ref: f0cc39bf0859cec692fbad1431b2c176fb14b4fb
parent: 75f25156e885c667bb15b862c9474c3d60f5e5c2
author: Olav Sørensen <olav.sorensen@live.no>
date: Sun Sep 7 12:00:36 EDT 2025
Pushed v1.98 code - Re-added audio dithering for 16-bit audio output - Tracker scopes changed yet again, reverted to original integer-based scaling
--- a/src/ft2_audio.c
+++ b/src/ft2_audio.c
@@ -23,10 +23,13 @@
#pragma warning(disable: 4996)
#endif
+#define INITIAL_DITHER_SEED 0x12345000
+
static int32_t smpShiftValue;
-static uint32_t oldAudioFreq, tickTimeLenInt;
+static uint32_t oldAudioFreq, tickTimeLenInt, randSeed = INITIAL_DITHER_SEED;
static uint64_t tickTimeLenFrac;
-static float fAudioNormalizeMul, fSqrtPanningTable[256+1];
+static float fSqrtPanningTable[256+1];
+static double dAudioNormalizeMul, dPrngStateL, dPrngStateR;
static voice_t voice[MAX_CHANNELS * 2];
// globalized
@@ -104,7 +107,7 @@
if (!bitDepth32Flag)
dAmp *= 32768.0;
- fAudioNormalizeMul = (float)dAmp;
+ dAudioNormalizeMul = dAmp;
}
void decreaseMasterVol(void)
@@ -368,7 +371,7 @@
if (status & IS_Vol)
{v->fVolume = ch->fFinalVol; // 0.0f .. 1.0f
- v->scopeVolume = (uint8_t)(v->fVolume * 255.0f);
+ v->scopeVolume = (uint8_t)((ch->fFinalVol * (SCOPE_HEIGHT*4.0f)) + 0.5f);
}
if (status & IS_Pan)
@@ -400,20 +403,47 @@
}
}
+void resetAudioDither(void)
+{+ randSeed = INITIAL_DITHER_SEED;
+ dPrngStateL = dPrngStateR = 0.0;
+}
+
+static inline int32_t random32(void)
+{+ // LCG 32-bit random
+ randSeed *= 134775813;
+ randSeed++;
+
+ return (int32_t)randSeed;
+}
+
static void sendSamples16BitStereo(void *stream, uint32_t sampleBlockLength)
{+ int32_t out32;
+ double dOut, dPrng;
+
int16_t *streamPtr16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{- int32_t L = (int32_t)(audio.fMixBufferL[i] * fAudioNormalizeMul);
- int32_t R = (int32_t)(audio.fMixBufferR[i] * fAudioNormalizeMul);
+ // left channel - 1-bit triangular dithering
+ dPrng = random32() * (1.0 / (UINT32_MAX+1.0)); // -0.5 .. 0.5
+ dOut = (double)audio.fMixBufferL[i] * dAudioNormalizeMul;
+ dOut = (dOut + dPrng) - dPrngStateL;
+ dPrngStateL = dPrng;
+ out32 = (int32_t)dOut;
+ CLAMP16(out32);
+ *streamPtr16++ = (int16_t)out32;
- CLAMP16(L);
- CLAMP16(R);
+ // right channel - 1-bit triangular dithering
+ dPrng = random32() * (1.0 / (UINT32_MAX+1.0)); // -0.5 .. 0.5
+ dOut = (double)audio.fMixBufferR[i] * dAudioNormalizeMul;
+ dOut = (dOut + dPrng) - dPrngStateR;
+ dPrngStateR = dPrng;
+ out32 = (int32_t)dOut;
+ CLAMP16(out32);
+ *streamPtr16++ = (int16_t)out32;
- *streamPtr16++ = (int16_t)L;
- *streamPtr16++ = (int16_t)R;
-
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = audio.fMixBufferR[i] = 0.0f;
}
@@ -421,14 +451,20 @@
static void sendSamples32BitFloatStereo(void *stream, uint32_t sampleBlockLength)
{+ double dOut;
+
float *fStreamPtr32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{- const float fL = audio.fMixBufferL[i] * fAudioNormalizeMul;
- const float fR = audio.fMixBufferR[i] * fAudioNormalizeMul;
+ // left channel
+ dOut = (double)audio.fMixBufferL[i] * dAudioNormalizeMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPtr32++ = (float)dOut;
- *fStreamPtr32++ = CLAMP(fL, -1.0f, 1.0f);
- *fStreamPtr32++ = CLAMP(fR, -1.0f, 1.0f);
+ // right channel
+ dOut = (double)audio.fMixBufferR[i] * dAudioNormalizeMul;
+ dOut = CLAMP(dOut, -1.0, 1.0);
+ *fStreamPtr32++ = (float)dOut;
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = audio.fMixBufferR[i] = 0.0f;
@@ -794,12 +830,17 @@
static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)
{if (editor.wavIsRendering)
+ {+ memset(stream, 0, len);
return;
+ }
len >>= smpShiftValue; // bytes -> samples
if (len <= 0)
return;
+ audio.callbackOngoing = true;
+
int32_t bufferPosition = 0;
uint32_t samplesLeft = len;
@@ -845,6 +886,8 @@
else
sendSamples32BitFloatStereo(stream, len);
+ audio.callbackOngoing = false;
+
(void)userdata;
}
@@ -1064,4 +1107,6 @@
}
freeAudioBuffers();
+
+ audio.callbackOngoing = false;
}
--- a/src/ft2_audio.h
+++ b/src/ft2_audio.h
@@ -35,7 +35,7 @@
{char *currInputDevice, *currOutputDevice, *lastWorkingAudioDeviceName;
char *inputDeviceNames[MAX_AUDIO_DEVICES], *outputDeviceNames[MAX_AUDIO_DEVICES];
- volatile bool locked, resetSyncTickTimeFlag, volumeRampingFlag;
+ volatile bool locked, resetSyncTickTimeFlag, volumeRampingFlag, callbackOngoing;
bool linearPeriodsFlag, rescanAudioDevicesSupported, sincInterpolation;
volatile uint8_t interpolationType;
int32_t inputDeviceNum, outputDeviceNum, lastWorkingAudioFreq, lastWorkingAudioBits;
@@ -140,6 +140,7 @@
void pauseAudio(void);
void resumeAudio(void);
bool setNewAudioSettings(void);
+void resetAudioDither(void);
void lockAudio(void);
void unlockAudio(void);
void lockMixerCallback(void);
--- a/src/ft2_replayer.c
+++ b/src/ft2_replayer.c
@@ -3145,6 +3145,7 @@
editor.curPlaySmp = 255;
stopAllScopes();
+ resetAudioDither();
// wait for scope thread to finish, making sure pointers aren't illegal
while (editor.scopeThreadBusy);
--- a/src/ft2_wav_renderer.c
+++ b/src/ft2_wav_renderer.c
@@ -201,7 +201,10 @@
if (wavRenderBuffer == NULL)
return false;
+ // wait for main audio callback to catch WAV render flag
editor.wavIsRendering = true;
+ while (audio.callbackOngoing)
+ SDL_Delay(5);
setPos(songPos, 0, true);
playMode = PLAYMODE_SONG;
--- a/src/scopes/ft2_scope_macros.h
+++ b/src/scopes/ft2_scope_macros.h
@@ -25,13 +25,11 @@
#define LINED_SCOPE_INIT \
SCOPE_INIT \
- float fSample; \
int32_t smpY1, smpY2; \
width--;
#define LINED_SCOPE_INIT_BIDI \
SCOPE_INIT_BIDI \
- float fSample; \
int32_t smpY1, smpY2; \
width--;
@@ -41,41 +39,41 @@
#define NEAREST_NEIGHGBOR8 \
{ \- fSample = s8[0]; \
+ sample = s8[0] << 8; \
} \
#define LINEAR_INTERPOLATION8(frac) \
{ \- const float f = (frac) * (1.0f / SCOPE_FRAC_SCALE); \
- fSample = s8[0] + ((s8[1] - s8[0]) * f); \
+ const int32_t f = (frac) >> (SCOPE_FRAC_BITS-15); \
+ sample = (s8[0] << 8) + ((((s8[1] - s8[0]) << 8) * f) >> 15); \
} \
#define NEAREST_NEIGHGBOR16 \
{ \- fSample = s16[0]; \
+ sample = s16[0]; \
} \
#define LINEAR_INTERPOLATION16(frac) \
{ \- const float f = (frac) * (1.0f / SCOPE_FRAC_SCALE); \
- fSample = s16[0] + ((s16[1] - s16[0]) * f); \
+ const int32_t f = (frac) >> (SCOPE_FRAC_BITS-15); \
+ sample = s16[0] + (((s16[1] - s16[0]) * f) >> 15); \
} \
#define CUBIC_SMP8(frac) \
- const float *t = fScopeIntrpLUT + (((frac) >> (SCOPE_FRAC_BITS-SCOPE_INTRP_PHASES_BITS)) << SCOPE_INTRP_WIDTH_BITS); \
+ const int16_t *t = scopeIntrpLUT + (((frac) >> (SCOPE_FRAC_BITS-SCOPE_INTRP_PHASES_BITS)) << SCOPE_INTRP_WIDTH_BITS); \
\
- fSample = (s8[-1] * t[0]) + \
+ sample = ((s8[-1] * t[0]) + \
( s8[0] * t[1]) + \
( s8[1] * t[2]) + \
- ( s8[2] * t[3]);
+ ( s8[2] * t[3])) >> (SCOPE_INTRP_SCALE_BITS-8);
#define CUBIC_SMP16(frac) \
- const float *t = fScopeIntrpLUT + (((frac) >> (SCOPE_FRAC_BITS-SCOPE_INTRP_PHASES_BITS)) << SCOPE_INTRP_WIDTH_BITS); \
+ const int16_t *t = scopeIntrpLUT + (((frac) >> (SCOPE_FRAC_BITS-SCOPE_INTRP_PHASES_BITS)) << SCOPE_INTRP_WIDTH_BITS); \
\
- fSample = (s16[-1] * t[0]) + \
+ sample = ((s16[-1] * t[0]) + \
( s16[0] * t[1]) + \
( s16[1] * t[2]) + \
- ( s16[2] * t[3]);
+ ( s16[2] * t[3])) >> SCOPE_INTRP_SCALE_BITS;
#define CUBIC_INTERPOLATION8(frac) \
{ \@@ -111,7 +109,7 @@
LINEAR_INTERPOLATION8(frac) \
else \
CUBIC_INTERPOLATION8(frac) \
- sample = (int32_t)((fSample * s->fVolume8) - 0.5f);
+ sample = (sample * s->volume) >> (16+2);
#define INTERPOLATE_SMP16(pos, frac) \
const int16_t *s16 = s->base16 + pos; \
@@ -121,7 +119,7 @@
LINEAR_INTERPOLATION16(frac) \
else \
CUBIC_INTERPOLATION16(frac) \
- sample = (int32_t)((fSample * s->fVolume16) - 0.5f);
+ sample = (sample * s->volume) >> (16+2);
#define INTERPOLATE_SMP8_LOOP(pos, frac) \
const int8_t *s8 = s->base8 + pos; \
@@ -131,7 +129,7 @@
LINEAR_INTERPOLATION8(frac) \
else \
CUBIC_INTERPOLATION8_LOOP(pos, frac) \
- sample = (int32_t)((fSample * s->fVolume8) - 0.5f);
+ sample = (sample * s->volume) >> (16+2);
#define INTERPOLATE_SMP16_LOOP(pos, frac) \
const int16_t *s16 = s->base16 + pos; \
@@ -141,17 +139,17 @@
LINEAR_INTERPOLATION16(frac) \
else \
CUBIC_INTERPOLATION16_LOOP(pos, frac) \
- sample = (int32_t)((fSample * s->fVolume16) - 0.5f);
+ sample = (sample * s->volume) >> (16+2);
#define SCOPE_GET_SMP8 \
if (s->active) \
- sample = (int32_t)((s->base8[position] * s->fVolume8) - 0.5f); \
+ sample = (s->base8[position] * s->volume) >> (8+2); \
else \
sample = 0;
#define SCOPE_GET_SMP16 \
if (s->active) \
- sample = (int32_t)((s->base16[position] * s->fVolume16) - 0.5f); \
+ sample = (s->base16[position] * s->volume) >> (16+2); \
else \
sample = 0;
@@ -159,7 +157,7 @@
if (s->active) \
{ \GET_BIDI_POSITION \
- sample = (int32_t)((s->base8[actualPos] * s->fVolume8) - 0.5f); \
+ sample = (s->base8[actualPos] * s->volume) >> (8+2); \
} \
else \
{ \@@ -170,7 +168,7 @@
if (s->active) \
{ \GET_BIDI_POSITION \
- sample = (int32_t)((s->base16[actualPos] * s->fVolume16) - 0.5f); \
+ sample = (s->base16[actualPos] * s->volume) >> (16+2); \
} \
else \
{ \--- a/src/scopes/ft2_scopedraw.c
+++ b/src/scopes/ft2_scopedraw.c
@@ -2,6 +2,7 @@
#include <stdint.h>
#include <stdbool.h>
#include <stdlib.h>
+#include <math.h>
#include "../ft2_config.h"
#include "../ft2_video.h"
#include "../ft2_palette.h"
@@ -9,14 +10,14 @@
#include "ft2_scopedraw.h"
#include "ft2_scope_macros.h"
-static float *fScopeIntrpLUT;
+static int16_t *scopeIntrpLUT;
static void scopeLine(int32_t x1, int32_t y1, int32_t y2, const uint32_t color);
bool calcScopeIntrpLUT(void)
{- fScopeIntrpLUT = (float *)malloc(SCOPE_INTRP_WIDTH * SCOPE_INTRP_PHASES * sizeof (float));
- if (fScopeIntrpLUT == NULL)
+ scopeIntrpLUT = (int16_t *)malloc(SCOPE_INTRP_WIDTH * SCOPE_INTRP_PHASES * sizeof (int16_t));
+ if (scopeIntrpLUT == NULL)
return false;
/* Several tests have been done to figure out what interpolation method is most suitable
@@ -26,7 +27,8 @@
*/
// 4-point cubic B-spline (no overshoot)
- float *fPtr = fScopeIntrpLUT;
+
+ int16_t *ptr16 = scopeIntrpLUT;
for (int32_t i = 0; i < SCOPE_INTRP_PHASES; i++)
{const double x1 = i * (1.0 / SCOPE_INTRP_PHASES);
@@ -38,10 +40,11 @@
const double t3 = (x1 * (1.0/2.0)) + (x2 * (1.0/2.0)) + (x3 * -(1.0/2.0)) + (1.0/6.0);
const double t4 = x3 * (1.0/6.0);
- *fPtr++ = (float)t1; // tap #1 at sample offset -1
- *fPtr++ = (float)t2; // tap #2 at sample offset 0 (center)
- *fPtr++ = (float)t3; // tap #3 at sample offset 1
- *fPtr++ = (float)t4; // tap #4 at sample offset 2
+ // truncate, do not round!
+ *ptr16++ = (int16_t)(t1 * SCOPE_INTRP_SCALE); // tap #1 at sample offset -1
+ *ptr16++ = (int16_t)(t2 * SCOPE_INTRP_SCALE); // tap #2 at sample offset 0 (center)
+ *ptr16++ = (int16_t)(t3 * SCOPE_INTRP_SCALE); // tap #3 at sample offset 1
+ *ptr16++ = (int16_t)(t4 * SCOPE_INTRP_SCALE); // tap #4 at sample offset 2
}
return true;
@@ -49,10 +52,10 @@
void freeScopeIntrpLUT(void)
{- if (fScopeIntrpLUT != NULL)
+ if (scopeIntrpLUT != NULL)
{- free(fScopeIntrpLUT);
- fScopeIntrpLUT = NULL;
+ free(scopeIntrpLUT);
+ scopeIntrpLUT = NULL;
}
}
--- a/src/scopes/ft2_scopes.c
+++ b/src/scopes/ft2_scopes.c
@@ -428,7 +428,7 @@
}
volatile scope_t s = scope[i]; // cache scope to lower thread race condition issues
- if (s.active && s.fVolume16 > 0.0f && !audio.locked)
+ if (s.active && s.volume > 0 && !audio.locked)
{// scope is active
scope[i].wasCleared = false;
@@ -486,10 +486,7 @@
const uint8_t status = scopeUpdateStatus[i];
if (status & IS_Vol)
- {- sc->fVolume8 = ch->scopeVolume * (((SCOPE_HEIGHT/2.0f) / 255.0f) / 128.0f);
- sc->fVolume16 = ch->scopeVolume * (((SCOPE_HEIGHT/2.0f) / 255.0f) / 32768.0f);
- }
+ sc->volume = ch->scopeVolume;
if (status & IS_Period)
{--- a/src/scopes/ft2_scopes.h
+++ b/src/scopes/ft2_scopes.h
@@ -19,6 +19,8 @@
#define SCOPE_INTRP_WIDTH 4
#define SCOPE_INTRP_WIDTH_BITS 2 /* log2(SCOPE_INTRP_WIDTH) */
+#define SCOPE_INTRP_SCALE 32768
+#define SCOPE_INTRP_SCALE_BITS 15 /* log2(SCOPE_INTRP_SCALE) */
#define SCOPE_INTRP_PHASES 512 /* enough for the scopes */
#define SCOPE_INTRP_PHASES_BITS 9 /* log2(SCOPE_INTRP_PHASES) */
@@ -38,9 +40,9 @@
const int16_t *base16;
bool wasCleared, sample16Bit, samplingBackwards, hasLooped;
uint8_t loopType;
+ int16_t volume;
int32_t loopStart, loopLength, loopEnd, sampleEnd, position;
uint64_t delta, drawDelta, positionFrac;
- float fVolume8, fVolume16;
// if (loopEnabled && hasLooped && samplingPos <= loopStart+MAX_LEFT_TAPS) readFixedTapsFromThisPointer();
const int8_t *leftEdgeTaps8;
--
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