ref: f2c806cb2ade67461ccb8d05c604ff2ad715e75a
dir: /src/pt2_audio.c/
// the audio filters and BLEP synthesis were coded by aciddose
// 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
#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"
#include "pt2_mod2wav.h"
#include "pt2_pat2smp.h"
#include "pt2_sync.h"
#include "pt2_structs.h"
#define INITIAL_DITHER_SEED 0x12345000
typedef struct ledFilter_t
{
double buffer[4];
double c, ci, feedback, bg, cg, c2;
} ledFilter_t;
static volatile int8_t filterFlags;
static int8_t defStereoSep;
static bool amigaPanFlag;
static uint16_t ch1Pan, ch2Pan, ch3Pan, ch4Pan;
static int32_t oldPeriod = -1, randSeed = INITIAL_DITHER_SEED;
static uint32_t sampleCounter, audLatencyPerfValInt, audLatencyPerfValFrac;
static uint64_t tickTime64, tickTime64Frac;
static double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned, dOldVoiceDelta, dOldVoiceDeltaMul;
static double dPrngStateL, dPrngStateR;
static blep_t blep[AMIGA_VOICES], blepVol[AMIGA_VOICES];
static rcFilter_t filterLoA500, filterLoA1200, filterHi;
static ledFilter_t filterLED;
static SDL_AudioDeviceID dev;
// for audio/video syncing
static uint32_t tickTimeLen, tickTimeLenFrac;
// globalized
audio_t audio;
uint32_t samplesPerTick;
paulaVoice_t paula[AMIGA_VOICES];
bool intMusic(void); // defined in pt_modplayer.c
static void calcAudioLatencyVars(int32_t audioBufferSize, int32_t audioFreq)
{
double dInt, dFrac;
if (audioFreq == 0)
return;
const double dAudioLatencySecs = audioBufferSize / (double)audioFreq;
dFrac = modf(dAudioLatencySecs * editor.dPerfFreq, &dInt);
// integer part
audLatencyPerfValInt = (int32_t)dInt;
// fractional part (scaled to 0..2^32-1)
dFrac *= UINT32_MAX;
dFrac += 0.5;
if (dFrac > UINT32_MAX)
dFrac = UINT32_MAX;
audLatencyPerfValFrac = (uint32_t)dFrac;
}
void setSyncTickTimeLen(uint32_t timeLen, uint32_t timeLenFrac)
{
tickTimeLen = timeLen;
tickTimeLenFrac = timeLenFrac;
}
static uint16_t bpm2SmpsPerTick(int32_t bpm, double dAudioFreq)
{
if (bpm == 0)
return 0;
const int32_t ciaVal = (int32_t)(1773447 / bpm); // yes, PT truncates here
const double dCiaHz = (double)CIA_PAL_CLK / ciaVal;
int32_t smpsPerTick = (int32_t)((dAudioFreq / dCiaHz) + 0.5); // rounded
return (uint16_t)smpsPerTick;
}
static void generateBpmTables(void)
{
for (int32_t i = 32; i <= 255; i++)
{
audio.bpmTab[i-32] = bpm2SmpsPerTick(i, audio.outputRate);
audio.bpmTab28kHz[i-32] = bpm2SmpsPerTick(i, PAT2SMP_HI_FREQ); // PAT2SMP hi quality
audio.bpmTab22kHz[i-32] = bpm2SmpsPerTick(i, PAT2SMP_LO_FREQ); // PAT2SMP low quality
audio.bpmTabMod2Wav[i-32] = bpm2SmpsPerTick(i, MOD2WAV_FREQ); // MOD2WAV
}
}
static void clearLEDFilterState(void)
{
filterLED.buffer[0] = 0.0; // left channel
filterLED.buffer[1] = 0.0;
filterLED.buffer[2] = 0.0; // right channel
filterLED.buffer[3] = 0.0;
}
void setLEDFilter(bool state, bool doLockAudio)
{
const bool audioWasntLocked = !audio.locked;
if (doLockAudio && audioWasntLocked)
lockAudio();
editor.useLEDFilter = state;
if (editor.useLEDFilter)
{
clearLEDFilterState();
filterFlags |= FILTER_LED_ENABLED;
}
else
{
filterFlags &= ~FILTER_LED_ENABLED;
}
if (doLockAudio && audioWasntLocked)
unlockAudio();
}
void toggleLEDFilter(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
editor.useLEDFilter ^= 1;
if (editor.useLEDFilter)
{
clearLEDFilterState();
filterFlags |= FILTER_LED_ENABLED;
}
else
{
filterFlags &= ~FILTER_LED_ENABLED;
}
if (audioWasntLocked)
unlockAudio();
}
/* Imperfect "LED" filter implementation. This may be further improved in the future.
** Based upon ideas posted by mystran @ the kvraudio.com forum.
**
** This filter may not function correctly used outside the fixed-cutoff context here!
*/
static double sigmoid(double x, double coefficient)
{
/* Coefficient from:
** 0.0 to inf (linear)
** -1.0 to -inf (linear)
*/
return x / (x + coefficient) * (coefficient + 1.0);
}
static void calcLEDFilterCoeffs(const double sr, const double hz, const double fb, ledFilter_t *filter)
{
/* tan() may produce NaN or other bad results in some cases!
** It appears to work correctly with these specific coefficients.
*/
const double c = (hz < (sr / 2.0)) ? tan((M_PI * hz) / sr) : 1.0;
const double g = 1.0 / (1.0 + c);
// dirty compensation
const double s = 0.5;
const double t = 0.5;
const double ic = c > t ? 1.0 / ((1.0 - s*t) + s*c) : 1.0;
const double cg = c * g;
const double fbg = 1.0 / (1.0 + fb * cg*cg);
filter->c = c;
filter->ci = g;
filter->feedback = 2.0 * sigmoid(fb, 0.5);
filter->bg = fbg * filter->feedback * ic;
filter->cg = cg;
filter->c2 = c * 2.0;
}
static inline void LEDFilter(ledFilter_t *f, const double *in, double *out)
{
const double in_1 = DENORMAL_OFFSET;
const double in_2 = DENORMAL_OFFSET;
const double c = f->c;
const double g = f->ci;
const double cg = f->cg;
const double bg = f->bg;
const double c2 = f->c2;
double *v = f->buffer;
// left channel
const double estimate_L = in_2 + g*(v[1] + c*(in_1 + g*(v[0] + c*in[0])));
const double y0_L = v[0]*g + in[0]*cg + in_1 + estimate_L * bg;
const double y1_L = v[1]*g + y0_L*cg + in_2;
v[0] += c2 * (in[0] - y0_L);
v[1] += c2 * (y0_L - y1_L);
out[0] = y1_L;
// right channel
const double estimate_R = in_2 + g*(v[3] + c*(in_1 + g*(v[2] + c*in[1])));
const double y0_R = v[2]*g + in[1]*cg + in_1 + estimate_R * bg;
const double y1_R = v[3]*g + y0_R*cg + in_2;
v[2] += c2 * (in[1] - y0_R);
v[3] += c2 * (y0_R - y1_R);
out[1] = y1_R;
}
void calcRCFilterCoeffs(double dSr, double dHz, rcFilter_t *f)
{
const double c = (dHz < (dSr / 2.0)) ? tan((M_PI * dHz) / dSr) : 1.0;
f->c = c;
f->c2 = f->c * 2.0;
f->g = 1.0 / (1.0 + f->c);
f->cg = f->c * f->g;
}
void clearRCFilterState(rcFilter_t *f)
{
f->buffer[0] = 0.0; // left channel
f->buffer[1] = 0.0; // right channel
}
// aciddose: input 0 is resistor side of capacitor (low-pass), input 1 is reference side (high-pass)
static inline double getLowpassOutput(rcFilter_t *f, const double input_0, const double input_1, const double buffer)
{
return buffer * f->g + input_0 * f->cg + input_1 * (1.0 - f->cg);
}
void RCLowPassFilter(rcFilter_t *f, const double *in, double *out)
{
double output;
// left channel RC low-pass
output = getLowpassOutput(f, in[0], 0.0, f->buffer[0]);
f->buffer[0] += (in[0] - output) * f->c2;
out[0] = output;
// right channel RC low-pass
output = getLowpassOutput(f, in[1], 0.0, f->buffer[1]);
f->buffer[1] += (in[1] - output) * f->c2;
out[1] = output;
}
void RCHighPassFilter(rcFilter_t *f, const double *in, double *out)
{
double low[2];
RCLowPassFilter(f, in, low);
out[0] = in[0] - low[0]; // left channel high-pass
out[1] = in[1] - low[1]; // right channel high-pass
}
/* These two are used for the filters in the SAMPLER screen, and
** also the 2x downsampling when loading samples whose frequency
** is above 22kHz.
*/
void RCLowPassFilterMono(rcFilter_t *f, const double in, double *out)
{
double output = getLowpassOutput(f, in, 0.0, f->buffer[0]);
f->buffer[0] += (in - output) * f->c2;
*out = output;
}
void RCHighPassFilterMono(rcFilter_t *f, const double in, double *out)
{
double low;
RCLowPassFilterMono(f, in, &low);
*out = in - low; // high-pass
}
void lockAudio(void)
{
if (dev != 0)
SDL_LockAudioDevice(dev);
audio.locked = true;
audio.resetSyncTickTimeFlag = true;
resetChSyncQueue();
}
void unlockAudio(void)
{
if (dev != 0)
SDL_UnlockAudioDevice(dev);
audio.resetSyncTickTimeFlag = true;
resetChSyncQueue();
audio.locked = false;
}
void mixerUpdateLoops(void) // updates Paula loop (+ scopes)
{
for (int32_t i = 0; i < AMIGA_VOICES; i++)
{
const moduleChannel_t *ch = &song->channels[i];
if (ch->n_samplenum == editor.currSample)
{
const moduleSample_t *s = &song->samples[editor.currSample];
paulaSetData(i, ch->n_start + s->loopStart);
paulaSetLength(i, s->loopLength >> 1);
if (!editor.songPlaying)
{
scopeSetData(i, ch->n_start + s->loopStart);
scopeSetLength(i, s->loopLength >> 1);
}
}
}
}
/* 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 x)
{
x = x * (2.0 - x);
return x * 1.09742972 + x * x * 0.31678383;
}
static double cosApx(double x)
{
x = (1.0 - x) * (1.0 + x);
return x * 1.09742972 + x * x * 0.31678383;
}
static void mixerSetVoicePan(uint8_t ch, uint16_t pan) // pan = 0..256
{
/* aciddose: 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);
*/
const double dPan = pan * (1.0 / 256.0); // 0.0..1.0
paula[ch].dPanL = cosApx(dPan);
paula[ch].dPanR = sinApx(dPan);
}
void mixerKillVoice(int32_t ch)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
// copy old pans
const double dOldPanL = paula[ch].dPanL;
const double dOldPanR = paula[ch].dPanR;
memset(&paula[ch], 0, sizeof (paulaVoice_t));
stopScope(ch);
// store old pans
paula[ch].dPanL = dOldPanL;
paula[ch].dPanR = dOldPanR;
memset(&blep[ch], 0, sizeof (blep_t));
memset(&blepVol[ch], 0, sizeof (blep_t));
if (audioWasntLocked)
unlockAudio();
}
void turnOffVoices(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
for (int32_t i = 0; i < AMIGA_VOICES; i++)
mixerKillVoice(i);
clearRCFilterState(&filterLoA500);
clearRCFilterState(&filterLoA1200);
clearRCFilterState(&filterHi);
clearLEDFilterState();
resetAudioDithering();
editor.tuningFlag = false;
if (audioWasntLocked)
unlockAudio();
}
void resetCachedMixerPeriod(void)
{
oldPeriod = -1;
}
// the following routines are only called from the mixer thread.
void paulaSetPeriod(int32_t ch, uint16_t period)
{
int32_t realPeriod;
double dPeriodToDeltaDiv;
paulaVoice_t *v;
v = &paula[ch];
v->syncPeriod = period; // used for pt2_sync.c
v->syncFlags |= UPDATE_PERIOD; // used for pt2_sync.c
if (period == 0)
realPeriod = 1+65535; // confirmed behavior on real Amiga
else if (period < 113)
realPeriod = 113; // close to what happens on real Amiga (and needed for BLEP synthesis)
else
realPeriod = period;
// if the new period was the same as the previous period, use cached deltas
if (realPeriod != oldPeriod)
{
oldPeriod = realPeriod;
// this period is not cached, calculate mixer deltas
// during PAT2SMP or doing MOD2WAV, use different audio output rates
if (editor.isSMPRendering)
dPeriodToDeltaDiv = editor.pat2SmpHQ ? (PAULA_PAL_CLK / PAT2SMP_HI_FREQ) : (PAULA_PAL_CLK / PAT2SMP_LO_FREQ);
else if (editor.isWAVRendering)
dPeriodToDeltaDiv = PAULA_PAL_CLK / (double)MOD2WAV_FREQ;
else
dPeriodToDeltaDiv = audio.dPeriodToDeltaDiv;
// cache these
dOldVoiceDelta = dPeriodToDeltaDiv / realPeriod;
dOldVoiceDeltaMul = 1.0 / dOldVoiceDelta; // for BLEP synthesis
}
v->dDelta = dOldVoiceDelta;
// for BLEP synthesis
v->dDeltaMul = dOldVoiceDeltaMul;
if (v->dLastDelta == 0.0) v->dLastDelta = v->dDelta;
if (v->dLastDeltaMul == 0.0) v->dLastDeltaMul = v->dDeltaMul;
}
void paulaSetVolume(int32_t ch, uint16_t vol)
{
paulaVoice_t *v;
v = &paula[ch];
vol &= 127; // confirmed behavior on real Amiga
if (vol > 64)
vol = 64; // confirmed behavior on real Amiga
v->dVolume = vol * (1.0 / 64.0);
v->syncVolume = (int8_t)vol; // used for pt2_sync.c
v->syncFlags |= UPDATE_VOLUME; // used for pt2_sync.c
}
void paulaSetLength(int32_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!
*/
}
paula[ch].newLength = len << 1; // our mixer works with bytes, not words
paula[ch].syncFlags |= UPDATE_LENGTH; // for pt2_sync.c
}
void paulaSetData(int32_t ch, const int8_t *src)
{
// set voice data
if (src == NULL)
src = &song->sampleData[RESERVED_SAMPLE_OFFSET]; // dummy sample
paula[ch].newData = src;
paula[ch].syncFlags |= UPDATE_DATA; // for pt2_sync.c
}
void paulaStopDMA(int32_t ch)
{
paula[ch].active = false;
}
void paulaStartDMA(int32_t ch)
{
const int8_t *dat;
int32_t length;
paulaVoice_t *v;
// trigger voice
v = &paula[ch];
dat = v->newData;
if (dat == NULL)
dat = &song->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;
// for pt2_sync.c
v->syncTriggerData = dat;
v->syncTriggerLength = (uint16_t)(length >> 1);
v->syncFlags |= TRIGGER_SAMPLE;
}
void toggleA500Filters(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
clearRCFilterState(&filterLoA500);
clearRCFilterState(&filterLoA1200);
clearRCFilterState(&filterHi);
clearLEDFilterState();
if (filterFlags & FILTER_A500)
{
filterFlags &= ~FILTER_A500;
displayMsg("LP FILTER: A1200");
}
else
{
filterFlags |= FILTER_A500;
displayMsg("LP FILTER: A500");
}
if (audioWasntLocked)
unlockAudio();
}
void mixChannels(int32_t numSamples)
{
double dSmp, dVol;
blep_t *bSmp, *bVol;
paulaVoice_t *v;
memset(dMixBufferL, 0, numSamples * sizeof (double));
memset(dMixBufferR, 0, numSamples * sizeof (double));
v = paula;
bSmp = blep;
bVol = blepVol;
for (int32_t i = 0; i < AMIGA_VOICES; i++, v++, bSmp++, bVol++)
{
if (!v->active || v->data == NULL)
continue;
for (int32_t j = 0; j < numSamples; j++)
{
assert(v->data != NULL);
dSmp = v->data[v->pos] * (1.0 / 128.0);
dVol = v->dVolume;
if (dSmp != bSmp->dLastValue)
{
if (v->dLastDelta > v->dLastPhase)
{
// div->mul trick: v->dLastDeltaMul is 1.0 / v->dLastDelta
blepAdd(bSmp, v->dLastPhase * v->dLastDeltaMul, bSmp->dLastValue - dSmp);
}
bSmp->dLastValue = dSmp;
}
if (dVol != bVol->dLastValue)
{
blepVolAdd(bVol, bVol->dLastValue - dVol);
bVol->dLastValue = dVol;
}
if (bSmp->samplesLeft > 0) dSmp = blepRun(bSmp, dSmp);
if (bVol->samplesLeft > 0) dVol = blepRun(bVol, dVol);
dSmp *= dVol;
dMixBufferL[j] += dSmp * v->dPanL;
dMixBufferR[j] += dSmp * v->dPanR;
v->dPhase += v->dDelta;
if (v->dPhase >= 1.0) // deltas can't be >= 1.0, so this is safe
{
v->dPhase -= 1.0;
v->dLastPhase = v->dPhase;
v->dLastDelta = v->dDelta;
v->dLastDeltaMul = v->dDeltaMul;
if (++v->pos >= v->length)
{
v->pos = 0;
// re-fetch new Paula register values now
v->length = v->newLength;
v->data = v->newData;
}
}
}
}
}
void mixChannelsMultiStep(int32_t numSamples) // for PAT2SMP
{
double dSmp, dVol;
blep_t *bSmp, *bVol;
paulaVoice_t *v;
memset(dMixBufferL, 0, numSamples * sizeof (double));
memset(dMixBufferR, 0, numSamples * sizeof (double));
v = paula;
bSmp = blep;
bVol = blepVol;
for (int32_t i = 0; i < AMIGA_VOICES; i++, v++, bSmp++, bVol++)
{
if (!v->active || v->data == NULL)
continue;
for (int32_t j = 0; j < numSamples; j++)
{
assert(v->data != NULL);
dSmp = v->data[v->pos] * (1.0 / 128.0);
dVol = v->dVolume;
if (dSmp != bSmp->dLastValue)
{
if (v->dLastDelta > v->dLastPhase)
{
// div->mul trick: v->dLastDeltaMul is 1.0 / v->dLastDelta
blepAdd(bSmp, v->dLastPhase * v->dLastDeltaMul, bSmp->dLastValue - dSmp);
}
bSmp->dLastValue = dSmp;
}
if (dVol != bVol->dLastValue)
{
blepVolAdd(bVol, bVol->dLastValue - dVol);
bVol->dLastValue = dVol;
}
if (bSmp->samplesLeft > 0) dSmp = blepRun(bSmp, dSmp);
if (bVol->samplesLeft > 0) dVol = blepRun(bVol, dVol);
dSmp *= dVol;
dMixBufferL[j] += dSmp * v->dPanL;
dMixBufferR[j] += dSmp * v->dPanR;
v->dPhase += v->dDelta;
while (v->dPhase >= 1.0) // deltas can be >= 1.0 here
{
v->dPhase -= 1.0;
v->dLastPhase = v->dPhase;
v->dLastDelta = v->dDelta;
v->dLastDeltaMul = v->dDeltaMul;
if (++v->pos >= v->length)
{
v->pos = 0;
// re-fetch new Paula register values now
v->length = v->newLength;
v->data = v->newData;
}
}
}
}
}
void resetAudioDithering(void)
{
randSeed = INITIAL_DITHER_SEED;
dPrngStateL = 0.0;
dPrngStateR = 0.0;
}
static inline int32_t random32(void)
{
// LCG random 32-bit generator (quite good and fast)
randSeed *= 134775813;
randSeed++;
return randSeed;
}
static inline void processMixedSamplesA1200(int32_t i, int16_t *out)
{
int32_t smp32;
double dOut[2], dPrng;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
if (audio.outputRate >= 96000) // cutoff is too high for 44.1kHz/48kHz
{
// process low-pass filter
RCLowPassFilter(&filterLoA1200, dOut, dOut);
}
// process high-pass filter
RCHighPassFilter(&filterHi, dOut, dOut);
// normalize and flip phase (A500/A1200 has an inverted audio signal)
dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[0] = (dOut[0] + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[1] = (dOut[1] + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dOut[1];
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamplesA1200LED(int32_t i, int16_t *out)
{
int32_t smp32;
double dOut[2], dPrng;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
if (audio.outputRate >= 96000) // cutoff is too high for 44.1kHz/48kHz
{
// process low-pass filter
RCLowPassFilter(&filterLoA1200, dOut, dOut);
}
// process "LED" filter
LEDFilter(&filterLED, dOut, dOut);
// process high-pass filter
RCHighPassFilter(&filterHi, dOut, dOut);
// normalize and flip phase (A500/A1200 has an inverted audio signal)
dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[0] = (dOut[0] + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[1] = (dOut[1] + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
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], dPrng;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
// process low-pass filter
RCLowPassFilter(&filterLoA500, dOut, dOut);
// process high-pass filter
RCHighPassFilter(&filterHi, dOut, dOut);
dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[0] = (dOut[0] + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[1] = (dOut[1] + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dOut[1];
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamplesA500LED(int32_t i, int16_t *out)
{
int32_t smp32;
double dOut[2], dPrng;
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
// process low-pass filter
RCLowPassFilter(&filterLoA500, dOut, dOut);
// process "LED" filter
LEDFilter(&filterLED, dOut, dOut);
// process high-pass filter
RCHighPassFilter(&filterHi, dOut, dOut);
dOut[0] *= (-INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= (-INT16_MAX / (double)AMIGA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[0] = (dOut[0] + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut[1] = (dOut[1] + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dOut[1];
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamplesRaw(int32_t i, int16_t *out) // for PAT2SMP
{
int32_t smp32;
double dOut[2];
dOut[0] = dMixBufferL[i];
dOut[1] = dMixBufferR[i];
// normalize (don't flip the phase this time)
dOut[0] *= (INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= (INT16_MAX / (double)AMIGA_VOICES);
dOut[0] = (dOut[0] + dOut[1]) * 0.5; // mix to mono
smp32 = (int32_t)dOut[0];
CLAMP16(smp32);
*out = (int16_t)smp32;
}
void outputAudio(int16_t *target, int32_t numSamples)
{
int16_t *outStream, out[2];
int32_t i;
if (editor.isSMPRendering)
{
// render to sample (PAT2SMP)
int32_t samplesTodo = numSamples;
if (editor.pat2SmpPos+samplesTodo > MAX_SAMPLE_LEN)
samplesTodo = MAX_SAMPLE_LEN-editor.pat2SmpPos;
mixChannelsMultiStep(samplesTodo);
outStream = &editor.pat2SmpBuf[editor.pat2SmpPos];
for (i = 0; i < samplesTodo; i++)
{
processMixedSamplesRaw(i, out);
outStream[i] = out[0];
}
editor.pat2SmpPos += samplesTodo;
if (editor.pat2SmpPos >= MAX_SAMPLE_LEN)
{
editor.smpRenderingDone = true;
updateWindowTitle(MOD_IS_MODIFIED);
}
}
else
{
// render to stream
mixChannels(numSamples);
outStream = target;
if (filterFlags & FILTER_A500)
{
// Amiga 500 filter model
if (filterFlags & FILTER_LED_ENABLED)
{
for (i = 0; i < numSamples; i++)
{
processMixedSamplesA500LED(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
else
{
for (i = 0; i < numSamples; i++)
{
processMixedSamplesA500(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
}
else
{
// Amiga 1200 filter model
if (filterFlags & FILTER_LED_ENABLED)
{
for (i = 0; i < numSamples; i++)
{
processMixedSamplesA1200LED(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
else
{
for (i = 0; i < numSamples; i++)
{
processMixedSamplesA1200(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
}
}
}
static void fillVisualsSyncBuffer(void)
{
chSyncData_t chSyncData;
if (audio.resetSyncTickTimeFlag)
{
audio.resetSyncTickTimeFlag = false;
tickTime64 = SDL_GetPerformanceCounter() + audLatencyPerfValInt;
tickTime64Frac = audLatencyPerfValFrac;
}
moduleChannel_t *c = song->channels;
paulaVoice_t *v = paula;
syncedChannel_t *s = chSyncData.channels;
for (int32_t i = 0; i < AMIGA_VOICES; i++, c++, s++, v++)
{
s->flags = v->syncFlags | c->syncFlags;
c->syncFlags = v->syncFlags = 0; // clear sync flags
s->volume = v->syncVolume;
s->period = v->syncPeriod;
s->triggerData = v->syncTriggerData;
s->triggerLength = v->syncTriggerLength;
s->newData = v->newData;
s->newLength = (uint16_t)(v->newLength >> 1);
s->vuVolume = c->syncVuVolume;
s->analyzerVolume = c->syncAnalyzerVolume;
s->analyzerPeriod = c->syncAnalyzerPeriod;
}
chSyncData.timestamp = tickTime64;
chQueuePush(chSyncData);
tickTime64 += tickTimeLen;
tickTime64Frac += tickTimeLenFrac;
if (tickTime64Frac > 0xFFFFFFFF)
{
tickTime64Frac &= 0xFFFFFFFF;
tickTime64++;
}
}
static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)
{
int16_t *streamOut;
uint32_t samplesLeft;
if (audio.forceMixerOff) // during MOD2WAV
{
memset(stream, 0, len);
return;
}
streamOut = (int16_t *)stream;
samplesLeft = len >> 2;
while (samplesLeft > 0)
{
if (sampleCounter == 0)
{
// new replayer tick
if (editor.songPlaying)
{
intMusic();
fillVisualsSyncBuffer();
}
sampleCounter = samplesPerTick;
}
uint32_t samplesTodo = sampleCounter;
if (samplesTodo > samplesLeft)
samplesTodo = samplesLeft;
outputAudio(streamOut, samplesTodo);
streamOut += samplesTodo << 1;
samplesLeft -= samplesTodo;
sampleCounter -= samplesTodo;
}
(void)userdata;
}
static void calculateFilterCoeffs(void)
{
/* 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 ~34kHz, 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.
**
** 8bitbubsy:
** Keep in mind that many of the Amiga schematics that are floating around on
** the internet have wrong RC values! They were most likely very early schematics
** that didn't change before production (or changes that never reached production).
** This has been confirmed by measuring the components on several Amiga motherboards.
**
** Correct values for A500 (A500_R6.pdf):
** - RC 6dB/oct low-pass: R=360 ohm, C=0.1uF (f=4420.970Hz)
** - Sallen-key low-pass ("LED"): R1/R2=10k ohm, C1=6800pF, C2=3900pF (f=3090.532Hz)
** - RC 6dB/oct high-pass: R=1390 ohm (1000+390), C=22.33uF (22+0.33) (f=5.127Hz)
**
** Correct values for A1200 (A1200_R2.pdf):
** - RC 6dB/oct low-pass: R=680 ohm, C=6800pF (f=34419.321Hz)
** - Sallen-key low-pass ("LED"): Same as A500 (f=3090.532Hz)
** - RC 6dB/oct high-pass: R=1390 ohm (1000+390), C=22uF (f=5.204Hz)
**
** Correct values for A600 (a600_schematics.pdf):
** - RC 6dB/oct low-pass: Same as A500 (f=4420.970Hz)
** - Sallen-key low-pass ("LED"): Same as A500 (f=3090.532Hz)
** - RC 6dB/oct high-pass: Same as A1200 (f=5.204Hz)
*/
double R, C, R1, R2, C1, C2, fc, fb;
if (audio.outputRate >= 96000) // cutoff is too high for 44.1kHz/48kHz
{
// A1200 one-pole 6db/oct static RC low-pass filter:
R = 680.0; // R321 (680 ohm resistor)
C = 6.8e-9; // C321 (6800pf capacitor)
fc = 1.0 / (2.0 * M_PI * R * C);
calcRCFilterCoeffs(audio.outputRate, fc, &filterLoA1200);
}
// A500 one-pole 6db/oct static RC low-pass filter:
R = 360.0; // R321 (360 ohm resistor)
C = 1e-7; // C321 (0.1uF capacitor)
fc = 1.0 / (2.0 * M_PI * R * C);
calcRCFilterCoeffs(audio.outputRate, fc, &filterLoA500);
// Sallen-Key filter ("LED"):
R1 = 10000.0; // R322 (10K ohm resistor)
R2 = 10000.0; // R323 (10K ohm resistor)
C1 = 6.8e-9; // C322 (6800pF capacitor)
C2 = 3.9e-9; // C323 (3900pF capacitor)
fc = 1.0 / (2.0 * M_PI * sqrt(R1 * R2 * C1 * C2));
fb = 0.125; // Fb = 0.125 : Q ~= 1/sqrt(2)
calcLEDFilterCoeffs(audio.outputRate, fc, fb, &filterLED);
// A1200 one-pole 6db/oct static RC high-pass filter:
R = 1390.0; // R324 (1K ohm resistor) + R325 (390 ohm resistor)
C = 2.2e-5; // C334 (22uF capacitor)
fc = 1.0 / (2.0 * M_PI * R * C);
calcRCFilterCoeffs(audio.outputRate, fc, &filterHi);
}
void mixerCalcVoicePans(uint8_t stereoSeparation)
{
const 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 = config.soundFrequency;
want.samples = (uint16_t)config.soundBufferSize;
want.format = AUDIO_S16;
want.channels = 2;
want.callback = audioCallback;
want.userdata = NULL;
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 the BLEP synthesis in the mixer
{
showErrorMsgBox("Unable to open audio: An audio rate below 32kHz can'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;
}
audio.outputRate = have.freq;
audio.audioBufferSize = have.samples;
audio.dPeriodToDeltaDiv = (double)PAULA_PAL_CLK / audio.outputRate;
generateBpmTables();
/* If the audio output rate is lower than MOD2WAV_FREQ, we need
** to allocate slightly more space so that MOD2WAV rendering
** won't overflow.
*/
uint32_t maxSamplesToMix;
if (MOD2WAV_FREQ > audio.outputRate)
maxSamplesToMix = audio.bpmTabMod2Wav[32-32]; // BPM 32
else
maxSamplesToMix = audio.bpmTab[32-32]; // BPM 32
dMixBufferLUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);
dMixBufferRUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double), 256);
if (dMixBufferLUnaligned == NULL || dMixBufferRUnaligned == NULL)
{
showErrorMsgBox("Out of memory!");
return false;
}
dMixBufferL = (double *)ALIGN_PTR(dMixBufferLUnaligned, 256);
dMixBufferR = (double *)ALIGN_PTR(dMixBufferRUnaligned, 256);
mixerCalcVoicePans(config.stereoSeparation);
defStereoSep = config.stereoSeparation;
filterFlags = config.a500LowPassFilter ? FILTER_A500 : 0;
calculateFilterCoeffs();
samplesPerTick = audio.bpmTab[125-32]; // BPM 125
sampleCounter = 0;
calcAudioLatencyVars(audio.audioBufferSize, audio.outputRate);
for (int32_t i = 0; i < 256-32; i++)
{
// number of samples per tick -> tick length for performance counter (syncing visuals to audio)
const double dTickTimeLenMul = (editor.dPerfFreq / audio.outputRate) * (UINT32_MAX + 1.0);
audio.tickTimeLengthTab[i] = (uint64_t)(audio.bpmTab[i] * dTickTimeLenMul);
}
audio.resetSyncTickTimeFlag = true;
SDL_PauseAudioDevice(dev, false);
return true;
}
void audioClose(void)
{
if (dev > 0)
{
SDL_PauseAudioDevice(dev, true);
SDL_CloseAudioDevice(dev);
dev = 0;
}
if (dMixBufferLUnaligned != NULL)
{
free(dMixBufferLUnaligned);
dMixBufferLUnaligned = NULL;
}
if (dMixBufferRUnaligned != NULL)
{
free(dMixBufferRUnaligned);
dMixBufferRUnaligned = NULL;
}
}
void mixerSetSamplesPerTick(uint32_t val)
{
samplesPerTick = val;
}
void mixerClearSampleCounter(void)
{
sampleCounter = 0;
}
void toggleAmigaPanMode(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
amigaPanFlag ^= 1;
if (!amigaPanFlag)
{
mixerCalcVoicePans(defStereoSep);
displayMsg("AMIGA PANNING OFF");
}
else
{
mixerCalcVoicePans(100);
displayMsg("AMIGA PANNING ON");
}
if (audioWasntLocked)
unlockAudio();
}
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;
}