shithub: pt2-clone

--- a/src/pt2_audio.c

+++ b/src/pt2_audio.c

@@ -30,6 +30,7 @@

 #include "pt2_textout.h"

 #include "pt2_visuals.h"

 #include "pt2_scopes.h"

+#include "pt2_mod2wav.h"

 #define INITIAL_DITHER_SEED 0x12345000

@@ -49,10 +50,10 @@

 static volatile int8_t filterFlags;

 static int8_t defStereoSep;

-static bool amigaPanFlag, wavRenderingDone;

+static bool amigaPanFlag;

 static uint16_t ch1Pan, ch2Pan, ch3Pan, ch4Pan;

-static int32_t oldPeriod, sampleCounter, randSeed = INITIAL_DITHER_SEED;

-static uint32_t oldScopeDelta;

+static int32_t oldPeriod, randSeed = INITIAL_DITHER_SEED;

+static uint32_t oldScopeDelta, sampleCounter;

 static double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned, dOldVoiceDelta, dOldVoiceDeltaMul;

 static double dPrngStateL, dPrngStateR;

 static blep_t blep[AMIGA_VOICES], blepVol[AMIGA_VOICES];

@@ -62,13 +63,10 @@

 static SDL_AudioDeviceID dev;

 // globalized

-int32_t samplesPerTick;

+uint32_t samplesPerTick;

 bool intMusic(void); // defined in pt_modplayer.c

-void storeTempVariables(void); // defined in pt_modplayer.c

-void calcMod2WavTotalRows(void);

 static uint16_t bpm2SmpsPerTick(uint32_t bpm, uint32_t audioFreq)

 	if (bpm == 0)

@@ -77,7 +75,7 @@

 	const uint32_t ciaVal = (uint32_t)(1773447 / bpm); // yes, PT truncates here

 	const double dCiaHz = (double)CIA_PAL_CLK / ciaVal;

-	int32_t smpsPerTick = (int32_t)((audioFreq / dCiaHz) + 0.5);

+	int32_t smpsPerTick = (int32_t)((audioFreq / dCiaHz) + 0.5); // rounded

 	return (uint16_t)smpsPerTick;

@@ -86,8 +84,9 @@

 	for (uint32_t i = 32; i <= 255; i++)

 		audio.bpmTab[i-32] = bpm2SmpsPerTick(i, audio.outputRate);

-		audio.bpmTab28kHz[i-32] = bpm2SmpsPerTick(i, 28836);

-		audio.bpmTab22kHz[i-32] = bpm2SmpsPerTick(i, 22168);

+		audio.bpmTab28kHz[i-32] = bpm2SmpsPerTick(i, 28836); // PAT2SMP hi quality

+		audio.bpmTab22kHz[i-32] = bpm2SmpsPerTick(i, 22168); // PAT2SMP low quality

+		audio.bpmTabMod2Wav[i-32] = bpm2SmpsPerTick(i, MOD2WAV_FREQ); // MOD2WAV

@@ -404,9 +403,11 @@

 #if SCOPE_HZ != 64

 #error Scope Hz is not 64 (2^n), change rate calc. to use doubles+round in pt2_scope.c

 #endif

-		// if we are rendering pattern to sample (PAT2SMP), use different frequencies

+		// during PAT2SMP or doing MOD2WAV, use different audio output rates

 		if (editor.isSMPRendering)

 			dPeriodToDeltaDiv = editor.pat2SmpHQ ? (PAULA_PAL_CLK / 28836.0) : (PAULA_PAL_CLK / 22168.0);

+		else if (editor.isWAVRendering)

+			dPeriodToDeltaDiv = PAULA_PAL_CLK / (double)MOD2WAV_FREQ;

 		else

 			dPeriodToDeltaDiv = audio.dPeriodToDeltaDiv;

@@ -594,8 +595,8 @@

 				bVol->dLastValue = dVol;

-			if (bSmp->samplesLeft > 0) dSmp += blepRun(bSmp);

-			if (bVol->samplesLeft > 0) dVol += blepRun(bVol);

+			if (bSmp->samplesLeft > 0) dSmp = blepRun(bSmp, dSmp);

+			if (bVol->samplesLeft > 0) dVol = blepRun(bVol, dVol);

 			dSmp *= dVol;

@@ -665,8 +666,8 @@

 				bVol->dLastValue = dVol;

-			if (bSmp->samplesLeft > 0) dSmp += blepRun(bSmp);

-			if (bVol->samplesLeft > 0) dVol += blepRun(bVol);

+			if (bSmp->samplesLeft > 0) dSmp = blepRun(bSmp, dSmp);

+			if (bVol->samplesLeft > 0) dVol = blepRun(bVol, dVol);

 			dSmp *= dVol;

@@ -724,8 +725,8 @@

 	RCHighPassFilter(&filterHi, dOut, dOut);

 	// normalize and flip phase (A500/A1200 has an inverted audio signal)

-	dOut[0] *= -(INT16_MAX / AMIGA_VOICES);

-	dOut[1] *= -(INT16_MAX / AMIGA_VOICES);

+	dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);

+	dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);

 	// left channel - 1-bit triangular dithering (high-pass filtered)

 	dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5

@@ -761,8 +762,8 @@

 	RCHighPassFilter(&filterHi, dOut, dOut);

 	// normalize and flip phase (A500/A1200 has an inverted audio signal)

-	dOut[0] *= -(INT16_MAX / AMIGA_VOICES);

-	dOut[1] *= -(INT16_MAX / AMIGA_VOICES);

+	dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);

+	dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);

 	// left channel - 1-bit triangular dithering (high-pass filtered)

 	dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5

@@ -795,8 +796,8 @@

 	// process high-pass filter

 	RCHighPassFilter(&filterHi, dOut, dOut);

-	dOut[0] *= -(INT16_MAX / AMIGA_VOICES);

-	dOut[1] *= -(INT16_MAX / AMIGA_VOICES);

+	dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);

+	dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);

 	// left channel - 1-bit triangular dithering (high-pass filtered)

 	dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5

@@ -832,8 +833,8 @@

 	// process high-pass filter

 	RCHighPassFilter(&filterHi, dOut, dOut);

-	dOut[0] *= -(INT16_MAX / AMIGA_VOICES);

-	dOut[1] *= -(INT16_MAX / AMIGA_VOICES);

+	dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);

+	dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);

 	// left channel - 1-bit triangular dithering (high-pass filtered)

 	dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5

@@ -852,28 +853,53 @@

 	out[1] = (int16_t)smp32;

+// for PAT2SMP

+static inline void processMixedSamplesRaw(int32_t i, int16_t *out)

+{

+	int32_t smp32;

+	double dOut[2];

+	dOut[0] = dMixBufferL[i];

+	dOut[1] = dMixBufferR[i];

+	// normalize

+	dOut[0] *= (INT16_MAX / (double)AMIGA_VOICES);

+	dOut[1] *= (INT16_MAX / (double)AMIGA_VOICES);

+	dOut[0] = (dOut[0] + dOut[1]) * 0.5; // mix to mono

+	smp32 = (int32_t)dOut[0];

+	CLAMP16(smp32);

+	out[0] = (int16_t)smp32;

+}

 void outputAudio(int16_t *target, int32_t numSamples)

 	int16_t *outStream, out[2];

-	int32_t j;

+	int32_t i;

 	if (editor.isSMPRendering)

 		// render to sample (PAT2SMP)

+		int32_t samplesTodo = numSamples;

+		if (editor.pat2SmpPos+samplesTodo > MAX_SAMPLE_LEN)

+			samplesTodo = MAX_SAMPLE_LEN-editor.pat2SmpPos;

 		mixChannelsMultiStep(numSamples);

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

+		outStream = &editor.pat2SmpBuf[editor.pat2SmpPos];

+		for (i = 0; i < samplesTodo; i++)

-			processMixedSamplesA1200(j, out);

-			editor.pat2SmpBuf[editor.pat2SmpPos++] = (int16_t)((out[0] + out[1]) >> 1); // mix to mono

+			processMixedSamplesA1200(i, out);

+			outStream[i] = (out[0] + out[1]) >> 1;

+		}

-			if (editor.pat2SmpPos >= MAX_SAMPLE_LEN)

-			{

-				editor.smpRenderingDone = true;

-				updateWindowTitle(MOD_IS_MODIFIED);

-				break;

-			}

+		editor.pat2SmpPos += samplesTodo;

+		if (editor.pat2SmpPos >= MAX_SAMPLE_LEN)

+		{

+			editor.smpRenderingDone = true;

+			updateWindowTitle(MOD_IS_MODIFIED);

 	else

@@ -889,9 +915,9 @@

 			if (filterFlags & FILTER_LED_ENABLED)

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

+				for (i = 0; i < numSamples; i++)

-					processMixedSamplesA500LED(j, out);

+					processMixedSamplesA500LED(i, out);

 					*outStream++ = out[0];

 					*outStream++ = out[1];

@@ -898,9 +924,9 @@

 			else

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

+				for (i = 0; i < numSamples; i++)

-					processMixedSamplesA500(j, out);

+					processMixedSamplesA500(i, out);

 					*outStream++ = out[0];

 					*outStream++ = out[1];

@@ -912,9 +938,9 @@

 			if (filterFlags & FILTER_LED_ENABLED)

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

+				for (i = 0; i < numSamples; i++)

-					processMixedSamplesA1200LED(j, out);

+					processMixedSamplesA1200LED(i, out);

 					*outStream++ = out[0];

 					*outStream++ = out[1];

@@ -921,9 +947,9 @@

 			else

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

+				for (i = 0; i < numSamples; i++)

-					processMixedSamplesA1200(j, out);

+					processMixedSamplesA1200(i, out);

 					*outStream++ = out[0];

 					*outStream++ = out[1];

@@ -934,8 +960,8 @@

 static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)

-	int16_t *out;

-	int32_t sampleBlock, samplesTodo;

+	int16_t *streamOut;

+	uint32_t samplesLeft;

 	if (audio.forceMixerOff) // during MOD2WAV

@@ -943,27 +969,28 @@

 		return;

-	out = (int16_t *)stream;

+	streamOut = (int16_t *)stream;

-	sampleBlock = len >> 2;

-	while (sampleBlock)

+	samplesLeft = len >> 2;

+	while (samplesLeft > 0)

-		samplesTodo = (sampleBlock < sampleCounter) ? sampleBlock : sampleCounter;

-		if (samplesTodo > 0)

+		if (sampleCounter == 0)

-			outputAudio(out, samplesTodo);

-			out += (uint32_t)samplesTodo * 2;

-			sampleBlock -= samplesTodo;

-			sampleCounter -= samplesTodo;

-		}

-		else

-		{

 			if (editor.songPlaying)

 				intMusic();

 			sampleCounter = samplesPerTick;

+		uint32_t samplesTodo = sampleCounter;

+		if (samplesTodo > samplesLeft)

+			samplesTodo = samplesLeft;

+		outputAudio(streamOut, samplesTodo);

+		streamOut += samplesTodo << 1;

+		samplesLeft -= samplesTodo;

+		sampleCounter -= samplesTodo;

 	(void)userdata;

@@ -1081,13 +1108,22 @@

 	audio.dPeriodToDeltaDiv = (double)PAULA_PAL_CLK / audio.outputRate;

 	generateBpmTables();

-	const int32_t maxSamplesToMix = audio.bpmTab[0]; // BPM 32

+	/* If the audio output rate is lower than MOD2WAV_FREQ, we need

+	** to allocate slightly more space so that MOD2WAV rendering

+	** won't overflow.

+	*/

+	uint32_t maxSamplesToMix;

+	if (MOD2WAV_FREQ > audio.outputRate)

+		 maxSamplesToMix = audio.bpmTabMod2Wav[32-32]; // BPM 32

+	else

+		maxSamplesToMix = audio.bpmTab[32-32]; // BPM 32

 	dMixBufferLUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);

 	dMixBufferRUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);

-	editor.mod2WavBuffer = (int16_t *)malloc(maxSamplesToMix * sizeof (int16_t));

-	if (dMixBufferLUnaligned == NULL || dMixBufferRUnaligned == NULL || editor.mod2WavBuffer == NULL)

+	if (dMixBufferLUnaligned == NULL || dMixBufferRUnaligned == NULL)

 		showErrorMsgBox("Out of memory!");

 		return false;

@@ -1102,7 +1138,7 @@

 	filterFlags = config.a500LowPassFilter ? FILTER_A500 : 0;

 	calculateFilterCoeffs();

-	samplesPerTick = 0;

+	samplesPerTick = audio.bpmTab[125-32]; // BPM 125

 	sampleCounter = 0;

 	SDL_PauseAudioDevice(dev, false);

@@ -1129,15 +1165,9 @@

 		free(dMixBufferRUnaligned);

 		dMixBufferRUnaligned = NULL;

-	if (editor.mod2WavBuffer != NULL)

-	{

-		free(editor.mod2WavBuffer);

-		editor.mod2WavBuffer = NULL;

-	}

-void mixerSetSamplesPerTick(int32_t val)

+void mixerSetSamplesPerTick(uint32_t val)

 	samplesPerTick = val;

@@ -1159,280 +1189,6 @@

 		mixerCalcVoicePans(100);

 		displayMsg("AMIGA PANNING ON");

-	}

-}

-// PAT2SMP RELATED STUFF

-uint32_t getAudioFrame(int16_t *outStream)

-{

-	int32_t smpCounter, samplesToMix;

-	if (!intMusic())

-		wavRenderingDone = true;

-	smpCounter = samplesPerTick;

-	while (smpCounter > 0)

-	{

-		samplesToMix = smpCounter;

-		outputAudio(outStream, samplesToMix);

-		outStream += (uint32_t)samplesToMix * 2;

-		smpCounter -= samplesToMix;

-	}

-	return (uint32_t)samplesPerTick * 2; // * 2 for stereo

-}

-static int32_t SDLCALL mod2WavThreadFunc(void *ptr)

-{

-	uint32_t size, totalSampleCounter, totalRiffChunkLen;

-	FILE *fOut;

-	wavHeader_t wavHeader;

-	fOut = (FILE *)ptr;

-	if (fOut == NULL)

-		return true;

-	// skip wav header place, render data first

-	fseek(fOut, sizeof (wavHeader_t), SEEK_SET);

-	wavRenderingDone = false;

-	totalSampleCounter = 0;

-	while (editor.isWAVRendering && !wavRenderingDone && !editor.abortMod2Wav)

-	{

-		size = getAudioFrame(editor.mod2WavBuffer);

-		if (size > 0)

-		{

-			fwrite(editor.mod2WavBuffer, sizeof (int16_t), size, fOut);

-			totalSampleCounter += size;

-		}

-		editor.ui.updateMod2WavDialog = true;

-	}

-	if (totalSampleCounter & 1)

-		fputc(0, fOut); // pad align byte

-	if ((ftell(fOut) - 8) > 0)

-		totalRiffChunkLen = ftell(fOut) - 8;

-	else

-		totalRiffChunkLen = 0;

-	editor.ui.mod2WavFinished = true;

-	editor.ui.updateMod2WavDialog = true;

-	// go back and fill the missing WAV header

-	fseek(fOut, 0, SEEK_SET);

-	wavHeader.chunkID = 0x46464952; // "RIFF"

-	wavHeader.chunkSize = totalRiffChunkLen;

-	wavHeader.format = 0x45564157; // "WAVE"

-	wavHeader.subchunk1ID = 0x20746D66; // "fmt "

-	wavHeader.subchunk1Size = 16;

-	wavHeader.audioFormat = 1;

-	wavHeader.numChannels = 2;

-	wavHeader.sampleRate = audio.outputRate;

-	wavHeader.bitsPerSample = 16;

-	wavHeader.byteRate = wavHeader.sampleRate * wavHeader.numChannels * (wavHeader.bitsPerSample / 8);

-	wavHeader.blockAlign = wavHeader.numChannels * (wavHeader.bitsPerSample / 8);

-	wavHeader.subchunk2ID = 0x61746164; // "data"

-	wavHeader.subchunk2Size = totalSampleCounter * (wavHeader.bitsPerSample / 8);

-	fwrite(&wavHeader, sizeof (wavHeader_t), 1, fOut);

-	fclose(fOut);

-	return true;

-}

-bool renderToWav(char *fileName, bool checkIfFileExist)

-{

-	FILE *fOut;

-	struct stat statBuffer;

-	if (checkIfFileExist)

-	{

-		if (stat(fileName, &statBuffer) == 0)

-		{

-			editor.ui.askScreenShown = true;

-			editor.ui.askScreenType = ASK_MOD2WAV_OVERWRITE;

-			pointerSetMode(POINTER_MODE_MSG1, NO_CARRY);

-			setStatusMessage("OVERWRITE FILE?", NO_CARRY);

-			renderAskDialog();

-			return false;

-		}

-	}

-	if (editor.ui.askScreenShown)

-	{

-		editor.ui.askScreenShown = false;

-		editor.ui.answerNo = false;

-		editor.ui.answerYes = false;

-	}

-	fOut = fopen(fileName, "wb");

-	if (fOut == NULL)

-	{

-		displayErrorMsg("FILE I/O ERROR");

-		return false;

-	}

-	storeTempVariables();

-	calcMod2WavTotalRows();

-	restartSong();

-	editor.blockMarkFlag = false;

-	pointerSetMode(POINTER_MODE_MSG2, NO_CARRY);

-	setStatusMessage("RENDERING MOD...", NO_CARRY);

-	editor.ui.disableVisualizer = true;

-	editor.isWAVRendering = true;

-	renderMOD2WAVDialog();

-	editor.abortMod2Wav = false;

-	editor.mod2WavThread = SDL_CreateThread(mod2WavThreadFunc, NULL, fOut);

-	if (editor.mod2WavThread != NULL)

-	{

-		SDL_DetachThread(editor.mod2WavThread);

-	}

-	else

-	{

-		editor.ui.disableVisualizer = false;

-		editor.isWAVRendering = false;

-		displayErrorMsg("THREAD ERROR");

-		pointerSetMode(POINTER_MODE_IDLE, DO_CARRY);

-		statusAllRight();

-		return false;

-	}

-	return true;

-}

-// for MOD2WAV - ONLY used for a visual percentage counter, so accuracy is not important

-void calcMod2WavTotalRows(void)

-{

-	bool pBreakFlag, posJumpAssert, calcingRows;

-	int8_t n_pattpos[AMIGA_VOICES], n_loopcount[AMIGA_VOICES];

-	uint8_t modRow, pBreakPosition, ch, pos;

-	int16_t modOrder;

-	uint16_t modPattern;

-	note_t *note;

-	// for pattern loop

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

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

-	modEntry->rowsCounter = 0;

-	modEntry->rowsInTotal = 0;

-	modRow = 0;

-	modOrder = 0;

-	modPattern = modEntry->head.order[0];

-	pBreakPosition = 0;

-	posJumpAssert = false;

-	pBreakFlag = false;

-	calcingRows = true;

-	memset(editor.rowVisitTable, 0, MOD_ORDERS * MOD_ROWS);

-	while (calcingRows)

-	{

-		editor.rowVisitTable[(modOrder * MOD_ROWS) + modRow] = true;

-		for (ch = 0; ch < AMIGA_VOICES; ch++)

-		{

-			note = &modEntry->patterns[modPattern][(modRow * AMIGA_VOICES) + ch];

-			if (note->command == 0x0B) // Bxx - Position Jump

-			{

-				modOrder = note->param - 1;

-				pBreakPosition = 0;

-				posJumpAssert = true;

-			}

-			else if (note->command == 0x0D) // Dxx - Pattern Break

-			{

-				pBreakPosition = (((note->param >> 4) * 10) + (note->param & 0x0F));

-				if (pBreakPosition > 63)

-					pBreakPosition = 0;

-				posJumpAssert = true;

-			}

-			else if (note->command == 0x0F && note->param == 0) // F00 - Set Speed 0 (stop)

-			{

-				calcingRows = false;

-				break;

-			}

-			else if (note->command == 0x0E && (note->param >> 4) == 0x06) // E6x - Pattern Loop

-			{

-				pos = note->param & 0x0F;

-				if (pos == 0)

-				{

-					n_pattpos[ch] = modRow;

-				}

-				else if (n_loopcount[ch] == 0)

-				{

-					n_loopcount[ch] = pos;

-					pBreakPosition = n_pattpos[ch];

-					pBreakFlag = true;

-					for (pos = pBreakPosition; pos <= modRow; pos++)

-						editor.rowVisitTable[(modOrder * MOD_ROWS) + pos] = false;

-				}

-				else if (--n_loopcount[ch])

-				{

-					pBreakPosition = n_pattpos[ch];

-					pBreakFlag = true;

-					for (pos = pBreakPosition; pos <= modRow; pos++)

-						editor.rowVisitTable[(modOrder * MOD_ROWS) + pos] = false;

-				}

-			}

-		}

-		modRow++;

-		modEntry->rowsInTotal++;

-		if (pBreakFlag)

-		{

-			modRow = pBreakPosition;

-			pBreakPosition = 0;

-			pBreakFlag = false;

-		}

-		if (modRow >= MOD_ROWS || posJumpAssert)

-		{

-			modRow = pBreakPosition;

-			pBreakPosition = 0;

-			posJumpAssert = false;

-			modOrder = (modOrder + 1) & 0x7F;

-			if (modOrder >= modEntry->head.orderCount)

-			{

-				modOrder = 0;

-				calcingRows = false;

-				break;

-			}

-			modPattern = modEntry->head.order[modOrder];

-			if (modPattern > MAX_PATTERNS-1)

-				modPattern = MAX_PATTERNS-1;

-		}

-		if (editor.rowVisitTable[(modOrder * MOD_ROWS) + modRow])

-		{

-			// row has been visited before, we're now done!

-			calcingRows = false;

-			break;

-		}

--- a/src/pt2_audio.h

+++ b/src/pt2_audio.h

@@ -29,7 +29,6 @@

 void normalize8bitDoubleSigned(double *dSampleData, uint32_t sampleLength);

 void setLEDFilter(bool state);

 void toggleLEDFilter(void);

-bool renderToWav(char *fileName, bool checkIfFileExist);

 void toggleAmigaPanMode(void);

 void toggleA500Filters(void);

 void paulaStopDMA(uint8_t ch);

@@ -44,6 +43,6 @@

 void mixerKillVoice(uint8_t ch);

 void turnOffVoices(void);

 void mixerCalcVoicePans(uint8_t stereoSeparation);

-void mixerSetSamplesPerTick(int32_t val);

+void mixerSetSamplesPerTick(uint32_t val);

 void mixerClearSampleCounter(void);

 void outputAudio(int16_t *target, int32_t numSamples);

--- a/src/pt2_blep.c

+++ b/src/pt2_blep.c

@@ -1,4 +1,4 @@

-// These BLEP routines were coded by aciddose/adejr

+// these BLEP routines were coded by aciddose

 #include <stdint.h>

 #include "pt2_blep.h"

@@ -5,50 +5,101 @@

 #include "pt2_helpers.h"

 /* Why this table is not represented as readable floating-point numbers:

-** Accurate float (double) representation in string format requires at least 14 digits and normalized

+** Accurate double representation in string format requires at least 14 digits and normalized

 ** (scientific) notation, notwithstanding compiler issues with precision or rounding error.

-** Also, don't touch this table ever, just keep it exactly identical! */

+** Also, don't touch this table ever, just keep it exactly identical!

+*/

-// TODO: get a proper double-precision table. This one is converted from float.

-static const uint64_t dBlepData[48] =

+static const uint64_t minblepdata[] =

-	0x3FEFFC3E20000000, 0x3FEFFAA900000000, 0x3FEFFAD460000000, 0x3FEFFA9C60000000,

-	0x3FEFF5B0A0000000, 0x3FEFE42A40000000, 0x3FEFB7F5C0000000, 0x3FEF599BE0000000,

-	0x3FEEA5E3C0000000, 0x3FED6E7080000000, 0x3FEB7F7960000000, 0x3FE8AB9E40000000,

-	0x3FE4DCA480000000, 0x3FE0251880000000, 0x3FD598FB80000000, 0x3FC53D0D60000000,

-	0x3F8383A520000000, 0xBFBC977CC0000000, 0xBFC755C080000000, 0xBFC91BDBA0000000,

-	0xBFC455AFC0000000, 0xBFB6461340000000, 0xBF7056C400000000, 0x3FB1028220000000,

-	0x3FBB5B7E60000000, 0x3FBC5903A0000000, 0x3FB55403E0000000, 0x3FA3CED340000000,

-	0xBF7822DAE0000000, 0xBFA2805D00000000, 0xBFA7140D20000000, 0xBFA18A7760000000,

-	0xBF87FF7180000000, 0x3F88CBFA40000000, 0x3F9D4AEC80000000, 0x3FA14A3AC0000000,

-	0x3F9D5C5AA0000000, 0x3F92558B40000000, 0x3F7C997EE0000000, 0x0000000000000000,

-	0x0000000000000000, 0x0000000000000000, 0x0000000000000000, 0x0000000000000000,

-	0x0000000000000000, 0x0000000000000000, 0x0000000000000000, 0x0000000000000000

+	0x3FF000320C7E95A6,0x3FF00049BE220FD5,0x3FF0001B92A41ACA,0x3FEFFF4425AA9724,

+	0x3FEFFDABDF6CF05C,0x3FEFFB5AF233EF1A,0x3FEFF837E2AE85F3,0x3FEFF4217B80E938,

+	0x3FEFEEECEB4E0444,0x3FEFE863A8358B5F,0x3FEFE04126292670,0x3FEFD63072A0D592,

+	0x3FEFC9C9CD36F56F,0x3FEFBA90594BD8C3,0x3FEFA7F008BA9F13,0x3FEF913BE2A0E0E2,

+	0x3FEF75ACCB01A327,0x3FEF5460F06A4E8F,0x3FEF2C5C0389BD3C,0x3FEEFC8859BF6BCB,

+	0x3FEEC3B916FD8D19,0x3FEE80AD74F0AD16,0x3FEE32153552E2C7,0x3FEDD69643CB9778,

+	0x3FED6CD380FFA864,0x3FECF374A4D2961A,0x3FEC692F19B34E54,0x3FEBCCCFA695DD5C,

+	0x3FEB1D44B168764A,0x3FEA59A8D8E4527F,0x3FE9814D9B10A9A3,0x3FE893C5B62135F2,

+	0x3FE790EEEBF9DABD,0x3FE678FACDEE27FF,0x3FE54C763699791A,0x3FE40C4F1B1EB7A3,

+	0x3FE2B9D863D4E0F3,0x3FE156CB86586B0B,0x3FDFCA8F5005B828,0x3FDCCF9C3F455DAC,

+	0x3FD9C2787F20D06E,0x3FD6A984CAD0F3E5,0x3FD38BB0C452732E,0x3FD0705EC7135366,

+	0x3FCABE86754E238F,0x3FC4C0801A6E9A04,0x3FBDECF490C5EA17,0x3FB2DFFACE9CE44B,

+	0x3FA0EFD4449F4620,0xBF72F4A65E22806D,0xBFA3F872D761F927,0xBFB1D89F0FD31F7C,

+	0xBFB8B1EA652EC270,0xBFBE79B82A37C92D,0xBFC1931B697E685E,0xBFC359383D4C8ADA,

+	0xBFC48F3BFF81B06B,0xBFC537BBA8D6B15C,0xBFC557CEF2168326,0xBFC4F6F781B3347A,

+	0xBFC41EF872F0E009,0xBFC2DB9F119D54D3,0xBFC13A7E196CB44F,0xBFBE953A67843504,

+	0xBFBA383D9C597E74,0xBFB57FBD67AD55D6,0xBFB08E18234E5CB3,0xBFA70B06D699FFD1,

+	0xBF9A1CFB65370184,0xBF7B2CEB901D2067,0x3F86D5DE2C267C78,0x3F9C1D9EF73F384D,

+	0x3FA579C530950503,0x3FABD1E5FFF9B1D0,0x3FB07DCDC3A4FB5B,0x3FB2724A856EEC1B,

+	0x3FB3C1F7199FC822,0x3FB46D0979F5043B,0x3FB47831387E0110,0x3FB3EC4A58A3D527,

+	0x3FB2D5F45F8889B3,0x3FB145113E25B749,0x3FAE9860D18779BC,0x3FA9FFD5F5AB96EA,

+	0x3FA4EC6C4F47777E,0x3F9F16C5B2604C3A,0x3F9413D801124DB7,0x3F824F668CBB5BDF,

+	0xBF55B3FA2EE30D66,0xBF86541863B38183,0xBF94031BBBD551DE,0xBF9BAFC27DC5E769,

+	0xBFA102B3683C57EC,0xBFA3731E608CC6E4,0xBFA520C9F5B5DEBD,0xBFA609DC89BE6ECE,

+	0xBFA632B83BC5F52F,0xBFA5A58885841AD4,0xBFA471A5D2FF02F3,0xBFA2AAD5CD0377C7,

+	0xBFA0686FFE4B9B05,0xBF9B88DE413ACB69,0xBF95B4EF6D93F1C5,0xBF8F1B72860B27FA,

+	0xBF8296A865CDF612,0xBF691BEEDABE928B,0x3F65C04E6AF9D4F1,0x3F8035D8FFCDB0F8,

+	0x3F89BED23C431BE3,0x3F90E737811A1D21,0x3F941C2040BD7CB1,0x3F967046EC629A09,

+	0x3F97DE27ECE9ED89,0x3F98684DE31E7040,0x3F9818C4B07718FA,0x3F97005261F91F60,

+	0x3F95357FDD157646,0x3F92D37C696C572A,0x3F8FF1CFF2BEECB5,0x3F898D20C7A72AC4,

+	0x3F82BC5B3B0AE2DF,0x3F7784A1B8E9E667,0x3F637BB14081726B,0xBF4B2DACA70C60A9,

+	0xBF6EFB00AD083727,0xBF7A313758DC6AE9,0xBF819D6A99164BE0,0xBF8533F57533403B,

+	0xBF87CD120DB5D340,0xBF89638549CD25DE,0xBF89FB8B8D37B1BB,0xBF89A21163F9204E,

+	0xBF886BA8931297D4,0xBF8673477783D71E,0xBF83D8E1CB165DB8,0xBF80BFEA7216142A,

+	0xBF7A9B9BC2E40EBF,0xBF7350E806435A7E,0xBF67D35D3734AB5E,0xBF52ADE8FEAB8DB9,

+	0x3F415669446478E4,0x3F60C56A092AFB48,0x3F6B9F4334A4561F,0x3F724FB908FD87AA,

+	0x3F75CC56DFE382EA,0x3F783A0C23969A7B,0x3F799833C40C3B82,0x3F79F02721981BF3,

+	0x3F7954212AB35261,0x3F77DDE0C5FC15C9,0x3F75AD1C98FE0777,0x3F72E5DACC0849F2,

+	0x3F6F5D7E69DFDE1B,0x3F685EC2CA09E1FD,0x3F611D750E54DF3A,0x3F53C6E392A46D17,

+	0x3F37A046885F3365,0xBF3BB034D2EE45C2,0xBF5254267B04B482,0xBF5C0516F9CECDC6,

+	0xBF61E5736853564D,0xBF64C464B9CC47AB,0xBF669C1AEF258F56,0xBF67739985DD0E60,

+	0xBF675AFD6446395B,0xBF666A0C909B4F78,0xBF64BE9879A7A07B,0xBF627AC74B119DBD,

+	0xBF5F86B04069DC9B,0xBF597BE8F754AF5E,0xBF531F3EAAE9A1B1,0xBF496D3DE6AD7EA3,

+	0xBF3A05FFDE4670CF,0xBF06DF95C93A85CA,0x3F31EE2B2C6547AC,0x3F41E694A378C129,

+	0x3F4930BF840E23C9,0x3F4EBB5D05A0D47D,0x3F51404DA0539855,0x3F524698F56B3F33,

+	0x3F527EF85309E28F,0x3F51FE70FE2513DE,0x3F50DF1642009B74,0x3F4E7CDA93517CAE,

+	0x3F4A77AE24F9A533,0x3F45EE226AA69E10,0x3F411DB747374F52,0x3F387F39D229D97F,

+	0x3F2E1B3D39AF5F8B,0x3F18F557BB082715,0xBEFAC04896E68DDB,0xBF20F5BC77DF558A,

+	0xBF2C1B6DF3EE94A4,0xBF3254602A816876,0xBF354E90F6EAC26B,0xBF3709F2E5AF1624,

+	0xBF379FCCB331CE8E,0xBF37327192ADDAD3,0xBF35EA998A894237,0xBF33F4C4977B3489,

+	0xBF317EC5F68E887B,0xBF2D6B1F793EB773,0xBF2786A226B076D9,0xBF219BE6CEC2CA36,

+	0xBF17D7F36D2A3A18,0xBF0AAEC5BBAB42AB,0xBEF01818DC224040,0x3EEF2F6E21093846,

+	0x3F049D6E0060B71F,0x3F0E598CCAFABEFD,0x3F128BC14BE97261,0x3F148703BC70EF6A,

+	0x3F1545E1579CAA25,0x3F14F7DDF5F8D766,0x3F13D10FF9A1BE0C,0x3F1206D5738ECE3A,

+	0x3F0F99F6BF17C5D4,0x3F0AA6D7EA524E96,0x3F0588DDF740E1F4,0x3F0086FB6FEA9839,

+	0x3EF7B28F6D6F5EED,0x3EEEA300DCBAF74A,0x3EE03F904789777C,0x3EC1BFEB320501ED,

+	0xBEC310D8E585A031,0xBED6F55ECA7E151F,0xBEDFDAA5DACDD0B7,0xBEE26944F3CF6E90,

+	0xBEE346894453BD1F,0xBEE2E099305CD5A8,0xBEE190385A7EA8B2,0xBEDF4D5FA2FB6BA2,

+	0xBEDAD4F371257BA0,0xBED62A9CDEB0AB32,0xBED1A6DF97B88316,0xBECB100096894E58,

+	0xBEC3E8A76257D275,0xBEBBF6C29A5150C9,0xBEB296292998088E,0xBEA70A10498F0E5E,

+	0xBE99E52D02F887A1,0xBE88C17F4066D432,0xBE702A716CFF56CA,0x3E409F820F781F78,

+	0x3E643EA99B770FE7,0x3E67DE40CDE0A550,0x3E64F4D534A2335C,0x3E5F194536BDDF7A,

+	0x3E5425CEBE1FA40A,0x3E46D7B7CC631E73,0x3E364746B6582E54,0x3E21FC07B13031DE,

+	0x3E064C3D91CF7665,0x3DE224F901A0AFC7,0x3DA97D57859C74A4,0x0000000000000000,

+	// extra padding needed for interpolation

+	0x0000000000000000

};

+const double *get_minblep_table(void) { return (const double *)minblepdata; }

 void blepAdd(blep_t *b, double dOffset, double dAmplitude)

-	int8_t n;

-	int32_t i;

-	const double *dBlepSrc;

-	double f;

 	assert(dOffset >= 0.0 && dOffset < 1.0);

-	f = dOffset * BLEP_SP;

+	double f = dOffset * BLEP_SP;

-	i = (int32_t)f; // get integer part of f

-	dBlepSrc = (const double *)dBlepData + i + BLEP_OS;

+	int32_t i = (int32_t)f; // get integer part of f

+	const double *dBlepSrc = get_minblep_table() + i;

 	f -= i; // remove integer part from f

 	i = b->index;

-	n = BLEP_NS;

-	while (n--)

+	for (int32_t n = 0; n < BLEP_NS; n++)

 		b->dBuffer[i] += dAmplitude * LERP(dBlepSrc[0], dBlepSrc[1], f);

-		i = (i + 1) & BLEP_RNS;

 		dBlepSrc += BLEP_SP;

+		i = (i + 1) & BLEP_RNS;

 	b->samplesLeft = BLEP_NS;

@@ -55,20 +106,14 @@

 /* 8bitbubsy: simplified, faster version of blepAdd for blep'ing voice volume.

-** Result is identical! (confirmed with binary comparison)

+** Result is identical! (confirmed with binary comparison w/ MOD2WAV)

*/

 void blepVolAdd(blep_t *b, double dAmplitude)

-	int8_t n;

-	int32_t i;

-	const double *dBlepSrc;

+	const double *dBlepSrc = get_minblep_table();

-	dBlepSrc = (const double *)dBlepData + BLEP_OS;

-	i = b->index;

-	n = BLEP_NS;

-	while (n--)

+	int32_t i = b->index;

+	for (int32_t n = 0; n < BLEP_NS; n++)

 		b->dBuffer[i] += dAmplitude * (*dBlepSrc);

 		i = (i + 1) & BLEP_RNS;

@@ -78,15 +123,13 @@

 	b->samplesLeft = BLEP_NS;

-double blepRun(blep_t *b)

+double blepRun(blep_t *b, double dInput)

-	double fBlepOutput;

-	fBlepOutput = b->dBuffer[b->index];

+	double dBlepOutput = dInput + b->dBuffer[b->index];

 	b->dBuffer[b->index] = 0.0;

 	b->index = (b->index + 1) & BLEP_RNS;

 	b->samplesLeft--;

-	return fBlepOutput;

+	return dBlepOutput;

--- a/src/pt2_blep.h

+++ b/src/pt2_blep.h

@@ -19,11 +19,11 @@

 ** the result of that is the filter cutoff is set at nyquist * (SP/OS), in this case nyquist/5.

*/

-#define BLEP_ZC 8

-#define BLEP_OS 5

-#define BLEP_SP 5

+#define BLEP_ZC 16

+#define BLEP_OS 16

+#define BLEP_SP 16

 #define BLEP_NS (BLEP_ZC * BLEP_OS / BLEP_SP)

-#define BLEP_RNS 7 // RNS = (2^ > NS) - 1

+#define BLEP_RNS 31 // RNS = (2^ > NS) - 1

 typedef struct blep_t

@@ -33,4 +33,4 @@

 void blepAdd(blep_t *b, double dOffset, double dAmplitude);

 void blepVolAdd(blep_t *b, double dAmplitude);

-double blepRun(blep_t *b);

+double blepRun(blep_t *b, double dInput);

--- a/src/pt2_header.h

+++ b/src/pt2_header.h

@@ -14,7 +14,7 @@

 #include "pt2_unicode.h"

 #include "pt2_palette.h"

-#define PROG_VER_STR "1.10"

+#define PROG_VER_STR "1.11"

 #ifdef _WIN32

 #define DIR_DELIMITER '\\'

@@ -291,7 +291,7 @@

 	volatile bool locked;

 	bool forceMixerOff;

-	uint16_t bpmTab[256-32], bpmTab28kHz[256-32], bpmTab22kHz[256-32];

+	uint16_t bpmTab[256-32], bpmTab28kHz[256-32], bpmTab22kHz[256-32], bpmTabMod2Wav[256-32];

 	uint32_t outputRate, audioBufferSize;

 	double dPeriodToDeltaDiv;

 } audio;

@@ -365,7 +365,7 @@

 	uint8_t blockFromPos, blockToPos, timingMode, f6Pos, f7Pos, f8Pos, f9Pos, f10Pos, keyOctave, pNoteFlag;

 	uint8_t tuningNote, resampleNote, initialTempo, initialSpeed, editMoveAdd;

-	int16_t *mod2WavBuffer, *pat2SmpBuf, modulateSpeed;

+	int16_t *pat2SmpBuf, modulateSpeed;

 	uint16_t metroSpeed, metroChannel, sampleVol, samplePos, chordLength;

 	uint16_t effectMacros[10], oldTempo, currPlayNote, vol1, vol2, lpCutOff, hpCutOff;

 	int32_t smpRedoLoopStarts[MOD_SAMPLES], smpRedoLoopLengths[MOD_SAMPLES], smpRedoLengths[MOD_SAMPLES];

--- /dev/null

+++ b/src/pt2_mod2wav.c

@@ -1,0 +1,320 @@

+// for finding memory leaks in debug mode with Visual Studio

+#if defined _DEBUG && defined _MSC_VER

+#include <crtdbg.h>

+#endif

+#include <stdio.h>

+#include <stdint.h>

+#include <stdbool.h>

+#include "pt2_header.h"

+#include "pt2_audio.h"

+#include "pt2_mouse.h"

+#include "pt2_textout.h"

+#include "pt2_visuals.h"

+#include "pt2_mod2wav.h"

+#define TICKS_PER_RENDER_CHUNK 32

+void storeTempVariables(void); // pt_modplayer.c

+bool intMusic(void); // pt_modplayer.c

+extern uint32_t samplesPerTick; // pt_audio.c

+static volatile bool wavRenderingDone;

+static int16_t *mod2WavBuffer;

+static void calcMod2WavTotalRows(void);

+static uint32_t getAudioFrame(int16_t *outStream)

+{

+	if (!intMusic())

+		wavRenderingDone = true;

+	outputAudio(outStream, samplesPerTick);

+	return samplesPerTick;

+}

+static int32_t SDLCALL mod2WavThreadFunc(void *ptr)

+{

+	wavHeader_t wavHeader;

+	FILE *f = (FILE *)ptr;

+	assert(mod2WavBuffer != NULL && f != NULL);

+	// skip wav header place, render data first

+	fseek(f, sizeof (wavHeader_t), SEEK_SET);

+	wavRenderingDone = false;

+	uint8_t loopCounter = 8;

+	uint32_t totalSampleCounter = 0;

+	bool renderDone = false;

+	while (!renderDone)

+	{

+		uint32_t samplesInChunk = 0;

+		// render several ticks at once to prevent frequent disk I/O (can speed up the process)

+		int16_t *ptr16 = mod2WavBuffer;

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

+		{

+			if (!editor.isWAVRendering || wavRenderingDone || editor.abortMod2Wav)

+			{

+				renderDone = true;

+				break;

+			}

+			uint32_t tickSamples = getAudioFrame(ptr16) << 1; // *2 for stereo

+			samplesInChunk += tickSamples;

+			totalSampleCounter += tickSamples;

+			// increase buffer pointer

+			ptr16 += tickSamples;

+			if (++loopCounter >= 8)

+			{

+				loopCounter = 0;

+				editor.ui.updateMod2WavDialog = true;

+			}

+		}

+		// write buffer to disk

+		if (samplesInChunk > 0)

+			fwrite(mod2WavBuffer, sizeof (int16_t), samplesInChunk, f);

+	}

+	free(mod2WavBuffer);

+	if (totalSampleCounter & 1)

+		fputc(0, f); // pad align byte

+	uint32_t totalRiffChunkLen = (uint32_t)ftell(f) - 8;

+	// go back and fill in WAV header

+	rewind(f);

+	wavHeader.chunkID = 0x46464952; // "RIFF"

+	wavHeader.chunkSize = totalRiffChunkLen;

+	wavHeader.format = 0x45564157; // "WAVE"

+	wavHeader.subchunk1ID = 0x20746D66; // "fmt "

+	wavHeader.subchunk1Size = 16;

+	wavHeader.audioFormat = 1;

+	wavHeader.numChannels = 2;

+	wavHeader.sampleRate = MOD2WAV_FREQ;

+	wavHeader.bitsPerSample = 16;

+	wavHeader.byteRate = (wavHeader.sampleRate * wavHeader.numChannels * wavHeader.bitsPerSample) / 8;

+	wavHeader.blockAlign = (wavHeader.numChannels * wavHeader.bitsPerSample) / 8;

+	wavHeader.subchunk2ID = 0x61746164; // "data"

+	wavHeader.subchunk2Size = totalSampleCounter * sizeof (int16_t);

+	// write main header

+	fwrite(&wavHeader, sizeof (wavHeader_t), 1, f);

+	fclose(f);

+	editor.ui.mod2WavFinished = true;

+	editor.ui.updateMod2WavDialog = true;

+	return true;

+}

+bool renderToWav(char *fileName, bool checkIfFileExist)

+{

+	FILE *fOut;

+	struct stat statBuffer;

+	if (checkIfFileExist)

+	{

+		if (stat(fileName, &statBuffer) == 0)

+		{

+			editor.ui.askScreenShown = true;

+			editor.ui.askScreenType = ASK_MOD2WAV_OVERWRITE;

+			pointerSetMode(POINTER_MODE_MSG1, NO_CARRY);

+			setStatusMessage("OVERWRITE FILE?", NO_CARRY);

+			renderAskDialog();

+			return false;

+		}

+	}

+	if (editor.ui.askScreenShown)

+	{

+		editor.ui.askScreenShown = false;

+		editor.ui.answerNo = false;

+		editor.ui.answerYes = false;

+	}

+	fOut = fopen(fileName, "wb");

+	if (fOut == NULL)

+	{

+		displayErrorMsg("FILE I/O ERROR");

+		return false;

+	}

+	const uint32_t maxSamplesToMix = TICKS_PER_RENDER_CHUNK * audio.bpmTabMod2Wav[32-32]; // BPM 32, stereo

+	mod2WavBuffer = (int16_t *)malloc(maxSamplesToMix * (2 * sizeof (int16_t)));

+	if (mod2WavBuffer == NULL)

+	{

+		statusOutOfMemory();

+		return false;

+	}

+	storeTempVariables();

+	calcMod2WavTotalRows();

+	restartSong();

+	editor.blockMarkFlag = false;

+	pointerSetMode(POINTER_MODE_MSG2, NO_CARRY);

+	setStatusMessage("RENDERING MOD...", NO_CARRY);

+	editor.ui.disableVisualizer = true;

+	editor.isWAVRendering = true;

+	renderMOD2WAVDialog();

+	editor.abortMod2Wav = false;

+	modSetTempo(modEntry->currBPM); // update BPM with MOD2WAV audio output rate

+	editor.mod2WavThread = SDL_CreateThread(mod2WavThreadFunc, NULL, fOut);

+	if (editor.mod2WavThread != NULL)

+	{

+		SDL_DetachThread(editor.mod2WavThread);

+	}

+	else

+	{

+		free(mod2WavBuffer);

+		editor.ui.disableVisualizer = false;

+		editor.isWAVRendering = false;

+		displayErrorMsg("THREAD ERROR");

+		pointerSetMode(POINTER_MODE_IDLE, DO_CARRY);

+		statusAllRight();

+		return false;

+	}

+	return true;

+}

+// ONLY used for a visual percentage counter, so accuracy is not very important

+static void calcMod2WavTotalRows(void)

+{

+	bool pBreakFlag, posJumpAssert, calcingRows;

+	int8_t n_pattpos[AMIGA_VOICES], n_loopcount[AMIGA_VOICES];

+	uint8_t modRow, pBreakPosition, ch, pos;

+	int16_t modOrder;

+	uint16_t modPattern;

+	note_t *note;

+	// for pattern loop

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

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

+	modEntry->rowsCounter = 0;

+	modEntry->rowsInTotal = 0;

+	modRow = 0;

+	modOrder = 0;

+	modPattern = modEntry->head.order[0];

+	pBreakPosition = 0;

+	posJumpAssert = false;

+	pBreakFlag = false;

+	calcingRows = true;

+	memset(editor.rowVisitTable, 0, MOD_ORDERS * MOD_ROWS);

+	while (calcingRows)

+	{

+		editor.rowVisitTable[(modOrder * MOD_ROWS) + modRow] = true;

+		for (ch = 0; ch < AMIGA_VOICES; ch++)

+		{

+			note = &modEntry->patterns[modPattern][(modRow * AMIGA_VOICES) + ch];

+			if (note->command == 0x0B) // Bxx - Position Jump

+			{

+				modOrder = note->param - 1;

+				pBreakPosition = 0;

+				posJumpAssert = true;

+			}

+			else if (note->command == 0x0D) // Dxx - Pattern Break

+			{

+				pBreakPosition = (((note->param >> 4) * 10) + (note->param & 0x0F));

+				if (pBreakPosition > 63)

+					pBreakPosition = 0;

+				posJumpAssert = true;

+			}

+			else if (note->command == 0x0F && note->param == 0) // F00 - Set Speed 0 (stop)

+			{

+				calcingRows = false;

+				break;

+			}

+			else if (note->command == 0x0E && (note->param >> 4) == 0x06) // E6x - Pattern Loop

+			{

+				pos = note->param & 0x0F;

+				if (pos == 0)

+				{

+					n_pattpos[ch] = modRow;

+				}

+				else if (n_loopcount[ch] == 0)

+				{

+					n_loopcount[ch] = pos;

+					pBreakPosition = n_pattpos[ch];

+					pBreakFlag = true;

+					for (pos = pBreakPosition; pos <= modRow; pos++)

+						editor.rowVisitTable[(modOrder * MOD_ROWS) + pos] = false;

+				}

+				else if (--n_loopcount[ch])

+				{

+					pBreakPosition = n_pattpos[ch];

+					pBreakFlag = true;

+					for (pos = pBreakPosition; pos <= modRow; pos++)

+						editor.rowVisitTable[(modOrder * MOD_ROWS) + pos] = false;

+				}

+			}

+		}

+		modRow++;

+		modEntry->rowsInTotal++;

+		if (pBreakFlag)

+		{

+			modRow = pBreakPosition;

+			pBreakPosition = 0;

+			pBreakFlag = false;

+		}

+		if (modRow >= MOD_ROWS || posJumpAssert)

+		{

+			modRow = pBreakPosition;

+			pBreakPosition = 0;

+			posJumpAssert = false;

+			modOrder = (modOrder + 1) & 0x7F;

+			if (modOrder >= modEntry->head.orderCount)

+			{

+				modOrder = 0;

+				calcingRows = false;

+				break;

+			}

+			modPattern = modEntry->head.order[modOrder];

+			if (modPattern > MAX_PATTERNS-1)

+				modPattern = MAX_PATTERNS-1;

+		}

+		if (editor.rowVisitTable[(modOrder * MOD_ROWS) + modRow])

+		{

+			// row has been visited before, we're now done!

+			calcingRows = false;

+			break;

+		}

+	}

+}

--- /dev/null

+++ b/src/pt2_mod2wav.h

@@ -1,0 +1,7 @@

+#pragma once

+#include <stdbool.h>

+#define MOD2WAV_FREQ 96000

+bool renderToWav(char *fileName, bool checkIfFileExist);

--- a/src/pt2_modloader.c

+++ b/src/pt2_modloader.c

@@ -1349,17 +1349,19 @@

 			if (autoPlay)

-				// start normal playback

 				editor.playMode = PLAY_MODE_NORMAL;

-				modPlay(DONT_SET_PATTERN, 0, 0);

 				editor.currMode = MODE_PLAY;

+				// start normal playback

+				modPlay(DONT_SET_PATTERN, 0, 0);

 				pointerSetMode(POINTER_MODE_PLAY, DO_CARRY);

-			else if ((oldMode == MODE_PLAY) || (oldMode == MODE_RECORD))

+			else if (oldMode == MODE_PLAY || oldMode == MODE_RECORD)

 				// use last mode

 				editor.playMode = oldPlayMode;

-				if ((oldPlayMode == PLAY_MODE_PATTERN) || (oldMode == MODE_RECORD))

+				if (oldPlayMode == PLAY_MODE_PATTERN || oldMode == MODE_RECORD)

 					modPlay(0, 0, 0);

 				else

 					modPlay(DONT_SET_PATTERN, 0, 0);

--- a/src/pt2_modplayer.c

+++ b/src/pt2_modplayer.c

@@ -1174,7 +1174,7 @@

 void modSetTempo(uint16_t bpm)

-	int16_t smpsPerTick;

+	uint32_t smpsPerTick;

 	if (bpm < 32)

 		return;

@@ -1190,6 +1190,8 @@

 	if (editor.isSMPRendering)

 		smpsPerTick = editor.pat2SmpHQ ? audio.bpmTab28kHz[bpm] : audio.bpmTab22kHz[bpm];

+	else if (editor.isWAVRendering)

+		smpsPerTick = audio.bpmTabMod2Wav[bpm];

 	else

 		smpsPerTick = audio.bpmTab[bpm];

--- /dev/null

+++ b/src/pt2_pat2smp.c

@@ -1,0 +1,102 @@

+// for finding memory leaks in debug mode with Visual Studio

+#if defined _DEBUG && defined _MSC_VER

+#include <crtdbg.h>

+#endif

+#include <stdint.h>

+#include <stdbool.h>

+#include "pt2_header.h"

+#include "pt2_helpers.h"

+#include "pt2_visuals.h"

+#include "pt2_mouse.h"

+#include "pt2_audio.h"

+#include "pt2_sampler.h"

+#include "pt2_textout.h"

+bool intMusic(void); // pt_modplayer.c

+extern uint32_t samplesPerTick; // pt_audio.c

+void storeTempVariables(void); // pt_modplayer.c

+void doPat2Smp(void)

+{

+	moduleSample_t *s;

+	editor.ui.pat2SmpDialogShown = false;

+	editor.pat2SmpBuf = (int16_t *)malloc(MAX_SAMPLE_LEN * sizeof (int16_t));

+	if (editor.pat2SmpBuf == NULL)

+	{

+		statusOutOfMemory();

+		return;

+	}

+	int8_t oldRow = editor.songPlaying ? 0 : modEntry->currRow;

+	uint32_t oldSamplesPerTick = samplesPerTick;

+	editor.isSMPRendering = true; // this must be set before restartSong()

+	storeTempVariables();

+	restartSong();

+	modEntry->row = oldRow;

+	modEntry->currRow = modEntry->row;

+	editor.blockMarkFlag = false;

+	pointerSetMode(POINTER_MODE_MSG2, NO_CARRY);

+	setStatusMessage("RENDERING...", NO_CARRY);

+	modSetTempo(modEntry->currBPM);

+	editor.pat2SmpPos = 0;

+	editor.smpRenderingDone = false;

+	while (!editor.smpRenderingDone)

+	{

+		if (!intMusic())

+			editor.smpRenderingDone = true;

+		outputAudio(NULL, samplesPerTick);

+	}

+	editor.isSMPRendering = false;

+	resetSong();

+	// set back old row and samplesPerTick

+	modEntry->row = oldRow;

+	modEntry->currRow = modEntry->row;

+	mixerSetSamplesPerTick(oldSamplesPerTick);

+	normalize16bitSigned(editor.pat2SmpBuf, MIN(editor.pat2SmpPos, MAX_SAMPLE_LEN));

+	s = &modEntry->samples[editor.currSample];

+	// quantize to 8-bit

+	for (uint32_t i = 0; i < editor.pat2SmpPos; i++)

+		modEntry->sampleData[s->offset+i] = editor.pat2SmpBuf[i] >> 8;

+	// clear the rest of the sample

+	if (editor.pat2SmpPos < MAX_SAMPLE_LEN)

+		memset(&modEntry->sampleData[s->offset+editor.pat2SmpPos], 0, MAX_SAMPLE_LEN - editor.pat2SmpPos);

+	free(editor.pat2SmpBuf);

+	memset(s->text, 0, sizeof (s->text));

+	if (editor.pat2SmpHQ)

+	{

+		strcpy(s->text, "pat2smp (a-3 tune:+5)");

+		s->fineTune = 5;

+	}

+	else

+	{

+		strcpy(s->text, "pat2smp (f-3 tune:+1)");

+		s->fineTune = 1;

+	}

+	s->length = editor.pat2SmpPos;

+	s->volume = 64;

+	s->loopStart = 0;

+	s->loopLength = 2;

+	editor.samplePos = 0;

+	fixSampleBeep(s);

+	updateCurrSample();

+	pointerSetMode(POINTER_MODE_IDLE, DO_CARRY);

+	displayMsg("ROWS RENDERED!");

+	setMsgPointer();

+}

--- /dev/null

+++ b/src/pt2_pat2smp.h

@@ -1,0 +1,3 @@

+#pragma once

+void doPat2Smp(void);

--- a/src/pt2_sampler.c

+++ b/src/pt2_sampler.c

@@ -44,6 +44,87 @@

};

 void setLoopSprites(void);

+void fixSampleBeep(moduleSample_t *s);

+void upSample(void)

+{

+	moduleSample_t *s = &modEntry->samples[editor.currSample];

+	uint32_t newLength = (s->length >> 1) & 0xFFFE;

+	if (newLength < 2)

+		return;

+	turnOffVoices();

+	// upsample

+	int8_t *ptr8 = &modEntry->sampleData[s->offset];

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

+		ptr8[i] = ptr8[i << 1];

+	// clear junk after shrunk sample

+	if (newLength < MAX_SAMPLE_LEN)

+		memset(&ptr8[newLength], 0, MAX_SAMPLE_LEN - newLength);

+	s->length = (uint16_t)newLength;

+	s->loopStart = (s->loopStart >> 1) & 0xFFFE;

+	s->loopLength = (s->loopLength >> 1) & 0xFFFE;

+	if (s->loopLength < 2)

+	{

+		s->loopStart = 0;

+		s->loopLength = 2;

+	}

+	fixSampleBeep(s);

+	updateCurrSample();

+	editor.ui.updateSongSize = true;

+	updateWindowTitle(MOD_IS_MODIFIED);

+}

+void downSample(void)

+{

+	moduleSample_t *s = &modEntry->samples[editor.currSample];

+	uint32_t newLength = s->length << 1;

+	if (newLength > MAX_SAMPLE_LEN)

+		newLength = MAX_SAMPLE_LEN;

+	turnOffVoices();

+	// downsample

+	int8_t *ptr8 = &modEntry->sampleData[s->offset];

+	int8_t *ptr8_2 = ptr8 - 1;

+	for (int32_t i = s->length-1; i > 0; i--)

+	{

+		ptr8[i<<1] = ptr8[i];

+		ptr8_2[i<<1] = ptr8_2[i];

+	}

+	s->length = newLength;

+	if (s->loopLength > 2)

+	{

+		uint32_t loopStart = s->loopStart << 1;

+		uint32_t loopLength = s->loopLength << 1;

+		if (loopStart+loopLength > s->length)

+		{

+			loopStart = 0;

+			loopLength = 2;

+		}

+		s->loopStart = (uint16_t)loopStart;

+		s->loopLength = (uint16_t)loopLength;

+	}

+	fixSampleBeep(s);

+	updateCurrSample();

+	editor.ui.updateSongSize = true;

+	updateWindowTitle(MOD_IS_MODIFIED);

+}

 void createSampleMarkTable(void)

--- a/src/pt2_sampler.h

+++ b/src/pt2_sampler.h

@@ -3,6 +3,8 @@

 #include <stdint.h>

 #include <stdbool.h>

+void downSample(void);

+void upSample(void);

 void createSampleMarkTable(void);

 int32_t smpPos2Scr(int32_t pos);

 int32_t scr2SmpPos(int32_t x);

--- a/src/pt2_visuals.c

+++ b/src/pt2_visuals.c

@@ -35,6 +35,8 @@

 #include "pt2_helpers.h"

 #include "pt2_scopes.h"

 #include "pt2_edit.h"

+#include "pt2_pat2smp.h"

+#include "pt2_mod2wav.h"

 typedef struct sprite_t

@@ -58,9 +60,7 @@

 	246, 270, 278, 286, 294, 302

};

-bool intMusic(void); // pt_modplayer.c

-extern int32_t samplesPerTick; // pt_audio.c

-void storeTempVariables(void); // pt_modplayer.c

 void updateSongInfo1(void);

 void updateSongInfo2(void);

 void updateSampler(void);

@@ -1449,6 +1449,7 @@

 			editor.isWAVRendering = false;

+			modSetTempo(modEntry->currBPM); // update BPM with normal audio output rate

 			displayMainScreen();

 		else

@@ -1785,8 +1786,7 @@

 void handleAskYes(void)

 	char fileName[20 + 4 + 1];

-	int8_t *tmpSmpBuffer, oldSample, oldRow;

-	int32_t j, newLength, oldSamplesPerTick, loopStart, loopLength;

+	int8_t oldSample;

 	uint32_t i;

 	moduleSample_t *s;

@@ -1816,90 +1816,7 @@

 		case ASK_PAT2SMP:

 			restoreStatusAndMousePointer();

-			editor.ui.pat2SmpDialogShown = false;

-			editor.pat2SmpBuf = (int16_t *)malloc(MAX_SAMPLE_LEN * sizeof (int16_t));

-			if (editor.pat2SmpBuf == NULL)

-			{

-				statusOutOfMemory();

-				return;

-			}

-			oldRow = editor.songPlaying ? 0 : modEntry->currRow;

-			oldSamplesPerTick = samplesPerTick;

-			editor.isSMPRendering = true; // this must be set before restartSong()

-			storeTempVariables();

-			restartSong();

-			modEntry->row = oldRow;

-			modEntry->currRow = modEntry->row;

-			editor.blockMarkFlag = false;

-			pointerSetMode(POINTER_MODE_MSG2, NO_CARRY);

-			setStatusMessage("RENDERING...", NO_CARRY);

-			modSetTempo(modEntry->currBPM);

-			editor.pat2SmpPos = 0;

-			editor.smpRenderingDone = false;

-			while (!editor.smpRenderingDone)

-			{

-				if (!intMusic())

-					editor.smpRenderingDone = true;

-				outputAudio(NULL, samplesPerTick);

-			}

-			editor.isSMPRendering = false;

-			resetSong();

-			// set back old row and samplesPerTick

-			modEntry->row = oldRow;

-			modEntry->currRow = modEntry->row;

-			mixerSetSamplesPerTick(oldSamplesPerTick);

-			// normalize 16-bit samples

-			normalize16bitSigned(editor.pat2SmpBuf, MIN(editor.pat2SmpPos, MAX_SAMPLE_LEN));

-			s = &modEntry->samples[editor.currSample];

-			// quantize to 8-bit

-			for (i = 0; i < editor.pat2SmpPos; i++)

-				modEntry->sampleData[s->offset+i] = editor.pat2SmpBuf[i] >> 8;

-			// clear the rest of the sample

-			if (editor.pat2SmpPos < MAX_SAMPLE_LEN)

-				memset(&modEntry->sampleData[s->offset+editor.pat2SmpPos], 0, MAX_SAMPLE_LEN - editor.pat2SmpPos);

-			// free temp mixing buffer

-			free(editor.pat2SmpBuf);

-			// zero out sample text

-			memset(s->text, 0, sizeof (s->text));

-			// set new sample text

-			if (editor.pat2SmpHQ)

-			{

-				strcpy(s->text, "pat2smp (a-3 tune:+5)");

-				s->fineTune = 5;

-			}

-			else

-			{

-				strcpy(s->text, "pat2smp (f-3 tune:+1)");

-				s->fineTune = 1;

-			}

-			// new sample attributes

-			s->length = editor.pat2SmpPos;

-			s->volume = 64;

-			s->loopStart = 0;

-			s->loopLength = 2;

-			pointerSetMode(POINTER_MODE_IDLE, DO_CARRY);

-			displayMsg("ROWS RENDERED!");

-			setMsgPointer();

-			editor.samplePos = 0;

-			fixSampleBeep(s);

-			updateCurrSample();

+			doPat2Smp();

 		break;

@@ -1961,48 +1878,7 @@

 		case ASK_UPSAMPLE:

 			restoreStatusAndMousePointer();

-			s = &modEntry->samples[editor.currSample];

-			tmpSmpBuffer = (int8_t *)malloc(s->length);

-			if (tmpSmpBuffer == NULL)

-			{

-				statusOutOfMemory();

-				return;

-			}

-			newLength = (s->length / 2) & 0xFFFE;

-			if (newLength < 2)

-				return;

-			turnOffVoices();

-			memcpy(tmpSmpBuffer, &modEntry->sampleData[s->offset], s->length);

-			// upsample

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

-				modEntry->sampleData[s->offset + j] = tmpSmpBuffer[j * 2];

-			if (newLength < MAX_SAMPLE_LEN)

-				memset(&modEntry->sampleData[s->offset + newLength], 0, MAX_SAMPLE_LEN - newLength);

-			free(tmpSmpBuffer);

-			s->length = newLength;

-			s->loopStart = (s->loopStart / 2) & 0xFFFE;

-			s->loopLength = (s->loopLength / 2) & 0xFFFE;

-			if (s->loopLength < 2)

-			{

-				s->loopStart = 0;

-				s->loopLength = 2;

-			}

-			fixSampleBeep(s);

-			updateCurrSample();

-			editor.ui.updateSongSize = true;

-			updateWindowTitle(MOD_IS_MODIFIED);

+			upSample();

 		break;

@@ -2009,55 +1885,7 @@

 		case ASK_DOWNSAMPLE:

 			restoreStatusAndMousePointer();

-			s = &modEntry->samples[editor.currSample];

-			tmpSmpBuffer = (int8_t *)malloc(s->length);

-			if (tmpSmpBuffer == NULL)

-			{

-				statusOutOfMemory();

-				return;

-			}

-			newLength = s->length * 2;

-			if (newLength > MAX_SAMPLE_LEN)

-				newLength = MAX_SAMPLE_LEN;

-			turnOffVoices();

-			memcpy(tmpSmpBuffer, &modEntry->sampleData[s->offset], s->length);

-			// downsample

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

-				modEntry->sampleData[s->offset+j] = tmpSmpBuffer[j >> 1];

-			if (newLength < MAX_SAMPLE_LEN)

-				memset(&modEntry->sampleData[s->offset+newLength], 0, MAX_SAMPLE_LEN - newLength);

-			free(tmpSmpBuffer);

-			s->length = newLength;

-			if (s->loopLength > 2)

-			{

-				loopStart = s->loopStart * 2;

-				loopLength = s->loopLength * 2;

-				if (loopStart+loopLength > s->length)

-				{

-					loopStart = 0;

-					loopLength = 2;

-				}

-				s->loopStart = (uint16_t)loopStart;

-				s->loopLength = (uint16_t)loopLength;

-			}

-			fixSampleBeep(s);

-			updateCurrSample();

-			editor.ui.updateSongSize = true;

-			updateWindowTitle(MOD_IS_MODIFIED);

+			downSample();

 		break;

--- a/vs2019_project/pt2-clone/pt2-clone.vcxproj

+++ b/vs2019_project/pt2-clone/pt2-clone.vcxproj

@@ -290,9 +290,11 @@

     <ClInclude Include="..\..\src\pt2_header.h" />

     <ClInclude Include="..\..\src\pt2_helpers.h" />

     <ClInclude Include="..\..\src\pt2_keyboard.h" />

+    <ClInclude Include="..\..\src\pt2_mod2wav.h" />

     <ClInclude Include="..\..\src\pt2_modloader.h" />

     <ClInclude Include="..\..\src\pt2_mouse.h" />

     <ClInclude Include="..\..\src\pt2_palette.h" />

+    <ClInclude Include="..\..\src\pt2_pat2smp.h" />

     <ClInclude Include="..\..\src\pt2_patternviewer.h" />

     <ClInclude Include="..\..\src\pt2_sampleloader.h" />

     <ClInclude Include="..\..\src\pt2_sampler.h" />

@@ -331,10 +333,12 @@

     <ClCompile Include="..\..\src\pt2_helpers.c" />

     <ClCompile Include="..\..\src\pt2_keyboard.c" />

     <ClCompile Include="..\..\src\pt2_main.c" />

+    <ClCompile Include="..\..\src\pt2_mod2wav.c" />

     <ClCompile Include="..\..\src\pt2_modloader.c" />

     <ClCompile Include="..\..\src\pt2_modplayer.c" />

     <ClCompile Include="..\..\src\pt2_mouse.c" />

     <ClCompile Include="..\..\src\pt2_palette.c" />

+    <ClCompile Include="..\..\src\pt2_pat2smp.c" />

     <ClCompile Include="..\..\src\pt2_patternviewer.c" />

     <ClCompile Include="..\..\src\pt2_sampleloader.c" />

     <ClCompile Include="..\..\src\pt2_sampler.c" />

--- a/vs2019_project/pt2-clone/pt2-clone.vcxproj.filters

+++ b/vs2019_project/pt2-clone/pt2-clone.vcxproj.filters

@@ -66,6 +66,12 @@

     <ClInclude Include="..\..\src\pt2_visuals.h">

       <Filter>headers</Filter>

     </ClInclude>

+    <ClInclude Include="..\..\src\pt2_mod2wav.h">

+      <Filter>headers</Filter>

+    </ClInclude>

+    <ClInclude Include="..\..\src\pt2_pat2smp.h">

+      <Filter>headers</Filter>

+    </ClInclude>

   </ItemGroup>

   <ItemGroup>

     <ClCompile Include="..\..\src\pt2_audio.c" />

@@ -142,6 +148,8 @@

     <ClCompile Include="..\..\src\gfx\pt2_gfx_yes_no_dialog.c">

       <Filter>gfx</Filter>

     </ClCompile>

+    <ClCompile Include="..\..\src\pt2_mod2wav.c" />

+    <ClCompile Include="..\..\src\pt2_pat2smp.c" />

   </ItemGroup>

   <ItemGroup>

     <ResourceCompile Include="..\..\src\pt2-clone.rc" />