ref: fbb9816f4f3ae74ef5db5f11127123f7efbecbd0
parent: 23923f12dcb6b945dc9f54f5556bd4d7278181cd
author: Olav Sørensen <[email protected]>
date: Sun Aug 16 12:03:42 EDT 2020
Pushed v1.28 code - Bugfix: Don't redraw the instrument switcher when pressing numpad keys while certain screens (about/config/help/nibbles) are open. - Slightly increased the volume precision in the audio channel mixer. This is most likely not measurable. - Fixed some errors in the help text - Small code cleanup
--- a/.gitignore
+++ b/.gitignore
@@ -20,3 +20,4 @@
*.opendb
*.db-shm
*.db-wal
+*.recipe
--- a/src/ft2_audio.c
+++ b/src/ft2_audio.c
@@ -27,7 +27,7 @@
static int32_t masterVol, oldAudioFreq, randSeed = INITIAL_DITHER_SEED;
static int32_t prngStateL, prngStateR;
static uint32_t tickTimeLen, tickTimeLenFrac;
-static float fAudioAmpMul;
+static double dAudioAmpMul, dPanningTab[256+1];
static voice_t voice[MAX_VOICES * 2];
static void (*sendAudSamplesFunc)(uint8_t *, uint32_t, uint8_t); // "send mixed samples" routines
@@ -123,18 +123,17 @@
ampFactor = CLAMP(ampFactor, 1, 32);
master = CLAMP(master, 0, 256);
- // voiceVolume = (vol(0..65535) * pan(0..65536)) >> 4
- const float fAudioNorm = 1.0f / (((65535UL * 65536) / 16) / MAX_VOICES);
+ const double dAudioNorm = 1.0 / (1UL << 25); // 2^25 = internal voice mixing range (per voice)
if (bitDepth32Flag)
{- // 32-bit floating point
- fAudioAmpMul = fAudioNorm * (master / 256.0f) * (ampFactor / 32.0f);
+ // 32-bit floating point mixing mode
+ dAudioAmpMul = dAudioNorm * (master / 256.0) * (ampFactor / 32.0);
}
else
{- // 16-bit integer
- masterVol = master * ampFactor * 2048;
+ // 16-bit integer mixing mode
+ masterVol = (master * ampFactor) * 512;
}
}
@@ -190,21 +189,23 @@
unlockMixerCallback();
}
+void calcPanningTable(void)
+{+ // same formula as FT2's panning table (with 0.0..1.0 range)
+ for (int32_t i = 0; i <= 256; i++)
+ dPanningTab[i] = sqrt(i * (1.0 / 256.0));
+}
+
static void voiceUpdateVolumes(int32_t i, uint8_t status)
{- int32_t volL, volR, destVolL, destVolR;
- uint32_t vol;
- voice_t *v, *f;
+ int32_t destVolL, destVolR;
- v = &voice[i];
+ voice_t *v = &voice[i];
- vol = v->SVol; // 0..4096
+ const int32_t volR = (int32_t)((v->SVol * dPanningTab[ v->SPan]) + 0.5);
+ const int32_t volL = (int32_t)((v->SVol * dPanningTab[256-v->SPan]) + 0.5);
- // 0..4096 * 0..65535 = 0..268431360 (about max safe volume range)
- volR = vol * panningTab[ v->SPan];
- volL = vol * panningTab[256-v->SPan];
-
- if (!audio.volumeRampingFlag)
+ if (!audio.volumeRampingFlag) // is volume ramping disabled?
{v->SVolIPLen = 0; // reset volume ramp length
v->SLVol2 = volL;
@@ -222,7 +223,7 @@
// setup "fade out" voice (only if current voice volume>0)
if (v->SLVol2 > 0 || v->SRVol2 > 0)
{- f = &voice[MAX_VOICES + i];
+ voice_t *f = &voice[MAX_VOICES + i];
*f = *v; // copy voice
@@ -231,7 +232,7 @@
destVolL = -f->SLVol2;
destVolR = -f->SRVol2;
- f->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32;
+ f->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32; // x = a / b
f->SRVolIP = ((int64_t)destVolR * audio.rampQuickVolMul) >> 32;
f->isFadeOutVoice = true;
@@ -262,13 +263,13 @@
if (status & IS_QuickVol)
{v->SVolIPLen = audio.quickVolSizeVal;
- v->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32;
+ v->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32; // x = a / b
v->SRVolIP = ((int64_t)destVolR * audio.rampQuickVolMul) >> 32;
}
else
{v->SVolIPLen = audio.samplesPerTick;
- v->SLVolIP = ((int64_t)destVolL * audio.rampSpeedValMul) >> 32;
+ v->SLVolIP = ((int64_t)destVolL * audio.rampSpeedValMul) >> 32; // x = a / b
v->SRVolIP = ((int64_t)destVolR * audio.rampSpeedValMul) >> 32;
}
}
@@ -424,12 +425,9 @@
static void sendSamples16BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- int16_t *streamPointer16;
int32_t out32;
- (void)numAudioChannels;
-
- streamPointer16 = (int16_t *)stream;
+ int16_t *streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel
@@ -442,16 +440,16 @@
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
}
+
+ (void)numAudioChannels;
}
static void sendSamples16BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- int16_t *streamPointer16;
int32_t out32;
- uint32_t i, j;
- streamPointer16 = (int16_t *)stream;
- for (i = 0; i < sampleBlockLength; i++)
+ int16_t *streamPointer16 = (int16_t *)stream;
+ for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel
out32 = ((int64_t)audio.mixBufferL[i] * masterVol) >> 32;
@@ -464,7 +462,7 @@
*streamPointer16++ = (int16_t)out32;
// send zeroes to the rest of the channels
- for (j = 2; j < numAudioChannels; j++)
+ for (uint32_t j = 2; j < numAudioChannels; j++)
*streamPointer16++ = 0;
}
}
@@ -471,12 +469,9 @@
static void sendSamples16BitDitherStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- int16_t *streamPointer16;
int32_t prng, out32;
- (void)numAudioChannels;
-
- streamPointer16 = (int16_t *)stream;
+ int16_t *streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel - 1-bit triangular dithering
@@ -493,14 +488,15 @@
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
}
+
+ (void)numAudioChannels;
}
static void sendSamples16BitDitherMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- int16_t *streamPointer16;
int32_t prng, out32;
- streamPointer16 = (int16_t *)stream;
+ int16_t *streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel - 1-bit triangular dithering
@@ -523,47 +519,47 @@
}
}
-static void sendSamples24BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
+static void sendSamples32BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- float fOut, *fStreamPointer24;
+ double dOut;
- (void)numAudioChannels;
-
- fStreamPointer24 = (float *)stream;
+ float *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel
- fOut = audio.mixBufferL[i] * fAudioAmpMul;
- fOut = CLAMP(fOut, -1.0f, 1.0f);
- *fStreamPointer24++ = fOut;
+ dOut = audio.mixBufferL[i] * dAudioAmpMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPointer32++ = (float)dOut;
// right channel
- fOut = audio.mixBufferR[i] * fAudioAmpMul;
- fOut = CLAMP(fOut, -1.0f, 1.0f);
- *fStreamPointer24++ = fOut;
+ dOut = audio.mixBufferR[i] * dAudioAmpMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPointer32++ = (float)dOut;
}
+
+ (void)numAudioChannels;
}
-static void sendSamples24BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
+static void sendSamples32BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{- float fOut, *fStreamPointer24;
+ double dOut;
- fStreamPointer24 = (float *)stream;
+ float *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{// left channel
- fOut = audio.mixBufferL[i] * fAudioAmpMul;
- fOut = CLAMP(fOut, -1.0f, 1.0f);
- *fStreamPointer24++ = fOut;
+ dOut = audio.mixBufferL[i] * dAudioAmpMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPointer32++ = (float)dOut;
// right channel
- fOut = audio.mixBufferR[i] * fAudioAmpMul;
- fOut = CLAMP(fOut, -1.0f, 1.0f);
- *fStreamPointer24++ = fOut;
+ dOut = audio.mixBufferR[i] * dAudioAmpMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPointer32++ = (float)dOut;
// send zeroes to the rest of the channels
for (uint32_t j = 2; j < numAudioChannels; j++)
- *fStreamPointer24++ = 0.0f;
+ *fStreamPointer32++ = 0.0f;
}
}
@@ -636,7 +632,7 @@
}
else
{- sendSamples24BitStereo(stream, samplesToMix, 2);
+ sendSamples32BitStereo(stream, samplesToMix, 2);
}
}
@@ -1061,9 +1057,9 @@
else
{if (pmpChannels > 2)
- sendAudSamplesFunc = sendSamples24BitMultiChan;
+ sendAudSamplesFunc = sendSamples32BitMultiChan;
else
- sendAudSamplesFunc = sendSamples24BitStereo;
+ sendAudSamplesFunc = sendSamples32BitStereo;
}
if (lockMixer)
--- a/src/ft2_audio.h
+++ b/src/ft2_audio.h
@@ -65,9 +65,8 @@
const int16_t *SBase16, *SRevBase16;
bool active, backwards, isFadeOutVoice;
uint8_t mixFuncOffset, SPan, SLoopType;
- uint16_t SVol;
int32_t SLVol1, SRVol1, SLVol2, SRVol2, SLVolIP, SRVolIP;
- int32_t SPos, SLen, SRepS, SRepL;
+ int32_t SVol, SPos, SLen, SRepS, SRepL;
uint32_t SVolIPLen, SFrqRev;
#if defined _WIN64 || defined __amd64__
@@ -116,6 +115,8 @@
bool chQueuePop(void);
chSyncData_t *chQueuePeek(void);
uint64_t getChQueueTimestamp(void);
+
+void calcPanningTable(void);
void setAudioAmp(int16_t ampFactor, int16_t master, bool bitDepth32Flag);
void setNewAudioFreq(uint32_t freq);
void setBackOldAudioFreq(void);
--- a/src/ft2_header.h
+++ b/src/ft2_header.h
@@ -12,7 +12,7 @@
#endif
#include "ft2_replayer.h"
-#define PROG_VER_STR "1.27"
+#define PROG_VER_STR "1.28"
// do NOT change these! It will only mess things up...
--- a/src/ft2_main.c
+++ b/src/ft2_main.c
@@ -440,38 +440,36 @@
#ifdef _WIN32
static void disableWasapi(void)
{- const char *audioDriver;
- int32_t i, numAudioDrivers;
-
// disable problematic WASAPI SDL2 audio driver on Windows (causes clicks/pops sometimes...)
- numAudioDrivers = SDL_GetNumAudioDrivers();
- for (i = 0; i < numAudioDrivers; i++)
+ const int32_t numAudioDrivers = SDL_GetNumAudioDrivers();
+ if (numAudioDrivers <= 1)
+ return;
+
+ // look for directsound and enable it if found
+ for (int32_t i = 0; i < numAudioDrivers; i++)
{- audioDriver = SDL_GetAudioDriver(i);
+ const char *audioDriver = SDL_GetAudioDriver(i);
if (audioDriver != NULL && strcmp("directsound", audioDriver) == 0) { SDL_setenv("SDL_AUDIODRIVER", "directsound", true);audio.rescanAudioDevicesSupported = false;
- break;
+ return;
}
}
- if (i == numAudioDrivers)
+ // directsound is not available, try winmm
+ for (int32_t i = 0; i < numAudioDrivers; i++)
{- // directsound is not available, try winmm
- for (i = 0; i < numAudioDrivers; i++)
+ const char *audioDriver = SDL_GetAudioDriver(i);
+ if (audioDriver != NULL && strcmp("winmm", audioDriver) == 0) {- audioDriver = SDL_GetAudioDriver(i);
- if (audioDriver != NULL && strcmp("winmm", audioDriver) == 0)- {- SDL_setenv("SDL_AUDIODRIVER", "winmm", true);- audio.rescanAudioDevicesSupported = false;
- break;
- }
+ SDL_setenv("SDL_AUDIODRIVER", "winmm", true);+ audio.rescanAudioDevicesSupported = false;
+ return;
}
}
- // maybe we didn't find directsound or winmm, let's use wasapi after all then...
+ // we didn't find directsound or winmm, let's use wasapi after all...
}
#endif
--- a/src/ft2_pattern_ed.c
+++ b/src/ft2_pattern_ed.c
@@ -2305,6 +2305,9 @@
int8_t i;
int16_t y;
+ if (ui.aboutScreenShown || ui.configScreenShown || ui.helpScreenShown || ui.nibblesShown)
+ return; // don't redraw instrument switcher when it's not shown!
+
if (ui.extended) // extended pattern editor
{//INSTRUMENTS
--- a/src/ft2_replayer.c
+++ b/src/ft2_replayer.c
@@ -21,13 +21,13 @@
#include "ft2_tables.h"
#include "ft2_structs.h"
-/* This is a *huge* mess, directly ported from the original FT2 code (and modified).
+/* 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!
*/
// non-FT2 precalced stuff
-static double dPeriod2HzTab[65536], dLogTab[768], dAudioRateFactor;
+static double dPeriod2HzTab[65536], dLogTab[768], d2nRevTab[32], dHz2MixDeltaMul;
static bool bxxOverflow;
static tonTyp nilPatternLine;
@@ -209,21 +209,28 @@
return i+1;
}
-static void calcPeriod2HzTab(void) // called every time you change linear/amiga period mode
+static void calc2nRevTable(void) // for calcPeriod2HzTable()
{- dPeriod2HzTab[0] = 0.0; // in FT2, a period of 0 yields 0Hz
+ d2nRevTab[0] = 1.0;
+ for (int32_t i = 0; i < 32; i++)
+ d2nRevTab[i] = 1.0 / (1UL << i);
+}
+static void calcPeriod2HzTable(void) // called every time "linear/amiga frequency" mode is changed
+{+ dPeriod2HzTab[0] = 0.0; // in FT2, a period of 0 converts to 0Hz
+
if (audio.linearFreqTable)
{// linear periods
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 uint16_t invPeriod = (12 * 192 * 4) - (uint16_t)i; // this intentionally 16-bit-underflows to be accurate to FT2
const int32_t octave = invPeriod / 768;
const int32_t period = invPeriod % 768;
- const int32_t bitshift = (14 - octave) & 0x1F; // 100% accurate to FT2
+ const int32_t invOct = (14 - octave) & 0x1F; // accurate to FT2
- dPeriod2HzTab[i] = dLogTab[period] / (1UL << bitshift);
+ dPeriod2HzTab[i] = dLogTab[period] * d2nRevTab[invOct]; // x = y / 2^invOct
}
}
else
@@ -245,7 +252,7 @@
else
note2Period = amigaPeriods;
- calcPeriod2HzTab();
+ calcPeriod2HzTable();
resumeAudio();
@@ -348,9 +355,9 @@
if (audioFreq == 0)
return;
- dAudioRateFactor = (double)MIXER_FRAC_SCALE / audioFreq;
+ dHz2MixDeltaMul = (double)MIXER_FRAC_SCALE / audioFreq;
- audio.quickVolSizeVal = audioFreq / 200; // FT2 truncates here
+ audio.quickVolSizeVal = (int32_t)((audioFreq / 200.0) + 0.5);
audio.rampQuickVolMul = (int32_t)(((UINT32_MAX + 1.0) / audio.quickVolSizeVal) + 0.5);
/* Calculate tables to prevent floating point operations on systems that
@@ -384,7 +391,7 @@
}
}
-double period2Hz(uint16_t period)
+double dPeriod2Hz(uint16_t period)
{return dPeriod2HzTab[period];
}
@@ -392,12 +399,12 @@
#if defined _WIN64 || defined __amd64__
int64_t getMixerDelta(uint16_t period)
{- return (int64_t)((dPeriod2HzTab[period] * dAudioRateFactor) + 0.5); // Hz -> rounded fixed-point mixer delta
+ return (int64_t)((dPeriod2Hz(period) * dHz2MixDeltaMul) + 0.5); // Hz -> rounded fixed-point mixer delta
}
#else
int32_t getMixerDelta(uint16_t period)
{- return (int32_t)((dPeriod2HzTab[period] * dAudioRateFactor) + 0.5); // Hz -> rounded fixed-point mixer delta
+ return (int32_t)((dPeriod2Hz(period) * dHz2MixDeltaMul) + 0.5); // Hz -> rounded fixed-point mixer delta
}
#endif
@@ -405,7 +412,7 @@
{finetune >>= 3; // FT2 does this in the replayer internally, so the actual range is -16..15
- const double dRelativeHz = dPeriod2HzTab[period] * (1.0 / (8363.0 / 16.0));
+ const double dRelativeHz = dPeriod2Hz(period) * (1.0 / (8363.0 / 16.0));
const double dNote = (log2(dRelativeHz) * 12.0) - (finetune * (1.0 / 16.0));
const int32_t note = (int32_t)(dNote + 0.5) - relativeNote; // rounded
@@ -1317,27 +1324,28 @@
}
}
- /* calculate with 16 times more precision than FT2 (vol = 0..4096)
- ** Also, vol envelope range is now 0..16384 instead of being shifted to 0..64
- */
+ // original FT2 shifts the vol env. range to 0..64, but we keep its full resolution (0..16384)
- int32_t vol1 = song.globVol * ch->outVol * ch->fadeOutAmp; // 0..64 * 0..64 * 0..32768 = 0..134217728
- int32_t vol2 = envVal << 3; // 0..16384 * 2^3 = 0..131072
+ // 0..64 * 0..64 * 0..32768 = 0..134217728
+ vol = song.globVol * ch->outVol * ch->fadeOutAmp;
- vol = ((int64_t)vol1 * vol2) >> 32; // 0..4096
+ // ((0..134217728 * 0..16384) + 2^10) / 2^11 = 0..1073741824 (rounded)
+ vol = (uint32_t)((((int64_t)vol * envVal) + (1UL << 10)) >> 11);
ch->status |= IS_Vol;
}
else
{- // calculate with 16 times more precision than FT2 (vol = 0..4096)
- vol = ((song.globVol * ch->outVol * ch->fadeOutAmp) + (1 << 14)) >> 15; // 0..64 * 0..64 * 0..32768 -> 0..4096 (rounded)
+ // (0..64 * 0..64 * 0..32768) * 2^3 = 0..1073741824
+ vol = (uint32_t)(song.globVol * ch->outVol * ch->fadeOutAmp) << 3;
}
- if (vol > 4096)
- vol = 4096;
+ // original FT2 calculates final volume with a range of 0..256. We do it with a range of 0..1073741824 (why not)
- ch->finalVol = (uint16_t)vol;
+ if (vol > (1UL<<30)) // shouldn't happen, but just in case...
+ vol = (1UL<<30);
+
+ ch->finalVol = vol;
}
else
{@@ -2727,7 +2735,9 @@
audio.linearFreqTable = true;
note2Period = linearPeriods;
- calcPeriod2HzTab();
+ calc2nRevTable();
+ calcPeriod2HzTable();
+ calcPanningTable();
setPos(0, 0, true);
--- a/src/ft2_replayer.h
+++ b/src/ft2_replayer.h
@@ -201,9 +201,10 @@
uint8_t volKolVol, tonNr, envPPos, eVibPos, envVPos, realVol, oldVol, outVol;
uint8_t oldPan, outPan, finalPan;
int16_t midiPitch, envVIPValue, envPIPValue;
- uint16_t finalVol, outPeriod, realPeriod, finalPeriod, tonTyp, wantPeriod, portaSpeed;
+ uint16_t outPeriod, realPeriod, finalPeriod, tonTyp, wantPeriod, portaSpeed;
uint16_t envVCnt, envVAmp, envPCnt, envPAmp, eVibAmp, eVibSweep;
uint16_t fadeOutAmp, fadeOutSpeed, midiVibDepth;
+ uint32_t finalVol;
int32_t smpStartPos;
sampleTyp *smpPtr;
instrTyp *instrSeg;
@@ -231,8 +232,9 @@
bool envSustainActive;
int8_t fineTune, relTonNr;
uint8_t status, sampleNr, instrNr;
- uint16_t finalPeriod, finalVol;
+ uint16_t finalPeriod;
int32_t smpStartPos;
+ uint32_t finalVol;
} syncedChannel_t;
void fixSongName(void); // removes spaces from right side of song name
@@ -241,7 +243,7 @@
void tuneSample(sampleTyp *s, int32_t midCFreq);
void calcReplayerLogTab(void);
-double period2Hz(uint16_t period);
+double dPeriod2Hz(uint16_t period);
#if defined _WIN64 || defined __amd64__
int64_t getMixerDelta(uint16_t period);
--- a/src/ft2_scopes.c
+++ b/src/ft2_scopes.c
@@ -58,7 +58,7 @@
int32_t getSamplePosition(uint8_t ch)
{- volatile bool active, sample16Bit;
+ volatile bool active, sampleIs16Bit;
volatile int32_t pos, len;
volatile scope_t *sc;
@@ -71,7 +71,7 @@
active = sc->active;
pos = sc->SPos;
len = sc->SLen;
- sample16Bit = sc->sample16Bit;
+ sampleIs16Bit = sc->sampleIs16Bit;
if (!active || len == 0)
return -1;
@@ -78,7 +78,7 @@
if (pos >= 0 && pos < len)
{- if (sample16Bit)
+ if (sampleIs16Bit)
pos <<= 1;
return pos;
@@ -357,7 +357,7 @@
else
tempState.sampleData8 = (const int8_t *)s->pek;
- tempState.sample16Bit = sampleIs16Bit;
+ tempState.sampleIs16Bit = sampleIs16Bit;
tempState.loopType = loopType;
tempState.backwards = false;
@@ -499,8 +499,8 @@
clearRect(scopeXOffs, scopeYOffs, scopeDrawLen, SCOPE_HEIGHT);
// draw scope
- bool linedScopes = !!(config.specialFlags & LINED_SCOPES);
- scopeDrawRoutineTable[(linedScopes * 6) + (s.sample16Bit * 3) + s.loopType](&s, scopeXOffs, scopeLineY, scopeDrawLen);
+ bool linedScopesFlag = !!(config.specialFlags & LINED_SCOPES);
+ scopeDrawRoutineTable[(linedScopesFlag * 6) + (s.sampleIs16Bit * 3) + s.loopType](&s, scopeXOffs, scopeLineY, scopeDrawLen);
}
else
{@@ -550,7 +550,7 @@
status = scopeUpdateStatus[i];
if (status & IS_Vol)
- sc->SVol = (int8_t)(((ch->finalVol * SCOPE_HEIGHT) + (1 << 11)) >> 12); // rounded
+ sc->SVol = (int8_t)((((ch->finalVol >> 6) * SCOPE_HEIGHT)) >> 24); // rounded
if (status & IS_Period)
{@@ -560,7 +560,7 @@
if (period != oldPeriod)
{oldPeriod = period;
- const double dHz = period2Hz(period);
+ const double dHz = dPeriod2Hz(period);
const double dScopeRateFactor = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;
oldSFrq = (int64_t)((dHz * dScopeRateFactor) + 0.5); // Hz -> rounded fixed-point delta
--- a/src/ft2_scopes.h
+++ b/src/ft2_scopes.h
@@ -30,7 +30,7 @@
const int8_t *sampleData8;
const int16_t *sampleData16;
int8_t SVol;
- bool wasCleared, sample16Bit, backwards;
+ bool wasCleared, sampleIs16Bit, backwards;
uint8_t loopType;
int32_t SRepS, SRepL, SLen, SPos;
uint32_t DFrq;
--- a/src/ft2_tables.c
+++ b/src/ft2_tables.c
@@ -317,31 +317,6 @@
};
/* ----------------------------------------------------------------------- */
-/* AUDIO MIXER TABLES */
-/* ----------------------------------------------------------------------- */
-
-const uint16_t panningTab[257] = // bit-exact to FT2 table (except last value 65536 -> 65535)
-{- 0, 4096, 5793, 7094, 8192, 9159,10033,10837,11585,12288,12953,13585,14189,14768,15326,15864,
- 16384,16888,17378,17854,18318,18770,19212,19644,20066,20480,20886,21283,21674,22058,22435,22806,
- 23170,23530,23884,24232,24576,24915,25249,25580,25905,26227,26545,26859,27170,27477,27780,28081,
- 28378,28672,28963,29251,29537,29819,30099,30377,30652,30924,31194,31462,31727,31991,32252,32511,
- 32768,33023,33276,33527,33776,34024,34270,34514,34756,34996,35235,35472,35708,35942,36175,36406,
- 36636,36864,37091,37316,37540,37763,37985,38205,38424,38642,38858,39073,39287,39500,39712,39923,
- 40132,40341,40548,40755,40960,41164,41368,41570,41771,41972,42171,42369,42567,42763,42959,43154,
- 43348,43541,43733,43925,44115,44305,44494,44682,44869,45056,45242,45427,45611,45795,45977,46160,
- 46341,46522,46702,46881,47059,47237,47415,47591,47767,47942,48117,48291,48465,48637,48809,48981,
- 49152,49322,49492,49661,49830,49998,50166,50332,50499,50665,50830,50995,51159,51323,51486,51649,
- 51811,51972,52134,52294,52454,52614,52773,52932,53090,53248,53405,53562,53719,53874,54030,54185,
- 54340,54494,54647,54801,54954,55106,55258,55410,55561,55712,55862,56012,56162,56311,56459,56608,
- 56756,56903,57051,57198,57344,57490,57636,57781,57926,58071,58215,58359,58503,58646,58789,58931,
- 59073,59215,59357,59498,59639,59779,59919,60059,60199,60338,60477,60615,60753,60891,61029,61166,
- 61303,61440,61576,61712,61848,61984,62119,62254,62388,62523,62657,62790,62924,63057,63190,63323,
- 63455,63587,63719,63850,63982,64113,64243,64374,64504,64634,64763,64893,65022,65151,65279,65408,
- 65535
-};
-
-/* ----------------------------------------------------------------------- */
/* GUI TABLES */
/* ----------------------------------------------------------------------- */
--- a/src/ft2_tables.h
+++ b/src/ft2_tables.h
@@ -17,8 +17,6 @@
extern const uint16_t linearPeriods[1936];
extern const uint16_t amigaPeriods[1936];
-extern const uint16_t panningTab[257];
-
extern const char *dec2StrTab[100];
extern const char *dec3StrTab[256];
--- a/src/helpdata/FT2.HLP
+++ b/src/helpdata/FT2.HLP
@@ -898,7 +898,7 @@
>@C001Sample editor:
>@C002
>@X010- When a looped sample is zoomed out in the sample editor, you could see
->@X021unwanted sample data at the loop-end point. This is because of a kludge
+>@X021unexpected sample data at the loop-end point. This is because of a kludge
for the resampling interpolation to work faster in the audio mixer, and the
original FT2 has the same problem. I have made it so that if you zoom in to
see the individual sample points, it will look like normal.
@@ -907,12 +907,9 @@
>
>@C002- Linux: The mouse cursor graphics can be glitchy at times...
>@X010
->@C002- The "clear sample" shortcut (shift + num-pad Del/',') only works if
->@X021num lock is off. There's no way I can fix this...
->@X010
>@C001Video:
>@C002
->@X010- Fullscreen mode can be unbearably slow on a Raspberry Pi (even an RPi 4)
+>@X010- Fullscreen mode can be unbearably slow on a Raspberry Pi (even on RPi 4)
>
>@X010- The scopes can mildly flicker depending on the waveform and pitch.
>@X021This is because their frequency is not clocked to exactly the same rate
--- 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 27295
+#define HELP_DATA_LEN 27158
-const uint8_t helpData[27295] =
+const uint8_t helpData[27158] =
{0x4C,0x3B,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,
0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,0x2A,
@@ -2187,101 +2187,90 @@
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};
#endif
--- a/src/mixer/ft2_mix_macros.h
+++ b/src/mixer/ft2_mix_macros.h
@@ -128,13 +128,13 @@
#define RENDER_8BIT_SMP \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = *smpPtr << 20; \
+ sample = *smpPtr << (12+8); \
*audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
#define RENDER_8BIT_SMP_MONO \
assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = *smpPtr << 20; \
+ sample = *smpPtr << (12+8); \
sample = ((int64_t)sample * CDA_LVol) >> 32; \
*audioMixL++ += sample; \
*audioMixR++ += sample; \