commit f71a1d5fc0085f357a6c5f12a72fad5926fc7ab5
parent 0cc01d0968f47593885846ac23878a89df22ba32
Author: Olav Sørensen <olav.sorensen@live.no>
Date: Mon, 22 Nov 2021 22:49:01 +0100
Fixed small bug in instrument editor piano (+ small cleanup)
Diffstat:
3 files changed, 46 insertions(+), 57 deletions(-)
diff --git a/src/ft2_header.h b/src/ft2_header.h
@@ -12,7 +12,7 @@
#endif
#include "ft2_replayer.h"
-#define PROG_VER_STR "1.47"
+#define PROG_VER_STR "1.48"
// do NOT change these! It will only mess things up...
@@ -56,11 +56,11 @@
#define CLAMP(x, low, high) (((x) > (high)) ? (high) : (((x) < (low)) ? (low) : (x)))
#define DROUND(x) \
- if (x < 0.0) x -= 0.5; \
+ if (x < 0.0) x -= 0.5; \
else if (x > 0.0) x += 0.5
#define FROUND(x) \
- if (x < 0.0f) x -= 0.5f; \
+ if (x < 0.0f) x -= 0.5f; \
else if (x > 0.0f) x += 0.5f
// fast 32-bit -> 8-bit clamp
diff --git a/src/ft2_replayer.c b/src/ft2_replayer.c
@@ -22,12 +22,7 @@
#include "ft2_structs.h"
#include "mixer/ft2_windowed_sinc.h"
-/* This is a mess, directly ported from the original FT2 code (with some modifications).
-** You will experience a lot of headaches if you dig into it...
-** If something looks to be off, it probably isn't!
-*/
-
-static double dLogTab[768], dExp2MulTab[32];
+static double dLogTab[4*12*16], dExp2MulTab[32];
static bool bxxOverflow;
static note_t nilPatternLine[MAX_CHANNELS];
@@ -43,7 +38,7 @@ const uint16_t *note2Period = NULL;
int16_t patternNumRows[MAX_PATTERNS];
channel_t channel[MAX_CHANNELS];
song_t song;
-instr_t *instr[132];
+instr_t *instr[128+4];
note_t *pattern[MAX_PATTERNS];
// ----------------------------------
@@ -111,10 +106,9 @@ void removeSongModifiedFlag(void)
editor.updateWindowTitle = true;
}
-// used on external sample load and during sample loading in some module formats
+// used on external sample load and during sample loading (on some module formats)
void tuneSample(sample_t *s, const int32_t midCFreq, bool linearPeriodsFlag)
{
- #define NOTE_C4 (4*12)
#define MIN_PERIOD (0)
#define MAX_PERIOD (((10*12*16)-1)-1) /* -1 (because of bugged amigaPeriods table values) */
@@ -257,19 +251,23 @@ int16_t getRealUsedSamples(int16_t smpNum)
double dLinearPeriod2Hz(int32_t period)
{
+ period &= 0xFFFF; // just in case (actual period range is 0..65535)
+
if (period == 0)
return 0.0; // in FT2, a period of 0 results in 0Hz
- const uint32_t invPeriod = ((12 * 192 * 4) - period) & 0xFFFF; // mask needed for FT2 frequency quirk
+ const uint32_t invPeriod = ((12 * 192 * 4) - period) & 0xFFFF; // mask needed for FT2 period overflow quirk
- const uint32_t quotient = invPeriod / 768;
- const uint32_t remainder = invPeriod % 768;
+ const uint32_t quotient = invPeriod / (12 * 16 * 4);
+ const uint32_t remainder = invPeriod % (12 * 16 * 4);
return dLogTab[remainder] * dExp2MulTab[(14-quotient) & 31]; // x = y >> ((14-quotient) & 31);
}
double dAmigaPeriod2Hz(int32_t period)
{
+ period &= 0xFFFF; // just in case (actual period range is 0..65535)
+
if (period == 0)
return 0.0; // in FT2, a period of 0 results in 0Hz
@@ -284,13 +282,16 @@ double dPeriod2Hz(int32_t period)
// returns *exact* FT2 C-4 voice rate (depending on finetune, relativeNote and linear/Amiga period mode)
double getSampleC4Rate(sample_t *s)
{
- int32_t note = (96/2) + s->relativeNote;
+ int32_t note = NOTE_C4 + s->relativeNote;
+ if (note < 0)
+ return -1; // shouldn't happen (just in case...)
+
if (note >= (10*12)-1)
- return -1; // B-9 (from relativeTone) = illegal! (won't play in replayer)
+ return -1; // B-9 (after relativeTone add) = illegal! (won't play in replayer)
const int32_t C4Period = (note << 4) + (((int8_t)s->finetune >> 3) + 16);
- const uint16_t period = audio.linearPeriodsFlag ? linearPeriods[C4Period] : amigaPeriods[C4Period];
+ const int32_t period = audio.linearPeriodsFlag ? linearPeriods[C4Period] : amigaPeriods[C4Period];
return dPeriod2Hz(period);
}
@@ -328,7 +329,7 @@ static void retrigVolume(channel_t *ch)
static void retrigEnvelopeVibrato(channel_t *ch)
{
- if (!(ch->waveCtrl & 0x04)) ch->vibPos = 0;
+ if (!(ch->waveCtrl & 0x04)) ch->vibPos = 0;
if (!(ch->waveCtrl & 0x40)) ch->tremPos = 0;
ch->retrigCnt = 0;
@@ -397,13 +398,13 @@ void keyOff(channel_t *ch)
}
}
-void calcReplayerLogTab(void)
+void calcReplayerLogTab(void) // for linear period -> hz calculation
{
for (int32_t i = 0; i < 32; i++)
dExp2MulTab[i] = 1.0 / exp2(i); // 1/(2^i)
- for (int32_t i = 0; i < 768; i++)
- dLogTab[i] = 8363.0 * 256.0 * exp2(i / 768.0);
+ for (int32_t i = 0; i < 4*12*16; i++)
+ dLogTab[i] = 8363.0 * exp2(i / 768.0) * 256.0;
}
void calcReplayerVars(int32_t audioFreq)
@@ -413,7 +414,7 @@ void calcReplayerVars(int32_t audioFreq)
return;
audio.dHz2MixDeltaMul = (double)MIXER_FRAC_SCALE / audioFreq;
- audio.quickVolRampSamples = (int32_t)round(audioFreq / 200.0);
+ audio.quickVolRampSamples = (int32_t)round(audioFreq / (double)FT2_QUICKRAMP_SAMPLES);
audio.fRampQuickVolMul = (float)(1.0 / audio.quickVolRampSamples);
for (int32_t bpm = MIN_BPM; bpm <= MAX_BPM; bpm++)
@@ -442,35 +443,23 @@ void calcReplayerVars(int32_t audioFreq)
}
}
-int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote) // for piano in Instr. Ed.
+// for piano in Instr. Ed. (values outside 0..95 can happen)
+int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote)
{
assert(note2Period != NULL);
-
if (period > note2Period[0])
- return -1; // outside lower range on piano
-
- 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;
+ return -1; // outside left piano edge
- return note;
- }
+ finetune = ((int8_t)finetune >> 3) + 16; // -128..127 -> 0..31
- /* Amiga periods require a slightly 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;
- int32_t hiPeriod = (10*12*16)+16;
+ int32_t hiPeriod = 10*12*16;
int32_t loPeriod = 0;
for (int32_t i = 0; i < 7; i++)
{
const int32_t tmpPeriod = (((loPeriod + hiPeriod) >> 1) & ~15) + finetune;
- int32_t lookUp = tmpPeriod - 8;
+ int32_t lookUp = tmpPeriod - 16;
if (lookUp < 0)
lookUp = 0;
@@ -480,11 +469,7 @@ int32_t getPianoKey(uint16_t period, int8_t finetune, int8_t relativeNote) // fo
loPeriod = (tmpPeriod - finetune) & ~15;
}
- int32_t note = loPeriod;
- note >>= 4;
- note -= relativeNote;
-
- return note;
+ return (loPeriod >> 4) - relativeNote;
}
static void startTone(uint8_t note, uint8_t efx, uint8_t efxData, channel_t *ch)
@@ -523,7 +508,7 @@ static void startTone(uint8_t note, uint8_t efx, uint8_t efxData, channel_t *ch)
ch->relativeNote = s->relativeNote;
note += ch->relativeNote;
- if (note >= 10*12)
+ if (note >= 10*12) // unsigned check (also handles note < 0)
return;
ch->oldVol = s->volume;
@@ -536,12 +521,10 @@ static void startTone(uint8_t note, uint8_t efx, uint8_t efxData, channel_t *ch)
if (note != 0)
{
- const uint16_t tmpTon = ((note-1) << 4) + (((int8_t)ch->finetune >> 3) + 16); // 0..1935
- if (tmpTon < MAX_NOTES) // tmpTon is *always* below MAX_NOTES here, so this check is not really needed
- {
- assert(note2Period != NULL);
- ch->outPeriod = ch->realPeriod = note2Period[tmpTon];
- }
+ const uint16_t noteIndex = ((note-1) << 4) + (((int8_t)ch->finetune >> 3) + 16); // 0..1920
+
+ assert(note2Period != NULL);
+ ch->outPeriod = ch->realPeriod = note2Period[noteIndex];
}
ch->status |= IS_Period + IS_Vol + IS_Pan + IS_Trigger + IS_QuickVol;
@@ -787,6 +770,8 @@ static void setEnvelopePos(channel_t *ch, uint8_t param)
instr_t *ins = ch->instrPtr;
assert(ins != NULL);
+ // (envelope precision has been updated from x.8fp to x.16fp)
+
// *** VOLUME ENVELOPE ***
if (ins->volEnvFlags & ENV_ENABLED)
{
@@ -854,7 +839,7 @@ static void setEnvelopePos(channel_t *ch, uint8_t param)
}
// *** PANNING ENVELOPE ***
- if (ins->volEnvFlags & ENV_SUSTAIN) // What..? Probably an FT2 bug!
+ if (ins->volEnvFlags & ENV_SUSTAIN) // What..? An FT2 bug!?
{
ch->panEnvTick = param-1;
@@ -1424,6 +1409,8 @@ static void updateChannel(channel_t *ch)
if (!ch->mute)
{
+ // (envelope precision has been updated from x.8fp to x.16fp)
+
// *** VOLUME ENVELOPE ***
envVal = 0;
if (ins->volEnvFlags & ENV_ENABLED)
@@ -1712,7 +1699,7 @@ static uint16_t relocateTon(uint16_t period, uint8_t arpNote, channel_t *ch)
int32_t lookUp = tmpPeriod - 8;
if (lookUp < 0)
- lookUp = 0; // safety fix (C-0 w/ finetune <= -65). This buggy read seems to return 0 in FT2 (TODO: Verify...)
+ lookUp = 0; // safety fix (C-0 w/ f.tune <= -65). This seems to result in 0 in FT2 (TODO: verify)
if (period >= note2Period[lookUp])
hiPeriod = (tmpPeriod - fineTune) & ~15;
diff --git a/src/ft2_replayer.h b/src/ft2_replayer.h
@@ -46,6 +46,7 @@ enum
#define TRACK_WIDTH (5 * MAX_CHANNELS)
#define MAX_FRQ 32000
#define C4_FREQ 8363
+#define NOTE_C4 (4*12)
#define NOTE_OFF 97
#define MAX_NOTES (10*12*16+16)
#define MAX_PATTERNS 256
@@ -57,6 +58,7 @@ enum
#define INSTR_HEADER_SIZE 263
#define INSTR_XI_HEADER_SIZE 298
#define MAX_SAMPLE_LEN 0x3FFFFFFF
+#define FT2_QUICKRAMP_SAMPLES 200
#define PROG_NAME_STR "Fasttracker II clone"
enum // sample flags
@@ -289,7 +291,7 @@ void calcReplayerVars(int32_t rate);
// used on external sample load and during sample loading in some module formats
void tuneSample(sample_t *s, const int32_t midCFreq, bool linearPeriodsFlag);
-void calcReplayerLogTab(void);
+void calcReplayerLogTab(void); // for linear period -> hz calculation
double dLinearPeriod2Hz(int32_t period);
double dAmigaPeriod2Hz(int32_t period);
@@ -352,5 +354,5 @@ extern const uint16_t *note2Period;
extern int16_t patternNumRows[MAX_PATTERNS];
extern channel_t channel[MAX_CHANNELS];
extern song_t song;
-extern instr_t *instr[132];
+extern instr_t *instr[128+4];
extern note_t *pattern[MAX_PATTERNS];
