shithub: ft2-clone

--- a/.gitignore

+++ b/.gitignore

@@ -18,3 +18,5 @@

 *.cod

 vs2019_project/ft2-clone/Debug/ft2-clone.vcxproj.FileListAbsolute.txt

 *.opendb

+*.db-shm

+*.db-wal

--- a/src/ft2_audio.c

+++ b/src/ft2_audio.c

@@ -18,14 +18,6 @@

 #define INITIAL_DITHER_SEED 0x12345000

-// globalized

-audio_t audio;

-pattSyncData_t *pattSyncEntry;

-chSyncData_t *chSyncEntry;

-chSync_t chSync;

-pattSync_t pattSync;

-volatile bool pattQueueClearing, chQueueClearing;

 static int8_t pmpCountDiv, pmpChannels = 2;

 static uint16_t smpBuffSize;

 static int32_t masterVol, oldAudioFreq, pmpLeft, randSeed = INITIAL_DITHER_SEED;

@@ -35,19 +27,29 @@

 static voice_t voice[MAX_VOICES * 2];

 static void (*sendAudSamplesFunc)(uint8_t *, uint32_t, uint8_t); // "send mixed samples" routines

-#if !defined __amd64__ && !defined _WIN64

 static int32_t oldPeriod;

+#if !defined __amd64__ && !defined _WIN64

 static uint32_t oldSFrq, oldSFrqRev;

+#else

+static uint64_t oldSFrq;

 #endif

-#if !defined __amd64__ && !defined _WIN64

+// globalized

+audio_t audio;

+pattSyncData_t *pattSyncEntry;

+chSyncData_t *chSyncEntry;

+chSync_t chSync;

+pattSync_t pattSync;

+volatile bool pattQueueClearing, chQueueClearing;

 void resetCachedMixerVars(void)

 	oldPeriod = -1;

 	oldSFrq = 0;

+#if !defined __amd64__ && !defined _WIN64

 	oldSFrqRev = 0xFFFFFFFF;

-}

 #endif

+}

 void stopVoice(int32_t i)

@@ -368,22 +370,23 @@

 		if (status & IS_Period)

-#if defined __amd64__ || defined _WIN64

-			v->SFrq = getFrequenceValue(ch->finalPeriod);

-#else

-			// use cached values to prevent a 32-bit divsion all the time

-			if (ch->finalPeriod != oldPeriod)

+			// use cached values if possible

+			const uint16_t period = ch->finalPeriod;

+			if (period != oldPeriod)

-				oldPeriod = ch->finalPeriod;

+				oldPeriod = period;

+				oldSFrq = getMixerDelta(period);

-				oldSFrq = getFrequenceValue(ch->finalPeriod);

+#if !defined __amd64__ && !defined _WIN64

 				oldSFrqRev = 0xFFFFFFFF;

 				if (oldSFrq != 0)

 					oldSFrqRev /= oldSFrq;

+#endif

 			v->SFrq = oldSFrq;

+#if !defined __amd64__ && !defined _WIN64

 			v->SFrqRev = oldSFrqRev;

 #endif

--- a/src/ft2_audio.h

+++ b/src/ft2_audio.h

@@ -54,7 +54,7 @@

 	uint32_t freq;

 	uint32_t audLatencyPerfValInt, audLatencyPerfValFrac, speedVal, musicTimeSpeedVal;

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

-	double dAudioLatencyMs, dSpeedValMul, dPianoDeltaMul;

+	double dAudioLatencyMs;

 	SDL_AudioDeviceID dev;

 	uint32_t wantFreq, haveFreq, wantSamples, haveSamples, wantChannels, haveChannels;

 } audio_t;

@@ -104,21 +104,7 @@

 	chSyncData_t data[SYNC_QUEUE_LEN + 1];

 } chSync_t;

-// in ft2_audio.c

-extern audio_t audio;

-extern pattSyncData_t *pattSyncEntry;

-extern chSyncData_t *chSyncEntry;

-extern chSync_t chSync;

-extern pattSync_t pattSync;

-extern volatile bool pattQueueClearing, chQueueClearing;

-#if !defined __amd64__ && !defined _WIN64

 void resetCachedMixerVars(void);

-#endif

-void calcAudioTables(void);

 int32_t pattQueueReadSize(void);

 int32_t pattQueueWriteSize(void);

 bool pattQueuePush(pattSyncData_t t);

@@ -153,3 +139,12 @@

 void mix_SaveIPVolumes(void);

 void mix_UpdateChannelVolPanFrq(void);

 uint32_t mixReplayerTickToBuffer(uint8_t *stream, uint8_t bitDepth);

+// in ft2_audio.c

+extern audio_t audio;

+extern pattSyncData_t *pattSyncEntry;

+extern chSyncData_t *chSyncEntry;

+extern chSync_t chSync;

+extern pattSync_t pattSync;

+extern volatile bool pattQueueClearing, chQueueClearing;

--- a/src/ft2_header.h

+++ b/src/ft2_header.h

@@ -12,7 +12,7 @@

 #endif

 #include "ft2_replayer.h"

-#define PROG_VER_STR "1.25"

+#define PROG_VER_STR "1.26"

 // do NOT change these! It will only mess things up...

--- a/src/ft2_main.c

+++ b/src/ft2_main.c

@@ -312,7 +312,7 @@

 	audio.linearFreqTable = true;

-	calcAudioTables();

+	calcReplayerLogTab();

 static void cleanUpAndExit(void) // never call this inside the main loop!

--- a/src/ft2_replayer.c

+++ b/src/ft2_replayer.c

@@ -26,20 +26,10 @@

 ** If something looks to be off, it probably isn't!

*/

-/* Tables for pre-calculated stuff on run time and when changing freq. and/or linear/amiga mode.

-** FT2 obviously didn't have such big tables.

-*/

+// non-FT2 precalced stuff

 static uint32_t musicTimeTab[MAX_BPM+1];

-static uint64_t period2ScopeDeltaTab[65536];

-static double dLogTab[768], dLogTabMul[32], dAudioRateFactor;

+static double dPeriod2HzTab[65536], dLogTab[768], dAudioRateFactor;

-#if defined _WIN64 || defined __amd64__

-static uint64_t period2DeltaTab[65536];

-#else

-static uint32_t period2DeltaTab[65536];

-#endif

 static bool bxxOverflow;

 static tonTyp nilPatternLine;

@@ -141,8 +131,8 @@

 		return;

-	double dFreq = log2(midCFreq * (1.0 / 8363.0)) * (12.0 * 128.0);

-	int32_t linearFreq = (int32_t)(dFreq + 0.5);

+	double dFreq = log2(midCFreq / 8363.0) * (12.0 * 128.0);

+	int32_t linearFreq = (int32_t)(dFreq + 0.5); // rounded

 	s->fine = ((linearFreq + 128) & 255) - 128;

 	int32_t relTon = (linearFreq - s->fine) >> 7;

@@ -220,61 +210,29 @@

 	return i+1;

-// called every time you change linear/amiga mode and mixing frequency

-static void calcPeriod2DeltaTables(void)

+static void calcPeriod2HzTab(void) // called every time you change linear/amiga period mode

-	int32_t baseDelta;

-	uint32_t i;

+	dPeriod2HzTab[0] = 0.0; // in FT2, a period of 0 yields 0Hz

-	period2DeltaTab[0] = 0; // in FT2, a period of 0 is converted to a delta of 0

-	const double dScopeRateFactor = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;

 	if (audio.linearFreqTable)

 		// linear periods

-		for (i = 1; i < 65536; i++)

+		for (int32_t i = 1; i < 65536; i++)

 			const uint16_t invPeriod = (12 * 192 * 4) - (uint16_t)i; // this intentionally overflows uint16_t to be accurate to FT2

 			const int32_t octave = invPeriod / 768;

 			const int32_t period = invPeriod % 768;

-			const int32_t shift = (14 - octave) & 0x1F; // 100% accurate to FT2!

+			const int32_t bitshift = (14 - octave) & 0x1F; // 100% accurate to FT2

-			const double dHz = dLogTab[period] * dLogTabMul[shift];

-#if defined _WIN64 || defined __amd64__

-			period2DeltaTab[i] = (uint64_t)((dHz * dAudioRateFactor) + 0.5);

-#else

-			period2DeltaTab[i] = (uint32_t)((dHz * dAudioRateFactor) + 0.5);

-#endif

-			period2ScopeDeltaTab[i] = (uint64_t)((dHz * dScopeRateFactor) + 0.5);

+			dPeriod2HzTab[i] = dLogTab[period] / (1UL << bitshift);

 	else

 		// Amiga periods

-		for (i = 1; i < 65536; i++)

-		{

-			double dHz = (8363.0 * 1712.0) / i;

-#if defined _WIN64 || defined __amd64__

-			period2DeltaTab[i] = (uint64_t)((dHz * dAudioRateFactor) + 0.5);

-#else

-			period2DeltaTab[i] = (uint32_t)((dHz * dAudioRateFactor) + 0.5);

-#endif

-			period2ScopeDeltaTab[i] = (uint64_t)((dHz * dScopeRateFactor) + 0.5);

-		}

+		for (int32_t i = 1; i < 65536; i++)

+			dPeriod2HzTab[i] = (8363.0 * 1712.0) / i;

-	// for piano in Instr. Ed.

-	// (this delta is small enough to fit in int32_t even with 32.32 deltas)

-	if (audio.linearFreqTable)

-		baseDelta = (int32_t)period2DeltaTab[7680];

-	else

-		baseDelta = (int32_t)period2DeltaTab[1712*16];

-	audio.dPianoDeltaMul = 1.0 / baseDelta;

 void setFrqTab(bool linear)

@@ -288,7 +246,7 @@

 	else

 		note2Period = amigaPeriods;

-	calcPeriod2DeltaTables();

+	calcPeriod2HzTab();

 	resumeAudio();

@@ -302,7 +260,7 @@

 	ch->realVol = ch->oldVol;

 	ch->outVol = ch->oldVol;

 	ch->outPan = ch->oldPan;

-	ch->status |= (IS_Vol + IS_Pan + IS_QuickVol);

+	ch->status |= IS_Vol + IS_Pan + IS_QuickVol;

 static void retrigEnvelopeVibrato(stmTyp *ch)

@@ -376,26 +334,18 @@

 		ch->realVol = 0;

 		ch->outVol = 0;

-		ch->status |= (IS_Vol + IS_QuickVol);

+		ch->status |= IS_Vol + IS_QuickVol;

-void calcAudioTables(void)

+void calcReplayerLogTab(void)

-	int32_t i;

-	for (i = 0; i < 768; i++)

+	for (int32_t i = 0; i < 768; i++)

 		dLogTab[i] = exp2(i / 768.0) * (8363.0 * 256.0);

-	dLogTabMul[0] = 1.0;

-	for (i = 1; i < 32; i++)

-		dLogTabMul[i] = exp2(-i);

 void calcReplayRate(int32_t audioFreq)

-	int32_t i;

 	if (audioFreq == 0)

 		return;

@@ -403,77 +353,66 @@

 	audio.quickVolSizeVal = audioFreq / 200; // FT2 truncates here

 	audio.rampQuickVolMul = (int32_t)(((UINT32_MAX + 1.0) / audio.quickVolSizeVal) + 0.5);

-	audio.dSpeedValMul = editor.dPerfFreq / audioFreq; // for audio/video sync

 	/* Calculate tables to prevent floating point operations on systems that

 	** might have a slow FPU. This is quite hackish and not really needed,

-	** but it doesn't take up THAT much RAM anyway.

+	** but it doesn't take up a lot of RAM, so why not.

*/

-	// calculate table used to count replayer time (displayed as hours/minutes/seconds)

-	const double dMul = (UINT32_MAX + 1.0) / audioFreq;

 	audio.speedValTab[0] = 0;

 	musicTimeTab[0] = UINT32_MAX;

-	audio.rampSpeedValMulTab[0] = UINT32_MAX;

-	audio.tickTimeLengthTab[0] = (uint64_t)UINT32_MAX << 32;

+	audio.tickTimeLengthTab[0] = UINT64_MAX;

+	audio.rampSpeedValMulTab[0] = INT32_MAX;

-	const double dTickTimeLenMul = audio.dSpeedValMul * (UINT32_MAX + 1.0);

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

-	{

-		int32_t samplesPerTick = (int32_t)(((audioFreq * 2.5) / i) + 0.5); // rounded

+	const double dMul1 = (UINT32_MAX + 1.0) / audioFreq;

+	const double dMul2 = (editor.dPerfFreq / audioFreq) * (UINT32_MAX + 1.0);

+	for (int32_t i = 1; i <= MAX_BPM; i++)

+	{

+		const int32_t samplesPerTick = (int32_t)(((audioFreq * 2.5) / i) + 0.5); // rounded

 		audio.speedValTab[i] = samplesPerTick;

 		// used for song playback counter (hh:mm:ss)

-		musicTimeTab[i] = (uint32_t)((samplesPerTick * dMul) + 0.5);

+		musicTimeTab[i] = (uint32_t)((samplesPerTick * dMul1) + 0.5);

 		// number of samples per tick -> tick length for performance counter (syncing visuals to audio)

-		audio.tickTimeLengthTab[i] = (uint64_t)(samplesPerTick * dTickTimeLenMul);

+		audio.tickTimeLengthTab[i] = (uint64_t)(samplesPerTick * dMul2);

 		// for calculating volume ramp length for "tick" ramps

 		audio.rampSpeedValMulTab[i] = (int32_t)(((UINT32_MAX + 1.0) / samplesPerTick) + 0.5);

+}

-	calcPeriod2DeltaTables();

+double period2Hz(uint16_t period)

+{

+	return dPeriod2HzTab[period];

 #if defined _WIN64 || defined __amd64__

-uint64_t getFrequenceValue(uint16_t period)

+int64_t getMixerDelta(uint16_t period)

-	return period2DeltaTab[period];

+	return (int64_t)((dPeriod2HzTab[period] * dAudioRateFactor) + 0.5); // Hz -> rounded fixed-point mixer delta

 #else

-uint32_t getFrequenceValue(uint16_t period)

+int32_t getMixerDelta(uint16_t period)

-	return period2DeltaTab[period];

+	return (int32_t)((dPeriod2HzTab[period] * dAudioRateFactor) + 0.5); // Hz -> rounded fixed-point mixer delta

 #endif

 int32_t getPianoKey(uint16_t period, int32_t finetune, int32_t relativeNote) // for piano in Instr. Ed.

-#if defined _WIN64 || defined __amd64__

-	uint64_t delta = period2DeltaTab[period];

-#else

-	uint32_t delta = period2DeltaTab[period];

-#endif

 	finetune >>= 3; // FT2 does this in the replayer internally, so the actual range is -16..15

-	const double dNote = (log2(delta * audio.dPianoDeltaMul) * 12.0) - (finetune * (1.0 / 16.0));

-	int32_t note = (int32_t)(dNote + 0.5);

+	const double dRelativeHz = dPeriod2HzTab[period] * (1.0 / (8363.0 / 16.0));

+	const double dNote = (log2(dRelativeHz) * 12.0) - (finetune * (1.0 / 16.0));

-	note -= relativeNote;

+	const int32_t note = (int32_t)(dNote + 0.5) - relativeNote; // rounded

-	// "note" is now the raw piano key number, unaffected by finetune/relativeNote

+	// "note" is now the raw piano key number, unaffected by finetune and relativeNote

 	return note;

-uint64_t getScopeFrequenceValue(uint16_t period)

-{

-	return period2ScopeDeltaTab[period];

-}

 static void startTone(uint8_t ton, uint8_t effTyp, uint8_t eff, stmTyp *ch)

 	uint8_t smp;

@@ -538,7 +477,7 @@

-	ch->status |= (IS_Period + IS_Vol + IS_Pan + IS_NyTon + IS_QuickVol);

+	ch->status |= IS_Period + IS_Vol + IS_Pan + IS_NyTon + IS_QuickVol;

 	if (effTyp == 9)

@@ -787,7 +726,7 @@

 				ch->realVol = 0;

 				ch->outVol = 0;

-				ch->status |= (IS_Vol + IS_QuickVol);

+				ch->status |= IS_Vol + IS_QuickVol;

@@ -978,7 +917,7 @@

 		ch->outVol = volKol;

 		ch->realVol = volKol;

-		ch->status |= (IS_Vol + IS_QuickVol);

+		ch->status |= IS_Vol + IS_QuickVol;

 	// fine volume slide down

@@ -1035,7 +974,7 @@

 			ch->realVol = 64;

 		ch->outVol = ch->realVol;

-		ch->status |= (IS_Vol + IS_QuickVol);

+		ch->status |= IS_Vol + IS_QuickVol;

 		return;

@@ -1939,7 +1878,7 @@

 				ch->outVol = 0;

 				ch->realVol = 0;

-				ch->status |= (IS_Vol + IS_QuickVol);

+				ch->status |= IS_Vol + IS_QuickVol;

@@ -2064,7 +2003,7 @@

 		ch->tremorPos = tremorSign | tremorData;

 		ch->outVol = (tremorSign == 0x80) ? ch->realVol : 0;

-		ch->status |= (IS_Vol + IS_QuickVol);

+		ch->status |= IS_Vol + IS_QuickVol;

@@ -2779,6 +2718,7 @@

 	audio.linearFreqTable = true;

 	note2Period = linearPeriods;

+	calcPeriod2HzTab();

 	setPos(0, 0, true);

@@ -3114,9 +3054,8 @@

 	stopAllScopes();

 	resetAudioDither();

-#if !defined __amd64__ && !defined _WIN64

 	resetCachedMixerVars();

-#endif

+	resetCachedScopeVars();

 	// wait for scope thread to finish, so that we know pointers aren't deprecated

 	while (editor.scopeThreadMutex);

--- a/src/ft2_replayer.h

+++ b/src/ft2_replayer.h

@@ -237,18 +237,18 @@

 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 calcReplayRate(int32_t rate);

 void tuneSample(sampleTyp *s, int32_t midCFreq);

+void calcReplayerLogTab(void);

+double period2Hz(uint16_t period);

 #if defined _WIN64 || defined __amd64__

-uint64_t getFrequenceValue(uint16_t period);

+int64_t getMixerDelta(uint16_t period);

 #else

-uint32_t getFrequenceValue(uint16_t period);

+int32_t getMixerDelta(uint16_t period);

 #endif

-uint64_t getScopeFrequenceValue(uint16_t period);

 int32_t getPianoKey(uint16_t period, int32_t finetune, int32_t relativeNote); // for piano in Instr. Ed.

 bool allocateInstr(int16_t nr);

--- a/src/ft2_scopedraw.c

+++ b/src/ft2_scopedraw.c

@@ -101,13 +101,13 @@

 #define SCOPE_UPDATE_DRAWPOS \

 	scopeDrawFrac += scopeDrawDelta; \

-	scopeDrawPos += scopeDrawFrac >> 16; \

-	scopeDrawFrac &= 0xFFFF; \

+	scopeDrawPos += scopeDrawFrac >> SCOPE_DRAW_FRAC_BITS; \

+	scopeDrawFrac &= SCOPE_DRAW_FRAC_MASK; \

 #define SCOPE_UPDATE_DRAWPOS_PINGPONG \

 	scopeDrawFrac += scopeDrawDelta; \

-	scopeDrawPos += (scopeDrawFrac >> 16) * drawPosDir; \

-	scopeDrawFrac &= 0xFFFF; \

+	scopeDrawPos += (scopeDrawFrac >> SCOPE_DRAW_FRAC_BITS) * drawPosDir; \

+	scopeDrawFrac &= SCOPE_DRAW_FRAC_MASK; \

 #define SCOPE_DRAW_SMP \

 	video.frameBuffer[((lineY - sample) * SCREEN_W) + x] = scopePixelColor;

--- a/src/ft2_scopedraw.h

+++ b/src/ft2_scopedraw.h

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

 #include <stdint.h>

 #include "ft2_scopes.h"

-typedef void (*scopeDrawRoutine)(scope_t *, uint32_t, uint32_t, uint32_t);

+typedef void (*scopeDrawRoutine)(const scope_t *, uint32_t, uint32_t, uint32_t);

 extern const scopeDrawRoutine scopeDrawRoutineTable[12]; // ft2_scopedraw.c

--- a/src/ft2_scopes.c

+++ b/src/ft2_scopes.c

@@ -41,8 +41,9 @@

 } scopeState_t;

 static volatile bool scopesUpdatingFlag, scopesDisplayingFlag;

-static uint32_t scopeTimeLen, scopeTimeLenFrac;

-static uint64_t timeNext64, timeNext64Frac;

+static int32_t oldPeriod;

+static uint32_t oldDFrq, scopeTimeLen, scopeTimeLenFrac;

+static uint64_t oldSFrq, timeNext64, timeNext64Frac;

 static volatile scope_t scope[MAX_VOICES];

 static SDL_Thread *scopeThread;

 static uint8_t *scopeMuteBMP_Ptrs[16];

@@ -49,6 +50,13 @@

 lastChInstr_t lastChInstr[MAX_VOICES]; // global

+void resetCachedScopeVars(void)

+{

+	oldPeriod = -1;

+	oldSFrq = 0;

+	oldDFrq = 0;

+}

 int32_t getSamplePosition(uint8_t ch)

 	volatile bool active, sample16Bit;

@@ -202,7 +210,7 @@

 void refreshScopes(void)

-	for (int16_t i = 0; i < MAX_VOICES; i++)

+	for (int32_t i = 0; i < MAX_VOICES; i++)

 		scope[i].wasCleared = false;

@@ -292,7 +300,7 @@

 			if (mouse.x >= x && mouse.x < x+scopeLens[i])

 				break;

-			x += scopeLens[i] + 3;

+			x += scopeLens[i]+3;

 		if (i == chansPerRow)

@@ -309,7 +317,7 @@

 	return false;

-static void scopeTrigger(int32_t ch, sampleTyp *s, int32_t playOffset)

+static void scopeTrigger(int32_t ch, const sampleTyp *s, int32_t playOffset)

 	bool sampleIs16Bit;

 	uint8_t loopType;

@@ -387,15 +395,13 @@

 static void updateScopes(void)

 	int32_t loopOverflowVal;

-	volatile scope_t *sc;

-	scope_t tempState;

 	scopesUpdatingFlag = true;

-	sc = scope;

+	volatile scope_t *sc = scope;

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

-		tempState = *sc; // cache it

+		scope_t tempState = *sc; // cache it

 		if (!tempState.active)

 			continue; // scope is not active, no need

@@ -402,17 +408,15 @@

 		// scope position update

 		tempState.SPosDec += tempState.SFrq;

 		const uint32_t posAdd = tempState.SPosDec >> SCOPE_FRAC_BITS;

+		tempState.SPosDec &= SCOPE_FRAC_MASK;

 		if (tempState.backwards)

 			tempState.SPos -= posAdd;

 		else

 			tempState.SPos += posAdd;

-		tempState.SPosDec &= SCOPE_FRAC_MASK;

 		// handle loop wrapping or sample end

 		if (tempState.backwards && tempState.SPos < tempState.SRepS) // sampling backwards (definitely pingpong loop)

 			tempState.backwards = false; // change direction to forwards

@@ -460,9 +464,7 @@

 	const uint16_t *scopeLens;

 	uint16_t scopeXOffs, scopeYOffs, scopeDrawLen;

 	int32_t chansPerRow;

-	volatile scope_t *sc;

-	scope_t s;

 	scopesDisplayingFlag = true;

 	chansPerRow = (uint32_t)song.antChn >> 1;

@@ -482,18 +484,16 @@

 		scopeDrawLen = scopeLens[i];

 		if (!editor.chnMode[i]) // scope muted (mute graphics blit()'ed elsewhere)

-			scopeXOffs += scopeDrawLen + 3; // align x to next scope

+			scopeXOffs += scopeDrawLen+3; // align x to next scope

 			continue;

-		s = scope[i]; // cache scope to lower thread race condition issues

+		const scope_t s = scope[i]; // cache scope to lower thread race condition issues

 		if (s.active && s.SVol > 0 && !audio.locked)

 			// scope is active

 			scope[i].wasCleared = false;

 			// clear scope background

@@ -506,8 +506,7 @@

 		else

 			// scope is inactive

-			sc = &scope[i];

+			volatile scope_t *sc = &scope[i];

 			if (!sc->wasCleared)

 				// clear scope background

@@ -528,7 +527,7 @@

 		if (config.multiRecChn[i])

 			blit(scopeXOffs + 1, scopeYOffs + 31, bmp.scopeRec, 13, 4);

-		scopeXOffs += scopeDrawLen + 3; // align x to next scope

+		scopeXOffs += scopeDrawLen+3; // align x to next scope

 	scopesDisplayingFlag = false;

@@ -537,7 +536,7 @@

 void drawScopeFramework(void)

 	drawFramework(0, 92, 291, 81, FRAMEWORK_TYPE1);

-	for (uint8_t i = 0; i < song.antChn; i++)

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

 		redrawScope(i);

@@ -544,11 +543,9 @@

 void handleScopesFromChQueue(chSyncData_t *chSyncData, uint8_t *scopeUpdateStatus)

 	uint8_t status;

-	syncedChannel_t *ch;

-	volatile scope_t *sc;

-	sc = scope;

-	ch = chSyncData->channels;

+	volatile scope_t *sc = scope;

+	syncedChannel_t *ch = chSyncData->channels;

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

 		status = scopeUpdateStatus[i];

@@ -558,8 +555,23 @@

 		if (status & IS_Period)

-			sc->SFrq = getScopeFrequenceValue(ch->finalPeriod);

-			sc->DFrq = (uint32_t)(sc->SFrq >> (SCOPE_FRAC_BITS - 10)); // amount of samples to skip after drawing a pixel

+			// use cached values if possible

+			const uint16_t period = ch->finalPeriod;

+			if (period != oldPeriod)

+			{

+				oldPeriod = period;

+				const double dHz = period2Hz(period);

+				const double dScopeRateFactor = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;

+				oldSFrq = (int64_t)((dHz * dScopeRateFactor) + 0.5); // Hz -> rounded fixed-point delta

+				const double dRelativeHz = dHz * (1.0 / (8363.0 / 2.0));

+				oldDFrq = (int32_t)((dRelativeHz * SCOPE_DRAW_FRAC_SCALE) + 0.5); // Hz -> rounded fixed-point draw delta

+			}

+			sc->SFrq = oldSFrq;

+			sc->DFrq = oldDFrq;

 		if (status & IS_NyTon)

--- a/src/ft2_scopes.h

+++ b/src/ft2_scopes.h

@@ -9,6 +9,12 @@

 #define SCOPE_FRAC_SCALE (1ULL << SCOPE_FRAC_BITS)

 #define SCOPE_FRAC_MASK (SCOPE_FRAC_SCALE-1)

+// just about max safe bits, don't mess with it!

+#define SCOPE_DRAW_FRAC_BITS 20

+#define SCOPE_DRAW_FRAC_SCALE (1UL << SCOPE_DRAW_FRAC_BITS)

+#define SCOPE_DRAW_FRAC_MASK (SCOPE_DRAW_FRAC_SCALE-1)

+void resetCachedScopeVars(void);

 int32_t getSamplePosition(uint8_t ch);

 void stopAllScopes(void);

 void refreshScopes(void);

--- a/vs2019_project/ft2-clone/ft2-clone.vcxproj.filters

+++ b/vs2019_project/ft2-clone/ft2-clone.vcxproj.filters

@@ -22,7 +22,6 @@

     <ClCompile Include="..\..\src\ft2_unicode.c" />

     <ClCompile Include="..\..\src\ft2_midi.c" />

     <ClCompile Include="..\..\src\ft2_wav_renderer.c" />

-    <ClCompile Include="..\..\src\ft2_about.c" />

     <ClCompile Include="..\..\src\ft2_trim.c" />

     <ClCompile Include="..\..\src\ft2_checkboxes.c" />

     <ClCompile Include="..\..\src\ft2_pushbuttons.c" />

@@ -77,6 +76,7 @@

       <Filter>graphics</Filter>

     </ClCompile>

     <ClCompile Include="..\..\src\ft2_structs.c" />

+    <ClCompile Include="..\..\src\ft2_about.c" />

   </ItemGroup>

   <ItemGroup>

     <ClInclude Include="..\..\src\ft2_audio.h">