ref: 63402619f6de96adeb1a45dff7a558d745448a1c
dir: /src/pt2_audio.c/
/* The "LED" filter and BLEP routines were coded by aciddose.
** Low-pass filter is based on https://bel.fi/alankila/modguide/interpolate.txt */
// for finding memory leaks in debug mode with Visual Studio
#if defined _DEBUG && defined _MSC_VER
#include <crtdbg.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <stdbool.h>
#include <SDL2/SDL.h>
#ifdef _WIN32
#include <io.h>
#else
#include <unistd.h>
#endif
#include <math.h> // sqrt(),tan(),M_PI,round(),roundf()
#include <fcntl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <limits.h>
#include "pt2_audio.h"
#include "pt2_header.h"
#include "pt2_helpers.h"
#include "pt2_blep.h"
#include "pt2_config.h"
#include "pt2_tables.h"
#include "pt2_palette.h"
#include "pt2_textout.h"
#include "pt2_visuals.h"
#include "pt2_scopes.h"
#define INITIAL_DITHER_SEED 0x12345000
#define DENORMAL_OFFSET 1e-10
typedef struct ledFilter_t
{
double dLed[4];
} ledFilter_t;
typedef struct ledFilterCoeff_t
{
double dLed, dLedFb;
} ledFilterCoeff_t;
typedef struct voice_t
{
volatile bool active;
const int8_t *data, *newData;
int32_t length, newLength, pos;
double dVolume, dDelta, dPhase, dLastDelta, dLastPhase, dPanL, dPanR;
} paulaVoice_t;
static volatile int8_t filterFlags;
static volatile bool audioLocked;
static int8_t defStereoSep;
static bool amigaPanFlag, wavRenderingDone;
static uint16_t ch1Pan, ch2Pan, ch3Pan, ch4Pan, oldPeriod;
static int32_t sampleCounter, maxSamplesToMix, randSeed = INITIAL_DITHER_SEED;
static uint32_t oldScopeDelta;
static double *dMixBufferL, *dMixBufferR, oldVoiceDelta;
static blep_t blep[AMIGA_VOICES], blepVol[AMIGA_VOICES];
static lossyIntegrator_t filterLo, filterHi;
static ledFilterCoeff_t filterLEDC;
static ledFilter_t filterLED;
static paulaVoice_t paula[AMIGA_VOICES];
static SDL_AudioDeviceID dev;
// globalized
bool forceMixerOff = false;
int32_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)
{
uint32_t ciaVal;
double dFreqMul;
if (bpm == 0)
return 0;
ciaVal = (uint32_t)(1773447 / bpm); // yes, PT truncates here
dFreqMul = ciaVal * (1.0 / CIA_PAL_CLK);
return (uint16_t)((audioFreq * dFreqMul) + 0.5);
}
static void generateBpmTables(void)
{
for (uint32_t i = 32; i <= 255; i++)
{
audio.bpmTab[i-32] = bpm2SmpsPerTick(i, audio.audioFreq);
audio.bpmTab28kHz[i-32] = bpm2SmpsPerTick(i, 28836);
audio.bpmTab22kHz[i-32] = bpm2SmpsPerTick(i, 22168);
}
}
void setLEDFilter(bool state)
{
editor.useLEDFilter = state;
if (editor.useLEDFilter)
filterFlags |= FILTER_LED_ENABLED;
else
filterFlags &= ~FILTER_LED_ENABLED;
}
void toggleLEDFilter(void)
{
editor.useLEDFilter ^= 1;
if (editor.useLEDFilter)
filterFlags |= FILTER_LED_ENABLED;
else
filterFlags &= ~FILTER_LED_ENABLED;
}
static void calcCoeffLED(double dSr, double dHz, ledFilterCoeff_t *filter)
{
static double dFb = 0.125;
#ifndef NO_FILTER_FINETUNING
/* 8bitbubsy: makes the filter curve sound (and look) much closer to the real deal.
** This has been tested against both an A500 and A1200. */
dFb *= 0.62;
#endif
if (dHz < dSr/2.0)
filter->dLed = ((2.0 * M_PI) * dHz) / dSr;
else
filter->dLed = 1.0;
filter->dLedFb = dFb + (dFb / (1.0 - filter->dLed)); // Q ~= 1/sqrt(2) (Butterworth)
}
void calcCoeffLossyIntegrator(double dSr, double dHz, lossyIntegrator_t *filter)
{
double dOmega = ((2.0 * M_PI) * dHz) / dSr;
filter->b0 = 1.0 / (1.0 + (1.0 / dOmega));
filter->b1 = 1.0 - filter->b0;
}
static void clearLossyIntegrator(lossyIntegrator_t *filter)
{
filter->dBuffer[0] = 0.0; // L
filter->dBuffer[1] = 0.0; // R
}
static void clearLEDFilter(ledFilter_t *filter)
{
filter->dLed[0] = 0.0; // L
filter->dLed[1] = 0.0;
filter->dLed[2] = 0.0; // R
filter->dLed[3] = 0.0;
}
static inline void lossyIntegratorLED(ledFilterCoeff_t filterC, ledFilter_t *filter, double *dIn, double *dOut)
{
// left channel "LED" filter
filter->dLed[0] += filterC.dLed * (dIn[0] - filter->dLed[0])
+ filterC.dLedFb * (filter->dLed[0] - filter->dLed[1]) + DENORMAL_OFFSET;
filter->dLed[1] += filterC.dLed * (filter->dLed[0] - filter->dLed[1]) + DENORMAL_OFFSET;
dOut[0] = filter->dLed[1];
// right channel "LED" filter
filter->dLed[2] += filterC.dLed * (dIn[1] - filter->dLed[2])
+ filterC.dLedFb * (filter->dLed[2] - filter->dLed[3]) + DENORMAL_OFFSET;
filter->dLed[3] += filterC.dLed * (filter->dLed[2] - filter->dLed[3]) + DENORMAL_OFFSET;
dOut[1] = filter->dLed[3];
}
void lossyIntegrator(lossyIntegrator_t *filter, double *dIn, double *dOut)
{
/* Low-pass filter implementation taken from:
** https://bel.fi/alankila/modguide/interpolate.txt
**
** This implementation has a less smooth cutoff curve compared to the old one, so it's
** maybe not the best. However, I stick to this one because it has a higher gain
** at the end of the curve (closer to Amiga 500). It also sounds much closer when
** comparing whitenoise on an A500. */
// left channel low-pass
filter->dBuffer[0] = (filter->b0 * dIn[0]) + (filter->b1 * filter->dBuffer[0]) + DENORMAL_OFFSET;
dOut[0] = filter->dBuffer[0];
// right channel low-pass
filter->dBuffer[1] = (filter->b0 * dIn[1]) + (filter->b1 * filter->dBuffer[1]) + DENORMAL_OFFSET;
dOut[1] = filter->dBuffer[1];
}
void lossyIntegratorHighPass(lossyIntegrator_t *filter, double *dIn, double *dOut)
{
double dLow[2];
lossyIntegrator(filter, dIn, dLow);
dOut[0] = dIn[0] - dLow[0]; // left channel high-pass
dOut[1] = dIn[1] - dLow[1]; // right channel high-pass
}
/* adejr/aciddose: these sin/cos approximations both use a 0..1
** parameter range and have 'normalized' (1/2 = 0db) coeffs
**
** the coeffs are for LERP(x, x * x, 0.224) * sqrt(2)
** max_error is minimized with 0.224 = 0.0013012886 */
static double sinApx(double fX)
{
fX = fX * (2.0 - fX);
return fX * 1.09742972 + fX * fX * 0.31678383;
}
static double cosApx(double fX)
{
fX = (1.0 - fX) * (1.0 + fX);
return fX * 1.09742972 + fX * fX * 0.31678383;
}
void lockAudio(void)
{
if (dev != 0)
SDL_LockAudioDevice(dev);
audioLocked = true;
}
void unlockAudio(void)
{
if (dev != 0)
SDL_UnlockAudioDevice(dev);
audioLocked = false;
}
void clearPaulaAndScopes(void)
{
uint8_t i;
double dOldPanL[4], dOldPanR[4];
// copy old pans
for (i = 0; i < AMIGA_VOICES; i++)
{
dOldPanL[i] = paula[i].dPanL;
dOldPanR[i] = paula[i].dPanR;
}
lockAudio();
memset(paula, 0, sizeof (paula));
unlockAudio();
// store old pans
for (i = 0; i < AMIGA_VOICES; i++)
{
paula[i].dPanL = dOldPanL[i];
paula[i].dPanR = dOldPanR[i];
}
clearScopes();
}
void mixerUpdateLoops(void) // updates Paula loop (+ scopes)
{
moduleChannel_t *ch;
moduleSample_t *s;
for (uint8_t i = 0; i < AMIGA_VOICES; i++)
{
ch = &modEntry->channels[i];
if (ch->n_samplenum == editor.currSample)
{
s = &modEntry->samples[editor.currSample];
paulaSetData(i, ch->n_start + s->loopStart);
paulaSetLength(i, s->loopLength / 2);
}
}
}
static void mixerSetVoicePan(uint8_t ch, uint16_t pan) // pan = 0..256
{
double dPan;
/* proper 'normalized' equal-power panning is (assuming pan left to right):
** L = cos(p * pi * 1/2) * sqrt(2);
** R = sin(p * pi * 1/2) * sqrt(2); */
dPan = pan * (1.0 / 256.0); // 0.0..1.0
paula[ch].dPanL = cosApx(dPan);
paula[ch].dPanR = sinApx(dPan);
}
void mixerKillVoice(uint8_t ch)
{
paulaVoice_t *v;
scopeChannelExt_t *s;
v = &paula[ch];
s = &scopeExt[ch];
v->active = false;
v->dVolume = 0.0;
s->active = false;
s->didSwapData = false;
memset(&blep[ch], 0, sizeof (blep_t));
memset(&blepVol[ch], 0, sizeof (blep_t));
}
void turnOffVoices(void)
{
for (uint8_t i = 0; i < AMIGA_VOICES; i++)
mixerKillVoice(i);
clearLossyIntegrator(&filterLo);
clearLossyIntegrator(&filterHi);
clearLEDFilter(&filterLED);
resetDitherSeed();
editor.tuningFlag = false;
}
void paulaStopDMA(uint8_t ch)
{
scopeExt[ch].active = paula[ch].active = false;
}
void paulaStartDMA(uint8_t ch)
{
const int8_t *dat;
int32_t length;
paulaVoice_t *v;
scopeChannel_t s, *sc;
scopeChannelExt_t *se;
// trigger voice
v = &paula[ch];
dat = v->newData;
if (dat == NULL)
dat = &modEntry->sampleData[RESERVED_SAMPLE_OFFSET]; // dummy sample
length = v->newLength;
if (length < 2)
length = 2; // for safety
v->dPhase = 0.0;
v->pos = 0;
v->data = dat;
v->length = length;
v->active = true;
// trigger scope
sc = &scope[ch];
se = &scopeExt[ch];
s = *sc; // cache it
dat = se->newData;
if (dat == NULL)
dat = &modEntry->sampleData[RESERVED_SAMPLE_OFFSET]; // dummy sample
s.length = length;
s.data = dat;
s.pos = 0;
s.posFrac = 0;
// data/length is already set from replayer thread (important)
s.loopFlag = se->newLoopFlag;
s.loopStart = se->newLoopStart;
se->didSwapData = false;
se->active = true;
*sc = s; // update it
}
void resetOldPeriods(void)
{
oldPeriod = 0;
}
void paulaSetPeriod(uint8_t ch, uint16_t period)
{
double dPeriodToDeltaDiv;
paulaVoice_t *v;
v = &paula[ch];
if (period == 0)
{
v->dDelta = 0.0; // confirmed behavior on real Amiga
setScopeDelta(ch, 0);
return;
}
if (period < 113)
period = 113; // confirmed behavior on real Amiga
// if the new period was the same as the previous period, use cached deltas
if (period == oldPeriod)
{
v->dDelta = oldVoiceDelta;
setScopeDelta(ch, oldScopeDelta);
}
else
{
oldPeriod = period;
// if we are rendering pattern to sample (PAT2SMP), use different frequencies
if (editor.isSMPRendering)
dPeriodToDeltaDiv = editor.pat2SmpHQ ? (PAULA_PAL_CLK / 28836.0) : (PAULA_PAL_CLK / 22168.0);
else
dPeriodToDeltaDiv = audio.dPeriodToDeltaDiv;
v->dDelta = dPeriodToDeltaDiv / period;
oldVoiceDelta = v->dDelta;
// set scope rate
#if SCOPE_HZ != 64
#error Scope Hz is not 64 (2^n), change rate calc. to use doubles+round in pt2_scope.c
#endif
oldScopeDelta = (PAULA_PAL_CLK * (65536UL / SCOPE_HZ)) / period;
setScopeDelta(ch, oldScopeDelta);
}
// for BLEP synthesis
if (v->dLastDelta == 0.0)
v->dLastDelta = v->dDelta;
}
void paulaSetVolume(uint8_t ch, uint16_t vol)
{
vol &= 127; // confirmed behavior on real Amiga
if (vol > 64)
vol = 64; // confirmed behavior on real Amiga
paula[ch].dVolume = vol * (1.0 / 64.0);
}
void paulaSetLength(uint8_t ch, uint16_t len)
{
if (len == 0)
{
len = 65535;
/* confirmed behavior on real Amiga (also needed for safety)
* And yes, we have room for this, it will never overflow! */
}
// our mixer works with bytes, not words. Multiply by two
scopeExt[ch].newLength = paula[ch].newLength = len * 2;
}
void paulaSetData(uint8_t ch, const int8_t *src)
{
uint8_t smp;
moduleSample_t *s;
scopeChannelExt_t *se, tmp;
smp = modEntry->channels[ch].n_samplenum;
assert(smp <= 30);
s = &modEntry->samples[smp];
// set voice data
if (src == NULL)
src = &modEntry->sampleData[RESERVED_SAMPLE_OFFSET]; // dummy sample
paula[ch].newData = src;
// set external scope data
se = &scopeExt[ch];
tmp = *se; // cache it
tmp.newData = src;
tmp.newLoopFlag = (s->loopStart + s->loopLength) > 2;
tmp.newLoopStart = s->loopStart;
*se = tmp; // update it
}
void toggleA500Filters(void)
{
if (filterFlags & FILTER_A500)
{
filterFlags &= ~FILTER_A500;
displayMsg("FILTER MOD: A1200");
}
else
{
filterFlags |= FILTER_A500;
clearLossyIntegrator(&filterLo);
displayMsg("FILTER MOD: A500");
}
}
void mixChannels(int32_t numSamples)
{
const int8_t *dataPtr;
double dTempSample, dTempVolume;
blep_t *bSmp, *bVol;
paulaVoice_t *v;
memset(dMixBufferL, 0, numSamples * sizeof (double));
memset(dMixBufferR, 0, numSamples * sizeof (double));
for (int32_t i = 0; i < AMIGA_VOICES; i++)
{
v = &paula[i];
bSmp = &blep[i];
bVol = &blepVol[i];
for (int32_t j = 0; v->active && j < numSamples; j++)
{
dataPtr = v->data;
if (dataPtr == NULL)
{
dTempSample = 0.0;
dTempVolume = 0.0;
}
else
{
dTempSample = dataPtr[v->pos] * (1.0 / 128.0);
dTempVolume = v->dVolume;
}
if (dTempSample != bSmp->dLastValue)
{
if (v->dLastDelta > 0.0 && v->dLastDelta > v->dLastPhase)
blepAdd(bSmp, v->dLastPhase / v->dLastDelta, bSmp->dLastValue - dTempSample);
bSmp->dLastValue = dTempSample;
}
if (dTempVolume != bVol->dLastValue)
{
blepAdd(bVol, 0.0, bVol->dLastValue - dTempVolume);
bVol->dLastValue = dTempVolume;
}
if (bSmp->samplesLeft > 0) dTempSample += blepRun(bSmp);
if (bVol->samplesLeft > 0) dTempVolume += blepRun(bVol);
dTempSample *= dTempVolume;
dMixBufferL[j] += dTempSample * v->dPanL;
dMixBufferR[j] += dTempSample * v->dPanR;
v->dPhase += v->dDelta;
while (v->dPhase >= 1.0) // PAT2SMP needs multi-step, so use while() here
{
v->dPhase -= 1.0;
v->dLastPhase = v->dPhase;
v->dLastDelta = v->dDelta;
if (++v->pos >= v->length)
{
v->pos = 0;
// re-fetch Paula register values now
v->length = v->newLength;
v->data = v->newData;
}
}
}
}
}
void resetDitherSeed(void)
{
randSeed = INITIAL_DITHER_SEED;
}
// Delphi/Pascal LCG Random() (without limit). Suitable for 32-bit random numbers
static inline int32_t random32(void)
{
randSeed = randSeed * 134775813 + 1;
return randSeed;
}
static inline void processMixedSamplesA1200(int32_t i, int16_t *out)
{
int32_t smp32;
double dOut[2], dDither;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
// don't process any low-pass filter since the cut-off is around 28-31kHz on A1200
// process "LED" filter
if (filterFlags & FILTER_LED_ENABLED)
lossyIntegratorLED(filterLEDC, &filterLED, dOut, dOut);
// process high-pass filter
lossyIntegratorHighPass(&filterHi, dOut, dOut);
// normalize and flip phase (A500/A1200 has an inverted audio signal)
dOut[0] *= -((INT16_MAX+1.0) / AMIGA_VOICES);
dOut[1] *= -((INT16_MAX+1.0) / AMIGA_VOICES);
// apply 0.5-bit dither
dDither = random32() * (0.5 / (INT32_MAX+1.0)); // -0.5..0.5
dOut[0] += dDither;
dDither = random32() * (0.5 / (INT32_MAX+1.0));
dOut[1] += dDither;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
smp32 = (int32_t)dOut[1];
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamplesA500(int32_t i, int16_t *out)
{
int32_t smp32;
double dOut[2], dDither;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
// process low-pass filter
lossyIntegrator(&filterLo, dOut, dOut);
// process "LED" filter
if (filterFlags & FILTER_LED_ENABLED)
lossyIntegratorLED(filterLEDC, &filterLED, dOut, dOut);
// process high-pass filter
lossyIntegratorHighPass(&filterHi, dOut, dOut);
// normalize and flip phase (A500/A1200 has an inverted audio signal)
dOut[0] *= -((INT16_MAX+1.0) / AMIGA_VOICES);
dOut[1] *= -((INT16_MAX+1.0) / AMIGA_VOICES);
// apply 0.5-bit dither
dDither = random32() * (0.5 / (INT32_MAX+1.0)); // -0.5..0.5
dOut[0] += dDither;
dDither = random32() * (0.5 / (INT32_MAX+1.0));
dOut[1] += dDither;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
smp32 = (int32_t)dOut[1];
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
void outputAudio(int16_t *target, int32_t numSamples)
{
int16_t *outStream, out[2];
int32_t j;
mixChannels(numSamples);
if (editor.isSMPRendering)
{
// render to sample (PAT2SMP)
for (j = 0; j < numSamples; j++)
{
processMixedSamplesA1200(j, out);
editor.pat2SmpBuf[editor.pat2SmpPos++] = (int16_t)((out[0] + out[1]) >> 1); // mix to mono
if (editor.pat2SmpPos >= MAX_SAMPLE_LEN)
{
editor.smpRenderingDone = true;
updateWindowTitle(MOD_IS_MODIFIED);
break;
}
}
}
else
{
// render to stream
outStream = target;
if (filterFlags & FILTER_A500)
{
for (j = 0; j < numSamples; j++)
{
processMixedSamplesA500(j, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
else
{
for (j = 0; j < numSamples; j++)
{
processMixedSamplesA1200(j, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
}
}
static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)
{
int16_t *out;
int32_t sampleBlock, samplesTodo;
(void)userdata;
if (forceMixerOff) // during MOD2WAV
{
memset(stream, 0, len);
return;
}
out = (int16_t *)stream;
sampleBlock = len >> 2;
while (sampleBlock)
{
samplesTodo = (sampleBlock < sampleCounter) ? sampleBlock : sampleCounter;
if (samplesTodo > 0)
{
outputAudio(out, samplesTodo);
out += (samplesTodo << 1);
sampleBlock -= samplesTodo;
sampleCounter -= samplesTodo;
}
else
{
if (editor.songPlaying)
intMusic();
sampleCounter = samplesPerTick;
}
}
}
static void calculateFilterCoeffs(void)
{
double dCutOffHz;
/* Amiga 500 filter emulation, by aciddose
**
** First comes a static low-pass 6dB formed by the supply current
** from the Paula's mixture of channels A+B / C+D into the opamp with
** 0.1uF capacitor and 360 ohm resistor feedback in inverting mode biased by
** dac vRef (used to center the output).
**
** R = 360 ohm
** C = 0.1uF
** Low Hz = 4420.97~ = 1 / (2pi * 360 * 0.0000001)
**
** Under spice simulation the circuit yields -3dB = 4400Hz.
** In the Amiga 1200, the low-pass cutoff is 26kHz+, so the
** static low-pass filter is disabled in the mixer in A1200 mode.
**
** Next comes a bog-standard Sallen-Key filter ("LED") with:
** R1 = 10K ohm
** R2 = 10K ohm
** C1 = 6800pF
** C2 = 3900pF
** Q ~= 1/sqrt(2)
**
** This filter is optionally bypassed by an MPF-102 JFET chip when
** the LED filter is turned off.
**
** Under spice simulation the circuit yields -3dB = 2800Hz.
** 90 degrees phase = 3000Hz (so, should oscillate at 3kHz!)
**
** The buffered output of the Sallen-Key passes into an RC high-pass with:
** R = 1.39K ohm (1K ohm + 390 ohm)
** C = 22uF (also C = 330nF, for improved high-frequency)
**
** High Hz = 5.2~ = 1 / (2pi * 1390 * 0.000022)
** Under spice simulation the circuit yields -3dB = 5.2Hz.
*/
// Amiga 500 rev6 RC low-pass filter:
const double dLp_R = 360.0; // R321 - 360 ohm resistor
const double dLp_C = 1e-7; // C321 - 0.1uF capacitor
dCutOffHz = 1.0 / ((2.0 * M_PI) * dLp_R * dLp_C); // ~4420.97Hz
#ifndef NO_FILTER_FINETUNING
dCutOffHz += 580.0; // 8bitbubsy: finetuning to better match A500 low-pass testing
#endif
calcCoeffLossyIntegrator(audio.dAudioFreq, dCutOffHz, &filterLo);
// Amiga Sallen-Key "LED" filter:
const double dLed_R1 = 10000.0; // R322 - 10K ohm resistor
const double dLed_R2 = 10000.0; // R323 - 10K ohm resistor
const double dLed_C1 = 6.8e-9; // C322 - 6800pF capacitor
const double dLed_C2 = 3.9e-9; // C323 - 3900pF capacitor
dCutOffHz = 1.0 / ((2.0 * M_PI) * sqrt(dLed_R1 * dLed_R2 * dLed_C1 * dLed_C2)); // ~3090.53Hz
#ifndef NO_FILTER_FINETUNING
dCutOffHz -= 300.0; // 8bitbubsy: finetuning to better match A500 & A1200 "LED" filter testing
#endif
calcCoeffLED(audio.dAudioFreq, dCutOffHz, &filterLEDC);
// Amiga RC high-pass filter:
const double dHp_R = 1000.0 + 390.0; // R324 - 1K ohm resistor + R325 - 390 ohm resistor
const double dHp_C = 2.2e-5; // C334 - 22uF capacitor
dCutOffHz = 1.0 / ((2.0 * M_PI) * dHp_R * dHp_C); // ~5.20Hz
#ifndef NO_FILTER_FINETUNING
dCutOffHz += 1.5; // 8bitbubsy: finetuning to better match A500 & A1200 high-pass testing
#endif
calcCoeffLossyIntegrator(audio.dAudioFreq, dCutOffHz, &filterHi);
}
void mixerCalcVoicePans(uint8_t stereoSeparation)
{
uint8_t scaledPanPos = (stereoSeparation * 128) / 100;
ch1Pan = 128 - scaledPanPos;
ch2Pan = 128 + scaledPanPos;
ch3Pan = 128 + scaledPanPos;
ch4Pan = 128 - scaledPanPos;
mixerSetVoicePan(0, ch1Pan);
mixerSetVoicePan(1, ch2Pan);
mixerSetVoicePan(2, ch3Pan);
mixerSetVoicePan(3, ch4Pan);
}
bool setupAudio(void)
{
SDL_AudioSpec want, have;
want.freq = ptConfig.soundFrequency;
want.format = AUDIO_S16;
want.channels = 2;
want.callback = audioCallback;
want.userdata = NULL;
want.samples = ptConfig.soundBufferSize;
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (dev == 0)
{
showErrorMsgBox("Unable to open audio device: %s", SDL_GetError());
return false;
}
if (have.freq < 32000) // lower than this is not safe for one-step mixer w/ BLEP
{
showErrorMsgBox("Unable to open audio: The audio output rate couldn't be used!");
return false;
}
if (have.format != want.format)
{
showErrorMsgBox("Unable to open audio: The sample format (signed 16-bit) couldn't be used!");
return false;
}
maxSamplesToMix = (int32_t)ceil((have.freq * 2.5) / 32.0);
dMixBufferL = (double *)calloc(maxSamplesToMix, sizeof (double));
dMixBufferR = (double *)calloc(maxSamplesToMix, sizeof (double));
editor.mod2WavBuffer = (int16_t *)malloc(sizeof (int16_t) * maxSamplesToMix);
if (dMixBufferL == NULL || dMixBufferR == NULL || editor.mod2WavBuffer == NULL)
{
showErrorMsgBox("Out of memory!");
return false;
}
audio.audioBufferSize = have.samples;
ptConfig.soundFrequency = have.freq;
audio.audioFreq = ptConfig.soundFrequency;
audio.dAudioFreq = (double)ptConfig.soundFrequency;
audio.dPeriodToDeltaDiv = PAULA_PAL_CLK / audio.dAudioFreq;
mixerCalcVoicePans(ptConfig.stereoSeparation);
defStereoSep = ptConfig.stereoSeparation;
filterFlags = ptConfig.a500LowPassFilter ? FILTER_A500 : 0;
calculateFilterCoeffs();
generateBpmTables();
samplesPerTick = 0;
sampleCounter = 0;
SDL_PauseAudioDevice(dev, false);
return true;
}
void audioClose(void)
{
if (dev > 0)
{
SDL_PauseAudioDevice(dev, true);
SDL_CloseAudioDevice(dev);
dev = 0;
}
if (dMixBufferL != NULL)
{
free(dMixBufferL);
dMixBufferL = NULL;
}
if (dMixBufferR != NULL)
{
free(dMixBufferR);
dMixBufferR = NULL;
}
if (editor.mod2WavBuffer != NULL)
{
free(editor.mod2WavBuffer);
editor.mod2WavBuffer = NULL;
}
}
void mixerSetSamplesPerTick(int32_t val)
{
samplesPerTick = val;
}
void mixerClearSampleCounter(void)
{
sampleCounter = 0;
}
void toggleAmigaPanMode(void)
{
amigaPanFlag ^= 1;
if (!amigaPanFlag)
{
mixerCalcVoicePans(defStereoSep);
displayMsg("AMIGA PANNING OFF");
}
else
{
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;
if (samplesToMix > maxSamplesToMix)
samplesToMix = maxSamplesToMix;
outputAudio(outStream, samplesToMix);
outStream += (samplesToMix << 1);
smpCounter -= samplesToMix;
}
return samplesPerTick << 1; // * 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.audioFreq;
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 * 4; // 16-bit stereo = * 4
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
{
// this is so ugly
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;
}
}
}
void normalize32bitSigned(int32_t *sampleData, uint32_t sampleLength)
{
int32_t sample, sampleVolPeak;
uint32_t i;
double dGain;
sampleVolPeak = 0;
for (i = 0; i < sampleLength; i++)
{
sample = ABS(sampleData[i]);
if (sampleVolPeak < sample)
sampleVolPeak = sample;
}
if (sampleVolPeak >= INT32_MAX)
return; // sample is already normalized
// prevent division by zero!
if (sampleVolPeak <= 0)
sampleVolPeak = 1;
dGain = (double)INT32_MAX / sampleVolPeak;
for (i = 0; i < sampleLength; i++)
{
sample = (int32_t)(sampleData[i] * dGain);
sampleData[i] = (int32_t)sample;
}
}
void normalize16bitSigned(int16_t *sampleData, uint32_t sampleLength)
{
uint32_t i;
int32_t sample, sampleVolPeak, gain;
sampleVolPeak = 0;
for (i = 0; i < sampleLength; i++)
{
sample = ABS(sampleData[i]);
if (sampleVolPeak < sample)
sampleVolPeak = sample;
}
if (sampleVolPeak >= INT16_MAX)
return; // sample is already normalized
if (sampleVolPeak < 1)
return;
gain = (INT16_MAX * 65536) / sampleVolPeak;
for (i = 0; i < sampleLength; i++)
sampleData[i] = (int16_t)((sampleData[i] * gain) >> 16);
}
void normalize8bitFloatSigned(float *fSampleData, uint32_t sampleLength)
{
uint32_t i;
float fSample, fSampleVolPeak, fGain;
fSampleVolPeak = 0.0f;
for (i = 0; i < sampleLength; i++)
{
fSample = fabsf(fSampleData[i]);
if (fSampleVolPeak < fSample)
fSampleVolPeak = fSample;
}
if (fSampleVolPeak <= 0.0f)
return;
fGain = INT8_MAX / fSampleVolPeak;
for (i = 0; i < sampleLength; i++)
fSampleData[i] *= fGain;
}
void normalize8bitDoubleSigned(double *dSampleData, uint32_t sampleLength)
{
uint32_t i;
double dSample, dSampleVolPeak, dGain;
dSampleVolPeak = 0.0;
for (i = 0; i < sampleLength; i++)
{
dSample = fabs(dSampleData[i]);
if (dSampleVolPeak < dSample)
dSampleVolPeak = dSample;
}
if (dSampleVolPeak <= 0.0)
return;
dGain = INT8_MAX / dSampleVolPeak;
for (i = 0; i < sampleLength; i++)
dSampleData[i] *= dGain;
}