3];

-				if (modFormat == FORMAT_STK)

-				{

-					if (ton->effTyp == 0xC || ton->effTyp == 0xD || ton->effTyp == 0xE)

-					{

-						// "TJC SoundTracker II" and later

-						lateSTKVerFlag = true;

-					}

-					if (ton->effTyp == 0xF)

-					{

-						// "DFJ SoundTracker III" and later

-						lateSTKVerFlag = true;

-						veryLateSTKVerFlag = true;

-					}

-				}

-				if (ton->effTyp == 0xC)

-				{

-					if (ton->eff > 64)

-						ton->eff = 64;

-				}

-				else if (ton->effTyp == 0x1)

-				{

-					if (ton->eff == 0)

-						ton->effTyp = 0;

-				}

-				else if (ton->effTyp == 0x2)

-				{

-					if (ton->eff == 0)

-						ton->effTyp = 0;

-				}

-				else if (ton->effTyp == 0x5)

-				{

-					if (ton->eff == 0)

-						ton->effTyp = 0x3;

-				}

-				else if (ton->effTyp == 0x6)

-				{

-					if (ton->eff == 0)

-						ton->effTyp = 0x4;

-				}

-				else if (ton->effTyp == 0xA)

-				{

-					if (ton->eff == 0)

-						ton->effTyp = 0;

-				}

-				else if (ton->effTyp == 0xE)

-				{

-					// check if certain E commands are empty

-					if (ton->eff == 0x10 || ton->eff == 0x20 || ton->eff == 0xA0 || ton->eff == 0xB0)

-					{

-						ton->effTyp = 0;

-						ton->eff = 0;

-					}

-				}

-			}

-		}

-		if (tmpPatternEmpty(a))

-		{

-			if (pattTmp[a] != NULL)

-			{

-				free(pattTmp[a]);

-				pattTmp[a] = NULL;

-			}

-		}

-	}

-	// pattern command conversion for non-PT formats

-	if (modFormat == FORMAT_STK || modFormat == FORMAT_NT || modFormat == FORMAT_HMNT || modFormat == FORMAT_FLT)

-	{

-		for (a = 0; a < b; a++)

-		{

-			if (pattTmp[a] == NULL)

-				continue;

-			for (j = 0; j < 64; j++)

-			{

-				for (k = 0; k < songTmp.antChn; k++)

-				{

-					ton = &pattTmp[a][(j * MAX_VOICES) + k];

-					if (modFormat == FORMAT_NT || modFormat == FORMAT_HMNT)

-					{

-						// any Dxx == D00 in NT/HMNT

-						if (ton->effTyp == 0xD)

-							ton->eff = 0;

-						// effect F with param 0x00 does nothing in NT/HMNT

-						if (ton->effTyp == 0xF && ton->eff == 0)

-							ton->effTyp = 0;

-					}

-					else if (modFormat == FORMAT_FLT) // Startrekker

-					{

-						if (ton->effTyp == 0xE) // remove unsupported "assembly macros" command

-						{

-							ton->effTyp = 0;

-							ton->eff = 0;

-						}

-						// Startrekker is always in vblank mode, and limits speed to 0x1F

-						if (ton->effTyp == 0xF && ton->eff > 0x1F)

-							ton->eff = 0x1F;

-					}

-					else if (modFormat == FORMAT_STK)

-					{

-						// convert STK effects to PT effects

-						if (!lateSTKVerFlag)

-						{

-							// old SoundTracker 1.x commands

-							if (ton->effTyp == 1)

-							{

-								// arpeggio

-								ton->effTyp = 0;

-							}

-							else if (ton->effTyp == 2)

-							{

-								// pitch slide

-								if (ton->eff & 0xF0)

-								{

-									// pitch slide down

-									ton->effTyp = 2;

-									ton->eff >>= 4;

-								}

-								else if (ton->eff & 0x0F)

-								{

-									// pitch slide up

-									ton->effTyp = 1;

-								}

-							}

-						}

-						else

-						{

-							// "DFJ SoundTracker II" or later

-							if (ton->effTyp == 0xD)

-							{

-								if (veryLateSTKVerFlag) // "DFJ SoundTracker III" or later

-								{

-									// pattern break w/ no param (param must be cleared to fix some songs)

-									ton->eff = 0;

-								}

-								else

-								{

-									// volume slide

-									ton->effTyp = 0xA;

-								}

-							}

-						}

-						// effect F with param 0x00 does nothing in UST/STK (I think?)

-						if (ton->effTyp == 0xF && ton->eff == 0)

-							ton->effTyp = 0;

-					}

-				}

-			}

-		}

-	}

-	for (a = 0; a < ai; a++)

-	{

-		if (h_MOD31.instr[a].len == 0)

-			continue;

-		if (!allocateTmpInstr(1+a))

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			goto modLoadError;

-		}

-		setNoEnvelope(instrTmp[1+a]);

-		s = &instrTmp[1+a]->samp[0];

-		s->len = 2 * SWAP16(h_MOD31.instr[a].len);

-		if (modFormat != FORMAT_HMNT) // most of "His Master's Noisetracker" songs have junk sample names, so let's not load them

-			memcpy(s->name, songTmp.instrName[1+a], 22);

-		if (modFormat == FORMAT_HMNT) // finetune in "His Master's NoiseTracker" is different

-			h_MOD31.instr[a].fine = (uint8_t)((-h_MOD31.instr[a].fine & 0x1F) >> 1); // one more bit of precision, + inverted

-		if (modFormat != FORMAT_STK)

-			s->fine = 8 * ((2 * ((h_MOD31.instr[a].fine & 0xF) ^ 8)) - 16);

-		s->pan = 128;

-		s->vol = h_MOD31.instr[a].vol;

-		if (s->vol > 64)

-			s->vol = 64;

-		s->repS = 2 * SWAP16(h_MOD31.instr[a].repS);

-		s->repL = 2 * SWAP16(h_MOD31.instr[a].repL);

-		if (s->repL < 2)

-			s->repL = 2;

-		// in The Ultimate SoundTracker, sample loop start is in bytes, not words

-		if (modFormat == FORMAT_STK)

-			s->repS >>= 1;

-		// fix for poorly converted STK (< v2.5) -> PT/NT modules (FIXME: Worth keeping or not?)

-		if (modFormat != FORMAT_STK && s->repL > 2 && s->repS+s->repL > s->len)

-		{

-			if ((s->repS>>1)+s->repL <= s->len)

-				s->repS >>= 1;

-		}

-		// fix overflown loop

-		if (s->repS+s->repL > s->len)

-		{

-			if (s->repS >= s->len)

-			{

-				s->repS = 0;

-				s->repL = 0;

-			}

-			else

-			{

-				s->repL = s->len - s->repS;

-			}

-		}

-		if (s->repS+s->repL > 2)

-			s->typ = 1; // enable loop

-		/* For Ultimate SoundTracker modules, only the loop area of a looped sample is played.

-		** Skip loading of eventual data present before loop start.

-		*/

-		if (modFormat == FORMAT_STK && s->repS > 0 && s->repL < s->len)

-		{

-			s->len -= s->repS;

-			fseek(f, s->repS, SEEK_CUR);

-			s->repS = 0;

-		}

-		s->pek = NULL;

-		s->origPek = (int8_t *)malloc(s->len + LOOP_FIX_LEN);

-		if (s->origPek == NULL)

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			goto modLoadError;

-		}

-		s->pek = s->origPek + SMP_DAT_OFFSET;

-		int32_t bytesRead = (int32_t)fread(s->pek, 1, s->len, f);

-		if (bytesRead < s->len)

-		{

-			int32_t bytesToClear = s->len - bytesRead;

-			memset(&s->pek[bytesRead], 0, bytesToClear);

-		}

-		// clear repL and repS on non-looping samples...

-		if (s->typ == 0)

-		{

-			s->repL = 0;

-			s->repS = 0;

-		}

-		fixSample(s);

-	}

-	fclose(f);

-	songTmp.initialTempo = songTmp.tempo = 6;

-	moduleLoaded = true;

-	return true;

-modLoadError:

-	fclose(f);

-	freeTmpModule();

-	moduleFailedToLoad = true;

-	return false;

-}

-static uint8_t stmTempoToBPM(uint8_t tempo) // ported from original ST2.3 replayer code

-{

-	const uint8_t slowdowns[16] = { 140, 50, 25, 15, 10, 7, 6, 4, 3, 3, 2, 2, 2, 2, 1, 1 };

-	uint16_t hz = 50;

-	hz -= ((slowdowns[tempo >> 4] * (tempo & 15)) >> 4); // can and will underflow

-	const uint32_t bpm = (hz << 1) + (hz >> 1); // BPM = hz * 2.5

-	return (uint8_t)CLAMP(bpm, 32, 255); // result can be slightly off, but close enough...

-}

-static bool loadMusicSTM(FILE *f, uint32_t fileLength, bool externalThreadFlag)

-{

-	uint8_t typ, tempo;

-	int16_t i, j, k, ap, tmp;

-	uint16_t a;

-	tonTyp *ton;

-	sampleTyp *s;

-	songSTMHeaderTyp h_STM;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

+	memset(D, 0, sizeof (D));

+	fread(D, 1, sizeof (D), f);

+	fseek(f, 1080, SEEK_SET); // MOD ID

+	I[0] = I[1] = I[2] = I[3] = 0;

+	fread(I, 1, 4, f);

 	rewind(f);

-	// start loading STM

+	// DIGI Booster (non-Pro)

+	if (!memcmp("DIGI Booster module", &D[0x00], 19+1) && D[0x19] >= 1 && D[0x19] <= 8)

+		return FORMAT_DIGI;

-	if (fread(&h_STM, 1, sizeof (h_STM), f) != sizeof (h_STM))

-		return loadMusicMOD(f, fileLength, externalThreadFlag); // file is not a .stm, try to load as .mod

+	// Scream Tracker 3 S3M (and compatible trackers)

+	if (!memcmp("SCRM", &D[0x2C], 4) && D[0x1D] == 16) // XXX: byte=16 in all cases?

+		return FORMAT_S3M;

-	if (memcmp(h_STM.sig, "!Scream!", 8) && memcmp(h_STM.sig, "BMOD2STM", 8) &&

-		memcmp(h_STM.sig, "WUZAMOD!", 8) && memcmp(h_STM.sig, "SWavePro", 8))

+	// Scream Tracker 2 STM

+	if ((!memcmp("!Scream!", &D[0x14], 8) || !memcmp("BMOD2STM", &D[0x14], 8) ||

+		 !memcmp("WUZAMOD!", &D[0x14], 8) || !memcmp("SWavePro", &D[0x14], 8)) && D[0x1D] == 2) // XXX: byte=2 for "WUZAMOD!"/"SWavePro" ?

-		return loadMusicMOD(f, fileLength, externalThreadFlag); // file is not a .stm, try to load as .mod

+		return FORMAT_STM;

-	loadedFormat = FORMAT_STM;

+	// Generic multi-channel MOD (1..9 channels)

+	if (isdigit(I[0]) && I[0] != '0' && I[1] == 'C' && I[2] == 'H' && I[3] == 'N') // xCHN

+		return FORMAT_MOD;

-	if (h_STM.verMinor == 0 || h_STM.typ != 2)

-	{

-		showMsg(0, "System message", "Error loading .stm: Incompatible module!");

-		goto stmLoadError;

-	}

+	// Generic multi-channel MOD (10..99 channels)

+	if (isdigit(I[0]) && isdigit(I[1]) && I[0] != '0' && I[2] == 'C' && I[3] == 'H') // xxCH

+		return FORMAT_MOD;

-	songTmp.antChn = 4;

-	memcpy(songTmp.songTab, h_STM.songTab, 128);

-	i = 0;

-	while (i < 128 && songTmp.songTab[i] < 99) i++;

-	songTmp.len = i + (i == 0);

-	if (songTmp.len < 255)

-		memset(&songTmp.songTab[songTmp.len], 0, 256 - songTmp.len);

-	// trim off spaces at end of name

-	for (i = 19; i >= 0; i--)

+	// Generic multi-channel MOD (10..99 channels)

+	if (isdigit(I[0]) && isdigit(I[1]) && I[0] != '0' && I[2] == 'C' && I[3] == 'N') // xxCN (same as xxCH)

+		return FORMAT_MOD;

+	// Generic 4-channel MOD (NoiseTracker/ProTracker or compatible)

+	if (!memcmp("M.K.", I, 4) || !memcmp("M!K!", I, 4) || !memcmp("NSMS", I, 4) ||

+		!memcmp("LARD", I, 4) || !memcmp("PATT", I, 4) || !memcmp("PATT", I, 4) ||

+		!memcmp("FLT4", I, 4) || !memcmp("FLT8", I, 4) || !memcmp("N.T.", I, 4) ||

+		!memcmp("M&K!", I, 4) || !memcmp("FEST", I, 4))

-		if (h_STM.name[i] == ' ' || h_STM.name[i] == 0x1A)

-			h_STM.name[i] = '\0';

-		else

-			break;

+		return FORMAT_MOD;

-	memcpy(songTmp.name, h_STM.name, 20);

-	songTmp.name[20] = '\0';

+	/* Check if the file is a .it module (Impulse Tracker, not supported).

+	** Some people may attempt to load .IT files in the FT2 clone, so

+	** reject them here instead of accidentally loading them as .STK

+	*/

+	if (!memcmp("IMPM", D, 4) && D[0x16] == 0)

+		return FORMAT_UNKNOWN;

-	tempo = h_STM.tempo;

-	if (h_STM.verMinor < 21)

-		tempo = ((tempo / 10) << 4) + (tempo % 10);

+	/* Fasttracker II XM and compatible trackers.

+	** Note: This test can falsely be true for STK modules (and non-supported files) where the

+	** first 17 bytes start with "Extended Module: ". This is unlikely to happen.

+	*/

+	if (!memcmp("Extended Module: ", &D[0x00], 17))

+		return FORMAT_XM;

-	if (tempo == 0)

-		tempo = 96;

+	/* Lastly, we assume that the file is either a 15-sample STK or an unsupported file.

+	** Let's assume it's an STK and do some sanity checks. If they fail, we have an

+	** unsupported file.

+	*/

-	songTmp.initialTempo = songTmp.tempo = CLAMP(h_STM.tempo >> 4, 1, 31);

-	songTmp.speed = stmTempoToBPM(tempo);

+	// minimum and maximum (?) possible size for a supported STK

+	if (fileLength < 1624 || fileLength > 984634)

+		return FORMAT_UNKNOWN;

-	if (h_STM.verMinor > 10)

-		songTmp.globVol = MIN(h_STM.vol, 64);

-	ap = h_STM.ap;

-	for (i = 0; i < ap; i++)

-	{

-		pattTmp[i] = (tonTyp *)calloc((MAX_PATT_LEN * TRACK_WIDTH) + 16, 1);

-		if (pattTmp[i] == NULL)

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			goto stmLoadError;

-		}

-		pattLensTmp[i] = 64;

-		if (fread(pattBuff, 64 * 4 * 4, 1, f) != 1)

-		{

-			showMsg(0, "System message", "General I/O error during loading!");

-			goto stmLoadError;

-		}

-		a = 0;

-		for (j = 0; j < 64; j++)

-		{

-			for (k = 0; k < 4; k++)

-			{

-				ton = &pattTmp[i][(j * MAX_VOICES) + k];

-				if (pattBuff[a] == 254)

-				{

-					ton->ton = 97;

-				}

-				else if (pattBuff[a] < 96)

-				{

-					ton->ton = (12 * (pattBuff[a] >> 4)) + (25 + (pattBuff[a] & 0x0F));

-					if (ton->ton > 96)

-						ton->ton = 0;

-				}

-				else

-				{

-					ton->ton = 0;

-				}

-				ton->instr = pattBuff[a + 1] >> 3;

-				typ = (pattBuff[a + 1] & 7) + ((pattBuff[a + 2] & 0xF0) >> 1);

-				if (typ <= 64)

-					ton->vol = typ + 0x10;

-				ton->eff = pattBuff[a + 3];

-				tmp = pattBuff[a + 2] & 0x0F;

-				if (tmp == 1)

-				{

-					ton->effTyp = 15;

-					if (h_STM.verMinor < 21)

-						ton->eff = ((ton->eff / 10) << 4) + (ton->eff % 10);

-					ton->eff >>= 4;

-				}

-				else if (tmp == 3)

-				{

-					ton->effTyp = 13;

-					ton->eff = 0;

-				}

-				else if (tmp == 2 || (tmp >= 4 && tmp <= 12))

-				{

-					ton->effTyp = stmEff[tmp];

-					if (ton->effTyp == 0xA)

-					{

-						if (ton->eff & 0x0F)

-							ton->eff &= 0x0F;

-						else

-							ton->eff &= 0xF0;

-					}

-				}

-				else

-				{

-					ton->eff = 0;

-				}

-				if (ton->effTyp > 35)

-				{

-					ton->effTyp = 0;

-					ton->eff = 0;

-				}

-				a += 4;

-			}

-		}

-		if (tmpPatternEmpty(i))

-		{

-			if (pattTmp[i] != NULL)

-			{

-				free(pattTmp[i]);

-				pattTmp[i] = NULL;

-			}

-		}

-	}

-	for (i = 0; i < 31; i++)

-	{

-		// trim off spaces at end of name

-		for (j = 11; j >= 0; j--)

-		{

-			if (h_STM.instr[i].name[j] == ' ' || h_STM.instr[i].name[j] == 0x1A)

-				h_STM.instr[i].name[j] = '\0';

-			else

-				break;

-		}

-		memset(&songTmp.instrName[1+i], 0, sizeof (songTmp.instrName[1+i]));

-		memcpy(&songTmp.instrName[1+i], h_STM.instr[i].name, 12);

-		if (h_STM.instr[i].len != 0 && h_STM.instr[i].reserved1 != 0)

-		{

-			allocateTmpInstr(1 + i);

-			setNoEnvelope(instrTmp[i]);

-			s = &instrTmp[1+i]->samp[0];

-			s->pek = NULL;

-			s->origPek = (int8_t *)malloc(h_STM.instr[i].len + LOOP_FIX_LEN);

-			if (s->origPek == NULL)

-			{

-				showMsg(0, "System message", "Not enough memory!");

-				goto stmLoadError;

-			}

-			s->pek = s->origPek + SMP_DAT_OFFSET;

-			s->len = h_STM.instr[i].len;

-			tuneSample(s, h_STM.instr[i].rate, false);

-			s->vol = h_STM.instr[i].vol;

-			s->repS = h_STM.instr[i].repS;

-			s->repL = h_STM.instr[i].repE - h_STM.instr[i].repS;

-			s->pan = 128;

-			if (s->repS < s->len && h_STM.instr[i].repE > s->repS && h_STM.instr[i].repE != 0xFFFF)

-			{

-				if (s->repS+s->repL > s->len)

-					s->repL = s->len - s->repS;

-				s->typ = 1; // enable loop

-			}

-			else

-			{

-				s->repS = 0;

-				s->repL = 0;

-				s->typ = 0;

-			}

-			if (s->vol > 64)

-				s->vol = 64;

-			if (fread(s->pek, s->len, 1, f) != 1)

-			{

-				showMsg(0, "System message", "General I/O error during loading! Possibly corrupt module?");

-				goto stmLoadError;

-			}

-			fixSample(s);

-		}

-	}

-	fclose(f);

-	moduleLoaded = true;

-	return true;

-stmLoadError:

-	fclose(f);

-	freeTmpModule();

-	moduleFailedToLoad = true;

-	return false;

-}

-static int8_t countS3MChannels(uint16_t antPtn)

-{

-	int32_t channels = 0;

-	for (int32_t i = 0; i < antPtn; i++)

-	{

-		if (pattTmp[i] == NULL)

-			continue;

-		tonTyp *ton = pattTmp[i];

-		for (int32_t j = 0; j < 64; j++)

-		{

-			for (int32_t k = 0; k < MAX_VOICES; k++, ton++)

-			{

-				if (ton->eff == 0 && ton->effTyp == 0 && ton->instr == 0 && ton->ton == 0 && ton->vol == 0)

-					continue;

-				if (k > channels)

-					channels = k;

-			}

-		}

-	}

-	channels++;

-	return (int8_t)channels;

-}

-static bool loadMusicS3M(FILE *f, uint32_t dataLength, bool externalThreadFlag)

-{

-	int8_t *tmpSmp;

-	uint8_t ha[2048];

-	uint8_t alastnfo[32], alastefx[32], alastvibnfo[32], s3mLastGInstr[32];

-	uint8_t typ;

-	int16_t ai, ap, ver, ii, kk, tmp;

-	uint16_t ptnOfs[256];

-	int32_t i, j, k, len;

-	tonTyp ton;

-	sampleTyp *s;

-	songS3MHeaderTyp h_S3M;

-	songS3MinstrHeaderTyp h_S3MInstr;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

+	// test STK numOrders+BPM for illegal values

+	fseek(f, 470, SEEK_SET);

+	D[0] = D[1] = 0;

+	fread(D, 1, 2, f);

 	rewind(f);

-	// start loading S3M

+	if (D[0] >= 0 && D[0] <= 128 && D[1] <= 220)

+		return FORMAT_POSSIBLY_STK;

-	if (fread(&h_S3M, 1, sizeof (h_S3M), f) != sizeof (h_S3M))

-		return loadMusicSTM(f, dataLength, externalThreadFlag); // not a .s3m, try loading as .stm

-	if (memcmp(h_S3M.id, "SCRM", 4))

-		return loadMusicSTM(f, dataLength, externalThreadFlag); // not a .s3m, try loading as .stm

-	loadedFormat = FORMAT_S3M;

-	if (h_S3M.antInstr > MAX_INST || h_S3M.songTabLen > 256 || h_S3M.antPatt > 256 ||

-		h_S3M.typ != 16 || h_S3M.ver < 1 || h_S3M.ver > 2)

-	{

-		showMsg(0, "System message", "Error loading .s3m: Incompatible module!");

-		goto s3mLoadError;

-	}

-	memset(songTmp.songTab, 255, sizeof (songTmp.songTab));

-	if (fread(songTmp.songTab, h_S3M.songTabLen, 1, f) != 1)

-	{

-		showMsg(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto s3mLoadError;

-	}

-	songTmp.len = h_S3M.songTabLen;

-	// remove pattern separators (254)

-	k = 0;

-	j = 0;

-	for (i = 0; i < songTmp.len; i++)

-	{

-		if (songTmp.songTab[i] != 254)

-			songTmp.songTab[j++] = songTmp.songTab[i];

-		else

-			k++;

-	}

-	if (k <= songTmp.len)

-		songTmp.len -= (uint16_t)k;

-	else

-		songTmp.len = 1;

-	for (i = 1; i < songTmp.len; i++)

-	{

-		if (songTmp.songTab[i] == 255)

-		{

-			songTmp.len = (uint16_t)i;

-			break;

-		}

-	}

-	// clear unused song table entries

-	if (songTmp.len < 255)

-		memset(&songTmp.songTab[songTmp.len], 0, 255-songTmp.len);

-	songTmp.speed = h_S3M.defTempo;

-	if (songTmp.speed < 32)

-		songTmp.speed = 32;

-	songTmp.tempo = h_S3M.defSpeed;

-	if (songTmp.tempo == 0)

-		songTmp.tempo = 6;

-	if (songTmp.tempo > 31)

-		songTmp.tempo = 31;

-	songTmp.initialTempo = songTmp.tempo;

-	// trim off spaces at end of name

-	for (i = 19; i >= 0; i--)

-	{

-		if (h_S3M.name[i] == ' ' || h_S3M.name[i] == 0x1A)

-			h_S3M.name[i] = '\0';

-		else

-			break;

-	}

-	memcpy(songTmp.name, h_S3M.name, 20);

-	songTmp.name[20] = '\0';

-	ap = h_S3M.antPatt;

-	ai = h_S3M.antInstr;

-	ver = h_S3M.ver;

-	k = 31;

-	while (k >= 0 && h_S3M.chanType[k] >= 16) k--;

-	songTmp.antChn = (k + 2) & 254;

-	if (fread(ha, ai + ai, 1, f) != 1)

-	{

-		showMsg(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto s3mLoadError;

-	}

-	if (fread(ptnOfs, ap + ap, 1, f) != 1)

-	{

-		showMsg(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto s3mLoadError;

-	}

-	// *** PATTERNS ***

-	k = 0;

-	for (i = 0; i < ap; i++)

-	{

-		if (ptnOfs[i]  == 0)

-			continue; // empty pattern

-		memset(alastnfo, 0, sizeof (alastnfo));

-		memset(alastefx, 0, sizeof (alastefx));

-		memset(alastvibnfo, 0, sizeof (alastvibnfo));

-		memset(s3mLastGInstr, 0, sizeof (s3mLastGInstr));

-		fseek(f, ptnOfs[i] << 4, SEEK_SET);

-		if (feof(f))

-			continue;

-		if (fread(&j, 2, 1, f) != 1)

-		{

-			showMsg(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto s3mLoadError;

-		}

-		if (j > 0 && j <= 12288)

-		{

-			pattTmp[i] = (tonTyp *)calloc((MAX_PATT_LEN * TRACK_WIDTH) + 16, 1);

-			if (pattTmp[i] == NULL)

-			{

-				showMsg(0, "System message", "Not enough memory!");

-				goto s3mLoadError;

-			}

-			pattLensTmp[i] = 64;

-			fread(pattBuff, j, 1, f);

-			k = 0;

-			kk = 0;

-			while (k < j && kk < 64)

-			{

-				typ = pattBuff[k++];

-				if (typ == 0)

-				{

-					kk++;

-				}

-				else

-				{

-					ii = typ & 31;

-					memset(&ton, 0, sizeof (ton));

-					// note and sample

-					if (typ & 32)

-					{

-						ton.ton = pattBuff[k++];

-						ton.instr = pattBuff[k++];

-						if (ton.instr > MAX_INST)

-							ton.instr = 0;

-						     if (ton.ton == 254) ton.ton = 97;

-						else if (ton.ton == 255) ton.ton = 0;

-						else

-						{

-							ton.ton = 1 + (ton.ton & 0xF) + (ton.ton >> 4) * 12;

-							if (ton.ton > 96)

-								ton.ton = 0;

-						}

-					}

-					// volume column

-					if (typ & 64)

-					{

-						ton.vol = pattBuff[k++];

-						if (ton.vol <= 64)

-							ton.vol += 0x10;

-						else

-							ton.vol = 0;

-					}

-					// effect

-					if (typ & 128)

-					{

-						ton.effTyp = pattBuff[k++];

-						ton.eff = pattBuff[k++];

-						if (ton.eff > 0)

-						{

-							alastnfo[ii] = ton.eff;

-							if (ton.effTyp == 8 || ton.effTyp == 21)

-								alastvibnfo[ii] = ton.eff; // H/U

-						}

-						// in ST3, a lot of effects directly share the same memory!

-						if (ton.eff == 0 && ton.effTyp != 7) // G

-						{

-							uint8_t efx = ton.effTyp;

-							if (efx == 8 || efx == 21) // H/U

-								ton.eff = alastvibnfo[ii];

-							else if ((efx >= 4 && efx <= 12) || (efx >= 17 && efx <= 19)) // D/E/F/I/J/K/L/Q/R/S

-								ton.eff = alastnfo[ii];

-							/* If effect data is zero and effect type was the same as last one, clear out

-							** data if it's not J or S (those have no memory in the equivalent XM effects).

-							** Also goes for extra fine pitch slides and fine volume slides,

-							** since they get converted to other effects.

-							*/

-							if (efx == alastefx[ii] && ton.effTyp != 10 && ton.effTyp != 19) // J/S

-							{

-								uint8_t nfo = ton.eff;

-								bool extraFinePitchSlides = (efx == 5 || efx == 6) && ((nfo & 0xF0) == 0xE0);

-								bool fineVolSlides = (efx == 4 || efx == 11) &&

-								     ((nfo > 0xF0) || (((nfo & 0xF) == 0xF) && ((nfo & 0xF0) > 0)));

-								if (!extraFinePitchSlides && !fineVolSlides)

-									ton.eff = 0;

-							}

-						}

-						if (ton.effTyp > 0)

-							alastefx[ii] = ton.effTyp;

-						switch (ton.effTyp)

-						{

-							case 1: // A

-							{

-								ton.effTyp = 0xF;

-								if (ton.eff == 0 || ton.eff > 0x1F)

-								{

-									ton.effTyp = 0;

-									ton.eff = 0;

-								}

-							}

-							break;

-							case 2: ton.effTyp = 0xB; break; // B

-							case 3: ton.effTyp = 0xD; break; // C

-							case 4: // D

-							{

-								if (ton.eff > 0xF0) // fine slide up

-								{

-									ton.effTyp = 0xE;

-									ton.eff = 0xB0 | (ton.eff & 0xF);

-								}

-								else if ((ton.eff & 0x0F) == 0x0F && (ton.eff & 0xF0) > 0) // fine slide down

-								{

-									ton.effTyp = 0xE;

-									ton.eff = 0xA0 | (ton.eff >> 4);

-								}

-								else

-								{

-									ton.effTyp = 0xA;

-									if (ton.eff & 0x0F) // on D/K, last nybble has first priority in ST3

-										ton.eff &= 0x0F;

-								}

-							}

-							break;

-							case 5: // E

-							case 6: // F

-							{

-								if ((ton.eff & 0xF0) >= 0xE0)

-								{

-									// convert to fine slide

-									if ((ton.eff & 0xF0) == 0xE0)

-										tmp = 0x21;

-									else

-										tmp = 0xE;

-									ton.eff &= 0x0F;

-									if (ton.effTyp == 0x05)

-										ton.eff |= 0x20;

-									else

-										ton.eff |= 0x10;

-									ton.effTyp = (uint8_t)tmp;

-									if (ton.effTyp == 0x21 && ton.eff == 0)

-									{

-										ton.effTyp = 0;

-									}

-								}

-								else

-								{

-									// convert to normal 1xx/2xx slide

-									ton.effTyp = 7 - ton.effTyp;

-								}

-							}

-							break;

-							case 7: // G

-							{

-								ton.effTyp = 0x03;

-								// fix illegal slides (to new instruments)

-								if (ton.instr != 0 && ton.instr != s3mLastGInstr[ii])

-									ton.instr = s3mLastGInstr[ii];

-							}

-							break;

-							case 11: // K

-							{

-								if (ton.eff > 0xF0) // fine slide up

-								{

-									ton.effTyp = 0xE;

-									ton.eff = 0xB0 | (ton.eff & 0xF);

-									// if volume column is unoccupied, set to vibrato

-									if (ton.vol == 0)

-										ton.vol = 0xB0;

-								}

-								else if ((ton.eff & 0x0F) == 0x0F && (ton.eff & 0xF0) > 0) // fine slide down

-								{

-									ton.effTyp = 0xE;

-									ton.eff = 0xA0 | (ton.eff >> 4);

-									// if volume column is unoccupied, set to vibrato

-									if (ton.vol == 0)

-										ton.vol = 0xB0;

-								}

-								else

-								{

-									ton.effTyp = 0x6;

-									if (ton.eff & 0x0F) // on D/K, last nybble has first priority in ST3

-										ton.eff &= 0x0F;

-								}

-							}

-							break;

-							case 8: ton.effTyp = 0x04; break; // H

-							case 9: ton.effTyp = 0x1D; break; // I

-							case 10: ton.effTyp = 0x00; break; // J

-							case 12: ton.effTyp = 0x05; break; // L

-							case 15: ton.effTyp = 0x09; break; // O

-							case 17: ton.effTyp = 0x1B; break; // Q

-							case 18: ton.effTyp = 0x07; break; // R

-							case 19: // S

-							{

-								ton.effTyp = 0xE;

-								tmp = ton.eff >> 4;

-								ton.eff &= 0x0F;

-								     if (tmp == 0x1) ton.eff |= 0x30;

-								else if (tmp == 0x2) ton.eff |= 0x50;

-								else if (tmp == 0x3) ton.eff |= 0x40;

-								else if (tmp == 0x4) ton.eff |= 0x70;

-								// ignore S8x becuase it's not compatible with FT2 panning

-								else if (tmp == 0xB) ton.eff |= 0x60;

-								else if (tmp == 0xC) // Note Cut

-								{

-									ton.eff |= 0xC0;

-									if (ton.eff == 0xC0)

-									{

-										// EC0 does nothing in ST3 but cuts voice in FT2, remove effect

-										ton.effTyp = 0;

-										ton.eff = 0;

-									}

-								}

-								else if (tmp == 0xD) // Note Delay

-								{

-									ton.eff |= 0xD0;

-									if (ton.ton == 0 || ton.ton == 97)

-									{

-										// EDx without a note does nothing in ST3 but retrigs in FT2, remove effect

-										ton.effTyp = 0;

-										ton.eff = 0;

-									}

-									else if (ton.eff == 0xD0)

-									{

-										// ED0 prevents note/smp/vol from updating in ST3, remove everything

-										ton.ton = 0;

-										ton.instr = 0;

-										ton.vol = 0;

-										ton.effTyp = 0;

-										ton.eff = 0;

-									}

-								}

-								else if (tmp == 0xE) ton.eff |= 0xE0;

-								else if (tmp == 0xF) ton.eff |= 0xF0;

-								else

-								{

-									ton.effTyp = 0;

-									ton.eff = 0;

-								}

-							}

-							break;

-							case 20: // T

-							{

-								ton.effTyp = 0x0F;

-								if (ton.eff < 0x21) // Txx with a value lower than 33 (0x21) does nothing in ST3, remove effect

-								{

-									ton.effTyp = 0;

-									ton.eff = 0;

-								}

-							}

-							break;

-							case 22: // V

-							{

-								ton.effTyp = 0x10;

-								if (ton.eff > 0x40)

-								{

-									// Vxx > 0x40 does nothing in ST3

-									ton.effTyp = 0;

-									ton.eff = 0;

-								}

-							}

-							break;

-							default:

-							{

-								ton.effTyp = 0;

-								ton.eff = 0;

-							}

-							break;

-						}

-					}

-					if (ton.instr != 0 && ton.effTyp != 0x3)

-						s3mLastGInstr[ii] = ton.instr;

-					if (ton.effTyp > 35)

-					{

-						ton.effTyp = 0;

-						ton.eff = 0;

-					}

-					pattTmp[i][(kk * MAX_VOICES) + ii] = ton;

-				}

-			}

-			if (tmpPatternEmpty((uint16_t)i))

-			{

-				if (pattTmp[i] != NULL)

-				{

-					free(pattTmp[i]);

-					pattTmp[i] = NULL;

-				}

-			}

-		}

-	}

-	// *** SAMPLES ***

-	bool adlibInsWarn = false;

-	memcpy(ptnOfs, ha, 512);

-	for (i = 0; i < ai; i++)

-	{

-		fseek(f, ptnOfs[i] << 4, SEEK_SET);

-		if (fread(&h_S3MInstr, 1, sizeof (h_S3MInstr), f) != sizeof (h_S3MInstr))

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			goto s3mLoadError;

-		}

-		// trim off spaces at end of name

-		for (j = 21; j >= 0; j--)

-		{

-			if (h_S3MInstr.name[j] == ' ' || h_S3MInstr.name[j] == 0x1A)

-				h_S3MInstr.name[j] = '\0';

-			else

-				break;

-		}

-		memcpy(songTmp.instrName[1+i], h_S3MInstr.name, 22);

-		songTmp.instrName[1+i][22] = '\0';

-		if (h_S3MInstr.typ == 2)

-		{

-			adlibInsWarn = true;

-		}

-		else if (h_S3MInstr.typ == 1)

-		{

-			if ((h_S3MInstr.flags & (255-1-2-4)) != 0 || h_S3MInstr.pack != 0)

-			{

-				showMsg(0, "System message", "Error loading .s3m: Incompatible module!");

-				goto s3mLoadError;

-			}

-			else if (h_S3MInstr.memSeg > 0 && h_S3MInstr.len > 0)

-			{

-				if (!allocateTmpInstr((int16_t)(1 + i)))

-				{

-					showMsg(0, "System message", "Not enough memory!");

-					goto s3mLoadError;

-				}

-				setNoEnvelope(instrTmp[1 + i]);

-				s = &instrTmp[1+i]->samp[0];

-				len = h_S3MInstr.len;

-				bool hasLoop = h_S3MInstr.flags & 1;

-				bool stereoSample = (h_S3MInstr.flags >> 1) & 1;

-				bool is16Bit = (h_S3MInstr.flags >> 2) & 1;

-				if (is16Bit) // 16-bit

-					len <<= 1;

-				if (stereoSample) // stereo

-					len <<= 1;

-				tmpSmp = (int8_t *)malloc(len + LOOP_FIX_LEN);

-				if (tmpSmp == NULL)

-				{

-					showMsg(0, "System message", "Not enough memory!");

-					goto s3mLoadError;

-				}

-				int8_t *newPtr = tmpSmp + SMP_DAT_OFFSET;

-				memcpy(s->name, h_S3MInstr.name, 21);

-				if (h_S3MInstr.c2Spd > 65535) // ST3 (and OpenMPT) does this

-					h_S3MInstr.c2Spd = 65535;

-				tuneSample(s, h_S3MInstr.c2Spd, false);

-				s->len = h_S3MInstr.len;

-				s->vol = h_S3MInstr.vol;

-				s->repS = h_S3MInstr.repS;

-				s->repL = h_S3MInstr.repE - h_S3MInstr.repS;

-				// non-FT2: fixes "miracle man.s3m"

-				if ((h_S3MInstr.memSeg<<4)+s->len > (int32_t)dataLength)

-					s->len = dataLength - (h_S3MInstr.memSeg << 4);

-				if (s->repL <= 2 || s->repS+s->repL > s->len)

-				{

-					s->repS = 0;

-					s->repL = 0;

-					hasLoop = false;

-				}

-				if (s->repL == 0)

-					hasLoop = false;

-				s->typ = hasLoop + (is16Bit << 4);

-				if (s->vol > 64)

-					s->vol = 64;

-				s->pan = 128;

-				fseek(f, h_S3MInstr.memSeg << 4, SEEK_SET);

-				// non-FT2: fixes "miracle man.s3m"

-				if ((h_S3MInstr.memSeg<<4)+len > (int32_t)dataLength)

-					len = dataLength - (h_S3MInstr.memSeg << 4);

-				if (ver == 1)

-				{

-					fseek(f, len, SEEK_CUR); // sample not supported

-				}

-				else

-				{

-					if (fread(newPtr, len, 1, f) != 1)

-					{

-						free(tmpSmp);

-						showMsg(0, "System message", "General I/O error during loading! Is the file in use?");

-						goto s3mLoadError;

-					}

-					if (is16Bit)

-					{

-						conv16BitSample(newPtr, len, stereoSample);

-						s->origPek = tmpSmp;

-						s->pek = s->origPek + SMP_DAT_OFFSET;

-						s->len <<= 1;

-						s->repS <<= 1;

-						s->repL <<= 1;

-					}

-					else

-					{

-						conv8BitSample(newPtr, len, stereoSample);

-						s->origPek = tmpSmp;

-						s->pek = s->origPek + SMP_DAT_OFFSET;

-					}

-					// if stereo sample: reduce memory footprint after sample was downmixed to mono

-					if (stereoSample)

-					{

-						newPtr = (int8_t *)realloc(s->origPek, s->len + LOOP_FIX_LEN);

-						if (newPtr != NULL)

-						{

-							s->origPek = newPtr;

-							s->pek = s->origPek + SMP_DAT_OFFSET;

-						}

-					}

-					fixSample(s);

-				}

-			}

-		}

-	}

-	fclose(f);

-	songTmp.antChn = countS3MChannels(ap);

-	if (adlibInsWarn)

-		showMsg(0, "System message", "Warning: The module contains unsupported AdLib instruments!");

-	if (!(config.dontShowAgainFlags & DONT_SHOW_S3M_LOAD_WARNING_FLAG))

-		showMsg(6, "System message", "Warning: S3M channel panning is ignored because it's not compatible with FT2.");

-	moduleLoaded = true;

-	return true;

-s3mLoadError:

-	fclose(f);

-	freeTmpModule();

-	moduleFailedToLoad = true;

-	return false;

+	return FORMAT_UNKNOWN;

-bool doLoadMusic(bool externalThreadFlag)

+static bool doLoadMusic(bool externalThreadFlag)

-	char tmpText[128];

-	int16_t k;

-	uint16_t i;

-	uint32_t filelength;

-	songHeaderTyp h;

-	FILE *f;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

+	// setup message box functions

+	loaderMsgBox = externalThreadFlag ? myLoaderMsgBoxThreadSafe : myLoaderMsgBox;

+	loaderSysReq = externalThreadFlag ? okBoxThreadSafe : okBox;

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

-	linearFreqTable = false;

 	if (editor.tmpFilenameU == NULL)

-		showMsg(0, "System message", "Generic memory fault during loading!");

-		moduleFailedToLoad = true;

-		return false;

+		loaderMsgBox("Generic memory fault during loading!");

+		goto loadError;

-	f = UNICHAR_FOPEN(editor.tmpFilenameU, "rb");

+	FILE *f = UNICHAR_FOPEN(editor.tmpFilenameU, "rb");

 	if (f == NULL)

-		showMsg(0, "System message", "General I/O error during loading! Is the file in use? Does it exist?");

-		moduleFailedToLoad = true;

-		return false;

+		loaderMsgBox("General I/O error during loading! Is the file in use? Does it exist?");

+		goto loadError;

+	int8_t format = detectModule(f);

 	fseek(f, 0, SEEK_END);

-	filelength = ftell(f);

-	rewind(f);

+	uint32_t filesize = ftell(f);

-	// start loading

-	if (fread(&h, 1, sizeof (h), f) != sizeof (h))

-		return loadMusicS3M(f, filelength, externalThreadFlag); // not a .xm file, try to load as .s3m

-	if (memcmp(h.sig, "Extended Module: ", 17))

-		return loadMusicS3M(f, filelength, externalThreadFlag); // not a .xm file, try to load as .s3m

-	loadedFormat = FORMAT_XM;

-	if (h.ver < 0x0102 || h.ver > 0x0104)

+	rewind(f);

+	switch (format)

-		fclose(f);

+		case FORMAT_XM: moduleLoaded = loadXM(f, filesize); break;

+		case FORMAT_S3M: moduleLoaded = loadS3M(f, filesize); break;

+		case FORMAT_STM: moduleLoaded = loadSTM(f, filesize); break;

+		case FORMAT_MOD: moduleLoaded = loadMOD(f, filesize); break;

+		case FORMAT_POSSIBLY_STK: moduleLoaded = loadSTK(f, filesize); break;

+		case FORMAT_DIGI: moduleLoaded = loadDIGI(f, filesize); break;

-		const int32_t major = (h.ver >> 8) & 0x0F;

-		const int32_t minor = h.ver & 0xFF;

-		sprintf(tmpText, "Error loading XM: Unsupported file version (v%01X.%02X).", major, minor);

-		showMsg(0, "System message", tmpText);

-		moduleFailedToLoad = true;

-		return false;

+		default:

+			loaderMsgBox("This file is not a supported module!");

+		break;

-	if (h.len > MAX_ORDERS)

-	{

-		showMsg(0, "System message", "Error loading XM: The song has more than 256 orders!");

-		goto xmLoadError;

-	}

-	if (h.antPtn > MAX_PATTERNS)

-	{

-		showMsg(0, "System message", "Error loading XM: The song has more than 256 patterns!");

-		goto xmLoadError;

-	}

-	if (h.antChn == 0)

-	{

-		showMsg(0, "System message", "Error loading XM: This file is corrupt.");

-		goto xmLoadError;

-	}

-	if (h.antInstrs > 256) // if >128 instruments, we fake-load up to 128 extra instruments and discard them

-	{

-		showMsg(0, "System message", "Error loading XM: This file is corrupt.");

-		goto xmLoadError;

-	}

-	fseek(f, 60 + h.headerSize, SEEK_SET);

-	if (filelength != 336 && feof(f)) // 336 in length at this point = empty XM

-	{

-		showMsg(0, "System message", "Error loading XM: The module is empty!");

-		goto xmLoadError;

-	}

-	// trim off spaces at end of name

-	for (k = 19; k >= 0; k--)

-	{

-		if (h.name[k] == ' ' || h.name[k] == 0x1A)

-			h.name[k] = '\0';

-		else

-			break;

-	}

-	memcpy(songTmp.name, h.name, 20);

-	songTmp.name[20] = '\0';

-	songTmp.len = h.len;

-	songTmp.repS = h.repS;

-	songTmp.antChn = (uint8_t)h.antChn;

-	songTmp.speed = h.defSpeed;

-	songTmp.tempo = h.defTempo;

-	songTmp.ver = h.ver;

-	linearFreqTable = h.flags & 1;

-	songTmp.speed = CLAMP(songTmp.speed, 1, 999);

-	songTmp.tempo = CLAMP(songTmp.tempo, 1, 99);

-	songTmp.initialTempo = songTmp.tempo;

-	if (songTmp.globVol > 64)

-		songTmp.globVol = 64;

-	if (songTmp.len == 0)

-		songTmp.len = 1; // songTmp.songTab is already empty

-	else

-		memcpy(songTmp.songTab, h.songTab, songTmp.len);

-	// some strange XMs have the order list padded with 0xFF, remove them!

-	for (int16_t j = 255; j >= 0; j--)

-	{

-		if (songTmp.songTab[j] != 0xFF)

-			break;

-		if (songTmp.len > j)

-			songTmp.len = j;

-	}

-	// even though XM supports 256 orders, FT2 supports only 255...

-	if (songTmp.len > 0xFF)

-		songTmp.len = 0xFF;

-	if (songTmp.ver < 0x0104)

-	{

-		// XM v1.02 and XM v1.03

-		for (i = 1; i <= h.antInstrs; i++)

-		{

-			if (!loadInstrHeader(f, i, externalThreadFlag))

-				goto xmLoadError;

-		}

-		if (!loadPatterns(f, h.antPtn, externalThreadFlag))

-			goto xmLoadError;

-		for (i = 1; i <= h.antInstrs; i++)

-		{

-			if (!loadInstrSample(f, i, externalThreadFlag))

-				goto xmLoadError;

-		}

-	}

-	else

-	{

-		// XM v1.04 (latest version)

-		if (!loadPatterns(f, h.antPtn, externalThreadFlag))

-			goto xmLoadError;

-		for (i = 1; i <= h.antInstrs; i++)

-		{

-			if (!loadInstrHeader(f, i, externalThreadFlag))

-				goto xmLoadError;

-			if (!loadInstrSample(f, i, externalThreadFlag))

-				goto xmLoadError;

-		}

-	}

-	// if we temporarily loaded more than 128 instruments, clear the extra allocated memory

-	if (h.antInstrs > MAX_INST)

-	{

-		for (i = MAX_INST+1; i <= h.antInstrs; i++)

-		{

-			if (instrTmp[i] != NULL)

-			{

-				free(instrTmp[i]);

-				instrTmp[i] = NULL;

-			}

-		}

-	}

 	fclose(f);

-	/* We support loading XMs with up to 32 samples per instrument (ModPlug/OpenMPT),

-	** but only the first 16 will be loaded. Now make sure we set the number of samples

-	** back to max 16 in the headers before loading is done.

-	*/

-	bool instrHasMoreThan16Samples = false;

-	for (i = 1; i <= MAX_INST; i++)

-	{

-		if (instrTmp[i] != NULL && instrTmp[i]->antSamp > MAX_SMP_PER_INST)

-		{

-			instrHasMoreThan16Samples = true;

-			instrTmp[i]->antSamp = MAX_SMP_PER_INST;

-		}

-	}

+	if (!moduleLoaded)

+		goto loadError;

-	if (songTmp.antChn > MAX_VOICES)

-	{

-		songTmp.antChn = MAX_VOICES;

-		showMsg(0, "System message", "Warning: This XM contains >32 channels. The extra channels will be discarded!");

-	}

-	if (h.antInstrs > MAX_INST)

-		showMsg(0, "System message", "Warning: This XM contains >128 instruments. The extra instruments will be discarded!");

-	if (instrHasMoreThan16Samples)

-		showMsg(0, "System message", "Warning: This XM contains instrument(s) with >16 samples. The extra samples will be discarded!");

 	moduleLoaded = true;

 	return true;

-xmLoadError:

-	fclose(f);

+loadError:

 	freeTmpModule();

 	moduleFailedToLoad = true;

 	return false;

+static void clearTmpModule(void)

+{

+	memset(pattTmp, 0, sizeof (pattTmp));

+	memset(instrTmp, 0, sizeof (instrTmp));

+	memset(&songTmp, 0, sizeof (songTmp));

+	for (uint32_t i = 0; i < MAX_PATTERNS; i++)

+		pattLensTmp[i] = 64;

+}

 static int32_t SDLCALL loadMusicThread(void *ptr)

 	return doLoadMusic(true);

 	(void)ptr;

 void loadMusic(UNICHAR *filenameU)

-	if (musicIsLoading)

+	if (musicIsLoading || filenameU == NULL)

 		return;

 	mouseAnimOn();

@@ -1882,20 +230,10 @@

 	musicIsLoading = true;

 	moduleLoaded = false;

 	moduleFailedToLoad = false;

-	loadedFormat = FORMAT_NONE;

+	clearTmpModule(); // clear stuff from last loading session (very important)

 	UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

-	// clear deprecated pointers from possible last loading session (super important)

-	memset(pattTmp,  0, sizeof (pattTmp));

-	memset(instrTmp, 0, sizeof (instrTmp));

-	// prevent stuck instrument names from previous module

-	memset(&songTmp, 0, sizeof (songTmp));

-	for (uint32_t i = 0; i < MAX_PATTERNS; i++)

-		pattLensTmp[i] = 64;

 	thread = SDL_CreateThread(loadMusicThread, NULL, NULL);

 	if (thread == NULL)

@@ -1910,19 +248,10 @@

 bool loadMusicUnthreaded(UNICHAR *filenameU, bool autoPlay)

-	if (filenameU == NULL || editor.tmpFilenameU == NULL)

+	if (filenameU == NULL)

 		return false;

-	// clear deprecated pointers from possible last loading session (super important)

-	memset(pattTmp, 0, sizeof (pattTmp));

-	memset(instrTmp, 0, sizeof (instrTmp));

-	// prevent stuck instrument names from previous module

-	memset(&songTmp, 0, sizeof (songTmp));

-	for (uint32_t i = 0; i < MAX_PATTERNS; i++)

-		pattLensTmp[i] = 64;

+	clearTmpModule(); // clear stuff from last loading session (very important)

 	UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

 	editor.loadMusicEvent = EVENT_NONE;

@@ -1940,6 +269,36 @@

 	return false;

+bool allocateTmpPatt(int32_t nr, uint16_t rows)

+{

+	pattTmp[nr] = (tonTyp *)calloc((MAX_PATT_LEN * TRACK_WIDTH) + 16, 1);

+	if (pattTmp[nr] == NULL)

+		return false;

+	pattLensTmp[nr] = rows;

+	return true;

+}

+bool allocateTmpInstr(int16_t nr)

+{

+	if (instrTmp[nr] != NULL)

+		return false; // already allocated

+	instrTyp *p = (instrTyp *)calloc(1, sizeof (instrTyp));

+	if (p == NULL)

+		return false;

+	sampleTyp *s = p->samp;

+	for (int32_t i = 0; i < MAX_SMP_PER_INST; i++, s++)

+	{

+		s->pan = 128;

+		s->vol = 64;

+	}

+	instrTmp[nr] = p;

+	return true;

+}

 static void freeTmpModule(void) // called on module load error

 	// free all patterns

@@ -1970,405 +329,8 @@

-static bool loadInstrHeader(FILE *f, uint16_t i, bool externalThreadFlag)

+bool tmpPatternEmpty(uint16_t nr)

-	int8_t k;

-	uint8_t j;

-	uint32_t readSize;

-	instrHeaderTyp ih;

-	instrTyp *ins;

-	sampleHeaderTyp *src;

-	sampleTyp *s;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

-	memset(extraSampleLengths, 0, sizeof (extraSampleLengths));

-	memset(&ih, 0, sizeof (ih));

-	fread(&ih.instrSize, 4, 1, f);

-	readSize = ih.instrSize;

-	// yes, some XMs can have a header size of 0, and it usually means 263 bytes (INSTR_HEADER_SIZE)

-	if (readSize == 0 || readSize > INSTR_HEADER_SIZE)

-		readSize = INSTR_HEADER_SIZE;

-	if (readSize < 4)

-	{

-		showMsg(0, "System message", "Error loading XM: This file is corrupt (or not supported)!");

-		return false;

-	}

-	// load instrument data into temp buffer

-	fread(ih.name, readSize-4, 1, f); // -4 = skip ih.instrSize

-	// FT2 bugfix: skip instrument header data if instrSize is above INSTR_HEADER_SIZE

-	if (ih.instrSize > INSTR_HEADER_SIZE)

-		fseek(f, ih.instrSize-INSTR_HEADER_SIZE, SEEK_CUR);

-	if (ih.antSamp < 0 || ih.antSamp > 32)

-	{

-		showMsg(0, "System message", "Error loading XM: This file is corrupt (or not supported)!");

-		return false;

-	}

-	if (i <= MAX_INST) // copy over instrument names

-	{

-		// trim off spaces at end of name

-		for (k = 21; k >= 0; k--)

-		{

-			if (ih.name[k] == ' ' || ih.name[k] == 0x1A)

-				ih.name[k] = '\0';

-			else

-				break;

-		}

-		memcpy(songTmp.instrName[i], ih.name, 22);

-		songTmp.instrName[i][22] = '\0';

-	}

-	if (ih.antSamp > 0)

-	{

-		if (!allocateTmpInstr(i))

-		{

-			showMsg(0, "System message", "Not enough memory!");

-			return false;

-		}

-		// copy instrument header elements to our instrument struct

-		ins = instrTmp[i];

-		memcpy(ins->ta, ih.ta, 96);

-		memcpy(ins->envVP, ih.envVP, 12*2*sizeof(int16_t));

-		memcpy(ins->envPP, ih.envPP, 12*2*sizeof(int16_t));

-		ins->envVPAnt = ih.envVPAnt;

-		ins->envPPAnt = ih.envPPAnt;

-		ins->envVSust = ih.envVSust;

-		ins->envVRepS = ih.envVRepS;

-		ins->envVRepE = ih.envVRepE;

-		ins->envPSust = ih.envPSust;

-		ins->envPRepS = ih.envPRepS;

-		ins->envPRepE = ih.envPRepE;

-		ins->envVTyp = ih.envVTyp;

-		ins->envPTyp = ih.envPTyp;

-		ins->vibTyp = ih.vibTyp;

-		ins->vibSweep = ih.vibSweep;

-		ins->vibDepth = ih.vibDepth;

-		ins->vibRate = ih.vibRate;

-		ins->fadeOut = ih.fadeOut;

-		ins->midiOn = (ih.midiOn == 1) ? true : false;

-		ins->midiChannel = ih.midiChannel;

-		ins->midiProgram = ih.midiProgram;

-		ins->midiBend = ih.midiBend;

-		ins->mute = (ih.mute == 1) ? true : false;

-		ins->antSamp = ih.antSamp; // used in loadInstrSample()

-		// sanitize stuff for broken/unsupported instruments

-		ins->midiProgram = CLAMP(ins->midiProgram, 0, 127);

-		ins->midiBend = CLAMP(ins->midiBend, 0, 36);

-		if (ins->midiChannel > 15) ins->midiChannel = 15;

-		if (ins->vibDepth > 0x0F) ins->vibDepth = 0x0F;

-		if (ins->vibRate > 0x3F) ins->vibRate = 0x3F;

-		if (ins->vibTyp > 3) ins->vibTyp = 0;

-		for (j = 0; j < 96; j++)

-		{

-			if (ins->ta[j] >= MAX_SMP_PER_INST)

-				ins->ta[j] = MAX_SMP_PER_INST-1;

-		}

-		if (ins->envVPAnt > 12) ins->envVPAnt = 12;

-		if (ins->envVRepS > 11) ins->envVRepS = 11;

-		if (ins->envVRepE > 11) ins->envVRepE = 11;

-		if (ins->envVSust > 11) ins->envVSust = 11;

-		if (ins->envPPAnt > 12) ins->envPPAnt = 12;

-		if (ins->envPRepS > 11) ins->envPRepS = 11;

-		if (ins->envPRepE > 11) ins->envPRepE = 11;

-		if (ins->envPSust > 11) ins->envPSust = 11;

-		for (j = 0; j < 12; j++)

-		{

-			if ((uint16_t)ins->envVP[j][0] > 32767) ins->envVP[j][0] = 32767;

-			if ((uint16_t)ins->envPP[j][0] > 32767) ins->envPP[j][0] = 32767;

-			if ((uint16_t)ins->envVP[j][1] > 64) ins->envVP[j][1] = 64;

-			if ((uint16_t)ins->envPP[j][1] > 63) ins->envPP[j][1] = 63;

-		}

-		int32_t sampleHeadersToRead = ih.antSamp;

-		if (sampleHeadersToRead > MAX_SMP_PER_INST)

-			sampleHeadersToRead = MAX_SMP_PER_INST;

-		if (fread(ih.samp, sampleHeadersToRead * sizeof (sampleHeaderTyp), 1, f) != 1)

-		{

-			showMsg(0, "System message", "General I/O error during loading!");

-			return false;

-		}

-		// if instrument contains more than 16 sample headers (unsupported), skip them

-		if (ih.antSamp > MAX_SMP_PER_INST) // can only be 0..32 at this point

-		{

-			const int32_t samplesToSkip = ih.antSamp-MAX_SMP_PER_INST;

-			for (j = 0; j < samplesToSkip; j++)

-			{

-				fread(&extraSampleLengths[j], 4, 1, f); // used for skipping data in loadInstrSample()

-				fseek(f, sizeof (sampleHeaderTyp)-4, SEEK_CUR);

-			}

-		}

-		for (j = 0; j < sampleHeadersToRead; j++)

-		{

-			s = &instrTmp[i]->samp[j];

-			src = &ih.samp[j];

-			// copy sample header elements to our sample struct

-			s->len = src->len;

-			s->repS = src->repS;

-			s->repL = src->repL;

-			s->vol = src->vol;

-			s->fine = src->fine;

-			s->typ = src->typ;

-			s->pan = src->pan;

-			s->relTon = src->relTon;

-			memcpy(s->name, src->name, 22);

-			s->name[22] = '\0';

-			// dst->pek is set up later

-			// trim off spaces at end of name

-			for (k = 21; k >= 0; k--)

-			{

-				if (s->name[k] == ' ' || s->name[k] == 0x1A)

-					s->name[k] = '\0';

-				else

-					break;

-			}

-			// sanitize stuff broken/unsupported samples

-			if (s->vol > 64)

-				s->vol = 64;

-			s->relTon = CLAMP(s->relTon, -48, 71);

-		}

-	}

-	return true;

-}

-void checkSampleRepeat(sampleTyp *s)

-{

-	if (s->repS < 0) s->repS = 0;

-	if (s->repL < 0) s->repL = 0;

-	if (s->repS > s->len) s->repS = s->len;

-	if (s->repS+s->repL > s->len) s->repL = s->len - s->repS;

-	if (s->repL == 0) s->typ &= ~3; // non-FT2 fix: force loop off if looplen is 0

-}

-static bool loadInstrSample(FILE *f, uint16_t i, bool externalThreadFlag)

-{

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

-	if (instrTmp[i] == NULL)

-		return true; // empty instrument, let's just pretend it got loaded successfully

-	uint16_t k = instrTmp[i]->antSamp;

-	if (k > MAX_SMP_PER_INST)

-		k = MAX_SMP_PER_INST;

-	sampleTyp *s = instrTmp[i]->samp;

-	if (i > MAX_INST) // insNum > 128, just skip sample data

-	{

-		for (uint16_t j = 0; j < k; j++, s++)

-		{

-			if (s->len > 0)

-				fseek(f, s->len, SEEK_CUR);

-		}

-	}

-	else

-	{

-		for (uint16_t j = 0; j < k; j++, s++)

-		{

-			// if a sample has both forward loop and pingpong loop set, make it pingpong loop only (FT2 mixer behavior)

-			if ((s->typ & 3) == 3)

-				s->typ &= 0xFE;

-			int32_t l = s->len;

-			if (l <= 0)

-			{

-				s->pek = NULL;

-				s->len = 0;

-				s->repL = 0;

-				s->repS = 0;

-				if (s->typ & 32)

-					s->typ &= ~32; // remove stereo flag

-			}

-			else

-			{

-				int32_t bytesToSkip = 0;

-				if (l > MAX_SAMPLE_LEN)

-				{

-					bytesToSkip = l - MAX_SAMPLE_LEN;

-					l = MAX_SAMPLE_LEN;

-				}

-				s->pek = NULL;

-				s->origPek = (int8_t *)malloc(l + LOOP_FIX_LEN);

-				if (s->origPek == NULL)

-				{

-					showMsg(0, "System message", "Not enough memory!");

-					return false;

-				}

-				s->pek = s->origPek + SMP_DAT_OFFSET;

-				const int32_t bytesRead = (int32_t)fread(s->pek, 1, l, f);

-				if (bytesRead < l)

-				{

-					const int32_t bytesToClear = l - bytesRead;

-					memset(&s->pek[bytesRead], 0, bytesToClear);

-				}

-				if (bytesToSkip > 0)

-					fseek(f, bytesToSkip, SEEK_CUR);

-				delta2Samp(s->pek, l, s->typ);

-				if (s->typ & 32) // stereo sample - already downmixed to mono in delta2samp()

-				{

-					s->typ &= ~32; // remove stereo flag

-					s->len >>= 1;

-					s->repL >>= 1;

-					s->repS >>= 1;

-					int8_t *newPtr = (int8_t *)realloc(s->origPek, s->len + LOOP_FIX_LEN);

-					if (newPtr != NULL)

-					{

-						s->origPek = newPtr;

-						s->pek = s->origPek + SMP_DAT_OFFSET;

-					}

-				}

-			}

-			// NON-FT2 FIX: Align to 2-byte if 16-bit sample

-			if (s->typ & 16)

-			{

-				s->repL &= 0xFFFFFFFE;

-				s->repS &= 0xFFFFFFFE;

-				s->len &= 0xFFFFFFFE;

-			}

-			checkSampleRepeat(s);

-			fixSample(s);

-		}

-	}

-	if (instrTmp[i]->antSamp > MAX_SMP_PER_INST)

-	{

-		const int32_t samplesToSkip = instrTmp[i]->antSamp-MAX_SMP_PER_INST;

-		for (i = 0; i < samplesToSkip; i++)

-		{

-			if (extraSampleLengths[i] > 0)

-				fseek(f, extraSampleLengths[i], SEEK_CUR);

-		}

-	}

-	return true;

-}

-void unpackPatt(uint8_t *dst, uint8_t *src, uint16_t len, int32_t antChn)

-{

-	int32_t j;

-	if (dst == NULL)

-		return;

-	const int32_t srcEnd = len * (sizeof (tonTyp) * antChn);

-	int32_t srcIdx = 0;

-	int32_t numChannels = antChn;

-	if (numChannels > MAX_VOICES)

-		numChannels = MAX_VOICES;

-	const int32_t pitch = sizeof (tonTyp) * (MAX_VOICES - antChn);

-	for (int32_t i = 0; i < len; i++)

-	{

-		for (j = 0; j < numChannels; j++)

-		{

-			if (srcIdx >= srcEnd)

-				return; // error!

-			const uint8_t note = *src++;

-			if (note & 0x80)

-			{

-				*dst++ = (note & 0x01) ? *src++ : 0;

-				*dst++ = (note & 0x02) ? *src++ : 0;

-				*dst++ = (note & 0x04) ? *src++ : 0;

-				*dst++ = (note & 0x08) ? *src++ : 0;

-				*dst++ = (note & 0x10) ? *src++ : 0;

-			}

-			else

-			{

-				*dst++ = note;

-				*dst++ = *src++;

-				*dst++ = *src++;

-				*dst++ = *src++;

-				*dst++ = *src++;

-			}

-			// if note is overflowing (>97), remove it

-			if (*(dst-5) > 97)

-				*(dst-5) = 0;

-			// non-FT2 security fix: if effect is above 35 (Z), clear effect and parameter

-			if (*(dst-2) > 35)

-			{

-				*(dst-2) = 0;

-				*(dst-1) = 0;

-			}

-			srcIdx += sizeof (tonTyp);

-		}

-		// if more than 32 channels, skip rest of the channels for this row

-		for (; j < antChn; j++)

-		{

-			if (srcIdx >= srcEnd)

-				return; // error!

-			const uint8_t note = *src++;

-			if (note & 0x80)

-			{

-				if (note & 0x01) src++;

-				if (note & 0x02) src++;

-				if (note & 0x04) src++;

-				if (note & 0x08) src++;

-				if (note & 0x10) src++;

-			}

-			else

-			{

-				src++;

-				src++;

-				src++;

-				src++;

-			}

-			srcIdx += sizeof (tonTyp);

-		}

-		// if song has <32 channels, align pointer to next row (skip unused channels)

-		if (antChn < MAX_VOICES)

-			dst += pitch;

-	}

-}

-static bool tmpPatternEmpty(uint16_t nr)

-{

 	if (pattTmp[nr] == NULL)

 		return true;

@@ -2396,97 +358,6 @@

 		memset(ptr, 0, width);

-static bool loadPatterns(FILE *f, uint16_t antPtn, bool externalThreadFlag)

-{

-	uint8_t tmpLen;

-	patternHeaderTyp ph;

-	int16_t (*showMsg)(int16_t, const char *, const char *);

-	showMsg = externalThreadFlag ? okBoxThreadSafe : okBox;

-	bool pattLenWarn = false;

-	for (uint16_t i = 0; i < antPtn; i++)

-	{

-		if (fread(&ph.patternHeaderSize, 4, 1, f) != 1)

-			goto pattCorrupt;

-		if (fread(&ph.typ, 1, 1, f) != 1)

-			goto pattCorrupt;

-		ph.pattLen = 0;

-		if (songTmp.ver == 0x0102)

-		{

-			if (fread(&tmpLen, 1, 1, f) != 1)

-				goto pattCorrupt;

-			if (fread(&ph.dataLen, 2, 1, f) != 1)

-				goto pattCorrupt;

-			ph.pattLen = tmpLen + 1; // +1 in v1.02

-			if (ph.patternHeaderSize > 8)

-				fseek(f, ph.patternHeaderSize - 8, SEEK_CUR);

-		}

-		else

-		{

-			if (fread(&ph.pattLen, 2, 1, f) != 1)

-				goto pattCorrupt;

-			if (fread(&ph.dataLen, 2, 1, f) != 1)

-				goto pattCorrupt;

-			if (ph.patternHeaderSize > 9)

-				fseek(f, ph.patternHeaderSize - 9, SEEK_CUR);

-		}

-		if (feof(f))

-			goto pattCorrupt;

-		pattLensTmp[i] = ph.pattLen;

-		if (pattLensTmp[i] > MAX_PATT_LEN)

-		{

-			pattLensTmp[i] = MAX_PATT_LEN;

-			pattLenWarn = true;

-		}

-		if (ph.dataLen > 0)

-		{

-			pattTmp[i] = (tonTyp *)calloc((MAX_PATT_LEN * TRACK_WIDTH) + 16, 1);

-			if (pattTmp[i] == NULL)

-			{

-				showMsg(0, "System message", "Not enough memory!");

-				return false;

-			}

-			if (fread(packedPattData, 1, ph.dataLen, f) != ph.dataLen)

-				goto pattCorrupt;

-			unpackPatt((uint8_t *)pattTmp[i], packedPattData, pattLensTmp[i], songTmp.antChn);

-			clearUnusedChannels(pattTmp[i], pattLensTmp[i], songTmp.antChn);

-		}

-		if (tmpPatternEmpty(i))

-		{

-			if (pattTmp[i] != NULL)

-			{

-				free(pattTmp[i]);

-				pattTmp[i] = NULL;

-			}

-			pattLensTmp[i] = 64;

-		}

-	}

-	if (pattLenWarn)

-		showMsg(0, "System message", "This module contains pattern(s) with a length above 256! They will be truncated.");

-	return true;

-pattCorrupt:

-	showMsg(0, "System message", "Error loading XM: This file is corrupt!");

-	return false;

-}

 // called from input/video thread after the module was done loading

 static void setupLoadedModule(void)

@@ -2513,17 +384,29 @@

 		pattLens[i] = pattLensTmp[i];

+	// copy over song struct

+	memcpy(&song, &songTmp, sizeof (songTyp));

+	fixSongName();

 	// copy over new instruments (includes sample pointers)

 	for (int16_t i = 1; i <= MAX_INST; i++)

 		instr[i] = instrTmp[i];

-		fixSampleName(i);

-	}

+		fixInstrAndSampleNames(i);

-	// copy over song struct

-	memcpy(&song, &songTmp, sizeof (songTyp));

-	fixSongName();

+		if (instr[i] != NULL)

+		{

+			for (int32_t j = 0; j < MAX_SMP_PER_INST; j++)

+			{

+				sampleTyp *s = &instr[i]->samp[j];

+				checkSampleRepeat(s);

+				if (s->pek != NULL)

+					fixSample(s); // prepare sample for branchless linear interpolation

+			}

+		}

+	}

 	// we are the owners of the allocated memory ptrs set by the loader thread now

 	// support non-even channel numbers

@@ -2555,10 +438,7 @@

 	editor.timer = song.timer;

 	editor.globalVol = song.globVol;

-	if (loadedFormat == FORMAT_XM)

-		setFrqTab(linearFreqTable);

-	else

-		setFrqTab(false);

+	setFrqTab(tmpLinearPeriodsFlag);

 	unlockMixerCallback();

@@ -2639,14 +519,9 @@

 	UNICHAR_CHDIR(editor.binaryPathU);

 	const int32_t filesize = getFileSize(filenameU);

-	if (filesize == -1 || filesize >= 512L*1024*1024) // >=2GB or >=512MB

+	if (filesize == -1 || filesize >= 512L*1024*1024) // 1) >=2GB   2) >=512MB

 		okBox(0, "System message", "Error: The module is too big to be loaded!");

-		/* This is not really true, but let's add this check to prevent accidentally

-		** passing really big files to the program. And how often do you really

-		** see a >=512MB .XM/.S3M module?

-		*/

 		free(filenameU);

 		UNICHAR_CHDIR(tmpPathU); // set old path back

 		return false;

@@ -2659,6 +534,64 @@

 	return result;

+static bool fileIsModule(UNICHAR *pathU)

+{

+	FILE *f = UNICHAR_FOPEN(pathU, "rb");

+	if (f == NULL)

+		return false;

+	int8_t modFormat = detectModule(f);

+	fclose(f);

+	/* If the module was not identified (possibly STK type),

+	** check the file extension and handle it as a module only

+	** if it starts with "mod."/"stk." or ends with ".mod"/".stk" (case insensitive).

+	*/

+	if (modFormat == FORMAT_POSSIBLY_STK)

+	{

+		char *path = unicharToCp437(pathU, false);

+		if (path == NULL)

+			return false;

+		int32_t pathLen = (int32_t)strlen(path);

+		// get filename from path

+		int32_t i = pathLen;

+		while (i--)

+		{

+			if (path[i] == DIR_DELIMITER)

+				break;

+		}

+		char *filename = path;

+		if (i > 0)

+			filename += i + 1;

+		int32_t filenameLen = (int32_t)strlen(filename);

+		// --------------------------

+		if (filenameLen > 5)

+		{

+			if (!_strnicmp("mod.", filename, 4) || !_strnicmp("stk.", filename, 4))

+			{

+				free(path);

+				return true;

+			}

+			if (!_strnicmp(".mod", &filename[filenameLen-4], 4) || !_strnicmp(".stk", &filename[filenameLen-4], 4))

+			{

+				free(path);

+				return true;

+			}

+		}

+		free(path);

+		return false;

+	}

+	return (modFormat != FORMAT_UNKNOWN);

+}

 void loadDroppedFile(char *fullPathUTF8, bool songModifiedCheck)

 	if (ui.sysReqShown || fullPathUTF8 == NULL)

@@ -2701,16 +634,12 @@

 	// pass UTF8 to these tests so that we can test file ending in ASCII/ANSI

-	if (fileIsInstrument(fullPathUTF8))

+	if (fileIsInstr(fullPathU))

 		loadInstr(fullPathU);

-	else if (fileIsSample(fullPathUTF8))

+	else if (fileIsModule(fullPathU))

-		loadSample(fullPathU, editor.curSmp, false);

-	}

-	else

-	{

 		SDL_RestoreWindow(video.window);

 		if (songModifiedCheck && song.isModified)

@@ -2728,6 +657,10 @@

 		editor.loadMusicEvent = EVENT_LOADMUSIC_DRAGNDROP;

 		loadMusic(fullPathU);

+	}

+	else

+	{

+		loadSample(fullPathU, editor.curSmp, false);

 	free(fullPathU);

--- a/src/ft2_module_loader.h

+++ b/src/ft2_module_loader.h

@@ -2,12 +2,24 @@

 #include <stdint.h>

 #include <stdbool.h>

+#include "ft2_header.h"

 #include "ft2_unicode.h"

+bool tmpPatternEmpty(uint16_t nr);

+void clearUnusedChannels(tonTyp *p, int16_t pattLen, int32_t antChn);

+bool allocateTmpInstr(int16_t nr);

+bool allocateTmpPatt(int32_t nr, uint16_t rows);

 void loadMusic(UNICHAR *filenameU);

 bool loadMusicUnthreaded(UNICHAR *filenameU, bool autoPlay);

 bool handleModuleLoadFromArg(int argc, char **argv);

 void loadDroppedFile(char *fullPathUTF8, bool songModifiedCheck);

 void handleLoadMusicEvents(void);

-void clearUnusedChannels(tonTyp *p, int16_t pattLen, int32_t antChn);

-void checkSampleRepeat(sampleTyp *s);

+// file extensions accepted by Disk Op. in module mode

+extern char *supportedModExtensions[];

+extern volatile bool tmpLinearPeriodsFlag;

+extern int16_t pattLensTmp[MAX_PATTERNS];

+extern tonTyp *pattTmp[MAX_PATTERNS];

+extern instrTyp *instrTmp[1+256];

+extern songTyp songTmp;

--- a/src/ft2_replayer.c

+++ b/src/ft2_replayer.c

@@ -27,9 +27,7 @@

 ** If something looks to be off, it probably isn't!

*/

-// non-FT2 precalced stuff (these are kinda big...)

-static double dPeriod2HzTab[65536], dLogTab[768], dExp2MulTab[32], dHz2MixDeltaMul;

-static uint32_t revMixDeltaTab[65536];

+static double dLogTab[768], dExp2MulTab[32];

 static bool bxxOverflow;

 static tonTyp nilPatternLine[MAX_VOICES];

@@ -49,41 +47,31 @@

 tonTyp *patt[MAX_PATTERNS];

 // ----------------------------------

-void fixSongName(void) // removes spaces from right side of song name

+void fixString(char *str, int32_t lastChrPos) // removes leading spaces and 0x1A chars

-	for (int32_t i = 20; i >= 0; i--)

+	for (int32_t i = lastChrPos; i >= 0; i--)

-		if (song.name[i] == ' ')

-			song.name[i] = '\0';

-		else

+		if (str[i] == ' ' || str[i] == 0x1A)

+			str[i] = '\0';

+		else if (str[i] != '\0')

 			break;

+	str[lastChrPos+1] = '\0';

-void fixSampleName(int16_t nr) // removes spaces from right side of ins/smp names

+void fixSongName(void)

-	for (int32_t i = 21; i >= 0; i--)

-	{

-		if (song.instrName[nr][i] == ' ')

-			song.instrName[nr][i] = '\0';

-		else

-			break;

-	}

+	fixString(song.name, 19);

+}

+void fixInstrAndSampleNames(int16_t nr)

+{

+	fixString(song.instrName[nr], 21);

 	if (instr[nr] != NULL)

 		sampleTyp *s = instr[nr]->samp;

 		for (int32_t i = 0; i < MAX_SMP_PER_INST; i++, s++)

-		{

-			for (int16_t j = 21; j >= 0; j--)

-			{

-				if (s->name[j] == ' ')

-					s->name[j] = '\0';

-				else

-					break;

-			}

-			s->name[22] = '\0'; // just in case (for tracker, not present in sample header when saving)

-		}

+			fixString(s->name, 21);

@@ -130,7 +118,7 @@

 	#define MIN_PERIOD (0)

 	#define MAX_PERIOD (((10*12*16)-1)-1) /* -1 (because of bugged amigaPeriods table values) */

-	double (*dGetHzFromPeriod)(int32_t) = linearPeriodsFlag ? &linearPeriod2Hz : &amigaPeriod2Hz;

+	double (*dGetHzFromPeriod)(int32_t) = linearPeriodsFlag ? dLinearPeriod2Hz : dAmigaPeriod2Hz;

 	const uint16_t *periodTab = linearPeriodsFlag ? linearPeriods : amigaPeriods;

 	if (midCFreq <= 0 || periodTab == NULL)

@@ -260,66 +248,32 @@

 	return i+1;

-double linearPeriod2Hz(int32_t period)

+double dLinearPeriod2Hz(int32_t period)

 	if (period == 0)

-		return 0.0; // in FT2, a period of 0 gives 0Hz

+		return 0.0; // in FT2, a period of 0 results in 0Hz

-	const uint16_t invPeriod = (uint16_t)(12 * 192 * 4) - (uint16_t)period; // this intentionally 16-bit-underflows to be accurate to FT2

+	const uint16_t invPeriod = (12 * 192 * 4) - (uint16_t)period; // intentionally underflows (uint16_t) to be FT2-accurate

 	const int32_t quotient = invPeriod / 768;

 	const int32_t remainder = invPeriod % 768;

-	return dLogTab[remainder] * dExp2MulTab[(14-quotient) & 31]; // x = y >> ((14-octave) & 31)

+	return dLogTab[remainder] * dExp2MulTab[(14-quotient) & 31]; // x = y / 2^((14-quotient) & 31)

-double amigaPeriod2Hz(int32_t period)

+double dAmigaPeriod2Hz(int32_t period)

 	if (period == 0)

-		return 0.0; // in FT2, a period of 0 gives 0Hz

+		return 0.0; // in FT2, a period of 0 results in 0Hz

-	return (double)(8363*1712) / period;

+	return (double)(8363 * 1712) / period;

-static void calcPeriod2HzTable(void) // called every time "linear/amiga frequency" mode is changed

+double dPeriod2Hz(int32_t period)

-	if (audio.linearPeriodsFlag)

-	{

-		// linear periods

-		for (int32_t i = 0; i < 65536; i++)

-			dPeriod2HzTab[i] = linearPeriod2Hz(i);

-	}

-	else

-	{

-		// Amiga periods

-		for (int32_t i = 0; i < 65536; i++)

-			dPeriod2HzTab[i] = amigaPeriod2Hz(i);

-	}

+	return audio.linearPeriodsFlag ? dLinearPeriod2Hz(period) : dAmigaPeriod2Hz(period);

-/* Called every time "linear/amiga frequency" mode or audio frequency is changed.

-**

-** Used to replace a DIV with a MUL in the outside audio mixer loop. This can actually

-** be beneficial if you are playing VERY tightly looped samples, and/or if the CPU has

-** no DIV instruction (certain ARM CPUs, for instance).

-**

-** A bit hackish and extreme considering it's 65536*4 bytes long, but why not.

-*/

-void calcRevMixDeltaTable(void)

-{

-	for (int32_t i = 0; i < 65536; i++)

-	{

-		const uint16_t period = (uint16_t)i;

-		const uint64_t delta = getMixerDelta(period);

-		uint32_t revDelta = UINT32_MAX;

-		if (delta != 0)

-			revDelta = (uint32_t)((UINT64_MAX / delta) >> 16); // MUST be truncated, not rounded!

-		revMixDeltaTab[i] = revDelta;

-	}

-}

-// returns *exact* FT2 C-4 voice rate (depending on finetune, relative note and Amiga/linear mode)

+// returns *exact* FT2 C-4 voice rate (depending on finetune, relative note and linear/Amiga period mode)

 double getSampleC4Rate(sampleTyp *s)

 	int32_t note = (96/2) + s->relTon;

@@ -343,16 +297,14 @@

 	else

 		note2Period = amigaPeriods;

-	calcPeriod2HzTable();

-	calcRevMixDeltaTable();

 	resumeAudio();

-	// update "frequency table" radiobutton, if it's shown

 	if (ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)

-		setConfigIORadioButtonStates();

+		// update "frequency table" radiobutton, if it's shown

+		setConfigIORadioButtonStates();

+		// update mid-C freq. in instr. editor (it can slightly differ between Amiga/linear)

 		if (ui.instEditorShown)

 			drawC4Rate();

@@ -452,7 +404,7 @@

 	if (audioFreq <= 0)

 		return;

-	dHz2MixDeltaMul = (double)MIXER_FRAC_SCALE / audioFreq;

+	audio.dHz2MixDeltaMul = (double)MIXER_FRAC_SCALE / audioFreq;

 	audio.quickVolRampSamples = (int32_t)((audioFreq / 200.0) + 0.5); // rounded

 	audio.dRampQuickVolMul = 1.0 / audio.quickVolRampSamples;

@@ -481,35 +433,27 @@

-double dPeriod2Hz(uint16_t period) // uses LUT for fast operation

+int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote) // for piano in Instr. Ed.

-	return dPeriod2HzTab[period];

-}

+	assert(note2Period != NULL);

-int64_t getMixerDelta(uint16_t period)

-{

-	/* Precision has been tested for most extreme case (Amiga period 1, 44100Hz),

-	** and there is no precision loss using 64-bit double-precision here, even

-	** though we can get VERY big numbers.

-	*/

-	return (int64_t)((dPeriod2Hz(period) * dHz2MixDeltaMul) + 0.5); // Hz -> rounded 32.32 fixed-point mixer delta

-}

+	if (period > note2Period[0])

+		return -1; // out of lower range on piano

-// used for calculating the max safe amount of samples to mix before entering inner mix loop

-uint32_t getRevMixerDelta(uint16_t period)

-{

-	return revMixDeltaTab[period];

-}

+	if (audio.linearPeriodsFlag)

+	{

+		int32_t note = ((10*12*16*4+64) - (period + 16*2)) >> 2;

+		note -= ((int8_t)finetune >> 3) + 16;

+		note = ((note >> 4) + 1) - relativeNote;

-int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote) // for piano in Instr. Ed.

-{

-	if (period > note2Period[0]) // don't show periods below C-0 (w/ lowest finetune) on piano

-		return -1;

+		return note;

+	}

+	/* Amiga periods require a slower method...

+	** This is not 100% accurate for all periods, but should be faster

+	** than using log2() and floating-point arithmetics.

+	*/

 	finetune = ((int8_t)finetune >> 3) + 16;

-	// this is not 100% accurate for all periods, but should be faster than using log2() and floating-point arithmetics

 	int32_t hiPeriod = (10*12*16)+16;

 	int32_t loPeriod = 0;

@@ -1561,10 +1505,12 @@

 			dVol = 1.0;

 		ch->dFinalVol = dVol;

+		ch->finalVol = (uint8_t)(int32_t)((ch->dFinalVol * 255) + 0.5); // 0.0 .. 1.0 -> 0..255 rounded, used for visuals

 	else

 		ch->dFinalVol = 0.0;

+		ch->finalVol = 0;

 	// *** PANNING ENVELOPE ***

@@ -2865,8 +2811,6 @@

 		return false;

-	calcPeriod2HzTable();

-	calcRevMixDeltaTable();

 	calcPanningTable();

 	setPos(0, 0, true);

--- a/src/ft2_replayer.h

+++ b/src/ft2_replayer.h

@@ -100,17 +100,6 @@

 #endif

 songMODInstrHeaderTyp;

-typedef struct songMOD15HeaderTyp_t

-{

-	char name[20];

-	songMODInstrHeaderTyp instr[15];

-	uint8_t len, repS, songTab[128];

-}

-#ifdef __GNUC__

-__attribute__ ((packed))

-#endif

-songMOD15HeaderTyp;

 typedef struct songMOD31HeaderTyp_t

 	char name[20];

@@ -214,11 +203,12 @@

 	int16_t midiPitch;

 	uint16_t outPeriod, realPeriod, finalPeriod, tonTyp, wantPeriod, portaSpeed;

 	uint16_t envVCnt, envPCnt, eVibAmp, eVibSweep;

-	uint16_t fadeOutAmp, fadeOutSpeed, midiVibDepth;

+	uint16_t fadeOutAmp, fadeOutSpeed, midiVibDepth;

 	int32_t envVIPValue, envPIPValue, envVAmp, envPAmp;

 	int32_t oldFinalPeriod, smpStartPos;

-	double dFinalVol;

+	uint8_t finalVol; // 0..255 (for visuals)

+	double dFinalVol; // 0.0 .. 1.0 (for mixer)

 	sampleTyp *smpPtr;

 	instrTyp *instrPtr;

@@ -241,38 +231,48 @@

 	uint8_t ton, instr, vol, effTyp, eff;

 } tonTyp;

+#ifdef _MSC_VER

+#pragma pack(push)

+#pragma pack(1)

+#endif

 typedef struct syncedChannel_t // used for audio/video sync queue

-	bool envSustainActive;

-	int8_t fineTune, relTonNr;

-	uint8_t status, sampleNr, instrNr;

-	uint16_t finalPeriod;

-	int32_t smpStartPos;

-	double dFinalVol;

-} syncedChannel_t;

+	uint8_t status;

+	uint8_t pianoNoteNr;

+	uint8_t sampleNr, instrNr;

+	uint16_t smpStartPos;

+	uint8_t vol;

+	double dHz;

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+syncedChannel_t;

+#ifdef _MSC_VER

+#pragma pack(pop)

+#endif

 double getSampleC4Rate(sampleTyp *s);

 void setNewSongPos(int32_t pos);

 void resetReplayerState(void);

-void fixSongName(void); // removes spaces from right side of song name

-void fixSampleName(int16_t nr); // removes spaces from right side of ins/smp names

+void fixString(char *str, int32_t lastChrPos); // removes leading spaces and 0x1A chars

+void fixSongName(void);

+void fixInstrAndSampleNames(int16_t nr);

 void calcReplayerVars(int32_t rate);

 // used on external sample load and during sample loading in some module formats

 void tuneSample(sampleTyp *s, const int32_t midCFreq, bool linearPeriodsFlag);

-void calcRevMixDeltaTable(void);

 void calcReplayerLogTab(void);

-double linearPeriod2Hz(int32_t period);

-double amigaPeriod2Hz(int32_t period);

-double dPeriod2Hz(uint16_t period); // uses LUT for fast operation

-int64_t getMixerDelta(uint16_t period);

-uint32_t getRevMixerDelta(uint16_t period);

+double dLinearPeriod2Hz(int32_t period);

+double dAmigaPeriod2Hz(int32_t period);

+double dPeriod2Hz(int32_t period);

 int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote); // for piano in Instr. Ed.

--- a/src/ft2_sample_ed.c

+++ b/src/ft2_sample_ed.c

@@ -47,6 +47,37 @@

 // globals

 int32_t smpEd_Rx1 = 0, smpEd_Rx2 = 0;

+// allocs sample with proper alignment and padding for branchless resampling interpolation

+bool allocateTmpSmpData(sampleTyp *s, int32_t length)

+{

+	s->origPek = (int8_t *)malloc(length + LOOP_FIX_LEN);

+	if (s->origPek == NULL)

+	{

+		s->pek = NULL;

+		return false;

+	}

+	s->pek = s->origPek + SMP_DAT_OFFSET;

+	return true;

+}

+// reallocs sample with proper alignment and padding for branchless resampling interpolation

+bool reallocateTmpSmpData(sampleTyp *s, int32_t length)

+{

+	if (s->origPek == NULL)

+		return allocateTmpSmpData(s, length);

+	s->origPek = (int8_t *)realloc(s->origPek, length + LOOP_FIX_LEN);

+	if (s->origPek == NULL)

+	{

+		s->pek = NULL;

+		return false;

+	}

+	s->pek = s->origPek + SMP_DAT_OFFSET;

+	return true;

+}

 sampleTyp *getCurSample(void)

 	if (editor.curInstr == 0 || instr[editor.curInstr] == NULL)

@@ -53,6 +84,20 @@

 		return NULL;

 	return &instr[editor.curInstr]->samp[editor.curSmp];

+}

+void checkSampleRepeat(sampleTyp *s)

+{

+	if (s->repS < 0) s->repS = 0;

+	if (s->repL < 0) s->repL = 0;

+	if (s->repS > s->len) s->repS = s->len;

+	if (s->repS+s->repL > s->len) s->repL = s->len - s->repS;

+	if (s->repL == 0) // non-FT2 fix: disable loop if loop length is 0

+	{

+		s->repS = 0;

+		s->typ &= ~3;

+	}

 // modifies samples before index 0, and after loop/end (for branchless mixer interpolation (kinda))

--- a/src/ft2_sample_ed.h

+++ b/src/ft2_sample_ed.h

@@ -7,7 +7,14 @@

 #define SAMPLE_AREA_WIDTH 632

 #define SAMPLE_AREA_Y_CENTER 250

+// allocs sample with proper alignment and padding for branchless resampling interpolation

+bool allocateTmpSmpData(sampleTyp *s, int32_t length);

+// reallocs sample with proper alignment and padding for branchless resampling interpolation

+bool reallocateTmpSmpData(sampleTyp *s, int32_t length);

 sampleTyp *getCurSample(void);

+void checkSampleRepeat(sampleTyp *s);

 void fixSample(sampleTyp *s); // modifies samples before index 0, and after loop/end (for branchless mixer interpolation)

 void restoreSample(sampleTyp *s); // restores samples after loop/end

 void clearSample(void);

--- a/src/ft2_sample_loader.c

+++ b/src/ft2_sample_loader.c

@@ -16,943 +16,124 @@

 #include "ft2_diskop.h"

 #include "ft2_structs.h"

-/* All of these routines were written from scratch and were not present

-** in original FT2.

-*/

+bool loadAIFF(FILE *f, uint32_t filesize);

+bool loadIFF(FILE *f, uint32_t filesize);

+bool loadRAW(FILE *f, uint32_t filesize);

+bool loadWAV(FILE *f, uint32_t filesize);

 enum

-	STEREO_SAMPLE_READ_LEFT = 1,

-	STEREO_SAMPLE_READ_RIGHT = 2,

-	STEREO_SAMPLE_CONVERT = 3,

-	WAV_FORMAT_PCM = 0x0001,

-	WAV_FORMAT_IEEE_FLOAT = 0x0003

+	FORMAT_UNKNOWN = 0,

+	FORMAT_IFF = 1,

+	FORMAT_WAV = 2,

+	FORMAT_AIFF = 3

};

-static volatile bool sampleIsLoading;

-static bool loadAsInstrFlag;

-static uint8_t sampleSlot;

-static int16_t stereoSampleLoadMode;

-static SDL_Thread *thread;

-static void normalize32bitSigned(int32_t *sampleData, uint32_t sampleLength);

-static void normalize16bitFloatSigned(float *fSampleData, uint32_t sampleLength);

-static void normalize64bitDoubleSigned(double *dSampleData, uint32_t sampleLength);

-void freeTmpSample(sampleTyp *s)

+// file extensions accepted by Disk Op. in sample mode

+char *supportedSmpExtensions[] =

-	if (s->origPek != NULL)

-	{

-		free(s->origPek);

-		s->origPek = NULL;

-		s->pek = NULL;

-	}

+	"iff", "raw", "wav", "snd", "smp", "sam", "aif", "pat", "aiff",

-	memset(s, 0, sizeof (sampleTyp));

-}

+	"END_OF_LIST" // do NOT move, remove or edit this line!

+};

-void removeSampleIsLoadingFlag(void)

-{

-	sampleIsLoading = false;

-}

+// globals for sample loaders

+bool loadAsInstrFlag, smpFilenameSet;

+char *smpFilename;

+uint8_t sampleSlot;

+sampleTyp tmpSmp;

+// --------------------------

-static int32_t getAIFFRate(uint8_t *in)

-{

-	int32_t exp = (int32_t)(((in[0] & 0x7F) << 8) | in[1]);

-	uint32_t lo = (in[2] << 24) | (in[3] << 16) | (in[4] << 8) | in[5];

-	uint32_t hi = (in[6] << 24) | (in[7] << 16) | (in[8] << 8) | in[9];

+static volatile bool sampleIsLoading;

+static SDL_Thread *thread;

-	if (exp == 0 && lo == 0 && hi == 0)

-		return 0;

+static void freeTmpSample(sampleTyp *s);

-	exp -= 16383;

-	double dOut = ldexp(lo, -31 + exp) + ldexp(hi, -63 + exp);

-	return (int32_t)(dOut + 0.5); // rounded

-}

-static bool aiffIsStereo(FILE *f) // only ran on files that are confirmed to be AIFFs

+// Crude sample detection routine. These aren't always accurate detections!

+static int8_t detectSample(FILE *f)

-	uint16_t numChannels;

-	uint32_t chunkID, chunkSize;

+	uint8_t D[256];

 	uint32_t oldPos = ftell(f);

+	rewind(f);

+	memset(D, 0, sizeof (D));

+	fread(D, 1, sizeof (D), f);

+	fseek(f, oldPos, SEEK_SET);

-	fseek(f, 0, SEEK_END);

-	int32_t filesize = ftell(f);

+	if (!memcmp("FORM", &D[0], 4) && (!memcmp("8SVX", &D[8], 4) || !memcmp("16SV", &D[8], 4)))

+		return FORMAT_IFF;

-	if (filesize < 12)

-	{

-		fseek(f, oldPos, SEEK_SET);

-		return false;

-	}

+	if (!memcmp("RIFF", &D[0], 4) && !memcmp("WAVE", &D[8], 4))

+		return FORMAT_WAV;

-	fseek(f, 12, SEEK_SET);

+	if (!memcmp("FORM", &D[0], 4) && (!memcmp("AIFF", &D[8], 4) || !memcmp("AIFC", &D[8], 4)))

+		return FORMAT_AIFF;

-	uint32_t commPtr = 0;

-	uint32_t commLen = 0;

-	int32_t bytesRead = 0;

-	while (!feof(f) && bytesRead < filesize-12)

-	{

-		fread(&chunkID, 4, 1, f); chunkID = SWAP32(chunkID); if (feof(f)) break;

-		fread(&chunkSize, 4, 1, f); chunkSize = SWAP32(chunkSize); if (feof(f)) break;

-		int32_t endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

-		switch (chunkID)

-		{

-			case 0x434F4D4D: // "COMM"

-			{

-				commPtr = ftell(f);

-				commLen = chunkSize;

-			}

-			break;

-			default: break;

-		}

-		bytesRead += (chunkSize + (chunkSize & 1));

-		fseek(f, endOfChunk, SEEK_SET);

-	}

-	if (commPtr == 0 || commLen < 2)

-	{

-		fseek(f, oldPos, SEEK_SET);

-		return false;

-	}

-	fseek(f, commPtr, SEEK_SET);

-	fread(&numChannels, 2, 1, f); numChannels = SWAP16(numChannels);

-	fseek(f, oldPos, SEEK_SET);

-	return (numChannels == 2);

+	return FORMAT_UNKNOWN;

-static bool wavIsStereo(FILE *f) // only ran on files that are confirmed to be WAVs

+static int32_t SDLCALL loadSampleThread(void *ptr)

-	uint16_t numChannels;

-	uint32_t chunkID, chunkSize;

-	uint32_t oldPos = ftell(f);

-	fseek(f, 0, SEEK_END);

-	int32_t filesize = ftell(f);

-	if (filesize < 12)

-	{

-		fseek(f, oldPos, SEEK_SET);

-		return false;

-	}

-	fseek(f, 12, SEEK_SET);

-	uint32_t fmtPtr = 0;

-	uint32_t fmtLen = 0;

-	int32_t bytesRead = 0;

-	while (!feof(f) && bytesRead < filesize-12)

+	if (editor.tmpFilenameU == NULL)

-		fread(&chunkID, 4, 1, f); if (feof(f)) break;

-		fread(&chunkSize, 4, 1, f); if (feof(f)) break;

-		int32_t endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

-		switch (chunkID)

-		{

-			case 0x20746D66: // "fmt "

-			{

-				fmtPtr = ftell(f);

-				fmtLen = chunkSize;

-			}

-			break;

-			default: break;

-		}

-		bytesRead += (chunkSize + (chunkSize & 1));

-		fseek(f, endOfChunk, SEEK_SET);

-	}

-	if (fmtPtr == 0 || fmtLen < 4)

-	{

-		fseek(f, oldPos, SEEK_SET);

+		loaderMsgBox("General I/O error during loading!");

 		return false;

-	fseek(f, fmtPtr + 2, SEEK_SET);

-	fread(&numChannels, 2, 1, f);

-	fseek(f, oldPos, SEEK_SET);

-	return (numChannels == 2);

-}

-static int32_t SDLCALL loadAIFFSample(void *ptr)

-{

-	char compType[4];

-	int8_t *audioDataS8;

-	uint8_t sampleRateBytes[10], *audioDataU8;

-	int16_t *audioDataS16, *audioDataS16_2, smp16;

-	uint16_t numChannels, bitDepth;

-	int32_t j, smp32, *audioDataS32;

-	uint32_t i, blockName, blockSize;

-	uint32_t offset, len32;

-	int64_t smp64;

-	sampleTyp tmpSmp;

-	// this is important for the "goto" on load error

-	FILE *f = NULL;

-	memset(&tmpSmp, 0, sizeof (tmpSmp));

-	if (editor.tmpFilenameU == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto aiffLoadError;

-	}

-	UNICHAR *filename = editor.tmpFilenameU;

-	f = UNICHAR_FOPEN(filename, "rb");

+	FILE *f = UNICHAR_FOPEN(editor.tmpFilenameU, "rb");

 	if (f == NULL)

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto aiffLoadError;

+		loaderMsgBox("General I/O error during loading! Is the file in use?");

+		return false;

+	int8_t format = detectSample(f);

 	fseek(f, 0, SEEK_END);

-	int32_t filesize = ftell(f);

-	if (filesize < 12)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto aiffLoadError;

-	}

+	uint32_t filesize = ftell(f);

-	// handle chunks

-	uint32_t commPtr = 0;

-	uint32_t commLen = 0;

-	uint32_t ssndPtr = 0;

-	uint32_t ssndLen = 0;

-	fseek(f, 12, SEEK_SET);

-	while (!feof(f) && ftell(f) < filesize-12)

+	if (filesize == 0)

-		fread(&blockName, 4, 1, f); if (feof(f)) break;

-		fread(&blockSize, 4, 1, f); if (feof(f)) break;

-		blockName = SWAP32(blockName);

-		blockSize = SWAP32(blockSize);

-		switch (blockName)

-		{

-			case 0x434F4D4D: // "COMM"

-			{

-				commPtr = ftell(f);

-				commLen = blockSize;

-			}

-			break;

-			case 0x53534E44: // "SSND"

-			{

-				ssndPtr = ftell(f);

-				ssndLen = blockSize;

-			}

-			break;

-			default: break;

-		}

-		fseek(f, blockSize + (blockSize & 1), SEEK_CUR);

+		fclose(f);

+		loaderMsgBox("Error loading sample: The file is empty!");

+		return false;

-	if (commPtr == 0 || commLen < 18 || ssndPtr == 0)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto aiffLoadError;

-	}

+	bool sampleLoaded = false;

-	// kludge for some really strange AIFFs

-	if (ssndLen == 0)

-		ssndLen = filesize - ssndPtr;

-	if (ssndPtr+ssndLen > (uint32_t)filesize)

-		ssndLen = filesize - ssndPtr;

-	fseek(f, commPtr, SEEK_SET);

-	fread(&numChannels, 2, 1, f); numChannels = SWAP16(numChannels);

-	fseek(f, 4, SEEK_CUR);

-	fread(&bitDepth, 2, 1, f); bitDepth = SWAP16(bitDepth);

-	fread(sampleRateBytes, 1, 10, f);

-	if (numChannels != 1 && numChannels != 2)

+	rewind(f);

+	switch (format)

-		okBoxThreadSafe(0, "System message", "Error loading sample: Unsupported amounts of channels!");

-		goto aiffLoadError;

+		case FORMAT_IFF: sampleLoaded = loadIFF(f, filesize); break;

+		case FORMAT_WAV: sampleLoaded = loadWAV(f, filesize); break;

+		case FORMAT_AIFF: sampleLoaded = loadAIFF(f, filesize); break;

+		default: sampleLoaded = loadRAW(f, filesize); break;

-	if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24 && bitDepth != 32)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: Unsupported bitdepth!");

-		goto aiffLoadError;

-	}

-	// read compression type (if present)

-	if (commLen > 18)

-	{

-		fread(&compType, 1, 4, f);

-		if (memcmp(compType, "NONE", 4))

-		{

-			okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-			goto aiffLoadError;

-		}

-	}

-	uint32_t sampleRate = getAIFFRate(sampleRateBytes);

-	// sample data chunk

-	fseek(f, ssndPtr, SEEK_SET);

-	fread(&offset, 4, 1, f);

-	if (offset > 0)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto aiffLoadError;

-	}

-	fseek(f, 4, SEEK_CUR);

-	ssndLen -= 8; // don't include offset and blockSize datas

-	uint32_t sampleLength = ssndLen;

-	if (sampleLength > MAX_SAMPLE_LEN)

-		sampleLength = MAX_SAMPLE_LEN;

-	freeTmpSample(&tmpSmp);

-	// read sample data

-	if (bitDepth == 8)

-	{

-		// 8-BIT SIGNED PCM

-		tmpSmp.origPek = (int8_t *)malloc(sampleLength + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto aiffLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto aiffLoadError;

-		}

-		audioDataS8 = (int8_t *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					for (i = 1; i < sampleLength; i++)

-						audioDataS8[i] = audioDataS8[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS8[i] = audioDataS8[(i * 2) + 1];

-					audioDataS8[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp16 = (audioDataS8[(i * 2) + 0] + audioDataS8[(i * 2) + 1]) >> 1;

-						audioDataS8[i] = (int8_t)smp16;

-					}

-					audioDataS8[i] = 0;

-				}

-				break;

-			}

-		}

-	}

-	else if (bitDepth == 16)

-	{

-		// 16-BIT SIGNED PCM

-		sampleLength /= 2;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 2) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto aiffLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 2, f) != 2)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto aiffLoadError;

-		}

-		// fix endianness

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		for (i = 0; i < sampleLength; i++)

-			audioDataS16[i] = SWAP16(audioDataS16[i]);

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					for (i = 1; i < sampleLength; i++)

-						audioDataS16[i] = audioDataS16[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS16[i] = audioDataS16[(i * 2) + 1];

-					audioDataS16[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp32 = (audioDataS16[(i * 2) + 0] + audioDataS16[(i * 2) + 1]) >> 1;

-						audioDataS16[i] = (int16_t)smp32;

-					}

-					audioDataS16[i] = 0;

-				}

-				break;

-			}

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16;

-	}

-	else if (bitDepth == 24)

-	{

-		// 24-BIT SIGNED PCM

-		sampleLength /= 3;

-		tmpSmp.origPek = (int8_t *)malloc(((sampleLength * 4) * 2) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto aiffLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(&tmpSmp.pek[sampleLength], sampleLength, 3, f) != 3)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto aiffLoadError;

-		}

-		audioDataS32 = (int32_t *)tmpSmp.pek;

-		// convert to 32-bit

-		audioDataU8 = (uint8_t *)tmpSmp.pek + sampleLength;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS32[i] = (audioDataU8[0] << 24) | (audioDataU8[1] << 16) | (audioDataU8[2] << 8);

-			audioDataU8 += 3;

-		}

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					for (i = 1; i < sampleLength; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 1];

-					audioDataS32[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp64 = audioDataS32[(i * 2) + 0];

-						smp64 += audioDataS32[(i * 2) + 1];

-						smp64 >>= 1;

-						audioDataS32[i] = (int32_t)smp64;

-					}

-					audioDataS32[i] = 0;

-				}

-				break;

-			}

-		}

-		normalize32bitSigned(audioDataS32, sampleLength);

-		// downscale to 16-bit (ultra fast method!)

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS16[i] = audioDataS16_2[i];

-			audioDataS16_2++;

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16;

-	}

-	else if (bitDepth == 32)

-	{

-		// 32-BIT SIGNED PCM

-		sampleLength /= 4;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 4) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto aiffLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto aiffLoadError;

-		}

-		// fix endianness

-		audioDataS32 = (int32_t *)tmpSmp.pek;

-		for (i = 0; i < sampleLength; i++)

-			audioDataS32[i] = SWAP32(audioDataS32[i]);

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					for (i = 1; i < sampleLength; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 1];

-					audioDataS32[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp64 = audioDataS32[(i * 2) + 0];

-						smp64 += audioDataS32[(i * 2) + 1];

-						smp64 >>= 1;

-						audioDataS32[i] = (int32_t)smp64;

-					}

-					audioDataS32[i] = 0;

-				}

-				break;

-			}

-		}

-		normalize32bitSigned(audioDataS32, sampleLength);

-		// downscale to 16-bit (ultra fast method!)

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS16[i] = audioDataS16_2[i];

-			audioDataS16_2++;

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16;

-	}

-	// adjust memory needed

-	int8_t *newPtr = (int8_t *)realloc(tmpSmp.origPek, sampleLength + LOOP_FIX_LEN);

-	if (newPtr != NULL)

-	{

-		tmpSmp.origPek = newPtr;

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-	}

-	// set sample attributes

-	tmpSmp.len = sampleLength;

-	tmpSmp.vol = 64;

-	tmpSmp.pan = 128;

-	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

-	// set sample name

-	char *tmpFilename = unicharToCp437(filename, true);

-	if (tmpFilename != NULL)

-	{

-		j = (int32_t)strlen(tmpFilename);

-		while (j--)

-		{

-			if (tmpFilename[j] == DIR_DELIMITER)

-				break;

-		}

-		char *tmpPtr = tmpFilename;

-		if (j > 0)

-			tmpPtr += j + 1;

-		sanitizeFilename(tmpPtr);

-		uint32_t filenameLen = (uint32_t)strlen(tmpPtr);

-		for (i = 0; i < 22; i++)

-		{

-			if (i < filenameLen)

-				tmpSmp.name[i] = tmpPtr[i];

-			else

-				tmpSmp.name[i] = '\0';

-		}

-		free(tmpFilename);

-	}

 	fclose(f);

-	lockMixerCallback();

-	if (loadAsInstrFlag) // if loaded in instrument mode

-	{

-		freeInstr(editor.curInstr);

-		memset(song.instrName[editor.curInstr], 0, 23);

-	}

+	if (!sampleLoaded)

+		goto loadError;

-	if (instr[editor.curInstr] == NULL)

-		allocateInstr(editor.curInstr);

-	if (instr[editor.curInstr] != NULL)

+	// sample loaded successfully!

+	if (!smpFilenameSet) // if we didn't set a custom sample name in the loader, set it to its filename

-		sampleTyp *s = &instr[editor.curInstr]->samp[sampleSlot];

-		freeSample(editor.curInstr, sampleSlot);

-		memcpy(s, &tmpSmp, sizeof (sampleTyp));

-		fixSample(s);

-	}

-	else

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto aiffLoadError;

-	}

-	unlockMixerCallback();

-	fixSampleName(editor.curInstr);

-	setSongModifiedFlag();

-	stereoSampleLoadMode = -1;

-	// also sets mouse busy to false when done

-	editor.updateCurSmp = true;

-	return true;

-aiffLoadError:

-	if (f != NULL) fclose(f);

-	if (tmpSmp.origPek != NULL) free(tmpSmp.origPek);

-	stereoSampleLoadMode = -1;

-	sampleIsLoading = false;

-	return false;

-	(void)ptr;

-}

-static int32_t SDLCALL loadIFFSample(void *ptr)

-{

-	char hdr[4+1];

-	uint8_t i;

-	int32_t j;

-	uint32_t blockName, blockSize;

-	sampleTyp tmpSmp;

-	// this is important for the "goto" on load error

-	FILE *f = NULL;

-	memset(&tmpSmp, 0, sizeof (tmpSmp));

-	if (editor.tmpFilenameU == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto iffLoadError;

-	}

-	UNICHAR *filename = editor.tmpFilenameU;

-	f = UNICHAR_FOPEN(filename, "rb");

-	if (f == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto iffLoadError;

-	}

-	fseek(f, 0, SEEK_END);

-	int32_t filesize = ftell(f);

-	if (filesize < 12)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto iffLoadError;

-	}

-	fseek(f, 8, SEEK_SET);

-	fread(hdr, 1, 4, f);

-	hdr[4] = '\0';

-	bool is16Bit = !strncmp(hdr, "16SV", 4);

-	uint32_t sampleLength = 0;

-	uint32_t sampleVol = 64;

-	uint32_t sampleLoopStart = 0;

-	uint32_t sampleLoopLength = 0;

-	uint16_t sampleRate = 16726;

-	uint32_t vhdrPtr = 0;

-	uint32_t vhdrLen = 0;

-	uint32_t bodyPtr = 0;

-	uint32_t bodyLen = 0;

-	uint32_t namePtr = 0;

-	uint32_t nameLen = 0;

-	fseek(f, 12, SEEK_SET);

-	while (!feof(f) && ftell(f) < filesize-12)

-	{

-		fread(&blockName, 4, 1, f); if (feof(f)) break;

-		fread(&blockSize, 4, 1, f); if (feof(f)) break;

-		blockName = SWAP32(blockName);

-		blockSize = SWAP32(blockSize);

-		switch (blockName)

-		{

-			case 0x56484452: // VHDR

-			{

-				vhdrPtr = ftell(f);

-				vhdrLen = blockSize;

-			}

-			break;

-			case 0x4E414D45: // NAME

-			{

-				namePtr = ftell(f);

-				nameLen = blockSize;

-			}

-			break;

-			case 0x424F4459: // BODY

-			{

-				bodyPtr = ftell(f);

-				bodyLen = blockSize;

-			}

-			break;

-			default: break;

-		}

-		fseek(f, blockSize + (blockSize & 1), SEEK_CUR);

-	}

-	if (vhdrPtr == 0 || vhdrLen < 20 || bodyPtr == 0)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto iffLoadError;

-	}

-	// kludge for some really strange IFFs

-	if (bodyLen == 0)

-		bodyLen = filesize - bodyPtr;

-	if (bodyPtr+bodyLen > (uint32_t)filesize)

-		bodyLen = filesize - bodyPtr;

-	fseek(f, vhdrPtr, SEEK_SET);

-	fread(&sampleLoopStart,  4, 1, f); sampleLoopStart = SWAP32(sampleLoopStart);

-	fread(&sampleLoopLength, 4, 1, f); sampleLoopLength = SWAP32(sampleLoopLength);

-	fseek(f, 4, SEEK_CUR);

-	fread(&sampleRate, 2, 1, f); sampleRate = SWAP16(sampleRate);

-	fseek(f, 1, SEEK_CUR);

-	if (fgetc(f) != 0) // sample type

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported!");

-		goto iffLoadError;

-	}

-	fread(&sampleVol, 4, 1, f); sampleVol = SWAP32(sampleVol);

-	if (sampleVol > 65535)

-		sampleVol = 65535;

-	sampleVol = (int32_t)((sampleVol / 1024.0) + 0.5);

-	if (sampleVol > 64)

-		sampleVol = 64;

-	sampleLength = bodyLen;

-	if (sampleLength > MAX_SAMPLE_LEN)

-		sampleLength = MAX_SAMPLE_LEN;

-	if (sampleLoopStart >= MAX_SAMPLE_LEN || sampleLoopLength > MAX_SAMPLE_LEN)

-	{

-		sampleLoopStart = 0;

-		sampleLoopLength = 0;

-	}

-	if (sampleLoopStart+sampleLoopLength > sampleLength)

-	{

-		sampleLoopStart = 0;

-		sampleLoopLength = 0;

-	}

-	if (sampleLoopStart > sampleLength-2)

-	{

-		sampleLoopStart = 0;

-		sampleLoopLength = 0;

-	}

-	tmpSmp.pan = 128;

-	tmpSmp.vol = 64;

-	tmpSmp.origPek = (int8_t *)malloc(sampleLength + LOOP_FIX_LEN);

-	if (tmpSmp.origPek == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto iffLoadError;

-	}

-	tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-	fseek(f, bodyPtr, SEEK_SET);

-	if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto iffLoadError;

-	}

-	// set sample attributes

-	tmpSmp.len = sampleLength;

-	if (sampleLoopLength > 2)

-	{

-		tmpSmp.repS = sampleLoopStart;

-		tmpSmp.repL = sampleLoopLength;

-		tmpSmp.typ |= 1;

-	}

-	if (is16Bit)

-	{

-		tmpSmp.len  &= 0xFFFFFFFE;

-		tmpSmp.repS &= 0xFFFFFFFE;

-		tmpSmp.repL &= 0xFFFFFFFE;

-		tmpSmp.typ |= 16;

-	}

-	tmpSmp.vol = (uint8_t)sampleVol;

-	tmpSmp.pan = 128;

-	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

-	// set name

-	if (namePtr != 0 && nameLen > 0)

-	{

-		fseek(f, namePtr, SEEK_SET);

-		if (nameLen > 21)

-		{

-			tmpSmp.name[21] = '\0';

-			fread(tmpSmp.name, 1, 21, f);

-		}

-		else

-		{

-			memset(tmpSmp.name, 0, 22);

-			fread(tmpSmp.name, 1, nameLen, f);

-		}

-	}

-	else

-	{

-		// set sample name from filename if we didn't load name from .wav

-		char *tmpFilename = unicharToCp437(filename, true);

+		char *tmpFilename = unicharToCp437(editor.tmpFilenameU, true);

 		if (tmpFilename != NULL)

-			j = (int32_t)strlen(tmpFilename);

-			while (j--)

+			int32_t i = (int32_t)strlen(tmpFilename);

+			while (i--)

-				if (tmpFilename[j] == DIR_DELIMITER)

+				if (tmpFilename[i] == DIR_DELIMITER)

 					break;

 			char *tmpPtr = tmpFilename;

-			if (j > 0)

-				tmpPtr += j + 1;

+			if (i > 0)

+				tmpPtr += i+1;

 			sanitizeFilename(tmpPtr);

-			uint32_t filenameLen = (uint32_t)strlen(tmpPtr);

+			int32_t filenameLen = (int32_t)strlen(tmpPtr);

 			for (i = 0; i < 22; i++)

 				if (i < filenameLen)

@@ -965,7 +146,7 @@

-	fclose(f);

+	fixString(tmpSmp.name, 21); // remove leading spaces from sample filename

 	lockMixerCallback();

 	if (loadAsInstrFlag) // if loaded in instrument mode

@@ -977,1114 +158,82 @@

 	if (instr[editor.curInstr] == NULL)

 		allocateInstr(editor.curInstr);

-	if (instr[editor.curInstr] != NULL)

+	if (instr[editor.curInstr] == NULL)

-		sampleTyp *s = &instr[editor.curInstr]->samp[sampleSlot];

-		freeSample(editor.curInstr, sampleSlot);

-		memcpy(s, &tmpSmp, sizeof (sampleTyp));

-		fixSample(s);

+		loaderMsgBox("Not enough memory!");

+		goto loadError;

-	else

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto iffLoadError;

-	}

-	unlockMixerCallback();

-	fixSampleName(editor.curInstr);

-	setSongModifiedFlag();

-	stereoSampleLoadMode = -1;

+	sampleTyp *s = &instr[editor.curInstr]->samp[sampleSlot];

-	// also sets mouse busy to false when done

-	editor.updateCurSmp = true;

+	freeSample(editor.curInstr, sampleSlot);

+	memcpy(s, &tmpSmp, sizeof (sampleTyp));

+	fixSample(s); // prepares sample for branchless resampling interpolation

+	fixInstrAndSampleNames(editor.curInstr);

-	return true;

-iffLoadError:

-	if (f != NULL) fclose(f);

-	if (tmpSmp.origPek != NULL) free(tmpSmp.origPek);

-	stereoSampleLoadMode = -1;

-	sampleIsLoading = false;

-	return false;

-	(void)ptr;

-}

-static int32_t SDLCALL loadRawSample(void *ptr)

-{

-	int32_t j;

-	uint32_t i;

-	sampleTyp tmpSmp;

-	// this is important for the "goto" on load error

-	FILE *f = NULL;

-	memset(&tmpSmp, 0, sizeof (tmpSmp));

-	if (editor.tmpFilenameU == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto rawLoadError;

-	}

-	UNICHAR *filename = editor.tmpFilenameU;

-	f = UNICHAR_FOPEN(filename, "rb");

-	if (f == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto rawLoadError;

-	}

-	fseek(f, 0, SEEK_END);

-	uint32_t filesize = ftell(f);

-	rewind(f);

-	if (filesize > MAX_SAMPLE_LEN)

-		filesize = MAX_SAMPLE_LEN;

-	if (filesize == 0)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto rawLoadError;

-	}

-	tmpSmp.pan = 128;

-	tmpSmp.vol = 64;

-	tmpSmp.origPek = (int8_t *)malloc(filesize + LOOP_FIX_LEN);

-	if (tmpSmp.origPek == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto rawLoadError;

-	}

-	tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-	if (fread(tmpSmp.pek, filesize, 1, f) != 1)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto rawLoadError;

-	}

-	fclose(f);

-	char *tmpFilename = unicharToCp437(filename, true);

-	if (tmpFilename != NULL)

-	{

-		j = (int32_t)strlen(tmpFilename);

-		while (j--)

-		{

-			if (tmpFilename[j] == DIR_DELIMITER)

-				break;

-		}

-		char *tmpPtr = tmpFilename;

-		if (j > 0)

-			tmpPtr += j + 1;

-		sanitizeFilename(tmpPtr);

-		uint32_t filenameLen = (uint32_t)strlen(tmpPtr);

-		for (i = 0; i < 22; i++)

-		{

-			if (i < filenameLen)

-				tmpSmp.name[i] = tmpPtr[i];

-			else

-				tmpSmp.name[i] = '\0';

-		}

-		free(tmpFilename);

-	}

-	tmpSmp.len = filesize;

-	tmpSmp.vol = 64;

-	tmpSmp.pan = 128;

-	lockMixerCallback();

-	if (loadAsInstrFlag) // if loaded in instrument mode

-	{

-		freeInstr(editor.curInstr);

-		memset(song.instrName[editor.curInstr], 0, 23);

-	}

-	if (instr[editor.curInstr] == NULL)

-		allocateInstr(editor.curInstr);

-	if (instr[editor.curInstr] != NULL)

-	{

-		sampleTyp *s = &instr[editor.curInstr]->samp[sampleSlot];

-		freeSample(editor.curInstr, sampleSlot);

-		memcpy(s, &tmpSmp, sizeof (sampleTyp));

-		fixSample(s);

-	}

-	else

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto rawLoadError;

-	}

 	unlockMixerCallback();

-	fixSampleName(editor.curInstr);

 	setSongModifiedFlag();

-	stereoSampleLoadMode = -1;

-	// also sets mouse busy to false when done

+	// when caught in main/video thread, it disables busy mouse and sets sampleIsLoading to true

 	editor.updateCurSmp = true;

 	return true;

-rawLoadError:

-	if (f != NULL) fclose(f);

-	if (tmpSmp.origPek != NULL) free(tmpSmp.origPek);

-	stereoSampleLoadMode = -1;

+loadError:

+	setMouseBusy(false);

+	freeTmpSample(&tmpSmp);

 	sampleIsLoading = false;

 	return false;

 	(void)ptr;

-static int32_t SDLCALL loadWAVSample(void *ptr)

+static void freeTmpSample(sampleTyp *s)

-	char chr, *tmpFilename, *tmpPtr;

-	int8_t *newPtr;

-	uint8_t *audioDataU8;

-	int16_t *audioDataS16, *audioDataS16_2, *ptr16;

-	uint16_t audioFormat, numChannels, bitsPerSample, tempPan, tempVol;

-	int32_t j, *audioDataS32, smp32;

-	uint32_t filenameLen, i, sampleRate, chunkID, chunkSize, sampleLength, filesize;

-	uint32_t numLoops, loopType, loopStart, loopEnd, bytesRead, endOfChunk, dataPtr, dataLen, fmtPtr;

-	uint32_t fmtLen, inamPtr, inamLen, smplPtr, smplLen, xtraPtr, xtraLen, xtraFlags, len32;

-	int64_t smp64;

-	float *fAudioDataFloat;

-	double *dAudioDataDouble;

-	sampleTyp tmpSmp, *s;

-	// this is important for the "goto" on load error

-	FILE *f = NULL;

-	memset(&tmpSmp, 0, sizeof (tmpSmp));

-	// zero out chunk pointers and lengths

-	fmtPtr = 0; fmtLen = 0;

-	dataPtr = 0; dataLen = 0;

-	inamPtr = 0; inamLen = 0;

-	xtraPtr = 0; xtraLen = 0;

-	smplPtr = 0; smplLen = 0;

-	if (editor.tmpFilenameU == NULL)

+	if (s->origPek != NULL)

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto wavLoadError;

+		free(s->origPek);

+		s->origPek = NULL;

-	UNICHAR *filename = editor.tmpFilenameU;

+	s->pek = NULL;

+}

-	f = UNICHAR_FOPEN(filename, "rb");

-	if (f == NULL)

-	{

-		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-		goto wavLoadError;

-	}

-	// get and check filesize

-	fseek(f, 0, SEEK_END);

-	filesize = ftell(f);

-	if (filesize < 12)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto wavLoadError;

-	}

-	// look for wanted chunks and set up pointers + lengths

-	fseek(f, 12, SEEK_SET);

-	bytesRead = 0;

-	while (!feof(f) && bytesRead < filesize-12)

-	{

-		fread(&chunkID, 4, 1, f); if (feof(f)) break;

-		fread(&chunkSize, 4, 1, f); if (feof(f)) break;

-		endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

-		switch (chunkID)

-		{

-			case 0x20746D66: // "fmt "

-			{

-				fmtPtr = ftell(f);

-				fmtLen = chunkSize;

-			}

-			break;

-			case 0x61746164: // "data"

-			{

-				dataPtr = ftell(f);

-				dataLen = chunkSize;

-			}

-			break;

-			case 0x5453494C: // "LIST"

-			{

-				if (chunkSize >= 4)

-				{

-					fread(&chunkID, 4, 1, f);

-					if (chunkID == 0x4F464E49) // "INFO"

-					{

-						bytesRead = 0;

-						while (!feof(f) && bytesRead < chunkSize)

-						{

-							fread(&chunkID, 4, 1, f);

-							fread(&chunkSize, 4, 1, f);

-							switch (chunkID)

-							{

-								case 0x4D414E49: // "INAM"

-								{

-									inamPtr = ftell(f);

-									inamLen = chunkSize;

-								}

-								break;

-								default: break;

-							}

-							bytesRead += (chunkSize + (chunkSize & 1));

-						}

-					}

-				}

-			}

-			break;

-			case 0x61727478: // "xtra"

-			{

-				xtraPtr = ftell(f);

-				xtraLen = chunkSize;

-			}

-			break;

-			case 0x6C706D73: // "smpl"

-			{

-				smplPtr = ftell(f);

-				smplLen = chunkSize;

-			}

-			break;

-			default: break;

-		}

-		bytesRead += (chunkSize + (chunkSize & 1));

-		fseek(f, endOfChunk, SEEK_SET);

-	}

-	// we need at least "fmt " and "data" - check if we found them sanely

-	if ((fmtPtr == 0 || fmtLen < 16) || (dataPtr == 0 || dataLen == 0))

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto wavLoadError;

-	}

-	// ---- READ "fmt " CHUNK ----

-	fseek(f, fmtPtr, SEEK_SET);

-	fread(&audioFormat, 2, 1, f);

-	fread(&numChannels, 2, 1, f);

-	fread(&sampleRate,  4, 1, f);

-	fseek(f, 6, SEEK_CUR); // unneeded

-	fread(&bitsPerSample, 2, 1, f);

-	sampleLength = dataLen;

-	// ---------------------------

-	// test if the WAV is compatible with our loader

-	if (sampleRate == 0 || sampleLength == 0 || sampleLength >= filesize)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported or is invalid!");

-		goto wavLoadError;

-	}

-	if (audioFormat != WAV_FORMAT_PCM && audioFormat != WAV_FORMAT_IEEE_FLOAT)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: The sample is not supported!");

-		goto wavLoadError;

-	}

-	if (numChannels == 0 || numChannels > 2)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: Unsupported number of channels!");

-		goto wavLoadError;

-	}

-	if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample != 32 && bitsPerSample != 64)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: Unsupported bitdepth!");

-		goto wavLoadError;

-	}

-	if (bitsPerSample != 8 && bitsPerSample != 16 && bitsPerSample != 24 && bitsPerSample != 32 && bitsPerSample != 64)

-	{

-		okBoxThreadSafe(0, "System message", "Error loading sample: Unsupported bitdepth!");

-		goto wavLoadError;

-	}

-	// ---- READ SAMPLE DATA ----

-	fseek(f, dataPtr, SEEK_SET);

-	s = &instr[editor.curInstr]->samp[editor.curSmp];

-	tmpSmp.pan = 128;

-	tmpSmp.vol = 64;

-	if (sampleLength > MAX_SAMPLE_LEN)

-		sampleLength = MAX_SAMPLE_LEN;

-	if (bitsPerSample == 8) // 8-BIT INTEGER SAMPLE

-	{

-		tmpSmp.origPek = (int8_t *)malloc(sampleLength + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		audioDataU8 = (uint8_t *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						audioDataU8[i] = audioDataU8[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataU8[i] = audioDataU8[(i * 2) + 1];

-					audioDataU8[i] = 128;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataU8[i] = (audioDataU8[(i * 2) + 0] + audioDataU8[(i * 2) + 1]) >> 1;

-					audioDataU8[i] = 128;

-				}

-				break;

-			}

-		}

-		// convert from unsigned to signed

-		for (i = 0; i < sampleLength; i++)

-			tmpSmp.pek[i] ^= 0x80;

-		tmpSmp.typ &= ~16; // 8-bit

-	}

-	else if (bitsPerSample == 16) // 16-BIT INTEGER SAMPLE

-	{

-		sampleLength /= 2;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 2) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 2, f) != 2)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						audioDataS16[i] = audioDataS16[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS16[i] = audioDataS16[(i * 2) + 1];

-					audioDataS16[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp32 = audioDataS16[(i * 2) + 0] + audioDataS16[(i * 2) + 1];

-						audioDataS16[i] = (int16_t)(smp32 >> 1);

-					}

-					audioDataS16[i] = 0;

-				}

-				break;

-			}

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16; // 16-bit

-	}

-	else if (bitsPerSample == 24) // 24-BIT INTEGER SAMPLE

-	{

-		sampleLength /= 3;

-		tmpSmp.origPek = (int8_t *)malloc(((sampleLength * 4) * 2) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(&tmpSmp.pek[sampleLength], sampleLength, 3, f) != 3)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		audioDataS32 = (int32_t *)tmpSmp.pek;

-		// convert to 32-bit

-		audioDataU8 = (uint8_t *)tmpSmp.pek + sampleLength;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS32[i] = (audioDataU8[2] << 24) | (audioDataU8[1] << 16) | (audioDataU8[0] << 8);

-			audioDataU8 += 3;

-		}

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 1];

-					audioDataS32[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp64 = audioDataS32[(i * 2) + 0];

-						smp64 += audioDataS32[(i * 2) + 1];

-						smp64 >>= 1;

-						audioDataS32[i] = (int32_t)smp64;

-					}

-					audioDataS32[i] = 0;

-				}

-				break;

-			}

-		}

-		normalize32bitSigned(audioDataS32, sampleLength);

-		// downscale to 16-bit (ultra fast method!)

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS16[i] = audioDataS16_2[i];

-			audioDataS16_2++;

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16; // 16-bit

-	}

-	else if (audioFormat == WAV_FORMAT_PCM && bitsPerSample == 32) // 32-BIT INTEGER SAMPLE

-	{

-		sampleLength /= 4;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 4) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		audioDataS32 = (int32_t *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						audioDataS32[i] = audioDataS32[(i * 2) + 1];

-					audioDataS32[i] = 0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-					{

-						smp64 = audioDataS32[(i * 2) + 0];

-						smp64 += audioDataS32[(i * 2) + 1];

-						smp64 >>= 1;

-						audioDataS32[i] = (int32_t)smp64;

-					}

-					audioDataS32[i] = 0;

-				}

-				break;

-			}

-		}

-		normalize32bitSigned(audioDataS32, sampleLength);

-		// downscale to 16-bit (ultra fast method!)

-		audioDataS16 = (int16_t *)tmpSmp.pek;

-		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1;

-		for (i = 0; i < sampleLength; i++)

-		{

-			audioDataS16[i] = audioDataS16_2[i];

-			audioDataS16_2++;

-		}

-		sampleLength *= 2;

-		tmpSmp.typ |= 16; // 16-bit

-	}

-	else if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample == 32) // 32-BIT FLOATING POINT SAMPLE

-	{

-		sampleLength /= 4;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 4) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		fAudioDataFloat = (float *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						fAudioDataFloat[i] = fAudioDataFloat[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						fAudioDataFloat[i] = fAudioDataFloat[(i * 2) + 1];

-					fAudioDataFloat[i] = 0.0f;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						fAudioDataFloat[i] = (fAudioDataFloat[(i * 2) + 0] + fAudioDataFloat[(i * 2) + 1]) * 0.5f;

-					fAudioDataFloat[i] = 0.0f;

-				}

-				break;

-			}

-		}

-		normalize16bitFloatSigned(fAudioDataFloat, sampleLength);

-		ptr16 = (int16_t *)tmpSmp.pek;

-		for (i = 0; i < sampleLength; i++)

-			ptr16[i] = (int16_t)fAudioDataFloat[i]; // should use SIMD if available

-		sampleLength *= 2;

-		tmpSmp.typ |= 16; // 16-bit

-	}

-	else if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample == 64) // 64-BIT FLOATING POINT SAMPLE

-	{

-		sampleLength /= 8;

-		tmpSmp.origPek = (int8_t *)malloc((sampleLength * 8) + LOOP_FIX_LEN);

-		if (tmpSmp.origPek == NULL)

-		{

-			okBoxThreadSafe(0, "System message", "Not enough memory!");

-			goto wavLoadError;

-		}

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-		if (fread(tmpSmp.pek, sampleLength, 8, f) != 8)

-		{

-			okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

-			goto wavLoadError;

-		}

-		dAudioDataDouble = (double *)tmpSmp.pek;

-		// stereo conversion

-		if (numChannels == 2)

-		{

-			sampleLength /= 2;

-			switch (stereoSampleLoadMode)

-			{

-				case STEREO_SAMPLE_READ_LEFT:

-				{

-					// remove right channel data

-					for (i = 1; i < sampleLength; i++)

-						dAudioDataDouble[i] = dAudioDataDouble[(i * 2) + 0];

-				}

-				break;

-				case STEREO_SAMPLE_READ_RIGHT:

-				{

-					// remove left channel data

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						dAudioDataDouble[i] = dAudioDataDouble[(i * 2) + 1];

-					dAudioDataDouble[i] = 0.0;

-				}

-				break;

-				default:

-				case STEREO_SAMPLE_CONVERT:

-				{

-					// mix stereo to mono

-					len32 = sampleLength - 1;

-					for (i = 0; i < len32; i++)

-						dAudioDataDouble[i] = (dAudioDataDouble[(i * 2) + 0] + dAudioDataDouble[(i * 2) + 1]) * 0.5;

-					dAudioDataDouble[i] = 0.0;

-				}

-				break;

-			}

-		}

-		normalize64bitDoubleSigned(dAudioDataDouble, sampleLength);

-		ptr16 = (int16_t *)tmpSmp.pek;

-		for (i = 0; i < sampleLength; i++)

-			ptr16[i] = (int16_t)dAudioDataDouble[i]; // should use SIMD if available

-		sampleLength *= 2;

-		tmpSmp.typ |= 16; // 16-bit

-	}

-	// adjust memory needed

-	newPtr = (int8_t *)realloc(tmpSmp.origPek, sampleLength + LOOP_FIX_LEN);

-	if (newPtr != NULL)

-	{

-		tmpSmp.origPek = newPtr;

-		tmpSmp.pek = tmpSmp.origPek + SMP_DAT_OFFSET;

-	}

-	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

-	tmpSmp.vol = 64;

-	tmpSmp.pan = 128;

-	tmpSmp.len = sampleLength;

-	// ---- READ "smpl" chunk ----

-	if (smplPtr != 0 && smplLen > 52)

-	{

-		fseek(f, smplPtr + 28, SEEK_SET); // seek to first wanted byte

-		fread(&numLoops, 4, 1, f);

-		if (numLoops == 1)

-		{

-			fseek(f, 4 + 4, SEEK_CUR); // skip "samplerData" and "identifier"

-			fread(&loopType, 4, 1, f);

-			fread(&loopStart, 4, 1, f);

-			fread(&loopEnd, 4, 1, f);

-			loopEnd++;

-			if (tmpSmp.typ & 16)

-			{

-				loopStart *= 2;

-				loopEnd *= 2;

-			}

-			if (loopEnd <= sampleLength)

-			{

-				tmpSmp.repS = loopStart;

-				tmpSmp.repL = loopEnd - loopStart;

-				tmpSmp.typ |= ((loopType == 0) ? 1 : 2);

-			}

-		}

-	}

-	// ---------------------------

-	// ---- READ "xtra" chunk ----

-	if (xtraPtr != 0 && xtraLen >= 8)

-	{

-		fseek(f, xtraPtr, SEEK_SET);

-		fread(&xtraFlags, 4, 1, f); // flags

-		// panning (0..256)

-		if (xtraFlags & 0x20) // set panning flag

-		{

-			fread(&tempPan, 2, 1, f);

-			if (tempPan > 255)

-				tempPan = 255;

-			tmpSmp.pan = (uint8_t)tempPan;

-		}

-		else

-		{

-			// don't read panning, skip it

-			fseek(f, 2, SEEK_CUR);

-		}

-		// volume (0..256)

-		fread(&tempVol, 2, 1, f);

-		if (tempVol > 256)

-			tempVol = 256;

-		tmpSmp.vol = (uint8_t)(tempVol / 4); // 0..256 -> 0..64

-	}

-	// ---------------------------

-	// ---- READ "INAM" chunk ----

-	if (inamPtr != 0 && inamLen > 0)

-	{

-		fseek(f, inamPtr, SEEK_SET);

-		memset(tmpSmp.name, 0, sizeof (tmpSmp.name));

-		for (i = 0; i < 22; i++)

-		{

-			if (i < inamLen)

-			{

-				chr = (char)fgetc(f);

-				if (chr == '\0')

-					break;

-				tmpSmp.name[i] = chr;

-			}

-		}

-	}

-	else

-	{

-		// set sample name from filename if we didn't load name from .wav

-		tmpFilename = unicharToCp437(filename, true);

-		if (tmpFilename != NULL)

-		{

-			j = (int32_t)strlen(tmpFilename);

-			while (j--)

-			{

-				if (tmpFilename[j] == DIR_DELIMITER)

-					break;

-			}

-			tmpPtr = tmpFilename;

-			if (j > 0)

-				tmpPtr += j + 1;

-			sanitizeFilename(tmpPtr);

-			filenameLen = (uint32_t)strlen(tmpPtr);

-			for (i = 0; i < 22; i++)

-			{

-				if (i < filenameLen)

-					tmpSmp.name[i] = tmpPtr[i];

-				else

-					tmpSmp.name[i] = '\0';

-			}

-			free(tmpFilename);

-		}

-	}

-	fclose(f);

-	lockMixerCallback();

-	if (loadAsInstrFlag) // if loaded in instrument mode

-	{

-		freeInstr(editor.curInstr);

-		memset(song.instrName[editor.curInstr], 0, 23);

-	}

-	if (instr[editor.curInstr] == NULL)

-		allocateInstr(editor.curInstr);

-	if (instr[editor.curInstr] != NULL)

-	{

-		s = &instr[editor.curInstr]->samp[sampleSlot];

-		freeSample(editor.curInstr, sampleSlot);

-		memcpy(s, &tmpSmp, sizeof (sampleTyp));

-		fixSample(s);

-	}

-	else

-	{

-		okBoxThreadSafe(0, "System message", "Not enough memory!");

-		goto wavLoadError;

-	}

-	unlockMixerCallback();

-	fixSampleName(editor.curInstr);

-	setSongModifiedFlag();

-	stereoSampleLoadMode = -1;

-	// also sets mouse busy to false when done

-	editor.updateCurSmp = true;

-	return true;

-wavLoadError:

-	if (f != NULL) fclose(f);

-	if (tmpSmp.origPek != NULL) free(tmpSmp.origPek);

-	stereoSampleLoadMode = -1;

+void removeSampleIsLoadingFlag(void)

+{

 	sampleIsLoading = false;

-	return false;

-	(void)ptr;

 bool loadSample(UNICHAR *filenameU, uint8_t smpNr, bool instrFlag)

-	char tmpBuffer[16+1];

-	if (sampleIsLoading)

+	if (sampleIsLoading || filenameU == NULL)

 		return false;

-	stereoSampleLoadMode = 0;

+	// setup message box functions

+	loaderMsgBox = myLoaderMsgBoxThreadSafe;

+	loaderSysReq = okBoxThreadSafe;

-	sampleSlot = smpNr;

-	loadAsInstrFlag = instrFlag;

 	if (editor.curInstr == 0)

-		okBox(0, "System message", "The zero-instrument cannot hold intrument data.");

+		loaderMsgBox("The zero-instrument cannot hold intrument data.");

 		return false;

-	FILE *f = UNICHAR_FOPEN(filenameU, "rb");

-	if (f == NULL)

-	{

-		okBox(0, "System message", "General I/O error during loading! Is the file in use?");

-		return false;

-	}

-	memset(tmpBuffer, 0, sizeof (tmpBuffer));

-	if (fread(tmpBuffer, sizeof (tmpBuffer) - 1, 1, f) == 1)

-	{

-		tmpBuffer[sizeof (tmpBuffer) - 1] = '\0';

-		// detect what sample format this is...

-		// WAV

-		if (!strncmp("RIFF", tmpBuffer, 4) && !strncmp("WAVE", tmpBuffer + 8, 4))

-		{

-			// let the user pick what to do with stereo samples...

-			if (wavIsStereo(f))

-				stereoSampleLoadMode = okBox(5, "System request", "This is a stereo sample...");

-			sampleIsLoading = true;

-			fclose(f);

-			UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

-			mouseAnimOn();

-			thread = SDL_CreateThread(loadWAVSample, NULL, NULL);

-			if (thread == NULL)

-			{

-				okBox(0, "System message", "Couldn't create thread!");

-				sampleIsLoading = false;

-				return false;

-			}

-			SDL_DetachThread(thread);

-			return true;

-		}

-		// AIFF or IFF

-		if (!strncmp("FORM", tmpBuffer, 4))

-		{

-			if (!strncmp("AIFC", tmpBuffer + 8, 4))

-			{

-				// AIFC (not supported)

-				fclose(f);

-				okBox(0, "System message", "Error loading sample: This AIFF type (AIFC) is not supported!");

-				return true;

-			}

-			else if (!strncmp("AIFF", tmpBuffer + 8, 4))

-			{

-				// AIFF

-				// let the user pick what to do with stereo samples...

-				if (aiffIsStereo(f))

-					stereoSampleLoadMode = okBox(5, "System request", "This is a stereo sample...");

-				sampleIsLoading = true;

-				fclose(f);

-				UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

-				mouseAnimOn();

-				thread = SDL_CreateThread(loadAIFFSample, NULL, NULL);

-				if (thread == NULL)

-				{

-					okBox(0, "System message", "Couldn't create thread!");

-					sampleIsLoading = false;

-					return false;

-				}

-				SDL_DetachThread(thread);

-				return true;

-			}

-			else if (!strncmp("8SVX", tmpBuffer + 8, 4) || !strncmp("16SV", tmpBuffer + 8, 4))

-			{

-				// IFF

-				sampleIsLoading = true;

-				fclose(f);

-				UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

-				mouseAnimOn();

-				thread = SDL_CreateThread(loadIFFSample, NULL, NULL);

-				if (thread == NULL)

-				{

-					okBox(0, "System message", "Couldn't create thread!");

-					sampleIsLoading = false;

-					return false;

-				}

-				SDL_DetachThread(thread);

-				return true;

-			}

-		}

-	}

-	// load as RAW sample

+	sampleSlot = smpNr;

+	loadAsInstrFlag = instrFlag;

 	sampleIsLoading = true;

+	smpFilenameSet = false;

-	fclose(f);

+	memset(&tmpSmp, 0, sizeof (tmpSmp));

 	UNICHAR_STRCPY(editor.tmpFilenameU, filenameU);

 	mouseAnimOn();

-	thread = SDL_CreateThread(loadRawSample, NULL, NULL);

+	thread = SDL_CreateThread(loadSampleThread, NULL, NULL);

 	if (thread == NULL)

-		okBox(0, "System message", "Couldn't create thread!");

 		sampleIsLoading = false;

+		loaderMsgBox("Couldn't create thread!");

 		return false;

@@ -2092,7 +241,7 @@

 	return true;

-static void normalize32bitSigned(int32_t *sampleData, uint32_t sampleLength)

+void normalizeSigned32Bit(int32_t *sampleData, uint32_t sampleLength)

 	uint32_t i;

@@ -2112,7 +261,7 @@

 		sampleData[i] = (int32_t)(sampleData[i] * dGain);

-static void normalize16bitFloatSigned(float *fSampleData, uint32_t sampleLength)

+void normalize32BitFloatToSigned16Bit(float *fSampleData, uint32_t sampleLength)

 	uint32_t i;

@@ -2132,7 +281,7 @@

 		fSampleData[i] *= fGain;

-static void normalize64bitDoubleSigned(double *dSampleData, uint32_t sampleLength)

+void normalize64BitFloatToSigned16Bit(double *dSampleData, uint32_t sampleLength)

 	uint32_t i;

@@ -2150,99 +299,4 @@

 	const double dGain = (double)INT16_MAX / dSampleVolPeak;

 	for (i = 0; i < sampleLength; i++)

 		dSampleData[i] *= dGain;

-}

-bool fileIsInstrument(char *fullPath)

-{

-	int32_t i, len;

-	// this assumes that fullPath is not empty

-	len = (int32_t)strlen(fullPath);

-	// get filename from path

-	i = len;

-	while (i--)

-	{

-		if (fullPath[i] == DIR_DELIMITER)

-			break;

-	}

-	char *filename = fullPath;

-	if (i > 0)

-		filename += i + 1;

-	// --------------------------

-	len = (int32_t)strlen(filename);

-	if (len < 4)

-		return true; // can't be an instrument

-	if (!_strnicmp("xi.", filename, 3) || (len >= 4 && !_strnicmp("pat.", filename, 4)))

-		return true;

-	int32_t extOffset = getExtOffset(filename, len);

-	if (extOffset != -1)

-	{

-		if ((extOffset <= len-4) && !_strnicmp(".pat", &filename[extOffset], 4)) return true;

-		if ((extOffset <= len-3) && !_strnicmp(".xi",  &filename[extOffset], 3)) return true;

-	}

-	return false;

-}

-bool fileIsSample(char *fullPath)

-{

-	int32_t i, len;

-	// this assumes that fullPath is not empty

-	len = (int32_t)strlen(fullPath);

-	// get filename from path

-	i = len;

-	while (i--)

-	{

-		if (fullPath[i] == DIR_DELIMITER)

-			break;

-	}

-	char *filename = fullPath;

-	if (i > 0)

-		filename += i + 1;

-	// --------------------------

-	len = (int32_t)strlen(filename);

-	if (len < 4)

-		return true; // can't be a module

-	if (!_strnicmp("xm.",  filename, 3) || !_strnicmp("ft.",  filename, 3) ||

-		!_strnicmp("mod.", filename, 4) || !_strnicmp("nst.", filename, 4) ||

-		!_strnicmp("s3m.", filename, 4) || !_strnicmp("stm.", filename, 4) ||

-		!_strnicmp("fst.", filename, 4) || !_strnicmp("it.",  filename, 3))

-	{

-		return false; // definitely a module

-	}

-	int32_t extOffset = getExtOffset(filename, len);

-	if (extOffset != -1)

-	{

-		if (extOffset <= len-4)

-		{

-			filename = &filename[extOffset];

-			if (!_strnicmp(".mod", filename, 4) || !_strnicmp(".nst", filename, 4) ||

-				!_strnicmp(".s3m", filename, 4) || !_strnicmp(".stm", filename, 4) ||

-				!_strnicmp(".fst", filename, 4))

-			{

-				return false; // definitely a module

-			}

-		}

-		else if (extOffset <= len-3)

-		{

-			filename = &filename[extOffset];

-			if (!_strnicmp(".xm", filename, 3) || !_strnicmp(".ft", filename, 3) || !_strnicmp(".it", filename, 3))

-				return false; // definitely a module

-		}

-	}

-	return true; // let's assume it's a sample

--- a/src/ft2_sample_loader.h

+++ b/src/ft2_sample_loader.h

@@ -2,9 +2,29 @@

 #include <stdint.h>

 #include <stdbool.h>

+#include "ft2_header.h"

 #include "ft2_unicode.h"

+enum

+{

+	STEREO_SAMPLE_READ_LEFT = 1,

+	STEREO_SAMPLE_READ_RIGHT = 2,

+	STEREO_SAMPLE_CONVERT = 3,

+};

+void normalizeSigned32Bit(int32_t *sampleData, uint32_t sampleLength);

+void normalize32BitFloatToSigned16Bit(float *fSampleData, uint32_t sampleLength);

+void normalize64BitFloatToSigned16Bit(double *dSampleData, uint32_t sampleLength);

 bool loadSample(UNICHAR *filenameU, uint8_t sampleSlot, bool loadAsInstrFlag);

-bool fileIsInstrument(char *fullPath);

-bool fileIsSample(char *fullPath);

 void removeSampleIsLoadingFlag(void);

+// globals for sample loaders

+extern bool loadAsInstrFlag, smpFilenameSet;

+extern char *smpFilename;

+extern uint8_t sampleSlot;

+extern sampleTyp tmpSmp;

+// --------------------------

+// file extensions accepted by Disk Op. in sample mode

+extern char *supportedSmpExtensions[];

--- a/src/ft2_scopes.c

+++ b/src/ft2_scopes.c

@@ -23,8 +23,6 @@

 #include "ft2_tables.h"

 #include "ft2_structs.h"

-#define SCOPE_HEIGHT 36

 static volatile bool scopesUpdatingFlag, scopesDisplayingFlag;

 static uint32_t scopeTimeLen, scopeTimeLenFrac;

 static uint64_t timeNext64, timeNext64Frac;

@@ -38,9 +36,9 @@

 	volatile scope_t *sc = scope;

 	for (int32_t i = 0; i < MAX_VOICES; i++, sc++)

-		sc->oldPeriod = -1;

-		sc->oldDrawDelta = 0;

+		sc->dOldHz = -1.0;

 		sc->oldDelta = 0;

+		sc->oldDrawDelta = 0;

@@ -353,10 +351,13 @@

 		return;

-	// these has to be read

+	// these has to be copied so that they are not lost

 	tempState.wasCleared = sc->wasCleared;

 	tempState.delta = sc->delta;

+	tempState.oldDelta = sc->oldDelta;

 	tempState.drawDelta = sc->drawDelta;

+	tempState.oldDrawDelta = sc->oldDrawDelta;

+	tempState.dOldHz = sc->dOldHz;

 	tempState.vol = sc->vol;

 	tempState.active = true;

@@ -367,6 +368,7 @@

 	** the trigger never happens. So far I have never seen it happen,

 	** so it's probably very rare. Yes, this is not good coding...

*/

 	*sc = tempState;

@@ -386,8 +388,8 @@

 		// scope position update

 		s.posFrac += s.delta;

-		const int32_t wholeSamples = (int32_t)(s.posFrac >> SCOPE_FRAC_BITS);

-		s.posFrac &= SCOPE_FRAC_MASK;

+		const int32_t wholeSamples = s.posFrac >> 32;

+		s.posFrac &= 0xFFFFFFFF;

 		if (s.direction == 1)

 			s.pos += wholeSamples; // forwards

@@ -522,23 +524,20 @@

 		const uint8_t status = scopeUpdateStatus[i];

 		if (status & IS_Vol)

-			sc->vol = (int32_t)((ch->dFinalVol * SCOPE_HEIGHT) + 0.5); // rounded

+			sc->vol = ((ch->vol * SCOPE_HEIGHT) + 128) >> 8; // rounded

 		if (status & IS_Period)

-			// use cached values if possible

-			const uint16_t period = ch->finalPeriod;

-			if (period != sc->oldPeriod)

+			// use cached values when possible

+			if (ch->dHz != sc->dOldHz)

-				sc->oldPeriod = period;

-				const double dHz = dPeriod2Hz(period);

+				sc->dOldHz = ch->dHz;

-				const double dScopeRateFactor = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;

-				sc->oldDelta = (int64_t)((dHz * dScopeRateFactor) + 0.5); // Hz -> rounded fixed-point delta

+				const double dHz2ScopeDeltaMul = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;

+				sc->oldDelta = (int64_t)((ch->dHz * dHz2ScopeDeltaMul) + 0.5); // Hz -> 32.32fp delta (rounded)

-				const double dRelativeHz = dHz * (1.0 / (8363.0 / 2.0));

-				sc->oldDrawDelta = (int32_t)((dRelativeHz * SCOPE_DRAW_FRAC_SCALE) + 0.5); // Hz -> rounded fixed-point draw delta

+				const double dRelativeHz = ch->dHz * (1.0 / (8363.0 / 2.0));

+				sc->oldDrawDelta = (int32_t)((dRelativeHz * SCOPE_DRAW_FRAC_SCALE) + 0.5); // Hz -> 13.19fp draw delta (rounded)

 			sc->delta = sc->oldDelta;

--- a/src/ft2_scopes.h

+++ b/src/ft2_scopes.h

@@ -5,6 +5,8 @@

 #include "ft2_header.h"

 #include "ft2_audio.h"

+#define SCOPE_HEIGHT 36

 #define SCOPE_FRAC_BITS 32

 #define SCOPE_FRAC_SCALE (1ULL << SCOPE_FRAC_BITS)

 #define SCOPE_FRAC_MASK (SCOPE_FRAC_SCALE-1)

@@ -31,9 +33,11 @@

 	const int16_t *base16;

 	bool wasCleared, sampleIs16Bit;

 	uint8_t loopType;

-	int32_t vol, loopStart, loopLength, end, pos, direction, oldPeriod;

+	int32_t vol, loopStart, loopLength, end, pos, direction;

 	uint32_t drawDelta, oldDrawDelta;

-	uint64_t delta, posFrac, oldDelta;

+	uint64_t delta, oldDelta, posFrac;

+	double dOldHz;

 } scope_t;

 typedef struct lastChInstr_t

--- a/src/ft2_sysreqs.c

+++ b/src/ft2_sysreqs.c

@@ -1,3 +1,4 @@

+#include <stdio.h> // vsnprintf()

 #include <stdint.h>

 #include <stdbool.h>

 #include "ft2_config.h"

@@ -16,7 +17,11 @@

 #define NUM_SYSREQ_TYPES 10

-okBoxData_t okBoxData; // globalized

+// globalized

+okBoxData_t okBoxData;

+void (*loaderMsgBox)(const char *, ...);

+int16_t (*loaderSysReq)(int16_t, const char *, const char *);

+// ----------------

 static char *buttonText[NUM_SYSREQ_TYPES][5] =

@@ -69,6 +74,32 @@

 	{ "Are we at the end of a Fasttracker round?", 2 },

 	{ "Hope you're doing the compulsory \"Exit ceremony\" before pressing >OK<.", 1 },

};

+void myLoaderMsgBoxThreadSafe(const char *fmt, ...)

+{

+	char strBuf[512];

+	va_list args;

+	// format the text string

+	va_start(args, fmt);

+	vsnprintf(strBuf, sizeof (strBuf), fmt, args);

+	va_end(args);

+	okBoxThreadSafe(0, "System message", fmt);

+}

+void myLoaderMsgBox(const char *fmt, ...)

+{

+	char strBuf[512];

+	va_list args;

+	// format the text string

+	va_start(args, fmt);

+	vsnprintf(strBuf, sizeof (strBuf), fmt, args);

+	va_end(args);

+	okBox(0, "System message", fmt);

+}

 static void drawWindow(uint16_t w)

--- a/src/ft2_sysreqs.h

+++ b/src/ft2_sysreqs.h

@@ -3,11 +3,11 @@

 #include <stdint.h>

 #include <stdbool.h>

-int16_t okBoxThreadSafe(int16_t typ, const char *headline, const char *text);

-int16_t okBox(int16_t typ, const char *headline, const char *text);

-int16_t quitBox(bool skipQuitMsg);

-int16_t inputBox(int16_t typ, const char *headline, char *edText, uint16_t maxStrLen);

-bool askUnsavedChanges(uint8_t type);

+enum

+{

+	ASK_TYPE_QUIT = 0,

+	ASK_TYPE_LOAD_SONG = 1,

+};

 // for thread-safe version of okBox()

 typedef struct okBoxData_t

@@ -17,10 +17,17 @@

 	const char *headline, *text;

 } okBoxData_t;

-extern okBoxData_t okBoxData; // ft2_sysreqs.c

+int16_t okBoxThreadSafe(int16_t typ, const char *headline, const char *text);

+int16_t okBox(int16_t typ, const char *headline, const char *text);

+int16_t quitBox(bool skipQuitMsg);

+int16_t inputBox(int16_t typ, const char *headline, char *edText, uint16_t maxStrLen);

+bool askUnsavedChanges(uint8_t type);

-enum

-{

-	ASK_TYPE_QUIT = 0,

-	ASK_TYPE_LOAD_SONG = 1,

-};

+void myLoaderMsgBoxThreadSafe(const char *fmt, ...);

+void myLoaderMsgBox(const char *fmt, ...);

+ // ft2_sysreqs.c

+extern okBoxData_t okBoxData;

+extern void (*loaderMsgBox)(const char *, ...);

+extern int16_t (*loaderSysReq)(int16_t, const char *, const char *);

+// ---------------

--- a/src/ft2_tables.c

+++ b/src/ft2_tables.c

@@ -10,6 +10,13 @@

 /*                             REPLAYER TABLES                             */

 /* ----------------------------------------------------------------------- */

+const uint16_t ptPeriods[3 * 12] =

+{

+	856,808,762,720,678,640,604,570,538,508,480,453,

+	428,404,381,360,339,320,302,285,269,254,240,226,

+	214,202,190,180,170,160,151,143,135,127,120,113

+};

 const uint8_t arpTab[256] =

 	0,1,2,0,1,2,0,1,2,0,1,2,0,1,2,0,

@@ -60,7 +67,7 @@

 	255,253,250,244,235,224,212,197,180,161,141,120, 97, 74, 49, 24

};

-const uint16_t amigaPeriod[12 * 8] = // used for .MOD loading/saving

+const uint16_t amigaPeriod[8 * 12] = // used for .MOD loading/saving

 	6848, 6464, 6096, 5760, 5424, 5120, 4832, 4560, 4304, 4064, 3840, 3624,

 	3424, 3232, 3048, 2880, 2712, 2560, 2416, 2280, 2152, 2032, 1920, 1812,

--- a/src/ft2_tables.h

+++ b/src/ft2_tables.h

@@ -10,10 +10,12 @@

 #define KEY2EFX_ENTRIES (signed)(sizeof (key2EfxTab) / sizeof (SDL_Keycode))

 #define KEY2HEX_ENTRIES (signed)(sizeof (key2HexTab) / sizeof (SDL_Keycode))

+extern const uint16_t ptPeriods[3 * 12];

 extern const uint8_t arpTab[256];

 extern const int8_t vibSineTab[256]; // for auto-vibrato

 extern const uint8_t vibTab[32];

-extern const uint16_t amigaPeriod[12 * 8];

+extern const uint16_t amigaPeriod[8 * 12];

 extern const uint16_t linearPeriods[1936];

 extern const uint16_t amigaPeriods[1936];

--- a/src/helpdata/FT2.HLP

+++ b/src/helpdata/FT2.HLP

@@ -20,21 +20,27 @@

 >- Full screen edit mode.

 >- Improved editing facilities.

-@X040@C001FT2 supports the following file formats:

+@X040@C001The FT2 clone supports the following file formats:

 @X040@C001Modules:

 >@X060@C002

->- Standard modules (15- > 31-instruments). (MOD,NST)

->- Fasttracker v1.0 2,4,6,8..32-channel formats. (MOD)

->- ScreamTracker modules. (STM,S3M)

+>- Standard modules (15/31 samples). (MOD,NST,STK)

+>- Fasttracker II modules. (XM,MOD)

+>- Fasttracker modules. (MOD/FST)

+@X040@C001Experimental module support:

+>@X060@C002

+>- Scream Tracker 3 modules. (S3M)

+>- Scream Tracker 2 modules. (STM)

+>- DIGI Booster (non-Pro) modules. (DIGI)

+>

 @X040@C001Samples:

 >@X060@C002

 >- Gravis Ultrasound Patches, PAT.

 >- SMP/SAM/RAW/SND data files, signed and unsigned.

->- Windows WAV-files.

->- Amiga IFF-files. (Interchange File Format)

->- Apple AIFF-files.

+>- WAV files.

+>- Amiga IFF.

+>- Apple AIFF.

 @X040@C001FT2 introduces several new file formats:

 >@X060@C002

--- a/src/helpdata/ft2_help_data.h

+++ b/src/helpdata/ft2_help_data.h

@@ -3,9 +3,9 @@

 #include <stdint.h>

-#define HELP_DATA_LEN 27371

+#define HELP_DATA_LEN 27476

-const uint8_t helpData[27371] =

+const uint8_t helpData[27476] =

 	0x4C,0x3B,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,

 	0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,

@@ -63,2231 +63,2240 @@

 	0x64,0x65,0x2E,0x1F,0x3E,0x2D,0x20,0x49,0x6D,0x70,0x72,0x6F,

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+	0x61,0x6D,0x2E,0x00,0x03,0x45,0x4E,0x44

};

 #endif

--- a/src/mixer/ft2_mix_macros.h

+++ b/src/mixer/ft2_mix_macros.h

@@ -253,18 +253,9 @@

 #define LIMIT_MIX_NUM \

 	i = (v->end - 1) - pos; \

-	if (i > 65535) \

-		i = 65535; \

+	const uint64_t tmp64 = ((uint64_t)i << 32) | ((uint32_t)posFrac ^ 0xFFFFFFFF); \

-	i = (i << 16) | ((uint32_t)(posFrac >> 16) ^ 0xFFFF); \

-	\

-	/* This is hackish, but fast. This is sometimes off by one (-1), so */ \

-	/* we need to do another cycle to reach the end of the sample. The */ \

-	/* error is never +1, it's always below (safe). */ \

-	\

-	samplesToMix = ((int64_t)i * v->revDelta) >> 32; \

-	samplesToMix++; \

-	\

+	samplesToMix = (uint32_t)(tmp64 / (uint64_t)v->delta) + 1; /* this can be slow on 32-bit systems... */ \

 	if (samplesToMix > samplesLeft) \

 		samplesToMix = samplesLeft; \

--- /dev/null

+++ b/src/modloaders/ft2_load_digi.c

@@ -1,0 +1,260 @@

+// DIGI Booster (non-Pro) module loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+#ifdef _MSC_VER  // please don't mess with this struct!

+#pragma pack(push)

+#pragma pack(1)

+#endif

+typedef struct digiHeaderTyp_t

+{

+	char sig[20];

+	char verStr[4];

+	uint8_t ver;

+	uint8_t numChannels;

+	uint8_t packedPatternsFlag;

+	char reserved[19];

+	uint8_t numPats;

+	uint8_t numOrders;

+	uint8_t orders[128];

+	uint32_t smpLength[31];

+	uint32_t smpLoopStart[31];

+	uint32_t smpLoopLength[31];

+	uint8_t smpVolume[31];

+	uint8_t smpFinetune[31];

+	char name[32];

+	char smpName[31][30];

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+digiHeaderTyp;

+#ifdef _MSC_VER

+#pragma pack(pop)

+#endif

+static void readPatternNote(FILE *f, tonTyp *ton);

+bool loadDIGI(FILE *f, uint32_t filesize)

+{

+	int16_t i, j, k;

+	tonTyp *ton;

+	sampleTyp *s;

+	digiHeaderTyp h_DIGI;

+	tmpLinearPeriodsFlag = false; // use Amiga periods

+	if (filesize < sizeof (h_DIGI))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	memset(&h_DIGI, 0, sizeof (digiHeaderTyp));

+	if (fread(&h_DIGI, 1, sizeof (h_DIGI), f) != sizeof (h_DIGI))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (h_DIGI.numChannels < 1 || h_DIGI.numChannels > 8)

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	h_DIGI.numOrders++;

+	h_DIGI.numPats++;

+	if (h_DIGI.numOrders < 1 || h_DIGI.numOrders > 128)

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	songTmp.antChn = h_DIGI.numChannels;

+	songTmp.len = h_DIGI.numOrders;

+	songTmp.initialTempo = songTmp.tempo = 6;

+	songTmp.speed = 125;

+	memcpy(songTmp.songTab, h_DIGI.orders, 128);

+	// load pattern data

+	for (i = 0; i < h_DIGI.numPats; i++)

+	{

+		if (!allocateTmpPatt(i, 64))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (h_DIGI.packedPatternsFlag)

+		{

+			uint16_t pattSize;

+			uint8_t bitMasks[64];

+			fread(&pattSize, 2, 1, f); pattSize = SWAP16(pattSize);

+			fread(bitMasks, 1, 64, f);

+			for (j = 0; j < 64; j++)

+			{

+				uint8_t bit = 128;

+				for (k = 0; k < songTmp.antChn; k++, bit >>= 1)

+				{

+					ton = &pattTmp[i][(j * MAX_VOICES) + k];

+					if (bitMasks[j] & bit)

+						readPatternNote(f, ton);

+				}

+			}

+		}

+		else

+		{

+			for (j = 0; j < songTmp.antChn; j++)

+			{

+				for (k = 0; k < 64; k++)

+				{

+					ton = &pattTmp[i][(k * MAX_VOICES) + j];

+					readPatternNote(f, ton);

+				}

+			}

+		}

+		if (tmpPatternEmpty(i))

+		{

+			if (pattTmp[i] != NULL)

+			{

+				free(pattTmp[i]);

+				pattTmp[i] = NULL;

+			}

+		}

+	}

+	// pattern command handling

+	for (i = 0; i < h_DIGI.numPats; i++)

+	{

+		if (pattTmp[i] == NULL)

+			continue;

+		for (j = 0; j < 64; j++)

+		{

+			for (k = 0; k < songTmp.antChn; k++)

+			{

+				ton = &pattTmp[i][(j * MAX_VOICES) + k];

+				if (ton->effTyp == 0x8) // Robot effect (not supported)

+				{

+					ton->effTyp = 0;

+					ton->eff = 0;

+				}

+				else if (ton->effTyp == 0xE)

+				{

+					switch (ton->eff >> 4)

+					{

+						case 0x3: ton->effTyp = ton->eff = 0; break; // backwards play (not supported)

+						case 0x4: ton->eff = 0xC0; break; // stop sample (convert to EC0)

+						case 0x8: ton->effTyp = ton->eff = 0; break; // high sample offset (not supported)

+						case 0x9: ton->effTyp = ton->eff = 0; break; // retrace (not supported)

+						default: break;

+					}

+				}

+			}

+		}

+	}

+	// load sample data

+	for (i = 0; i < 31; i++)

+	{

+		memcpy(songTmp.instrName[1+i], h_DIGI.smpName[i], 22);

+		if (h_DIGI.smpLength[i] == 0)

+			continue;

+		if (!allocateTmpInstr(1+i))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		setNoEnvelope(instrTmp[1+i]);

+		s = &instrTmp[1+i]->samp[0];

+		memcpy(s->name, h_DIGI.smpName[i], 22);

+		s->len = SWAP32(h_DIGI.smpLength[i]);

+		s->fine = 8 * ((2 * ((h_DIGI.smpFinetune[i] & 0xF) ^ 8)) - 16);

+		s->vol = h_DIGI.smpVolume[i];

+		s->repS = SWAP32(h_DIGI.smpLoopStart[i]);

+		s->repL = SWAP32(h_DIGI.smpLoopLength[i]);

+		if (s->vol > 64)

+			s->vol = 64;

+		if (s->repL < 2)

+			s->repL = 2;

+		// fix overflown loop

+		if (s->repS+s->repL > s->len)

+		{

+			if (s->repS >= s->len)

+			{

+				s->repS = 0;

+				s->repL = 0;

+			}

+			else

+			{

+				s->repL = s->len - s->repS;

+			}

+		}

+		if (s->repS+s->repL > 2)

+		{

+			s->typ = 1; // enable loop

+		}

+		else

+		{

+			s->repS = 0;

+			s->repL = 0;

+		}

+		if (!allocateTmpSmpData(s, s->len))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		int32_t bytesRead = (int32_t)fread(s->pek, 1, s->len, f);

+		if (bytesRead < s->len)

+		{

+			int32_t bytesToClear = s->len - bytesRead;

+			memset(&s->pek[bytesRead], 0, bytesToClear);

+		}

+	}

+	return true;

+}

+static void readPatternNote(FILE *f, tonTyp *ton)

+{

+	uint8_t bytes[4];

+	fread(bytes, 1, 4, f);

+	// period to note

+	uint16_t period = ((bytes[0] & 0x0F) << 8) | bytes[1];

+	for (uint8_t i = 0; i < 3*12; i++)

+	{

+		if (period >= ptPeriods[i])

+		{

+			ton->ton = 1 + (3*12) + i;

+			break;

+		}

+	}

+	ton->instr = (bytes[0] & 0xF0) | (bytes[2] >> 4);

+	ton->effTyp = bytes[2] & 0x0F;

+	ton->eff = bytes[3];

+}

--- /dev/null

+++ b/src/modloaders/ft2_load_mod.c

@@ -1,0 +1,408 @@

+// NoiseTracker/ProTracker (or compatible) MOD loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+enum // supported 31-sample .MOD types

+{

+	FORMAT_MK,   // ProTracker or compatible

+	FORMAT_FLT4, // StarTrekker (4ch modules)

+	FORMAT_FLT8, // StarTrekker (8ch modules)

+	FORMAT_FT2,  // FT2 or compatible (multi-channel)

+	FORMAT_NT,   // NoiseTracker

+	FORMAT_HMNT, // His Master's NoiseTracker (special one)

+	FORMAT_UNKNOWN

+};

+static uint8_t getModType(uint8_t *numChannels, const char *id);

+bool loadMOD(FILE *f, uint32_t filesize)

+{

+	uint8_t bytes[4], modFormat, numChannels;

+	int16_t i, j, k;

+	uint16_t a, b, period;

+	tonTyp *ton;

+	sampleTyp *s;

+	songMOD31HeaderTyp h_MOD31;

+	tmpLinearPeriodsFlag = false; // use Amiga periods

+	if (filesize < sizeof (h_MOD31))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	memset(&h_MOD31, 0, sizeof (songMOD31HeaderTyp));

+	if (fread(&h_MOD31, 1, sizeof (h_MOD31), f) != sizeof (h_MOD31))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	modFormat = getModType(&numChannels, h_MOD31.sig);

+	if (modFormat == FORMAT_UNKNOWN)

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	songTmp.antChn = numChannels;

+	songTmp.len = h_MOD31.len;

+	songTmp.repS = h_MOD31.repS;

+	songTmp.initialTempo = songTmp.tempo = 6;

+	songTmp.speed = 125;

+	memcpy(songTmp.songTab, h_MOD31.songTab, 128);

+	if (modFormat == FORMAT_MK && songTmp.len == 129)

+		songTmp.len = 127; // fixes a specific copy of beatwave.mod (FIXME: Do we want to keep this hack?)

+	if (songTmp.repS >= songTmp.len)

+		songTmp.repS = 0;

+	if (songTmp.antChn == 0 || songTmp.len < 1 || songTmp.len > 128)

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	for (a = 0; a < 31; a++)

+	{

+		songMODInstrHeaderTyp *smp = &h_MOD31.instr[a];

+		// copy over sample name if format isn't "His Master's Noisetracker" (junk sample names)

+		if (modFormat != FORMAT_HMNT)

+			memcpy(songTmp.instrName[1+a], smp->name, 22);

+	}

+	memcpy(songTmp.name, h_MOD31.name, 20);

+	// count number of patterns

+	b = 0;

+	for (a = 0; a < 128; a++)

+	{

+		if (modFormat == FORMAT_FLT8)

+			songTmp.songTab[a] >>= 1;

+		if (songTmp.songTab[a] > b)

+			b = songTmp.songTab[a];

+	}

+	b++;

+	// load pattern data

+	if (modFormat != FORMAT_FLT8)

+	{

+		// normal pattern loading

+		for (a = 0; a < b; a++)

+		{

+			if (!allocateTmpPatt(a, 64))

+			{

+				loaderMsgBox("Not enough memory!");

+				return false;

+			}

+			for (j = 0; j < 64; j++)

+			{

+				for (k = 0; k < songTmp.antChn; k++)

+				{

+					ton = &pattTmp[a][(j * MAX_VOICES) + k];

+					fread(bytes, 1, 4, f);

+					// period to note

+					period = ((bytes[0] & 0x0F) << 8) | bytes[1];

+					for (i = 0; i < 8*12; i++)

+					{

+						if (period >= amigaPeriod[i])

+						{

+							ton->ton = (uint8_t)i + 1;

+							break;

+						}

+					}

+					ton->instr = (bytes[0] & 0xF0) | (bytes[2] >> 4);

+					ton->effTyp = bytes[2] & 0x0F;

+					ton->eff = bytes[3];

+				}

+			}

+			if (tmpPatternEmpty(a))

+			{

+				if (pattTmp[a] != NULL)

+				{

+					free(pattTmp[a]);

+					pattTmp[a] = NULL;

+				}

+			}

+		}

+	}

+	else

+	{

+		// FLT8 pattern loading

+		for (a = 0; a < b; a++)

+		{

+			if (!allocateTmpPatt(a, 64))

+			{

+				loaderMsgBox("Not enough memory!");

+				return false;

+			}

+		}

+		for (a = 0; a < b*2; a++)

+		{

+			int32_t pattern = a >> 1;

+			int32_t chnOffset = (a & 1) * 4;

+			for (j = 0; j < 64; j++)

+			{

+				for (k = 0; k < 4; k++)

+				{

+					ton = &pattTmp[pattern][(j * MAX_VOICES) + (k+chnOffset)];

+					fread(bytes, 1, 4, f);

+					// period to note

+					period = ((bytes[0] & 0x0F) << 8) | bytes[1];

+					for (i = 0; i < 8*12; i++)

+					{

+						if (period >= amigaPeriod[i])

+						{

+							ton->ton = (uint8_t)i + 1;

+							break;

+						}

+					}

+					ton->instr = (bytes[0] & 0xF0) | (bytes[2] >> 4);

+					ton->effTyp = bytes[2] & 0x0F;

+					ton->eff = bytes[3];

+				}

+			}

+		}

+		for (a = 0; a < b; a++)

+		{

+			if (tmpPatternEmpty(a))

+			{

+				if (pattTmp[a] != NULL)

+				{

+					free(pattTmp[a]);

+					pattTmp[a] = NULL;

+				}

+			}

+		}

+	}

+	// pattern command conversion

+	for (a = 0; a < b; a++)

+	{

+		if (pattTmp[a] == NULL)

+			continue;

+		for (j = 0; j < 64; j++)

+		{

+			for (k = 0; k < songTmp.antChn; k++)

+			{

+				ton = &pattTmp[a][(j * MAX_VOICES) + k];

+				if (ton->effTyp == 0xC)

+				{

+					if (ton->eff > 64)

+						ton->eff = 64;

+				}

+				else if (ton->effTyp == 0x1)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0x2)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0x5)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0x3;

+				}

+				else if (ton->effTyp == 0x6)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0x4;

+				}

+				else if (ton->effTyp == 0xA)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0xE)

+				{

+					// check if certain E commands are empty

+					if (ton->eff == 0x10 || ton->eff == 0x20 || ton->eff == 0xA0 || ton->eff == 0xB0)

+					{

+						ton->effTyp = 0;

+						ton->eff = 0;

+					}

+				}

+				if (modFormat == FORMAT_NT || modFormat == FORMAT_HMNT)

+				{

+					// any Dxx == D00 in NT/HMNT

+					if (ton->effTyp == 0xD)

+						ton->eff = 0;

+					// effect F with param 0x00 does nothing in NT/HMNT

+					if (ton->effTyp == 0xF && ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (modFormat == FORMAT_FLT4 || modFormat == FORMAT_FLT8) // Startrekker

+				{

+					if (ton->effTyp == 0xE) // remove unsupported "assembly macros" command

+					{

+						ton->effTyp = 0;

+						ton->eff = 0;

+					}

+					// Startrekker is always in vblank mode, and limits speed to 0x1F

+					if (ton->effTyp == 0xF && ton->eff > 0x1F)

+						ton->eff = 0x1F;

+				}

+			}

+		}

+	}

+	for (a = 0; a < 31; a++)

+	{

+		if (h_MOD31.instr[a].len == 0)

+			continue;

+		if (!allocateTmpInstr(1+a))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		setNoEnvelope(instrTmp[1+a]);

+		s = &instrTmp[1+a]->samp[0];

+		// copy over sample name if format isn't "His Master's Noisetracker" (junk sample names)

+		if (modFormat != FORMAT_HMNT)

+			memcpy(s->name, songTmp.instrName[1+a], 22);

+		if (modFormat == FORMAT_HMNT) // finetune in "His Master's NoiseTracker" is different

+			h_MOD31.instr[a].fine = (uint8_t)((-h_MOD31.instr[a].fine & 0x1F) >> 1); // one more bit of precision, + inverted

+		s->len = 2 * SWAP16(h_MOD31.instr[a].len);

+		s->fine = 8 * ((2 * ((h_MOD31.instr[a].fine & 0xF) ^ 8)) - 16);

+		s->vol = h_MOD31.instr[a].vol;

+		s->repS = 2 * SWAP16(h_MOD31.instr[a].repS);

+		s->repL = 2 * SWAP16(h_MOD31.instr[a].repL);

+		if (s->vol > 64)

+			s->vol = 64;

+		if (s->repL < 2)

+			s->repL = 2;

+		// fix for poorly converted STK (< v2.5) -> PT/NT modules (FIXME: Worth keeping or not?)

+		if (s->repL > 2 && s->repS+s->repL > s->len)

+		{

+			if ((s->repS>>1)+s->repL <= s->len)

+				s->repS >>= 1;

+		}

+		// fix overflown loop

+		if (s->repS+s->repL > s->len)

+		{

+			if (s->repS >= s->len)

+			{

+				s->repS = 0;

+				s->repL = 0;

+			}

+			else

+			{

+				s->repL = s->len - s->repS;

+			}

+		}

+		if (s->repS+s->repL > 2)

+			s->typ = 1; // enable loop

+		if (!allocateTmpSmpData(s, s->len))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		int32_t bytesRead = (int32_t)fread(s->pek, 1, s->len, f);

+		if (bytesRead < s->len)

+		{

+			int32_t bytesToClear = s->len - bytesRead;

+			memset(&s->pek[bytesRead], 0, bytesToClear);

+		}

+		if (s->typ == 0) // clear repL and repS on non-looping samples...

+		{

+			s->repL = 0;

+			s->repS = 0;

+		}

+	}

+	return true;

+}

+static uint8_t getModType(uint8_t *numChannels, const char *id)

+{

+#define IS_ID(s, b) !strncmp(s, b, 4)

+	uint8_t modFormat = FORMAT_UNKNOWN;

+	*numChannels = 4;

+	if (IS_ID("M.K.", id) || IS_ID("M!K!", id) || IS_ID("NSMS", id) ||

+		IS_ID("LARD", id) || IS_ID("PATT", id))

+	{

+		modFormat = FORMAT_MK; // ProTracker or compatible

+	}

+	else if (isdigit(id[0]) && id[1] == 'C' && id[2] == 'H' && id[3] == 'N') // xCHN

+	{

+		modFormat = FORMAT_FT2; // FT2 or compatible (multi-channel)

+		*numChannels = id[0] - '0';

+	}

+	else if (isdigit(id[0]) && isdigit(id[1]) && id[2] == 'C' && id[3] == 'H') // xxCH

+	{

+		modFormat = FORMAT_FT2; // FT2 or compatible (multi-channel)

+		*numChannels = ((id[0] - '0') * 10) + (id[1] - '0');

+	}

+	else if (isdigit(id[0]) && isdigit(id[1]) && id[2] == 'C' && id[3] == 'N') // xxCN (load as xxCH)

+	{

+		modFormat = FORMAT_FT2; // FT2 or compatible (multi-channel)

+		*numChannels = ((id[0] - '0') * 10) + (id[1] - '0');

+	}

+	else if (IS_ID("FLT4", id))

+	{

+		modFormat = FORMAT_FLT4; // StarTrekker (4ch modules)

+	}

+	else if (IS_ID("FLT8", id))

+	{

+		modFormat = FORMAT_FLT8; // StarTrekker (8ch modules)

+		*numChannels = 8;

+	}

+	else if (IS_ID("N.T.", id))

+	{

+		modFormat = FORMAT_NT; // NoiseTracker

+	}

+	else if (IS_ID("M&K!", id) || IS_ID("FEST", id))

+	{

+		modFormat = FORMAT_HMNT; // His Master's NoiseTracker (special one)

+	}

+	return modFormat;

+}

--- /dev/null

+++ b/src/modloaders/ft2_load_s3m.c

@@ -1,0 +1,663 @@

+// Scream Tracker 3 (or compatible) S3M loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+#ifdef _MSC_VER // please don't mess with these structs!

+#pragma pack(push)

+#pragma pack(1)

+#endif

+typedef struct songS3MinstrHeaderTyp_t

+{

+	uint8_t typ;

+	char dosName[12];

+	uint8_t memSegH;

+	uint16_t memSeg;

+	int32_t len, repS, repE;

+	uint8_t vol, dsk, pack, flags;

+	int32_t c2Spd, res1;

+	uint16_t gusPos;

+	uint8_t res2[6];

+	char name[28], id[4];

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+songS3MinstrHeaderTyp;

+typedef struct songS3MHeaderTyp_t

+{

+	char name[28];

+	uint8_t id1a, typ;

+	uint16_t res1;

+	int16_t songTabLen, antInstr, antPatt;

+	uint16_t flags, trackerID, ver;

+	char id[4];

+	uint8_t globalVol, defSpeed, defTempo, masterVol, res2[12], chanType[32];

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+songS3MHeaderTyp;

+#ifdef _MSC_VER

+#pragma pack(pop)

+#endif

+static uint8_t pattBuff[12288];

+static int8_t countS3MChannels(uint16_t antPtn);

+bool loadS3M(FILE *f, uint32_t filesize)

+{

+	uint8_t ha[2048];

+	uint8_t alastnfo[32], alastefx[32], alastvibnfo[32], s3mLastGInstr[32];

+	uint8_t typ;

+	int16_t ai, ap, ver, ii, kk, tmp;

+	uint16_t ptnOfs[256];

+	int32_t i, j, k, len;

+	tonTyp ton;

+	sampleTyp *s;

+	songS3MHeaderTyp h_S3M;

+	songS3MinstrHeaderTyp h_S3MInstr;

+	tmpLinearPeriodsFlag = false; // use Amiga periods

+	if (filesize < sizeof (h_S3M))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	memset(&h_S3M, 0, sizeof (h_S3M));

+	if (fread(&h_S3M, 1, sizeof (h_S3M), f) != sizeof (h_S3M))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (h_S3M.antInstr > MAX_INST || h_S3M.songTabLen > 256 || h_S3M.antPatt > 256 ||

+		h_S3M.typ != 16 || h_S3M.ver < 1 || h_S3M.ver > 2)

+	{

+		loaderMsgBox("Error loading .s3m: Incompatible module!");

+		return false;

+	}

+	memset(songTmp.songTab, 255, sizeof (songTmp.songTab));

+	if (fread(songTmp.songTab, h_S3M.songTabLen, 1, f) != 1)

+	{

+		loaderMsgBox("General I/O error during loading! Is the file in use?");

+		return false;

+	}

+	songTmp.len = h_S3M.songTabLen;

+	// remove pattern separators (254)

+	k = 0;

+	j = 0;

+	for (i = 0; i < songTmp.len; i++)

+	{

+		if (songTmp.songTab[i] != 254)

+			songTmp.songTab[j++] = songTmp.songTab[i];

+		else

+			k++;

+	}

+	if (k <= songTmp.len)

+		songTmp.len -= (uint16_t)k;

+	else

+		songTmp.len = 1;

+	for (i = 1; i < songTmp.len; i++)

+	{

+		if (songTmp.songTab[i] == 255)

+		{

+			songTmp.len = (uint16_t)i;

+			break;

+		}

+	}

+	// clear unused song table entries

+	if (songTmp.len < 255)

+		memset(&songTmp.songTab[songTmp.len], 0, 255-songTmp.len);

+	songTmp.speed = h_S3M.defTempo;

+	if (songTmp.speed < 32)

+		songTmp.speed = 32;

+	songTmp.tempo = h_S3M.defSpeed;

+	if (songTmp.tempo == 0)

+		songTmp.tempo = 6;

+	if (songTmp.tempo > 31)

+		songTmp.tempo = 31;

+	songTmp.initialTempo = songTmp.tempo;

+	memcpy(songTmp.name, h_S3M.name, 20);

+	ap = h_S3M.antPatt;

+	ai = h_S3M.antInstr;

+	ver = h_S3M.ver;

+	k = 31;

+	while (k >= 0 && h_S3M.chanType[k] >= 16) k--;

+	songTmp.antChn = (k + 2) & 254;

+	if (fread(ha, ai*2, 1, f) != 1)

+	{

+		loaderMsgBox("General I/O error during loading! Is the file in use?");

+		return false;

+	}

+	if (fread(ptnOfs, ap*2, 1, f) != 1)

+	{

+		loaderMsgBox("General I/O error during loading! Is the file in use?");

+		return false;

+	}

+	// *** PATTERNS ***

+	k = 0;

+	for (i = 0; i < ap; i++)

+	{

+		if (ptnOfs[i]  == 0)

+			continue; // empty pattern

+		memset(alastnfo, 0, sizeof (alastnfo));

+		memset(alastefx, 0, sizeof (alastefx));

+		memset(alastvibnfo, 0, sizeof (alastvibnfo));

+		memset(s3mLastGInstr, 0, sizeof (s3mLastGInstr));

+		fseek(f, ptnOfs[i] << 4, SEEK_SET);

+		if (feof(f))

+			continue;

+		if (fread(&j, 2, 1, f) != 1)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		if (j > 0 && j <= 12288)

+		{

+			if (!allocateTmpPatt(i, 64))

+			{

+				loaderMsgBox("Not enough memory!");

+				return false;

+			}

+			fread(pattBuff, j, 1, f);

+			k = 0;

+			kk = 0;

+			while (k < j && kk < 64)

+			{

+				typ = pattBuff[k++];

+				if (typ == 0)

+				{

+					kk++;

+				}

+				else

+				{

+					ii = typ & 31;

+					memset(&ton, 0, sizeof (ton));

+					// note and sample

+					if (typ & 32)

+					{

+						ton.ton = pattBuff[k++];

+						ton.instr = pattBuff[k++];

+						if (ton.instr > MAX_INST)

+							ton.instr = 0;

+						     if (ton.ton == 254) ton.ton = 97;

+						else if (ton.ton == 255) ton.ton = 0;

+						else

+						{

+							ton.ton = 1 + (ton.ton & 0xF) + (ton.ton >> 4) * 12;

+							if (ton.ton > 96)

+								ton.ton = 0;

+						}

+					}

+					// volume column

+					if (typ & 64)

+					{

+						ton.vol = pattBuff[k++];

+						if (ton.vol <= 64)

+							ton.vol += 0x10;

+						else

+							ton.vol = 0;

+					}

+					// effect

+					if (typ & 128)

+					{

+						ton.effTyp = pattBuff[k++];

+						ton.eff = pattBuff[k++];

+						if (ton.eff > 0)

+						{

+							alastnfo[ii] = ton.eff;

+							if (ton.effTyp == 8 || ton.effTyp == 21)

+								alastvibnfo[ii] = ton.eff; // H/U

+						}

+						// in ST3, a lot of effects directly share the same memory!

+						if (ton.eff == 0 && ton.effTyp != 7) // G

+						{

+							uint8_t efx = ton.effTyp;

+							if (efx == 8 || efx == 21) // H/U

+								ton.eff = alastvibnfo[ii];

+							else if ((efx >= 4 && efx <= 12) || (efx >= 17 && efx <= 19)) // D/E/F/I/J/K/L/Q/R/S

+								ton.eff = alastnfo[ii];

+							/* If effect data is zero and effect type was the same as last one, clear out

+							** data if it's not J or S (those have no memory in the equivalent XM effects).

+							** Also goes for extra fine pitch slides and fine volume slides,

+							** since they get converted to other effects.

+							*/

+							if (efx == alastefx[ii] && ton.effTyp != 10 && ton.effTyp != 19) // J/S

+							{

+								uint8_t nfo = ton.eff;

+								bool extraFinePitchSlides = (efx == 5 || efx == 6) && ((nfo & 0xF0) == 0xE0);

+								bool fineVolSlides = (efx == 4 || efx == 11) &&

+								     ((nfo > 0xF0) || (((nfo & 0xF) == 0xF) && ((nfo & 0xF0) > 0)));

+								if (!extraFinePitchSlides && !fineVolSlides)

+									ton.eff = 0;

+							}

+						}

+						if (ton.effTyp > 0)

+							alastefx[ii] = ton.effTyp;

+						switch (ton.effTyp)

+						{

+							case 1: // A

+							{

+								ton.effTyp = 0xF;

+								if (ton.eff == 0 || ton.eff > 0x1F)

+								{

+									ton.effTyp = 0;

+									ton.eff = 0;

+								}

+							}

+							break;

+							case 2: ton.effTyp = 0xB; break; // B

+							case 3: ton.effTyp = 0xD; break; // C

+							case 4: // D

+							{

+								if (ton.eff > 0xF0) // fine slide up

+								{

+									ton.effTyp = 0xE;

+									ton.eff = 0xB0 | (ton.eff & 0xF);

+								}

+								else if ((ton.eff & 0x0F) == 0x0F && (ton.eff & 0xF0) > 0) // fine slide down

+								{

+									ton.effTyp = 0xE;

+									ton.eff = 0xA0 | (ton.eff >> 4);

+								}

+								else

+								{

+									ton.effTyp = 0xA;

+									if (ton.eff & 0x0F) // on D/K, last nybble has first priority in ST3

+										ton.eff &= 0x0F;

+								}

+							}

+							break;

+							case 5: // E

+							case 6: // F

+							{

+								if ((ton.eff & 0xF0) >= 0xE0)

+								{

+									// convert to fine slide

+									if ((ton.eff & 0xF0) == 0xE0)

+										tmp = 0x21;

+									else

+										tmp = 0xE;

+									ton.eff &= 0x0F;

+									if (ton.effTyp == 0x05)

+										ton.eff |= 0x20;

+									else

+										ton.eff |= 0x10;

+									ton.effTyp = (uint8_t)tmp;

+									if (ton.effTyp == 0x21 && ton.eff == 0)

+									{

+										ton.effTyp = 0;

+									}

+								}

+								else

+								{

+									// convert to normal 1xx/2xx slide

+									ton.effTyp = 7 - ton.effTyp;

+								}

+							}

+							break;

+							case 7: // G

+							{

+								ton.effTyp = 0x03;

+								// fix illegal slides (to new instruments)

+								if (ton.instr != 0 && ton.instr != s3mLastGInstr[ii])

+									ton.instr = s3mLastGInstr[ii];

+							}

+							break;

+							case 11: // K

+							{

+								if (ton.eff > 0xF0) // fine slide up

+								{

+									ton.effTyp = 0xE;

+									ton.eff = 0xB0 | (ton.eff & 0xF);

+									// if volume column is unoccupied, set to vibrato

+									if (ton.vol == 0)

+										ton.vol = 0xB0;

+								}

+								else if ((ton.eff & 0x0F) == 0x0F && (ton.eff & 0xF0) > 0) // fine slide down

+								{

+									ton.effTyp = 0xE;

+									ton.eff = 0xA0 | (ton.eff >> 4);

+									// if volume column is unoccupied, set to vibrato

+									if (ton.vol == 0)

+										ton.vol = 0xB0;

+								}

+								else

+								{

+									ton.effTyp = 0x6;

+									if (ton.eff & 0x0F) // on D/K, last nybble has first priority in ST3

+										ton.eff &= 0x0F;

+								}

+							}

+							break;

+							case 8: ton.effTyp = 0x04; break; // H

+							case 9: ton.effTyp = 0x1D; break; // I

+							case 10: ton.effTyp = 0x00; break; // J

+							case 12: ton.effTyp = 0x05; break; // L

+							case 15: ton.effTyp = 0x09; break; // O

+							case 17: ton.effTyp = 0x1B; break; // Q

+							case 18: ton.effTyp = 0x07; break; // R

+							case 19: // S

+							{

+								ton.effTyp = 0xE;

+								tmp = ton.eff >> 4;

+								ton.eff &= 0x0F;

+								     if (tmp == 0x1) ton.eff |= 0x30;

+								else if (tmp == 0x2) ton.eff |= 0x50;

+								else if (tmp == 0x3) ton.eff |= 0x40;

+								else if (tmp == 0x4) ton.eff |= 0x70;

+								// ignore S8x becuase it's not compatible with FT2 panning

+								else if (tmp == 0xB) ton.eff |= 0x60;

+								else if (tmp == 0xC) // Note Cut

+								{

+									ton.eff |= 0xC0;

+									if (ton.eff == 0xC0)

+									{

+										// EC0 does nothing in ST3 but cuts voice in FT2, remove effect

+										ton.effTyp = 0;

+										ton.eff = 0;

+									}

+								}

+								else if (tmp == 0xD) // Note Delay

+								{

+									ton.eff |= 0xD0;

+									if (ton.ton == 0 || ton.ton == 97)

+									{

+										// EDx without a note does nothing in ST3 but retrigs in FT2, remove effect

+										ton.effTyp = 0;

+										ton.eff = 0;

+									}

+									else if (ton.eff == 0xD0)

+									{

+										// ED0 prevents note/smp/vol from updating in ST3, remove everything

+										ton.ton = 0;

+										ton.instr = 0;

+										ton.vol = 0;

+										ton.effTyp = 0;

+										ton.eff = 0;

+									}

+								}

+								else if (tmp == 0xE) ton.eff |= 0xE0;

+								else if (tmp == 0xF) ton.eff |= 0xF0;

+								else

+								{

+									ton.effTyp = 0;

+									ton.eff = 0;

+								}

+							}

+							break;

+							case 20: // T

+							{

+								ton.effTyp = 0x0F;

+								if (ton.eff < 0x21) // Txx with a value lower than 33 (0x21) does nothing in ST3, remove effect

+								{

+									ton.effTyp = 0;

+									ton.eff = 0;

+								}

+							}

+							break;

+							case 22: // V

+							{

+								ton.effTyp = 0x10;

+								if (ton.eff > 0x40)

+								{

+									// Vxx > 0x40 does nothing in ST3

+									ton.effTyp = 0;

+									ton.eff = 0;

+								}

+							}

+							break;

+							default:

+							{

+								ton.effTyp = 0;

+								ton.eff = 0;

+							}

+							break;

+						}

+					}

+					if (ton.instr != 0 && ton.effTyp != 0x3)

+						s3mLastGInstr[ii] = ton.instr;

+					if (ton.effTyp > 35)

+					{

+						ton.effTyp = 0;

+						ton.eff = 0;

+					}

+					pattTmp[i][(kk * MAX_VOICES) + ii] = ton;

+				}

+			}

+			if (tmpPatternEmpty((uint16_t)i))

+			{

+				if (pattTmp[i] != NULL)

+				{

+					free(pattTmp[i]);

+					pattTmp[i] = NULL;

+				}

+			}

+		}

+	}

+	// *** SAMPLES ***

+	bool adlibInsWarn = false;

+	memcpy(ptnOfs, ha, 512);

+	for (i = 0; i < ai; i++)

+	{

+		fseek(f, ptnOfs[i] << 4, SEEK_SET);

+		if (fread(&h_S3MInstr, 1, sizeof (h_S3MInstr), f) != sizeof (h_S3MInstr))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		memcpy(songTmp.instrName[1+i], h_S3MInstr.name, 22);

+		if (h_S3MInstr.typ == 2)

+		{

+			adlibInsWarn = true;

+		}

+		else if (h_S3MInstr.typ == 1)

+		{

+			if ((h_S3MInstr.flags & (255-1-2-4)) != 0 || h_S3MInstr.pack != 0)

+			{

+				loaderMsgBox("Error loading .s3m: Incompatible module!");

+				return false;

+			}

+			else if (h_S3MInstr.memSeg > 0 && h_S3MInstr.len > 0)

+			{

+				if (!allocateTmpInstr((int16_t)(1 + i)))

+				{

+					loaderMsgBox("Not enough memory!");

+					return false;

+				}

+				setNoEnvelope(instrTmp[1 + i]);

+				s = &instrTmp[1+i]->samp[0];

+				// non-FT2: fixes "miracle man.s3m" and other broken S3Ms

+				if ((h_S3MInstr.memSeg<<4)+h_S3MInstr.len > (int32_t)filesize)

+					h_S3MInstr.len = filesize - (h_S3MInstr.memSeg << 4);

+				len = h_S3MInstr.len;

+				bool hasLoop = h_S3MInstr.flags & 1;

+				bool stereoSample = (h_S3MInstr.flags >> 1) & 1;

+				bool is16Bit = (h_S3MInstr.flags >> 2) & 1;

+				if (is16Bit) // 16-bit

+					len <<= 1;

+				if (stereoSample) // stereo

+					len <<= 1;

+				if (!allocateTmpSmpData(s, len))

+				{

+					loaderMsgBox("Not enough memory!");

+					return false;

+				}

+				memcpy(s->name, h_S3MInstr.name, 21);

+				if (h_S3MInstr.c2Spd > 65535) // ST3 (and OpenMPT) does this

+					h_S3MInstr.c2Spd = 65535;

+				s->len = h_S3MInstr.len;

+				s->vol = h_S3MInstr.vol;

+				s->repS = h_S3MInstr.repS;

+				s->repL = h_S3MInstr.repE - h_S3MInstr.repS;

+				tuneSample(s, h_S3MInstr.c2Spd, tmpLinearPeriodsFlag);

+				if (s->vol > 64)

+					s->vol = 64;

+				if (s->repL <= 2 || s->repS+s->repL > s->len || s->repL == 0)

+				{

+					s->repS = 0;

+					s->repL = 0;

+					hasLoop = false;

+				}

+				s->typ = hasLoop + (is16Bit << 4);

+				fseek(f, h_S3MInstr.memSeg << 4, SEEK_SET);

+				if (ver == 1)

+				{

+					fseek(f, len, SEEK_CUR); // sample not supported

+				}

+				else

+				{

+					if (fread(s->pek, len, 1, f) != 1)

+					{

+						loaderMsgBox("General I/O error during loading! Is the file in use?");

+						return false;

+					}

+					if (is16Bit)

+					{

+						conv16BitSample(s->pek, len, stereoSample);

+						s->len <<= 1;

+						s->repS <<= 1;

+						s->repL <<= 1;

+					}

+					else

+					{

+						conv8BitSample(s->pek, len, stereoSample);

+					}

+					// if stereo sample: reduce memory footprint after sample was downmixed to mono

+					if (stereoSample)

+						reallocateTmpSmpData(s, s->len);

+				}

+			}

+		}

+	}

+	songTmp.antChn = countS3MChannels(ap);

+	if (adlibInsWarn)

+		loaderMsgBox("Warning: The module contains unsupported AdLib instruments!");

+	if (!(config.dontShowAgainFlags & DONT_SHOW_S3M_LOAD_WARNING_FLAG))

+		loaderSysReq(6, "System message", "Warning: S3M channel panning is ignored because it's not compatible with FT2.");

+	return true;

+}

+static int8_t countS3MChannels(uint16_t antPtn)

+{

+	int32_t channels = 0;

+	for (int32_t i = 0; i < antPtn; i++)

+	{

+		if (pattTmp[i] == NULL)

+			continue;

+		tonTyp *ton = pattTmp[i];

+		for (int32_t j = 0; j < 64; j++)

+		{

+			for (int32_t k = 0; k < MAX_VOICES; k++, ton++)

+			{

+				if (ton->eff == 0 && ton->effTyp == 0 && ton->instr == 0 && ton->ton == 0 && ton->vol == 0)

+					continue;

+				if (k > channels)

+					channels = k;

+			}

+		}

+	}

+	channels++;

+	return (int8_t)channels;

+}

--- /dev/null

+++ b/src/modloaders/ft2_load_stk.c

@@ -1,0 +1,350 @@

+// Ultimate SoundTracker (or compatible) STK loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+#ifdef _MSC_VER  // please don't mess with this struct!

+#pragma pack(push)

+#pragma pack(1)

+#endif

+typedef struct songMOD15HeaderTyp_t

+{

+	char name[20];

+	songMODInstrHeaderTyp instr[15];

+	uint8_t len, CIAVal, songTab[128];

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+songMOD15HeaderTyp;

+#ifdef _MSC_VER

+#pragma pack(pop)

+#endif

+bool loadSTK(FILE *f, uint32_t filesize)

+{

+	uint8_t bytes[4];

+	int16_t i, j, k;

+	uint16_t a, b, period;

+	tonTyp *ton;

+	sampleTyp *s;

+	songMOD15HeaderTyp h_MOD15;

+	tmpLinearPeriodsFlag = false; // use Amiga periods

+	bool veryLateSTKVerFlag = false; // "DFJ SoundTracker III" nad later

+	bool lateSTKVerFlag = false; // "TJC SoundTracker II" and later

+	if (filesize < sizeof (h_MOD15))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	memset(&h_MOD15, 0, sizeof (songMOD15HeaderTyp));

+	if (fread(&h_MOD15, 1, sizeof (h_MOD15), f) != sizeof (h_MOD15))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (h_MOD15.CIAVal == 0) // a CIA value of 0 results in 120

+		h_MOD15.CIAVal = 120;

+	songTmp.antChn = 4;

+	songTmp.len = h_MOD15.len;

+	songTmp.speed = 125;

+	songTmp.initialTempo = songTmp.tempo = 6;

+	memcpy(songTmp.songTab, h_MOD15.songTab, 128);

+	if (songTmp.len < 1 || songTmp.len > 128 || h_MOD15.CIAVal > 220)

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	for (a = 0; a < 15; a++)

+	{

+		songMODInstrHeaderTyp *smp = &h_MOD15.instr[a];

+		memcpy(songTmp.instrName[1+a], smp->name, 22);

+		/* Only late versions of Ultimate SoundTracker supports samples larger than 9999 bytes.

+		** If found, we know for sure that this is a late STK module.

+		*/

+		const int32_t sampleLen = 2*SWAP16(smp->len);

+		if (sampleLen > 9999)

+			lateSTKVerFlag = true;

+	}

+	// jjk55.mod by Jesper Kyd has a bogus STK tempo value that should be ignored (hackish!)

+	if (!strcmp("jjk55", h_MOD15.name))

+		h_MOD15.CIAVal = 120;

+	if (h_MOD15.CIAVal != 120) // 120 is a special case and means 50Hz (125BPM)

+	{

+		// convert UST tempo to BPM

+		uint16_t ciaPeriod = (240 - h_MOD15.CIAVal) * 122;

+		double dHz = 709379.0 / ciaPeriod;

+		int32_t BPM = (int32_t)((dHz * 2.5) + 0.5);

+		songTmp.speed = (uint16_t)BPM;

+	}

+	memcpy(songTmp.name, h_MOD15.name, 20);

+	// count number of patterns

+	b = 0;

+	for (a = 0; a < 128; a++)

+	{

+		if (songTmp.songTab[a] > b)

+			b = songTmp.songTab[a];

+	}

+	b++;

+	for (a = 0; a < b; a++)

+	{

+		if (!allocateTmpPatt(a, 64))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		for (j = 0; j < 64; j++)

+		{

+			for (k = 0; k < songTmp.antChn; k++)

+			{

+				ton = &pattTmp[a][(j * MAX_VOICES) + k];

+				if (fread(bytes, 1, 4, f) != 4)

+				{

+					loaderMsgBox("Error: This file is either not a module, or is not supported.");

+					return false;

+				}

+				// period to note

+				period = ((bytes[0] & 0x0F) << 8) | bytes[1];

+				for (i = 0; i < 3*12; i++)

+				{

+					if (period >= ptPeriods[i])

+					{

+						ton->ton = 1 + (3*12) + (uint8_t)i;

+						break;

+					}

+				}

+				ton->instr = (bytes[0] & 0xF0) | (bytes[2] >> 4);

+				ton->effTyp = bytes[2] & 0x0F;

+				ton->eff = bytes[3];

+				if (ton->effTyp == 0xC || ton->effTyp == 0xD || ton->effTyp == 0xE)

+				{

+					// "TJC SoundTracker II" and later

+					lateSTKVerFlag = true;

+				}

+				if (ton->effTyp == 0xF)

+				{

+					// "DFJ SoundTracker III" and later

+					lateSTKVerFlag = true;

+					veryLateSTKVerFlag = true;

+				}

+				if (ton->effTyp == 0xC)

+				{

+					if (ton->eff > 64)

+						ton->eff = 64;

+				}

+				else if (ton->effTyp == 0x1)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0x2)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0x5)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0x3;

+				}

+				else if (ton->effTyp == 0x6)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0x4;

+				}

+				else if (ton->effTyp == 0xA)

+				{

+					if (ton->eff == 0)

+						ton->effTyp = 0;

+				}

+				else if (ton->effTyp == 0xE)

+				{

+					// check if certain E commands are empty

+					if (ton->eff == 0x10 || ton->eff == 0x20 || ton->eff == 0xA0 || ton->eff == 0xB0)

+					{

+						ton->effTyp = 0;

+						ton->eff = 0;

+					}

+				}

+			}

+		}

+		if (tmpPatternEmpty(a))

+		{

+			if (pattTmp[a] != NULL)

+			{

+				free(pattTmp[a]);

+				pattTmp[a] = NULL;

+			}

+		}

+	}

+	// pattern command conversion for non-PT formats

+	for (a = 0; a < b; a++)

+	{

+		if (pattTmp[a] == NULL)

+			continue;

+		for (j = 0; j < 64; j++)

+		{

+			for (k = 0; k < songTmp.antChn; k++)

+			{

+				ton = &pattTmp[a][(j * MAX_VOICES) + k];

+				// convert STK effects to PT effects

+				if (!lateSTKVerFlag)

+				{

+					// old SoundTracker 1.x commands

+					if (ton->effTyp == 1)

+					{

+						// arpeggio

+						ton->effTyp = 0;

+					}

+					else if (ton->effTyp == 2)

+					{

+						// pitch slide

+						if (ton->eff & 0xF0)

+						{

+							// pitch slide down

+							ton->effTyp = 2;

+							ton->eff >>= 4;

+						}

+						else if (ton->eff & 0x0F)

+						{

+							// pitch slide up

+							ton->effTyp = 1;

+						}

+					}

+				}

+				else

+				{

+					// "DFJ SoundTracker II" or later

+					if (ton->effTyp == 0xD)

+					{

+						if (veryLateSTKVerFlag) // "DFJ SoundTracker III" or later

+						{

+							// pattern break w/ no param (param must be cleared to fix some songs)

+							ton->eff = 0;

+						}

+						else

+						{

+							// volume slide

+							ton->effTyp = 0xA;

+						}

+					}

+				}

+				// effect F with param 0x00 does nothing in UST/STK (I think)

+				if (ton->effTyp == 0xF && ton->eff == 0)

+					ton->effTyp = 0;

+			}

+		}

+	}

+	for (a = 0; a < 15; a++)

+	{

+		if (h_MOD15.instr[a].len == 0)

+			continue;

+		if (!allocateTmpInstr(1+a))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		setNoEnvelope(instrTmp[1+a]);

+		s = &instrTmp[1+a]->samp[0];

+		s->vol = h_MOD15.instr[a].vol;

+		s->len  = 2 * SWAP16(h_MOD15.instr[a].len);

+		s->repS =     SWAP16(h_MOD15.instr[a].repS); // in STK, loopStart = bytes, not words

+		s->repL = 2 * SWAP16(h_MOD15.instr[a].repL);

+		if (s->vol > 64)

+			s->vol = 64;

+		if (s->repL < 2)

+			s->repL = 2;

+		// fix overflown loop

+		if (s->repS+s->repL > s->len)

+		{

+			if (s->repS >= s->len)

+			{

+				s->repS = 0;

+				s->repL = 0;

+			}

+			else

+			{

+				s->repL = s->len - s->repS;

+			}

+		}

+		if (s->repS+s->repL > 2)

+		{

+			s->typ = 1; // enable loop

+		}

+		else

+		{

+			s->repL = 0;

+			s->repS = 0;

+		}

+		/* In STK, only the loop area of a looped sample is played.

+		** Skip loading of eventual data present before loop start.

+		*/

+		if (s->repS > 0 && s->repL < s->len)

+		{

+			s->len -= s->repS;

+			fseek(f, s->repS, SEEK_CUR);

+			s->repS = 0;

+		}

+		if (!allocateTmpSmpData(s, s->len))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		int32_t bytesRead = (int32_t)fread(s->pek, 1, s->len, f);

+		if (bytesRead < s->len)

+		{

+			int32_t bytesToClear = s->len - bytesRead;

+			memset(&s->pek[bytesRead], 0, bytesToClear);

+		}

+	}

+	return true;

+}

--- /dev/null

+++ b/src/modloaders/ft2_load_stm.c

@@ -1,0 +1,263 @@

+// Scream Tracker 2 STM loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+#ifdef _MSC_VER // please don't mess with these structs!

+#pragma pack(push)

+#pragma pack(1)

+#endif

+typedef struct songSTMinstrHeaderTyp_t

+{

+	char name[12];

+	uint8_t nul, insDisk;

+	uint16_t reserved1, len, repS, repE;

+	uint8_t vol, reserved2;

+	uint16_t rate;

+	int32_t reserved3;

+	uint16_t paraLen;

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+songSTMinstrHeaderTyp;

+typedef struct songSTMHeaderTyp_t

+{

+	char name[20], sig[8];

+	uint8_t id1a, typ;

+	uint8_t verMajor, verMinor;

+	uint8_t tempo, ap, vol, reserved[13];

+	songSTMinstrHeaderTyp instr[31];

+	uint8_t songTab[128];

+}

+#ifdef __GNUC__

+__attribute__ ((packed))

+#endif

+songSTMHeaderTyp;

+#ifdef _MSC_VER

+#pragma pack(pop)

+#endif

+static const uint8_t stmEff[16] = { 0, 0, 11, 0, 10, 2, 1, 3, 4, 7, 0, 5, 6, 0, 0, 0 };

+static uint8_t stmTempoToBPM(uint8_t tempo);

+bool loadSTM(FILE *f, uint32_t filesize)

+{

+	uint8_t typ, tempo, pattBuff[1024];

+	int16_t i, j, k, ap, tmp;

+	uint16_t a;

+	tonTyp *ton;

+	sampleTyp *s;

+	songSTMHeaderTyp h_STM;

+	tmpLinearPeriodsFlag = false; // use Amiga periods

+	if (filesize < sizeof (h_STM))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (fread(&h_STM, 1, sizeof (h_STM), f) != sizeof (h_STM))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (h_STM.verMinor == 0 || h_STM.typ != 2)

+	{

+		loaderMsgBox("Error loading STM: Incompatible module!");

+		return false;

+	}

+	songTmp.antChn = 4;

+	memcpy(songTmp.songTab, h_STM.songTab, 128);

+	i = 0;

+	while (i < 128 && songTmp.songTab[i] < 99) i++;

+	songTmp.len = i + (i == 0);

+	if (songTmp.len < 255)

+		memset(&songTmp.songTab[songTmp.len], 0, 256 - songTmp.len);

+	memcpy(songTmp.name, h_STM.name, 20);

+	tempo = h_STM.tempo;

+	if (h_STM.verMinor < 21)

+		tempo = ((tempo / 10) << 4) + (tempo % 10);

+	if (tempo == 0)

+		tempo = 96;

+	songTmp.initialTempo = songTmp.tempo = CLAMP(h_STM.tempo >> 4, 1, 31);

+	songTmp.speed = stmTempoToBPM(tempo);

+	if (h_STM.verMinor > 10)

+		songTmp.globVol = MIN(h_STM.vol, 64);

+	ap = h_STM.ap;

+	for (i = 0; i < ap; i++)

+	{

+		if (!allocateTmpPatt(i, 64))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(pattBuff, 64 * 4 * 4, 1, f) != 1)

+		{

+			loaderMsgBox("General I/O error during loading!");

+			return false;

+		}

+		a = 0;

+		for (j = 0; j < 64; j++)

+		{

+			for (k = 0; k < 4; k++)

+			{

+				ton = &pattTmp[i][(j * MAX_VOICES) + k];

+				if (pattBuff[a] == 254)

+				{

+					ton->ton = 97;

+				}

+				else if (pattBuff[a] < 96)

+				{

+					ton->ton = (12 * (pattBuff[a] >> 4)) + (25 + (pattBuff[a] & 0x0F));

+					if (ton->ton > 96)

+						ton->ton = 0;

+				}

+				else

+				{

+					ton->ton = 0;

+				}

+				ton->instr = pattBuff[a + 1] >> 3;

+				typ = (pattBuff[a + 1] & 7) + ((pattBuff[a + 2] & 0xF0) >> 1);

+				if (typ <= 64)

+					ton->vol = typ + 0x10;

+				ton->eff = pattBuff[a + 3];

+				tmp = pattBuff[a + 2] & 0x0F;

+				if (tmp == 1)

+				{

+					ton->effTyp = 15;

+					if (h_STM.verMinor < 21)

+						ton->eff = ((ton->eff / 10) << 4) + (ton->eff % 10);

+					ton->eff >>= 4;

+				}

+				else if (tmp == 3)

+				{

+					ton->effTyp = 13;

+					ton->eff = 0;

+				}

+				else if (tmp == 2 || (tmp >= 4 && tmp <= 12))

+				{

+					ton->effTyp = stmEff[tmp];

+					if (ton->effTyp == 0xA)

+					{

+						if (ton->eff & 0x0F)

+							ton->eff &= 0x0F;

+						else

+							ton->eff &= 0xF0;

+					}

+				}

+				else

+				{

+					ton->eff = 0;

+				}

+				if (ton->effTyp > 35)

+				{

+					ton->effTyp = 0;

+					ton->eff = 0;

+				}

+				a += 4;

+			}

+		}

+		if (tmpPatternEmpty(i))

+		{

+			if (pattTmp[i] != NULL)

+			{

+				free(pattTmp[i]);

+				pattTmp[i] = NULL;

+			}

+		}

+	}

+	for (i = 0; i < 31; i++)

+	{

+		memcpy(&songTmp.instrName[1+i], h_STM.instr[i].name, 12);

+		if (h_STM.instr[i].len != 0 && h_STM.instr[i].reserved1 != 0)

+		{

+			allocateTmpInstr(1 + i);

+			setNoEnvelope(instrTmp[i]);

+			s = &instrTmp[1+i]->samp[0];

+			if (!allocateTmpSmpData(s, h_STM.instr[i].len))

+			{

+				loaderMsgBox("Not enough memory!");

+				return false;

+			}

+			s->len = h_STM.instr[i].len;

+			s->vol = h_STM.instr[i].vol;

+			s->repS = h_STM.instr[i].repS;

+			s->repL = h_STM.instr[i].repE - h_STM.instr[i].repS;

+			memcpy(s->name, h_STM.instr[i].name, 12);

+			tuneSample(s, h_STM.instr[i].rate, tmpLinearPeriodsFlag);

+			if (s->repS < s->len && h_STM.instr[i].repE > s->repS && h_STM.instr[i].repE != 0xFFFF)

+			{

+				if (s->repS+s->repL > s->len)

+					s->repL = s->len - s->repS;

+				s->typ = 1; // enable loop

+			}

+			else

+			{

+				s->repS = 0;

+				s->repL = 0;

+				s->typ = 0;

+			}

+			if (s->vol > 64)

+				s->vol = 64;

+			if (fread(s->pek, s->len, 1, f) != 1)

+			{

+				loaderMsgBox("General I/O error during loading! Possibly corrupt module?");

+				return false;

+			}

+		}

+	}

+	return true;

+}

+static uint8_t stmTempoToBPM(uint8_t tempo) // ported from original ST2.3 replayer code

+{

+	const uint8_t slowdowns[16] = { 140, 50, 25, 15, 10, 7, 6, 4, 3, 3, 2, 2, 2, 2, 1, 1 };

+	uint16_t hz = 50;

+	hz -= ((slowdowns[tempo >> 4] * (tempo & 15)) >> 4); // can and will underflow

+	const uint32_t bpm = (hz << 1) + (hz >> 1); // BPM = hz * 2.5

+	return (uint8_t)CLAMP(bpm, 32, 255); // result can be slightly off, but close enough...

+}

--- /dev/null

+++ b/src/modloaders/ft2_load_xm.c

@@ -1,0 +1,626 @@

+// Fasttracker II (or compatible) XM loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_module_loader.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_tables.h"

+#include "../ft2_sysreqs.h"

+static uint8_t packedPattData[65536];

+/* ModPlug Tracker & OpenMPT supports up to 32 samples per instrument for XMs -  we don't.

+** For such modules, we use a temporary array here to store the extra sample data lengths

+** we need to skip to be able to load the file (we lose the extra samples, though...).

+*/

+static uint32_t extraSampleLengths[32-MAX_SMP_PER_INST];

+static bool loadInstrHeader(FILE *f, uint16_t i);

+static bool loadInstrSample(FILE *f, uint16_t i);

+static void unpackPatt(uint8_t *dst, uint8_t *src, uint16_t len, int32_t antChn);

+static bool loadPatterns(FILE *f, uint16_t antPtn);

+static void unpackPatt(uint8_t *dst, uint8_t *src, uint16_t len, int32_t antChn);

+static void sanitizeInstrument(instrTyp *ins);

+bool loadXM(FILE *f, uint32_t filesize)

+{

+	uint16_t i;

+	songHeaderTyp h;

+	if (filesize < sizeof (h))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (fread(&h, 1, sizeof (h), f) != sizeof (h))

+	{

+		loaderMsgBox("Error: This file is either not a module, or is not supported.");

+		return false;

+	}

+	if (h.ver < 0x0102 || h.ver > 0x0104)

+	{

+		loaderMsgBox("Error loading XM: Unsupported file version (v%01X.%02X).", (h.ver >> 8) & 15, h.ver & 0xFF);

+		return false;

+	}

+	if (h.len > MAX_ORDERS)

+	{

+		loaderMsgBox("Error loading XM: The song has more than 256 orders!");

+		return false;

+	}

+	if (h.antPtn > MAX_PATTERNS)

+	{

+		loaderMsgBox("Error loading XM: The song has more than 256 patterns!");

+		return false;

+	}

+	if (h.antChn == 0)

+	{

+		loaderMsgBox("Error loading XM: This file is corrupt.");

+		return false;

+	}

+	if (h.antInstrs > 256) // if >128 instruments, we fake-load up to 128 extra instruments and discard them

+	{

+		loaderMsgBox("Error loading XM: This file is corrupt.");

+		return false;

+	}

+	fseek(f, 60 + h.headerSize, SEEK_SET);

+	if (filesize != 336 && feof(f)) // 336 in length at this point = empty XM

+	{

+		loaderMsgBox("Error loading XM: The module is empty!");

+		return false;

+	}

+	memcpy(songTmp.name, h.name, 20);

+	songTmp.len = h.len;

+	songTmp.repS = h.repS;

+	songTmp.antChn = (uint8_t)h.antChn;

+	songTmp.speed = h.defSpeed;

+	songTmp.tempo = h.defTempo;

+	songTmp.ver = h.ver;

+	tmpLinearPeriodsFlag = h.flags & 1;

+	// non-FT2: clamp to max numbers that are okay for GUI

+	songTmp.speed = CLAMP(songTmp.speed, 1, 999);

+	songTmp.initialTempo = songTmp.tempo = CLAMP(songTmp.tempo, 1, 99);

+	if (songTmp.len == 0)

+		songTmp.len = 1; // songTmp.songTab is already empty

+	else

+		memcpy(songTmp.songTab, h.songTab, songTmp.len);

+	// some strange XMs have the order list padded with 0xFF, remove them!

+	for (int16_t j = 255; j >= 0; j--)

+	{

+		if (songTmp.songTab[j] != 0xFF)

+			break;

+		if (songTmp.len > j)

+			songTmp.len = j;

+	}

+	// even though XM supports 256 orders, FT2 supports only 255...

+	if (songTmp.len > 0xFF)

+		songTmp.len = 0xFF;

+	if (songTmp.ver < 0x0104)

+	{

+		// XM v1.02 and XM v1.03

+		for (i = 1; i <= h.antInstrs; i++)

+		{

+			if (!loadInstrHeader(f, i))

+				return false;

+		}

+		if (!loadPatterns(f, h.antPtn))

+			return false;

+		for (i = 1; i <= h.antInstrs; i++)

+		{

+			if (!loadInstrSample(f, i))

+				return false;

+		}

+	}

+	else

+	{

+		// XM v1.04 (latest version)

+		if (!loadPatterns(f, h.antPtn))

+			return false;

+		for (i = 1; i <= h.antInstrs; i++)

+		{

+			if (!loadInstrHeader(f, i))

+				return false;

+			if (!loadInstrSample(f, i))

+				return false;

+		}

+	}

+	// if we temporarily loaded more than 128 instruments, clear the extra allocated memory

+	if (h.antInstrs > MAX_INST)

+	{

+		for (i = MAX_INST+1; i <= h.antInstrs; i++)

+		{

+			if (instrTmp[i] != NULL)

+			{

+				free(instrTmp[i]);

+				instrTmp[i] = NULL;

+			}

+		}

+	}

+	/* We support loading XMs with up to 32 samples per instrument (ModPlug/OpenMPT),

+	** but only the first 16 will be loaded. Now make sure we set the number of samples

+	** back to max 16 in the headers before loading is done.

+	*/

+	bool instrHasMoreThan16Samples = false;

+	for (i = 1; i <= MAX_INST; i++)

+	{

+		if (instrTmp[i] != NULL && instrTmp[i]->antSamp > MAX_SMP_PER_INST)

+		{

+			instrHasMoreThan16Samples = true;

+			instrTmp[i]->antSamp = MAX_SMP_PER_INST;

+		}

+	}

+	if (songTmp.antChn > MAX_VOICES)

+	{

+		songTmp.antChn = MAX_VOICES;

+		loaderMsgBox("Warning: This XM contains >32 channels. The extra channels will be discarded!");

+	}

+	if (h.antInstrs > MAX_INST)

+		loaderMsgBox("Warning: This XM contains >128 instruments. The extra instruments will be discarded!");

+	if (instrHasMoreThan16Samples)

+		loaderMsgBox("Warning: This XM contains instrument(s) with >16 samples. The extra samples will be discarded!");

+	return true;

+}

+static bool loadInstrHeader(FILE *f, uint16_t i)

+{

+	uint8_t j;

+	uint32_t readSize;

+	instrHeaderTyp ih;

+	instrTyp *ins;

+	sampleHeaderTyp *src;

+	sampleTyp *s;

+	memset(extraSampleLengths, 0, sizeof (extraSampleLengths));

+	memset(&ih, 0, sizeof (ih));

+	fread(&ih.instrSize, 4, 1, f);

+	readSize = ih.instrSize;

+	// yes, some XMs can have a header size of 0, and it usually means 263 bytes (INSTR_HEADER_SIZE)

+	if (readSize == 0 || readSize > INSTR_HEADER_SIZE)

+		readSize = INSTR_HEADER_SIZE;

+	if (readSize < 4)

+	{

+		loaderMsgBox("Error loading XM: This file is corrupt (or not supported)!");

+		return false;

+	}

+	// load instrument data into temp buffer

+	fread(ih.name, readSize-4, 1, f); // -4 = skip ih.instrSize

+	// FT2 bugfix: skip instrument header data if instrSize is above INSTR_HEADER_SIZE

+	if (ih.instrSize > INSTR_HEADER_SIZE)

+		fseek(f, ih.instrSize-INSTR_HEADER_SIZE, SEEK_CUR);

+	if (ih.antSamp < 0 || ih.antSamp > 32)

+	{

+		loaderMsgBox("Error loading XM: This file is corrupt (or not supported)!");

+		return false;

+	}

+	if (i <= MAX_INST) // copy over instrument names

+		memcpy(songTmp.instrName[i], ih.name, 22);

+	if (ih.antSamp > 0)

+	{

+		if (!allocateTmpInstr(i))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		// copy instrument header elements to our instrument struct

+		ins = instrTmp[i];

+		memcpy(ins->ta, ih.ta, 96);

+		memcpy(ins->envVP, ih.envVP, 12*2*sizeof(int16_t));

+		memcpy(ins->envPP, ih.envPP, 12*2*sizeof(int16_t));

+		ins->envVPAnt = ih.envVPAnt;

+		ins->envPPAnt = ih.envPPAnt;

+		ins->envVSust = ih.envVSust;

+		ins->envVRepS = ih.envVRepS;

+		ins->envVRepE = ih.envVRepE;

+		ins->envPSust = ih.envPSust;

+		ins->envPRepS = ih.envPRepS;

+		ins->envPRepE = ih.envPRepE;

+		ins->envVTyp = ih.envVTyp;

+		ins->envPTyp = ih.envPTyp;

+		ins->vibTyp = ih.vibTyp;

+		ins->vibSweep = ih.vibSweep;

+		ins->vibDepth = ih.vibDepth;

+		ins->vibRate = ih.vibRate;

+		ins->fadeOut = ih.fadeOut;

+		ins->midiOn = (ih.midiOn == 1) ? true : false;

+		ins->midiChannel = ih.midiChannel;

+		ins->midiProgram = ih.midiProgram;

+		ins->midiBend = ih.midiBend;

+		ins->mute = (ih.mute == 1) ? true : false; // correct logic, don't change this!

+		ins->antSamp = ih.antSamp; // used in loadInstrSample()

+		sanitizeInstrument(ins);

+		int32_t sampleHeadersToRead = ih.antSamp;

+		if (sampleHeadersToRead > MAX_SMP_PER_INST)

+			sampleHeadersToRead = MAX_SMP_PER_INST;

+		if (fread(ih.samp, sampleHeadersToRead * sizeof (sampleHeaderTyp), 1, f) != 1)

+		{

+			loaderMsgBox("General I/O error during loading!");

+			return false;

+		}

+		// if instrument contains more than 16 sample headers (unsupported), skip them

+		if (ih.antSamp > MAX_SMP_PER_INST) // can only be 0..32 at this point

+		{

+			const int32_t samplesToSkip = ih.antSamp-MAX_SMP_PER_INST;

+			for (j = 0; j < samplesToSkip; j++)

+			{

+				fread(&extraSampleLengths[j], 4, 1, f); // used for skipping data in loadInstrSample()

+				fseek(f, sizeof (sampleHeaderTyp)-4, SEEK_CUR);

+			}

+		}

+		for (j = 0; j < sampleHeadersToRead; j++)

+		{

+			s = &instrTmp[i]->samp[j];

+			src = &ih.samp[j];

+			// copy sample header elements to our sample struct

+			s->len = src->len;

+			s->repS = src->repS;

+			s->repL = src->repL;

+			s->vol = src->vol;

+			s->fine = src->fine;

+			s->typ = src->typ;

+			s->pan = src->pan;

+			s->relTon = src->relTon;

+			memcpy(s->name, src->name, 22);

+			// dst->pek is set up later

+			// sanitize stuff broken/unsupported samples (FT2 doesn't do this, but we do!)

+			if (s->vol > 64)

+				s->vol = 64;

+			s->relTon = CLAMP(s->relTon, -48, 71);

+		}

+	}

+	return true;

+}

+static bool loadInstrSample(FILE *f, uint16_t i)

+{

+	if (instrTmp[i] == NULL)

+		return true; // empty instrument, let's just pretend it got loaded successfully

+	uint16_t k = instrTmp[i]->antSamp;

+	if (k > MAX_SMP_PER_INST)

+		k = MAX_SMP_PER_INST;

+	sampleTyp *s = instrTmp[i]->samp;

+	if (i > MAX_INST) // insNum > 128, just skip sample data

+	{

+		for (uint16_t j = 0; j < k; j++, s++)

+		{

+			if (s->len > 0)

+				fseek(f, s->len, SEEK_CUR);

+		}

+	}

+	else

+	{

+		for (uint16_t j = 0; j < k; j++, s++)

+		{

+			// FT2: a sample with both forward loop and pingpong loop set results in pingpong

+			if ((s->typ & 3) == 3)

+				s->typ &= 0xFE; // remove forward loop flag

+			int32_t l = s->len;

+			if (l <= 0)

+			{

+				s->len = 0;

+				s->repL = 0;

+				s->repS = 0;

+				if (s->typ & 32) // remove stereo flag if present

+					s->typ &= ~32;

+			}

+			else

+			{

+				int32_t bytesToSkip = 0;

+				if (l > MAX_SAMPLE_LEN)

+				{

+					bytesToSkip = l - MAX_SAMPLE_LEN;

+					l = MAX_SAMPLE_LEN;

+				}

+				if (!allocateTmpSmpData(s, l))

+				{

+					loaderMsgBox("Not enough memory!");

+					return false;

+				}

+				const int32_t bytesRead = (int32_t)fread(s->pek, 1, l, f);

+				if (bytesRead < l)

+				{

+					const int32_t bytesToClear = l - bytesRead;

+					memset(&s->pek[bytesRead], 0, bytesToClear);

+				}

+				if (bytesToSkip > 0)

+					fseek(f, bytesToSkip, SEEK_CUR);

+				delta2Samp(s->pek, l, s->typ);

+				if (s->typ & 32) // stereo sample - already downmixed to mono in delta2samp()

+				{

+					s->typ &= ~32; // remove stereo flag

+					s->len >>= 1;

+					s->repL >>= 1;

+					s->repS >>= 1;

+					reallocateTmpSmpData(s, s->len); // dealloc unused memory

+				}

+			}

+			// NON-FT2 FIX: Align to 2-byte if 16-bit sample

+			if (s->typ & 16)

+			{

+				s->repL &= 0xFFFFFFFE;

+				s->repS &= 0xFFFFFFFE;

+				s->len &= 0xFFFFFFFE;

+			}

+		}

+	}

+	if (instrTmp[i]->antSamp > MAX_SMP_PER_INST)

+	{

+		const int32_t samplesToSkip = instrTmp[i]->antSamp-MAX_SMP_PER_INST;

+		for (i = 0; i < samplesToSkip; i++)

+		{

+			if (extraSampleLengths[i] > 0)

+				fseek(f, extraSampleLengths[i], SEEK_CUR);

+		}

+	}

+	return true;

+}

+static bool loadPatterns(FILE *f, uint16_t antPtn)

+{

+	uint8_t tmpLen;

+	patternHeaderTyp ph;

+	bool pattLenWarn = false;

+	for (uint16_t i = 0; i < antPtn; i++)

+	{

+		if (fread(&ph.patternHeaderSize, 4, 1, f) != 1)

+			goto pattCorrupt;

+		if (fread(&ph.typ, 1, 1, f) != 1)

+			goto pattCorrupt;

+		ph.pattLen = 0;

+		if (songTmp.ver == 0x0102)

+		{

+			if (fread(&tmpLen, 1, 1, f) != 1)

+				goto pattCorrupt;

+			if (fread(&ph.dataLen, 2, 1, f) != 1)

+				goto pattCorrupt;

+			ph.pattLen = tmpLen + 1; // +1 in v1.02

+			if (ph.patternHeaderSize > 8)

+				fseek(f, ph.patternHeaderSize - 8, SEEK_CUR);

+		}

+		else

+		{

+			if (fread(&ph.pattLen, 2, 1, f) != 1)

+				goto pattCorrupt;

+			if (fread(&ph.dataLen, 2, 1, f) != 1)

+				goto pattCorrupt;

+			if (ph.patternHeaderSize > 9)

+				fseek(f, ph.patternHeaderSize - 9, SEEK_CUR);

+		}

+		if (feof(f))

+			goto pattCorrupt;

+		pattLensTmp[i] = ph.pattLen;

+		if (pattLensTmp[i] > MAX_PATT_LEN)

+		{

+			pattLensTmp[i] = MAX_PATT_LEN;

+			pattLenWarn = true;

+		}

+		if (ph.dataLen > 0)

+		{

+			if (!allocateTmpPatt(i, pattLensTmp[i]))

+			{

+				loaderMsgBox("Not enough memory!");

+				return false;

+			}

+			if (fread(packedPattData, 1, ph.dataLen, f) != ph.dataLen)

+				goto pattCorrupt;

+			unpackPatt((uint8_t *)pattTmp[i], packedPattData, pattLensTmp[i], songTmp.antChn);

+			clearUnusedChannels(pattTmp[i], pattLensTmp[i], songTmp.antChn);

+		}

+		if (tmpPatternEmpty(i))

+		{

+			if (pattTmp[i] != NULL)

+			{

+				free(pattTmp[i]);

+				pattTmp[i] = NULL;

+			}

+			pattLensTmp[i] = 64;

+		}

+	}

+	if (pattLenWarn)

+		loaderMsgBox("This module contains pattern(s) with a length above 256! They will be truncated.");

+	return true;

+pattCorrupt:

+	loaderMsgBox("Error loading XM: This file is corrupt!");

+	return false;

+}

+static void unpackPatt(uint8_t *dst, uint8_t *src, uint16_t len, int32_t antChn)

+{

+	int32_t j;

+	if (dst == NULL)

+		return;

+	const int32_t srcEnd = len * (sizeof (tonTyp) * antChn);

+	int32_t srcIdx = 0;

+	int32_t numChannels = antChn;

+	if (numChannels > MAX_VOICES)

+		numChannels = MAX_VOICES;

+	const int32_t pitch = sizeof (tonTyp) * (MAX_VOICES - antChn);

+	for (int32_t i = 0; i < len; i++)

+	{

+		for (j = 0; j < numChannels; j++)

+		{

+			if (srcIdx >= srcEnd)

+				return; // error!

+			const uint8_t note = *src++;

+			if (note & 0x80)

+			{

+				*dst++ = (note & 0x01) ? *src++ : 0;

+				*dst++ = (note & 0x02) ? *src++ : 0;

+				*dst++ = (note & 0x04) ? *src++ : 0;

+				*dst++ = (note & 0x08) ? *src++ : 0;

+				*dst++ = (note & 0x10) ? *src++ : 0;

+			}

+			else

+			{

+				*dst++ = note;

+				*dst++ = *src++;

+				*dst++ = *src++;

+				*dst++ = *src++;

+				*dst++ = *src++;

+			}

+			// if note is overflowing (>97), remove it

+			if (*(dst-5) > 97)

+				*(dst-5) = 0;

+			// non-FT2 security fix: if effect is above 35 (Z), clear effect and parameter

+			if (*(dst-2) > 35)

+			{

+				*(dst-2) = 0;

+				*(dst-1) = 0;

+			}

+			srcIdx += sizeof (tonTyp);

+		}

+		// if more than 32 channels, skip rest of the channels for this row

+		for (; j < antChn; j++)

+		{

+			if (srcIdx >= srcEnd)

+				return; // error!

+			const uint8_t note = *src++;

+			if (note & 0x80)

+			{

+				if (note & 0x01) src++;

+				if (note & 0x02) src++;

+				if (note & 0x04) src++;

+				if (note & 0x08) src++;

+				if (note & 0x10) src++;

+			}

+			else

+			{

+				src++;

+				src++;

+				src++;

+				src++;

+			}

+			srcIdx += sizeof (tonTyp);

+		}

+		// if song has <32 channels, align pointer to next row (skip unused channels)

+		if (antChn < MAX_VOICES)

+			dst += pitch;

+	}

+}

+static void sanitizeInstrument(instrTyp *ins) // FT2 doesn't do this, but we do!

+{

+	ins->midiProgram = CLAMP(ins->midiProgram, 0, 127);

+	ins->midiBend = CLAMP(ins->midiBend, 0, 36);

+	if (ins->midiChannel > 15) ins->midiChannel = 15;

+	if (ins->vibDepth > 0x0F) ins->vibDepth = 0x0F;

+	if (ins->vibRate > 0x3F) ins->vibRate = 0x3F;

+	if (ins->vibTyp > 3) ins->vibTyp = 0;

+	for (int32_t i = 0; i < 96; i++)

+	{

+		if (ins->ta[i] >= MAX_SMP_PER_INST)

+			ins->ta[i] = MAX_SMP_PER_INST-1;

+	}

+	if (ins->envVPAnt > 12) ins->envVPAnt = 12;

+	if (ins->envVRepS > 11) ins->envVRepS = 11;

+	if (ins->envVRepE > 11) ins->envVRepE = 11;

+	if (ins->envVSust > 11) ins->envVSust = 11;

+	if (ins->envPPAnt > 12) ins->envPPAnt = 12;

+	if (ins->envPRepS > 11) ins->envPRepS = 11;

+	if (ins->envPRepE > 11) ins->envPRepE = 11;

+	if (ins->envPSust > 11) ins->envPSust = 11;

+	for (int32_t  i= 0; i < 12; i++)

+	{

+		if ((uint16_t)ins->envVP[i][0] > 32767) ins->envVP[i][0] = 32767;

+		if ((uint16_t)ins->envPP[i][0] > 32767) ins->envPP[i][0] = 32767;

+		if ((uint16_t)ins->envVP[i][1] > 64) ins->envVP[i][1] = 64;

+		if ((uint16_t)ins->envPP[i][1] > 63) ins->envPP[i][1] = 63;

+	}

+}

--- /dev/null

+++ b/src/smploaders/ft2_load_aiff.c

@@ -1,0 +1,510 @@

+// Apple AIFF sample loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_audio.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_sysreqs.h"

+#include "../ft2_sample_loader.h"

+static int32_t getAIFFRate(uint8_t *in);

+static bool aiffIsStereo(FILE *f); // only ran on files that are confirmed to be AIFFs

+bool loadAIFF(FILE *f, uint32_t filesize)

+{

+	char compType[4];

+	int8_t *audioDataS8;

+	uint8_t sampleRateBytes[10], *audioDataU8;

+	int16_t *audioDataS16, *audioDataS16_2, smp16;

+	uint16_t numChannels, bitDepth;

+	int32_t smp32, *audioDataS32;

+	uint32_t i, blockName, blockSize;

+	uint32_t offset, len32;

+	fseek(f, 8, SEEK_SET);

+	fread(compType, 1, 4, f);

+	rewind(f);

+	if (!memcmp(compType, "AIFC", 4))

+	{

+		loaderMsgBox("Error loading sample: This AIFF type (AIFC) is not supported!");

+		return false;

+	}

+	if (filesize < 12)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	uint32_t commPtr = 0, commLen = 0;

+	uint32_t ssndPtr = 0, ssndLen = 0;

+	fseek(f, 12, SEEK_SET);

+	while (!feof(f) && (uint32_t)ftell(f) < filesize-12)

+	{

+		fread(&blockName, 4, 1, f); if (feof(f)) break;

+		fread(&blockSize, 4, 1, f); if (feof(f)) break;

+		blockName = SWAP32(blockName);

+		blockSize = SWAP32(blockSize);

+		switch (blockName)

+		{

+			case 0x434F4D4D: // "COMM"

+			{

+				commPtr = ftell(f);

+				commLen = blockSize;

+			}

+			break;

+			case 0x53534E44: // "SSND"

+			{

+				ssndPtr = ftell(f);

+				ssndLen = blockSize;

+			}

+			break;

+			default: break;

+		}

+		fseek(f, blockSize + (blockSize & 1), SEEK_CUR);

+	}

+	if (commPtr == 0 || commLen < 18 || ssndPtr == 0)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	// kludge for strange AIFFs

+	if (ssndLen == 0)

+		ssndLen = filesize - ssndPtr;

+	if (ssndPtr+ssndLen > (uint32_t)filesize)

+		ssndLen = filesize - ssndPtr;

+	fseek(f, commPtr, SEEK_SET);

+	fread(&numChannels, 2, 1, f); numChannels = SWAP16(numChannels);

+	fseek(f, 4, SEEK_CUR);

+	fread(&bitDepth, 2, 1, f); bitDepth = SWAP16(bitDepth);

+	fread(sampleRateBytes, 1, 10, f);

+	if (numChannels != 1 && numChannels != 2)

+	{

+		loaderMsgBox("Error loading sample: Unsupported amounts of channels!");

+		return false;

+	}

+	if (bitDepth != 8 && bitDepth != 16 && bitDepth != 24 && bitDepth != 32)

+	{

+		loaderMsgBox("Error loading sample: Unsupported bitdepth!");

+		return false;

+	}

+	// read compression type (if present)

+	if (commLen > 18)

+	{

+		fread(&compType, 1, 4, f);

+		if (memcmp(compType, "NONE", 4) != 0)

+		{

+			loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+			return false;

+		}

+	}

+	uint32_t sampleRate = getAIFFRate(sampleRateBytes);

+	// sample data chunk

+	fseek(f, ssndPtr, SEEK_SET);

+	fread(&offset, 4, 1, f);

+	if (offset > 0)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	fseek(f, 4, SEEK_CUR);

+	ssndLen -= 8; // don't include offset and blockSize datas

+	uint32_t sampleLength = ssndLen;

+	if (sampleLength > MAX_SAMPLE_LEN)

+		sampleLength = MAX_SAMPLE_LEN;

+	int16_t stereoSampleLoadMode = -1;

+	if (aiffIsStereo(f))

+		stereoSampleLoadMode = loaderSysReq(5, "System request", "This is a stereo sample...");

+	// read sample data

+	if (bitDepth == 8)

+	{

+		// 8-BIT SIGNED PCM

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataS8 = (int8_t *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					for (i = 1; i < sampleLength; i++)

+						audioDataS8[i] = audioDataS8[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS8[i] = audioDataS8[(i * 2) + 1];

+					audioDataS8[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						smp16 = (audioDataS8[(i * 2) + 0] + audioDataS8[(i * 2) + 1]) >> 1;

+						audioDataS8[i] = (int8_t)smp16;

+					}

+					audioDataS8[i] = 0;

+				}

+				break;

+			}

+		}

+	}

+	else if (bitDepth == 16)

+	{

+		// 16-BIT SIGNED PCM

+		sampleLength /= 2;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength*2))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 2, f) != 2)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		// fix endianness

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		for (i = 0; i < sampleLength; i++)

+			audioDataS16[i] = SWAP16(audioDataS16[i]);

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					for (i = 1; i < sampleLength; i++)

+						audioDataS16[i] = audioDataS16[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS16[i] = audioDataS16[(i * 2) + 1];

+					audioDataS16[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						smp32 = (audioDataS16[(i * 2) + 0] + audioDataS16[(i * 2) + 1]) >> 1;

+						audioDataS16[i] = (int16_t)smp32;

+					}

+					audioDataS16[i] = 0;

+				}

+				break;

+			}

+		}

+		sampleLength *= 2;

+		tmpSmp.typ |= 16;

+	}

+	else if (bitDepth == 24)

+	{

+		// 24-BIT SIGNED PCM

+		sampleLength /= 3;

+		if (!allocateTmpSmpData(&tmpSmp, (sampleLength * 4) * 2))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(&tmpSmp.pek[sampleLength], sampleLength, 3, f) != 3)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataS32 = (int32_t *)tmpSmp.pek;

+		// convert to 32-bit

+		audioDataU8 = (uint8_t *)tmpSmp.pek + sampleLength;

+		for (i = 0; i < sampleLength; i++)

+		{

+			audioDataS32[i] = (audioDataU8[0] << 24) | (audioDataU8[1] << 16) | (audioDataU8[2] << 8);

+			audioDataU8 += 3;

+		}

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					for (i = 1; i < sampleLength; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 1];

+					audioDataS32[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						int64_t smp64 = audioDataS32[(i * 2) + 0];

+						smp64 += audioDataS32[(i * 2) + 1];

+						smp64 >>= 1;

+						audioDataS32[i] = (int32_t)smp64;

+					}

+					audioDataS32[i] = 0;

+				}

+				break;

+			}

+		}

+		normalizeSigned32Bit(audioDataS32, sampleLength);

+		// downscale to 16-bit (ultra fast method!)

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1; // yes, this is aligned to words

+		for (i = 0; i < sampleLength; i++, audioDataS16_2++)

+			audioDataS16[i] = audioDataS16_2[i];

+		sampleLength *= 2;

+		tmpSmp.typ |= 16;

+	}

+	else if (bitDepth == 32)

+	{

+		// 32-BIT SIGNED PCM

+		sampleLength /= 4;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength * 4))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		// fix endianness

+		audioDataS32 = (int32_t *)tmpSmp.pek;

+		for (i = 0; i < sampleLength; i++)

+			audioDataS32[i] = SWAP32(audioDataS32[i]);

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					for (i = 1; i < sampleLength; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 1];

+					audioDataS32[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						int64_t smp64 = audioDataS32[(i * 2) + 0];

+						smp64 += audioDataS32[(i * 2) + 1];

+						smp64 >>= 1;

+						audioDataS32[i] = (int32_t)smp64;

+					}

+					audioDataS32[i] = 0;

+				}

+				break;

+			}

+		}

+		normalizeSigned32Bit(audioDataS32, sampleLength);

+		// downscale to 16-bit (ultra fast method!)

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1; // yes, this is aligned to words

+		for (i = 0; i < sampleLength; i++, audioDataS16_2++)

+			audioDataS16[i] = audioDataS16_2[i];

+		sampleLength *= 2;

+		tmpSmp.typ |= 16;

+	}

+	reallocateTmpSmpData(&tmpSmp, sampleLength); // readjust memory needed

+	tmpSmp.len = sampleLength;

+	tmpSmp.vol = 64;

+	tmpSmp.pan = 128;

+	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

+	return true;

+}

+static int32_t getAIFFRate(uint8_t *in)

+{

+	int32_t exp = (int32_t)(((in[0] & 0x7F) << 8) | in[1]);

+	uint32_t lo = (in[2] << 24) | (in[3] << 16) | (in[4] << 8) | in[5];

+	uint32_t hi = (in[6] << 24) | (in[7] << 16) | (in[8] << 8) | in[9];

+	if (exp == 0 && lo == 0 && hi == 0)

+		return 0;

+	exp -= 16383;

+	double dOut = ldexp(lo, -31 + exp) + ldexp(hi, -63 + exp);

+	return (int32_t)(dOut + 0.5); // rounded

+}

+static bool aiffIsStereo(FILE *f) // only ran on files that are confirmed to be AIFFs

+{

+	uint16_t numChannels;

+	uint32_t chunkID, chunkSize;

+	uint32_t oldPos = ftell(f);

+	fseek(f, 0, SEEK_END);

+	int32_t filesize = ftell(f);

+	if (filesize < 12)

+	{

+		fseek(f, oldPos, SEEK_SET);

+		return false;

+	}

+	fseek(f, 12, SEEK_SET);

+	uint32_t commPtr = 0;

+	uint32_t commLen = 0;

+	int32_t bytesRead = 0;

+	while (!feof(f) && bytesRead < filesize-12)

+	{

+		fread(&chunkID, 4, 1, f); chunkID = SWAP32(chunkID); if (feof(f)) break;

+		fread(&chunkSize, 4, 1, f); chunkSize = SWAP32(chunkSize); if (feof(f)) break;

+		int32_t endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

+		switch (chunkID)

+		{

+			case 0x434F4D4D: // "COMM"

+			{

+				commPtr = ftell(f);

+				commLen = chunkSize;

+			}

+			break;

+			default: break;

+		}

+		bytesRead += (chunkSize + (chunkSize & 1));

+		fseek(f, endOfChunk, SEEK_SET);

+	}

+	if (commPtr == 0 || commLen < 2)

+	{

+		fseek(f, oldPos, SEEK_SET);

+		return false;

+	}

+	fseek(f, commPtr, SEEK_SET);

+	fread(&numChannels, 2, 1, f); numChannels = SWAP16(numChannels);

+	fseek(f, oldPos, SEEK_SET);

+	return (numChannels == 2);

+}

--- /dev/null

+++ b/src/smploaders/ft2_load_iff.c

@@ -1,0 +1,179 @@

+// IFF (Amiga/FT2) sample loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_audio.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_sysreqs.h"

+#include "../ft2_sample_loader.h"

+bool loadIFF(FILE *f, uint32_t filesize)

+{

+	char hdr[4+1];

+	uint32_t sampleLength, sampleVol, sampleLoopStart, sampleLoopLength, sampleRate;

+	if (filesize < 12)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	fseek(f, 8, SEEK_SET);

+	fread(hdr, 1, 4, f);

+	hdr[4] = '\0';

+	bool is16Bit = !strncmp(hdr, "16SV", 4);

+	uint32_t vhdrPtr = 0, vhdrLen = 0;

+	uint32_t bodyPtr = 0, bodyLen = 0;

+	uint32_t namePtr = 0, nameLen = 0;

+	fseek(f, 12, SEEK_SET);

+	while (!feof(f) && (uint32_t)ftell(f) < filesize-12)

+	{

+		uint32_t blockName, blockSize;

+		fread(&blockName, 4, 1, f); if (feof(f)) break;

+		fread(&blockSize, 4, 1, f); if (feof(f)) break;

+		blockName = SWAP32(blockName);

+		blockSize = SWAP32(blockSize);

+		switch (blockName)

+		{

+			case 0x56484452: // VHDR

+			{

+				vhdrPtr = ftell(f);

+				vhdrLen = blockSize;

+			}

+			break;

+			case 0x4E414D45: // NAME

+			{

+				namePtr = ftell(f);

+				nameLen = blockSize;

+			}

+			break;

+			case 0x424F4459: // BODY

+			{

+				bodyPtr = ftell(f);

+				bodyLen = blockSize;

+			}

+			break;

+			default: break;

+		}

+		fseek(f, blockSize + (blockSize & 1), SEEK_CUR);

+	}

+	if (vhdrPtr == 0 || vhdrLen < 20 || bodyPtr == 0)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	// kludge for some really strange IFFs

+	if (bodyLen == 0)

+		bodyLen = filesize - bodyPtr;

+	if (bodyPtr+bodyLen > (uint32_t)filesize)

+		bodyLen = filesize - bodyPtr;

+	fseek(f, vhdrPtr, SEEK_SET);

+	fread(&sampleLoopStart,  4, 1, f); sampleLoopStart = SWAP32(sampleLoopStart);

+	fread(&sampleLoopLength, 4, 1, f); sampleLoopLength = SWAP32(sampleLoopLength);

+	fseek(f, 4, SEEK_CUR);

+	fread(&sampleRate, 2, 1, f); sampleRate = SWAP16(sampleRate);

+	fseek(f, 1, SEEK_CUR);

+	if (fgetc(f) != 0) // sample type

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported!");

+		return false;

+	}

+	fread(&sampleVol, 4, 1, f); sampleVol = SWAP32(sampleVol);

+	if (sampleVol > 65535)

+		sampleVol = 65535;

+	sampleVol = (sampleVol + 512) / 1024; // rounded

+	sampleLength = bodyLen;

+	if (sampleLength > MAX_SAMPLE_LEN)

+		sampleLength = MAX_SAMPLE_LEN;

+	if (sampleLoopStart >= MAX_SAMPLE_LEN || sampleLoopLength > MAX_SAMPLE_LEN)

+	{

+		sampleLoopStart = 0;

+		sampleLoopLength = 0;

+	}

+	if (sampleLoopStart+sampleLoopLength > sampleLength)

+	{

+		sampleLoopStart = 0;

+		sampleLoopLength = 0;

+	}

+	if (sampleLoopStart > sampleLength-2)

+	{

+		sampleLoopStart = 0;

+		sampleLoopLength = 0;

+	}

+	if (!allocateTmpSmpData(&tmpSmp, sampleLength))

+	{

+		loaderMsgBox("Not enough memory!");

+		return false;

+	}

+	fseek(f, bodyPtr, SEEK_SET);

+	if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

+	{

+		loaderMsgBox("General I/O error during loading! Is the file in use?");

+		return false;

+	}

+	tmpSmp.len = sampleLength;

+	if (sampleLoopLength > 2)

+	{

+		tmpSmp.repS = sampleLoopStart;

+		tmpSmp.repL = sampleLoopLength;

+		tmpSmp.typ |= 1;

+	}

+	if (is16Bit)

+	{

+		tmpSmp.len  &= 0xFFFFFFFE;

+		tmpSmp.repS &= 0xFFFFFFFE;

+		tmpSmp.repL &= 0xFFFFFFFE;

+		tmpSmp.typ |= 16;

+	}

+	tmpSmp.vol = (uint8_t)sampleVol;

+	tmpSmp.pan = 128;

+	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

+	// set name

+	if (namePtr != 0 && nameLen > 0)

+	{

+		fseek(f, namePtr, SEEK_SET);

+		if (nameLen > 21)

+		{

+			fread(tmpSmp.name, 1, 21, f);

+			tmpSmp.name[21] = '\0';

+		}

+		else

+		{

+			memset(tmpSmp.name, 0, 22);

+			fread(tmpSmp.name, 1, nameLen, f);

+		}

+		smpFilenameSet = true;

+	}

+	return true;

+}

--- /dev/null

+++ b/src/smploaders/ft2_load_raw.c

@@ -1,0 +1,30 @@

+// RAW (header-less) sample loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_sysreqs.h"

+#include "../ft2_sample_loader.h"

+bool loadRAW(FILE *f, uint32_t filesize)

+{

+	if (!allocateTmpSmpData(&tmpSmp, filesize))

+	{

+		loaderMsgBox("Not enough memory!");

+		return false;

+	}

+	if (fread(tmpSmp.pek, filesize, 1, f) != 1)

+	{

+		okBoxThreadSafe(0, "System message", "General I/O error during loading! Is the file in use?");

+		return false;

+	}

+	tmpSmp.len = filesize;

+	tmpSmp.vol = 64;

+	tmpSmp.pan = 128;

+	return true;

+}

--- /dev/null

+++ b/src/smploaders/ft2_load_wav.c

@@ -1,0 +1,746 @@

+// WAV sample loader

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "../ft2_header.h"

+#include "../ft2_audio.h"

+#include "../ft2_sample_ed.h"

+#include "../ft2_sysreqs.h"

+#include "../ft2_sample_loader.h"

+enum

+{

+	WAV_FORMAT_PCM = 1,

+	WAV_FORMAT_IEEE_FLOAT = 3

+};

+static bool wavIsStereo(FILE *f);

+bool loadWAV(FILE *f, uint32_t filesize)

+{

+	uint8_t *audioDataU8;

+	int16_t *audioDataS16, *audioDataS16_2, *ptr16;

+	uint16_t audioFormat, numChannels, bitsPerSample;

+	int32_t *audioDataS32;

+	uint32_t i, sampleRate, sampleLength;

+	uint32_t len32;

+	float *fAudioDataFloat;

+	double *dAudioDataDouble;

+	if (filesize < 12)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	uint32_t fmtPtr  = 0, fmtLen  = 0;

+	uint32_t dataPtr = 0, dataLen = 0;

+	uint32_t inamPtr = 0, inamLen = 0;

+	uint32_t xtraPtr = 0, xtraLen = 0;

+	uint32_t smplPtr = 0, smplLen = 0;

+	// look for wanted chunks and set up pointers + lengths

+	fseek(f, 12, SEEK_SET);

+	uint32_t bytesRead = 0;

+	while (!feof(f) && bytesRead < filesize-12)

+	{

+		uint32_t chunkID, chunkSize;

+		fread(&chunkID, 4, 1, f); if (feof(f)) break;

+		fread(&chunkSize, 4, 1, f); if (feof(f)) break;

+		uint32_t endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

+		switch (chunkID)

+		{

+			case 0x20746D66: // "fmt "

+			{

+				fmtPtr = ftell(f);

+				fmtLen = chunkSize;

+			}

+			break;

+			case 0x61746164: // "data"

+			{

+				dataPtr = ftell(f);

+				dataLen = chunkSize;

+			}

+			break;

+			case 0x5453494C: // "LIST"

+			{

+				if (chunkSize >= 4)

+				{

+					fread(&chunkID, 4, 1, f);

+					if (chunkID == 0x4F464E49) // "INFO"

+					{

+						bytesRead = 0;

+						while (!feof(f) && bytesRead < chunkSize)

+						{

+							fread(&chunkID, 4, 1, f);

+							fread(&chunkSize, 4, 1, f);

+							switch (chunkID)

+							{

+								case 0x4D414E49: // "INAM"

+								{

+									inamPtr = ftell(f);

+									inamLen = chunkSize;

+								}

+								break;

+								default: break;

+							}

+							bytesRead += (chunkSize + (chunkSize & 1));

+						}

+					}

+				}

+			}

+			break;

+			case 0x61727478: // "xtra"

+			{

+				xtraPtr = ftell(f);

+				xtraLen = chunkSize;

+			}

+			break;

+			case 0x6C706D73: // "smpl"

+			{

+				smplPtr = ftell(f);

+				smplLen = chunkSize;

+			}

+			break;

+			default: break;

+		}

+		bytesRead += (chunkSize + (chunkSize & 1));

+		fseek(f, endOfChunk, SEEK_SET);

+	}

+	// we need at least "fmt " and "data" - check if we found them sanely

+	if (fmtPtr == 0 || fmtLen < 16 || dataPtr == 0 || dataLen == 0)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	// ---- READ "fmt " CHUNK ----

+	fseek(f, fmtPtr, SEEK_SET);

+	fread(&audioFormat, 2, 1, f);

+	fread(&numChannels, 2, 1, f);

+	fread(&sampleRate,  4, 1, f);

+	fseek(f, 6, SEEK_CUR); // unneeded

+	fread(&bitsPerSample, 2, 1, f);

+	sampleLength = dataLen;

+	// ---------------------------

+	// test if the WAV is compatible with our loader

+	if (sampleRate == 0 || sampleLength == 0 || sampleLength >= filesize)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported or is invalid!");

+		return false;

+	}

+	if (audioFormat != WAV_FORMAT_PCM && audioFormat != WAV_FORMAT_IEEE_FLOAT)

+	{

+		loaderMsgBox("Error loading sample: The sample is not supported!");

+		return false;

+	}

+	if (numChannels == 0 || numChannels > 2)

+	{

+		loaderMsgBox("Error loading sample: Unsupported number of channels!");

+		return false;

+	}

+	if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample != 32 && bitsPerSample != 64)

+	{

+		loaderMsgBox("Error loading sample: Unsupported bitdepth!");

+		return false;

+	}

+	if (bitsPerSample != 8 && bitsPerSample != 16 && bitsPerSample != 24 && bitsPerSample != 32 && bitsPerSample != 64)

+	{

+		loaderMsgBox("Error loading sample: Unsupported bitdepth!");

+		return false;

+	}

+	// ---- READ SAMPLE DATA ----

+	fseek(f, dataPtr, SEEK_SET);

+	if (sampleLength > MAX_SAMPLE_LEN)

+		sampleLength = MAX_SAMPLE_LEN;

+	int16_t stereoSampleLoadMode = -1;

+	if (wavIsStereo(f))

+		stereoSampleLoadMode = loaderSysReq(5, "System request", "This is a stereo sample...");

+	if (bitsPerSample == 8) // 8-BIT INTEGER SAMPLE

+	{

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 1, f) != 1)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataU8 = (uint8_t *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						audioDataU8[i] = audioDataU8[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataU8[i] = audioDataU8[(i * 2) + 1];

+					audioDataU8[i] = 128;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataU8[i] = (audioDataU8[(i * 2) + 0] + audioDataU8[(i * 2) + 1]) >> 1;

+					audioDataU8[i] = 128;

+				}

+				break;

+			}

+		}

+		// convert from unsigned to signed

+		for (i = 0; i < sampleLength; i++)

+			tmpSmp.pek[i] ^= 0x80;

+	}

+	else if (bitsPerSample == 16) // 16-BIT INTEGER SAMPLE

+	{

+		sampleLength /= 2;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength*2))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 2, f) != 2)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						audioDataS16[i] = audioDataS16[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS16[i] = audioDataS16[(i * 2) + 1];

+					audioDataS16[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						int32_t smp32 = audioDataS16[(i * 2) + 0] + audioDataS16[(i * 2) + 1];

+						audioDataS16[i] = (int16_t)(smp32 >> 1);

+					}

+					audioDataS16[i] = 0;

+				}

+				break;

+			}

+		}

+		sampleLength *= 2;

+		tmpSmp.typ |= 16; // 16-bit

+	}

+	else if (bitsPerSample == 24) // 24-BIT INTEGER SAMPLE

+	{

+		sampleLength /= 3;

+		if (!allocateTmpSmpData(&tmpSmp, (sampleLength * 4) * 2))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(&tmpSmp.pek[sampleLength], sampleLength, 3, f) != 3)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataS32 = (int32_t *)tmpSmp.pek;

+		// convert to 32-bit

+		audioDataU8 = (uint8_t *)tmpSmp.pek + sampleLength;

+		for (i = 0; i < sampleLength; i++)

+		{

+			audioDataS32[i] = (audioDataU8[2] << 24) | (audioDataU8[1] << 16) | (audioDataU8[0] << 8);

+			audioDataU8 += 3;

+		}

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 1];

+					audioDataS32[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						int64_t smp64 = audioDataS32[(i * 2) + 0];

+						smp64 += audioDataS32[(i * 2) + 1];

+						smp64 >>= 1;

+						audioDataS32[i] = (int32_t)smp64;

+					}

+					audioDataS32[i] = 0;

+				}

+				break;

+			}

+		}

+		normalizeSigned32Bit(audioDataS32, sampleLength);

+		// downscale to 16-bit (ultra fast method!)

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1; // yes, this is aligned to words

+		for (i = 0; i < sampleLength; i++, audioDataS16_2++)

+			audioDataS16[i] = audioDataS16_2[i];

+		sampleLength *= 2;

+		tmpSmp.typ |= 16; // 16-bit

+	}

+	else if (audioFormat == WAV_FORMAT_PCM && bitsPerSample == 32) // 32-BIT INTEGER SAMPLE

+	{

+		sampleLength /= 4;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength * 4))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		audioDataS32 = (int32_t *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						audioDataS32[i] = audioDataS32[(i * 2) + 1];

+					audioDataS32[i] = 0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+					{

+						int64_t smp64 = audioDataS32[(i * 2) + 0];

+						smp64 += audioDataS32[(i * 2) + 1];

+						smp64 >>= 1;

+						audioDataS32[i] = (int32_t)smp64;

+					}

+					audioDataS32[i] = 0;

+				}

+				break;

+			}

+		}

+		normalizeSigned32Bit(audioDataS32, sampleLength);

+		// downscale to 16-bit (ultra fast method!)

+		audioDataS16 = (int16_t *)tmpSmp.pek;

+		audioDataS16_2 = (int16_t *)tmpSmp.pek + 1; // yes, this is aligned to words

+		for (i = 0; i < sampleLength; i++, audioDataS16_2++)

+			audioDataS16[i] = audioDataS16_2[i];

+		sampleLength *= 2;

+		tmpSmp.typ |= 16; // 16-bit

+	}

+	else if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample == 32) // 32-BIT FLOATING POINT SAMPLE

+	{

+		sampleLength /= 4;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength * 4))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 4, f) != 4)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		fAudioDataFloat = (float *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						fAudioDataFloat[i] = fAudioDataFloat[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						fAudioDataFloat[i] = fAudioDataFloat[(i * 2) + 1];

+					fAudioDataFloat[i] = 0.0f;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						fAudioDataFloat[i] = (fAudioDataFloat[(i * 2) + 0] + fAudioDataFloat[(i * 2) + 1]) * 0.5f;

+					fAudioDataFloat[i] = 0.0f;

+				}

+				break;

+			}

+		}

+		normalize32BitFloatToSigned16Bit(fAudioDataFloat, sampleLength);

+		ptr16 = (int16_t *)tmpSmp.pek;

+		for (i = 0; i < sampleLength; i++)

+			ptr16[i] = (int16_t)fAudioDataFloat[i]; // should use SIMD if available

+		sampleLength *= 2;

+		tmpSmp.typ |= 16; // 16-bit

+	}

+	else if (audioFormat == WAV_FORMAT_IEEE_FLOAT && bitsPerSample == 64) // 64-BIT FLOATING POINT SAMPLE

+	{

+		sampleLength /= 8;

+		if (!allocateTmpSmpData(&tmpSmp, sampleLength * 8))

+		{

+			loaderMsgBox("Not enough memory!");

+			return false;

+		}

+		if (fread(tmpSmp.pek, sampleLength, 8, f) != 8)

+		{

+			loaderMsgBox("General I/O error during loading! Is the file in use?");

+			return false;

+		}

+		dAudioDataDouble = (double *)tmpSmp.pek;

+		// stereo conversion

+		if (numChannels == 2)

+		{

+			sampleLength /= 2;

+			switch (stereoSampleLoadMode)

+			{

+				case STEREO_SAMPLE_READ_LEFT:

+				{

+					// remove right channel data

+					for (i = 1; i < sampleLength; i++)

+						dAudioDataDouble[i] = dAudioDataDouble[(i * 2) + 0];

+				}

+				break;

+				case STEREO_SAMPLE_READ_RIGHT:

+				{

+					// remove left channel data

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						dAudioDataDouble[i] = dAudioDataDouble[(i * 2) + 1];

+					dAudioDataDouble[i] = 0.0;

+				}

+				break;

+				default:

+				case STEREO_SAMPLE_CONVERT:

+				{

+					// mix stereo to mono

+					len32 = sampleLength - 1;

+					for (i = 0; i < len32; i++)

+						dAudioDataDouble[i] = (dAudioDataDouble[(i * 2) + 0] + dAudioDataDouble[(i * 2) + 1]) * 0.5;

+					dAudioDataDouble[i] = 0.0;

+				}

+				break;

+			}

+		}

+		normalize64BitFloatToSigned16Bit(dAudioDataDouble, sampleLength);

+		ptr16 = (int16_t *)tmpSmp.pek;

+		for (i = 0; i < sampleLength; i++)

+			ptr16[i] = (int16_t)dAudioDataDouble[i]; // should use SIMD if available

+		sampleLength *= 2;

+		tmpSmp.typ |= 16; // 16-bit

+	}

+	reallocateTmpSmpData(&tmpSmp, sampleLength); // readjust memory needed

+	tuneSample(&tmpSmp, sampleRate, audio.linearPeriodsFlag);

+	tmpSmp.vol = 64;

+	tmpSmp.pan = 128;

+	tmpSmp.len = sampleLength;

+	// ---- READ "smpl" chunk ----

+	if (smplPtr != 0 && smplLen > 52)

+	{

+		uint32_t numLoops, loopType, loopStart, loopEnd;

+		fseek(f, smplPtr+28, SEEK_SET); // seek to first wanted byte

+		fread(&numLoops, 4, 1, f);

+		if (numLoops == 1)

+		{

+			fseek(f, 4+4, SEEK_CUR); // skip "samplerData" and "identifier"

+			fread(&loopType, 4, 1, f);

+			fread(&loopStart, 4, 1, f);

+			fread(&loopEnd, 4, 1, f); loopEnd++;

+			if (tmpSmp.typ & 16)

+			{

+				loopStart *= 2;

+				loopEnd *= 2;

+			}

+			if (loopEnd <= sampleLength)

+			{

+				tmpSmp.repS = loopStart;

+				tmpSmp.repL = loopEnd - loopStart;

+				tmpSmp.typ |= (loopType == 0) ? 1 : 2;

+			}

+		}

+	}

+	// ---------------------------

+	// ---- READ "xtra" chunk ----

+	if (xtraPtr != 0 && xtraLen >= 8)

+	{

+		uint16_t tmpPan, tmpVol;

+		uint32_t xtraFlags;

+		fseek(f, xtraPtr, SEEK_SET);

+		fread(&xtraFlags, 4, 1, f); // flags

+		// panning (0..256)

+		if (xtraFlags & 0x20) // set panning flag

+		{

+			fread(&tmpPan, 2, 1, f);

+			if (tmpPan > 255)

+				tmpPan = 255;

+			tmpSmp.pan = (uint8_t)tmpPan;

+		}

+		else

+		{

+			// don't read panning, skip it

+			fseek(f, 2, SEEK_CUR);

+		}

+		// volume (0..256)

+		fread(&tmpVol, 2, 1, f);

+		if (tmpVol > 256)

+			tmpVol = 256;

+		tmpSmp.vol = (uint8_t)((tmpVol + 2) / 4); // 0..256 -> 0..64 (rounded)

+	}

+	// ---------------------------

+	// ---- READ "INAM" chunk ----

+	if (inamPtr != 0 && inamLen > 0)

+	{

+		fseek(f, inamPtr, SEEK_SET);

+		uint32_t len = 22;

+		if (len > inamLen)

+			len = inamLen;

+		memset(tmpSmp.name, 0, sizeof (tmpSmp.name));

+		for (i = 0; i < len; i++)

+		{

+			char chr = (char)fgetc(f);

+			if (chr == '\0')

+				break;

+			tmpSmp.name[i] = chr;

+		}

+		smpFilenameSet = true;

+	}

+	return true;

+}

+static bool wavIsStereo(FILE *f)

+{

+	uint16_t numChannels;

+	uint32_t chunkID, chunkSize;

+	uint32_t oldPos = ftell(f);

+	fseek(f, 0, SEEK_END);

+	int32_t filesize = ftell(f);

+	if (filesize < 12)

+	{

+		fseek(f, oldPos, SEEK_SET);

+		return false;

+	}

+	fseek(f, 12, SEEK_SET);

+	uint32_t fmtPtr = 0;

+	uint32_t fmtLen = 0;

+	int32_t bytesRead = 0;

+	while (!feof(f) && bytesRead < filesize-12)

+	{

+		fread(&chunkID, 4, 1, f); if (feof(f)) break;

+		fread(&chunkSize, 4, 1, f); if (feof(f)) break;

+		int32_t endOfChunk = (ftell(f) + chunkSize) + (chunkSize & 1);

+		switch (chunkID)

+		{

+			case 0x20746D66: // "fmt "

+			{

+				fmtPtr = ftell(f);

+				fmtLen = chunkSize;

+			}

+			break;

+			default: break;

+		}

+		bytesRead += (chunkSize + (chunkSize & 1));

+		fseek(f, endOfChunk, SEEK_SET);

+	}

+	if (fmtPtr == 0 || fmtLen < 4)

+	{

+		fseek(f, oldPos, SEEK_SET);

+		return false;

+	}

+	fseek(f, fmtPtr + 2, SEEK_SET);

+	fread(&numChannels, 2, 1, f);

+	fseek(f, oldPos, SEEK_SET);

+	return (numChannels == 2);

+}

--- a/vs2019_project/ft2-clone/ft2-clone.vcxproj

+++ b/vs2019_project/ft2-clone/ft2-clone.vcxproj

@@ -342,6 +342,12 @@

     <ClCompile Include="..\..\src\mixer\ft2_mix.c" />

     <ClCompile Include="..\..\src\mixer\ft2_center_mix.c" />

     <ClCompile Include="..\..\src\mixer\ft2_silence_mix.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_digi.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_mod.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_s3m.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_stk.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_stm.c" />

+    <ClCompile Include="..\..\src\modloaders\ft2_load_xm.c" />

     <ClCompile Include="..\..\src\rtmidi\RtMidi.cpp">

       <ExceptionHandling Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">Sync</ExceptionHandling>

       <ExceptionHandling Condition="'$(Configuration)|$(Platform)'=='Release|x64'">Sync</ExceptionHandling>

@@ -358,6 +364,10 @@

       <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">4245;4996;4267</DisableSpecificWarnings>

       <DisableSpecificWarnings Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">4245;4996;4267</DisableSpecificWarnings>

     </ClCompile>

+    <ClCompile Include="..\..\src\smploaders\ft2_load_aiff.c" />

+    <ClCompile Include="..\..\src\smploaders\ft2_load_iff.c" />

+    <ClCompile Include="..\..\src\smploaders\ft2_load_raw.c" />

+    <ClCompile Include="..\..\src\smploaders\ft2_load_wav.c" />

   </ItemGroup>

   <ItemGroup>

     <ClInclude Include="..\..\src\ft2_about.h" />

@@ -412,6 +422,9 @@

   </ItemGroup>

   <ItemGroup>

     <ResourceCompile Include="..\..\src\ft2-clone.rc" />

+  </ItemGroup>

+  <ItemGroup>

+    <None Include="D:\dos\unreal ][.s3m" />

   </ItemGroup>

   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />

   <ImportGroup Label="ExtensionTargets">

--- a/vs2019_project/ft2-clone/ft2-clone.vcxproj.filters

+++ b/vs2019_project/ft2-clone/ft2-clone.vcxproj.filters

@@ -87,6 +87,36 @@

     <ClCompile Include="..\..\src\mixer\ft2_windowed_sinc.c">

       <Filter>mixer</Filter>

     </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_mod.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_s3m.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_stm.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_xm.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_stk.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\modloaders\ft2_load_digi.c">

+      <Filter>modloaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\smploaders\ft2_load_raw.c">

+      <Filter>smploaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\smploaders\ft2_load_iff.c">

+      <Filter>smploaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\smploaders\ft2_load_aiff.c">

+      <Filter>smploaders</Filter>

+    </ClCompile>

+    <ClCompile Include="..\..\src\smploaders\ft2_load_wav.c">

+      <Filter>smploaders</Filter>

+    </ClCompile>

   </ItemGroup>

   <ItemGroup>

     <ClInclude Include="..\..\src\rtmidi\RtMidi.h">

@@ -250,8 +280,17 @@

     <Filter Include="mixer">

       <UniqueIdentifier>{5c40c417-c4bb-4cf2-b71b-c557bf0a86cd}</UniqueIdentifier>

     </Filter>

+    <Filter Include="modloaders">

+      <UniqueIdentifier>{9948839a-30e1-4ef0-a25a-e88002c0ace3}</UniqueIdentifier>

+    </Filter>

+    <Filter Include="smploaders">

+      <UniqueIdentifier>{62e7f7c0-f7ae-4b85-bd62-2c83f74e3953}</UniqueIdentifier>

+    </Filter>

   </ItemGroup>

   <ItemGroup>

     <ResourceCompile Include="..\..\src\ft2-clone.rc" />

+  </ItemGroup>

+  <ItemGroup>

+    <None Include="D:\dos\unreal ][.s3m" />

   </ItemGroup>

 </Project>

\ No newline at end of file