ref: 0699e2c43053653db63616e42ae2080ffd2dc73c
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()
#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_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"
#include "pt2_rcfilter.h"
#include "pt2_ledfilter.h"
#include "pt2_downsamplers2x.h"
#define INITIAL_DITHER_SEED 0x12345000
static volatile bool ledFilterEnabled;
static volatile uint8_t filterModel;
static int8_t defStereoSep;
static bool amigaPanFlag;
static int32_t oldPeriod = -1, randSeed = INITIAL_DITHER_SEED;
static uint32_t audLatencyPerfValInt, audLatencyPerfValFrac;
static uint64_t tickTime64, tickTime64Frac;
static double *dMixBufferL, *dMixBufferR, *dMixBufferLUnaligned, *dMixBufferRUnaligned, dOldVoiceDelta, dOldVoiceDeltaMul;
static double dPrngStateL, dPrngStateR, dLState[2], dRState[2];
static blep_t blep[AMIGA_VOICES], blepVol[AMIGA_VOICES];
static rcFilter_t filterLoA500, filterHiA500, filterHiA1200;
static ledFilter_t filterLED;
static SDL_AudioDeviceID dev;
// for audio/video syncing
static uint32_t tickTimeLen, tickTimeLenFrac;
// globalized
audio_t audio;
paulaVoice_t paula[AMIGA_VOICES];
bool intMusic(void); // defined in pt_modplayer.c
void setLEDFilter(bool state, bool doLockAudio)
{
const bool audioWasntLocked = !audio.locked;
if (doLockAudio && audioWasntLocked)
lockAudio();
clearLEDFilterState(&filterLED);
editor.useLEDFilter = state;
ledFilterEnabled = editor.useLEDFilter;
if (doLockAudio && audioWasntLocked)
unlockAudio();
}
void toggleLEDFilter(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
clearLEDFilterState(&filterLED);
editor.useLEDFilter ^= 1;
ledFilterEnabled = editor.useLEDFilter;
if (audioWasntLocked)
unlockAudio();
}
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+1.0;
audLatencyPerfValFrac = (uint32_t)dFrac;
}
void setSyncTickTimeLen(uint32_t timeLen, uint32_t timeLenFrac)
{
tickTimeLen = timeLen;
tickTimeLenFrac = timeLenFrac;
}
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);
}
}
}
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));
memset(&blep[ch], 0, sizeof (blep_t));
memset(&blepVol[ch], 0, sizeof (blep_t));
stopScope(ch); // it should be safe to clear the scope now
memset(&scope[ch], 0, sizeof (scope_t));
// restore old pans
paula[ch].dPanL = dOldPanL;
paula[ch].dPanR = dOldPanR;
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(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
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)
{
double dPeriodToDeltaDiv;
paulaVoice_t *v = &paula[ch];
int32_t realPeriod = period;
if (realPeriod == 0)
realPeriod = 1+65535; // confirmed behavior on real Amiga
else if (realPeriod < 113)
realPeriod = 113; // close to what happens on real Amiga (and needed for BLEP synthesis)
if (editor.songPlaying)
{
v->syncPeriod = realPeriod;
v->syncFlags |= SET_SCOPE_PERIOD;
}
else
{
scopeSetPeriod(ch, realPeriod);
}
// 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 = &paula[ch];
int32_t realVol = vol;
// confirmed behavior on real Amiga
realVol &= 127;
if (realVol > 64)
realVol = 64;
v->dVolume = realVol * (1.0 / 64.0);
if (editor.songPlaying)
{
v->syncVolume = (uint8_t)realVol;
v->syncFlags |= SET_SCOPE_VOLUME;
}
else
{
scope[ch].volume = (uint8_t)realVol;
}
}
void paulaSetLength(int32_t ch, uint16_t len)
{
int32_t realLength = len;
if (realLength == 0)
{
realLength = 1+65535;
/* Confirmed behavior on real Amiga. We have room for this
** even at the last sample slot, so it will never overflow!
**
** PS: I don't really know if it's possible for ProTracker to
** set a Paula length of 0, but I fully support this Paula
** behavior just in case.
*/
}
realLength <<= 1; // we work with bytes, not words
paula[ch].newLength = realLength;
if (editor.songPlaying)
paula[ch].syncFlags |= SET_SCOPE_LENGTH;
else
scope[ch].newLength = realLength;
}
void paulaSetData(int32_t ch, const int8_t *src)
{
if (src == NULL)
src = &song->sampleData[RESERVED_SAMPLE_OFFSET]; // 128K reserved sample
paula[ch].newData = src;
if (editor.songPlaying)
paula[ch].syncFlags |= SET_SCOPE_DATA;
else
scope[ch].newData = src;
}
void paulaStopDMA(int32_t ch)
{
paula[ch].active = false;
if (editor.songPlaying)
paula[ch].syncFlags |= STOP_SCOPE;
else
scope[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]; // 128K reserved sample
length = v->newLength; // in bytes, not words
if (length < 2)
length = 2; // for safety
v->dPhase = 0.0;
v->pos = 0;
v->data = dat;
v->length = length;
v->active = true;
if (editor.songPlaying)
{
v->syncTriggerData = dat;
v->syncTriggerLength = length;
v->syncFlags |= TRIGGER_SCOPE;
}
else
{
scope[ch].newData = dat;
scope[ch].newLength = length;
scopeTrigger(ch);
}
}
void toggleFilterModel(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
clearRCFilterState(&filterLoA500);
clearRCFilterState(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
filterModel ^= 1;
if (filterModel == FILTERMODEL_A500)
displayMsg("AUDIO: AMIGA 500");
else
displayMsg("AUDIO: AMIGA 1200");
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 resetAudioDithering(void)
{
randSeed = INITIAL_DITHER_SEED;
dPrngStateL = 0.0;
dPrngStateR = 0.0;
}
void resetAudioDownsamplingStates(void)
{
dLState[0] = dLState[1] = 0.0;
dRState[0] = dRState[1] = 0.0;
}
static inline int32_t random32(void)
{
// LCG random 32-bit generator (quite good and fast)
randSeed *= 134775813;
randSeed++;
return randSeed;
}
static void processMixedSamples(int32_t i, int16_t *out)
{
int32_t smp32;
double dPrng, dOut[2], dMixL[2], dMixR[2];
// we run the filters at 2x the audio output rate for more precision
for (int32_t j = 0; j < 2; j++)
{
// zero-padding (yes, this makes sense)
dOut[0] = (j == 0) ? dMixBufferL[i] : 0.0;
dOut[1] = (j == 0) ? dMixBufferR[i] : 0.0;
if (filterModel == FILTERMODEL_A500)
{
// A500 low-pass RC filter
RCLowPassFilterStereo(&filterLoA500, dOut, dOut);
// "LED" Sallen-Key filter
if (ledFilterEnabled)
LEDFilter(&filterLED, dOut, dOut);
// A500 high-pass RC filter
RCHighPassFilterStereo(&filterHiA500, dOut, dOut);
}
else
{
// A1200 low-pass filter is ignored (we don't want it)
// "LED" Sallen-Key filter
if (ledFilterEnabled)
LEDFilter(&filterLED, dOut, dOut);
// A1200 high-pass RC filter
RCHighPassFilterStereo(&filterHiA1200, dOut, dOut);
}
dMixL[j] = dOut[0];
dMixR[j] = dOut[1];
}
#define NORMALIZE_DOWNSAMPLE 2.0
// 2x "all-pass halfband" downsampling
dOut[0] = d2x(dMixL, dLState);
dOut[1] = d2x(dMixR, dRState);
// normalize and invert phase (A500/A1200 has a phase-inverted audio signal)
dOut[0] *= NORMALIZE_DOWNSAMPLE * (-INT16_MAX / (double)AMIGA_VOICES);
dOut[1] *= NORMALIZE_DOWNSAMPLE * (-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;
}
void outputAudio(int16_t *target, int32_t numSamples)
{
int16_t out[2];
int32_t i;
if (editor.isSMPRendering)
{
// render to sample (PAT2SMP)
int32_t samplesTodo = numSamples;
if (editor.pat2SmpPos+samplesTodo > MAX_SAMPLE_LEN*2)
samplesTodo = (MAX_SAMPLE_LEN*2)-editor.pat2SmpPos;
mixChannels(samplesTodo);
double *dOutStream = &editor.dPat2SmpBuf[editor.pat2SmpPos];
for (i = 0; i < samplesTodo; i++)
dOutStream[i] = dMixBufferL[i] + dMixBufferR[i]; // normalized to -128..127 later
editor.pat2SmpPos += samplesTodo;
if (editor.pat2SmpPos >= MAX_SAMPLE_LEN*2)
{
editor.smpRenderingDone = true;
updateWindowTitle(MOD_IS_MODIFIED);
}
}
else
{
// render to stream
mixChannels(numSamples);
int16_t *outStream = target;
for (i = 0; i < numSamples; i++)
{
processMixedSamples(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 = v->newLength;
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)
{
if (audio.forceMixerOff) // during MOD2WAV
{
memset(stream, 0, len);
return;
}
int16_t *streamOut = (int16_t *)stream;
int32_t samplesLeft = len >> 2;
while (samplesLeft > 0)
{
if (audio.tickSampleCounter64 <= 0)
{
// new replayer tick
if (editor.songPlaying)
{
intMusic();
fillVisualsSyncBuffer();
}
audio.tickSampleCounter64 += audio.samplesPerTick64;
}
const int32_t remainingTick = (audio.tickSampleCounter64 + UINT32_MAX) >> 32; // ceil rounding (upwards)
int32_t samplesToMix = samplesLeft;
if (samplesToMix > remainingTick)
samplesToMix = remainingTick;
outputAudio(streamOut, samplesToMix);
streamOut += samplesToMix<<1;
samplesLeft -= samplesToMix;
audio.tickSampleCounter64 -= (int64_t)samplesToMix << 32;
}
(void)userdata;
}
static void calculateFilterCoeffs(void)
{
/* Amiga 500/1200 filter emulation
**
** 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)
*/
// we run the filters at twice the frequency for improved precision (zero-padding)
const uint32_t audioFreq = audio.outputRate * 2;
double R, C, R1, R2, C1, C2, fc, fb;
const double pi = 4.0 * atan(1.0); // M_PI can not be trusted
/*
** 8bitbubsy:
** Hackish low-pass cutoff compensation to better match Amiga 500 when
** we use "lower" audio output rates. This has been loosely hand-picked
** after looking at many frequency analyses on a sine-sweep test module
** rendered on 7 different Amiga 500 machines (and taking the average).
** Don't try to make sense of this magic constant, and it should only be
** used within this very specific application!
**
** The reason we want this bias is because our digital RC filter is not
** that precise at lower audio output rates. It would otherwise lead to a
** slight unwanted cut of treble near the cutoff we aim for. It was easily
** audible, and especially visible on a plotted frequency spectrum.
**
** 1100Hz is the magic value I found that seems to be good. Higher than that
** would allow too much treble to pass.
**
** Scaling it like this is 'acceptable' (confirmed with further frequency analyses
** at output rates of 48, 96 and 192).
*/
double dLPCutoffBias = 1100.0 * (44100.0 / audio.outputRate);
// A500 1-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 * pi * R * C)) + dLPCutoffBias;
calcRCFilterCoeffs(audioFreq, fc, &filterLoA500);
/*
** 8bitbubsy:
** We don't handle Amiga 1200's ~34kHz low-pass filter as it's not really
** needed. The reason it was still present in the A1200 (despite its high
** non-audible cutoff) was to filter away high-frequency noise from Paula's
** PWM (volume modulation). We don't do PWM for volume in the PT2 clone.
*/
// Sallen-Key filter ("LED" filter, same RC values on A500 and A1200):
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 * pi * sqrt(R1 * R2 * C1 * C2));
fb = 0.125; // Fb = 0.125 : Q ~= 1/sqrt(2)
calcLEDFilterCoeffs(audioFreq, fc, fb, &filterLED);
// A500 1-pole (6dB/oct) static RC high-pass filter:
R = 1390.0; // R324 (1K ohm resistor) + R325 (390 ohm resistor)
C = 2.233e-5; // C334 (22uF capacitor) + C335 (0.33�F capacitor)
fc = 1.0 / (2.0 * pi * R * C);
calcRCFilterCoeffs(audioFreq, fc, &filterHiA500);
// A1200 1-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 * pi * R * C);
calcRCFilterCoeffs(audioFreq, fc, &filterHiA1200);
}
void recalcFilterCoeffs(int32_t outputRate) // for MOD2WAV
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
const int32_t oldOutputRate = audio.outputRate;
audio.outputRate = outputRate;
clearRCFilterState(&filterLoA500);
clearRCFilterState(&filterHiA500);
clearRCFilterState(&filterHiA1200);
clearLEDFilterState(&filterLED);
calculateFilterCoeffs();
audio.outputRate = oldOutputRate;
if (audioWasntLocked)
unlockAudio();
}
static void setVoicePan(int32_t ch, double pan) // pan = 0.0 .. 1.0
{
// constant power panning
const double pi = 4.0 * atan(1.0); // M_PI can not be trusted
paula[ch].dPanL = cos(pan * pi * 0.5) * sqrt(2.0);
paula[ch].dPanR = sin(pan * pi * 0.5) * sqrt(2.0);
}
void mixerCalcVoicePans(uint8_t stereoSeparation) // 0..100 (percentage)
{
assert(stereoSeparation <= 100);
const double panMid = 0.5;
const double panR = panMid + (stereoSeparation / (100.0 * 2.0));
const double panL = 1.0 - panR;
setVoicePan(0, panL);
setVoicePan(1, panR);
setVoicePan(2, panR);
setVoicePan(3, panL);
}
static double ciaBpm2Hz(int32_t bpm)
{
if (bpm == 0)
return 0.0;
const uint32_t ciaPeriod = 1773447 / bpm; // yes, PT truncates here
return (double)CIA_PAL_CLK / ciaPeriod;
}
static void generateBpmTables(bool vblankTimingFlag)
{
for (int32_t bpm = 32; bpm <= 255; bpm++)
{
double dBpmHz;
if (vblankTimingFlag)
dBpmHz = AMIGA_PAL_VBLANK_HZ;
else
dBpmHz = ciaBpm2Hz(bpm);
const double dSamplesPerTick = audio.outputRate / dBpmHz;
const double dSamplesPerTick28kHz = PAT2SMP_HI_FREQ / dBpmHz; // PAT2SMP hi quality
const double dSamplesPerTick22kHz = PAT2SMP_LO_FREQ / dBpmHz; // PAT2SMP low quality
const double dSamplesPerTickMod2Wav = MOD2WAV_FREQ / dBpmHz; // MOD2WAV
// convert to rounded 32.32 fixed-point
const int32_t i = bpm-32;
audio.bpmTable[i] = (int64_t)((dSamplesPerTick * (UINT32_MAX+1.0)) + 0.5);
audio.bpmTable28kHz[i] = (int64_t)((dSamplesPerTick28kHz * (UINT32_MAX+1.0)) + 0.5);
audio.bpmTable22kHz[i] = (int64_t)((dSamplesPerTick22kHz * (UINT32_MAX+1.0)) + 0.5);
audio.bpmTableMod2Wav[i] = (int64_t)((dSamplesPerTickMod2Wav * (UINT32_MAX+1.0)) + 0.5);
}
}
static void generateTickLengthTable(bool vblankTimingFlag)
{
for (int32_t bpm = 32; bpm <= 255; bpm++)
{
double dHz;
if (vblankTimingFlag)
dHz = AMIGA_PAL_VBLANK_HZ;
else
dHz = ciaBpm2Hz(bpm);
// BPM -> Hz -> tick length for performance counter (syncing visuals to audio)
double dTimeInt;
double dTimeFrac = modf(editor.dPerfFreq / dHz, &dTimeInt);
const int32_t timeInt = (int32_t)dTimeInt;
dTimeFrac = floor((UINT32_MAX+1.0) * dTimeFrac); // fractional part (scaled to 0..2^32-1)
audio.tickLengthTable[bpm-32] = ((uint64_t)timeInt << 32) | (uint32_t)dTimeFrac;
}
}
void updateReplayerTimingMode(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
const bool vblankTimingMode = (editor.timingMode == TEMPO_MODE_VBLANK);
generateBpmTables(vblankTimingMode);
generateTickLengthTable(vblankTimingMode);
if (audioWasntLocked)
unlockAudio();
}
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;
updateReplayerTimingMode();
const int32_t lowestBPM = 32;
const int32_t pat2SmpMaxSamples = (audio.bpmTable22kHz[lowestBPM-32] + (1LL + 31)) >> 32; // ceil (rounded upwards)
const int32_t mod2WavMaxSamples = (audio.bpmTableMod2Wav[lowestBPM-32] + (1LL + 31)) >> 32; // ceil (rounded upwards)
const int32_t renderMaxSamples = (audio.bpmTable[lowestBPM-32] + (1LL + 31)) >> 32; // ceil (rounded upwards)
const int32_t maxSamplesToMix = MAX(pat2SmpMaxSamples, MAX(mod2WavMaxSamples, renderMaxSamples));
dMixBufferLUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double) * 8, 256);
dMixBufferRUnaligned = (double *)MALLOC_PAD(maxSamplesToMix * sizeof (double) * 8, 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;
filterModel = config.filterModel;
ledFilterEnabled = false;
calculateFilterCoeffs();
audio.samplesPerTick64 = audio.bpmTable[125-32]; // BPM 125
audio.tickSampleCounter64 = 0; // zero tick sample counter so that it will instantly initiate a tick
calcAudioLatencyVars(audio.audioBufferSize, audio.outputRate);
resetAudioDownsamplingStates();
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 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();
}
uint16_t get16BitPeak(int16_t *sampleData, uint32_t sampleLength)
{
uint16_t samplePeak = 0;
for (uint32_t i = 0; i < sampleLength; i++)
{
uint16_t sample = ABS(sampleData[i]);
if (samplePeak < sample)
samplePeak = sample;
}
return samplePeak;
}
uint32_t get32BitPeak(int32_t *sampleData, uint32_t sampleLength)
{
uint32_t samplePeak = 0;
for (uint32_t i = 0; i < sampleLength; i++)
{
uint32_t sample = ABS(sampleData[i]);
if (samplePeak < sample)
samplePeak = sample;
}
return samplePeak;
}
float getFloatPeak(float *fSampleData, uint32_t sampleLength)
{
float fSamplePeak = 0.0f;
for (uint32_t i = 0; i < sampleLength; i++)
{
const float fSample = fabsf(fSampleData[i]);
if (fSamplePeak < fSample)
fSamplePeak = fSample;
}
return fSamplePeak;
}
double getDoublePeak(double *dSampleData, uint32_t sampleLength)
{
double dSamplePeak = 0.0;
for (uint32_t i = 0; i < sampleLength; i++)
{
const double dSample = fabs(dSampleData[i]);
if (dSamplePeak < dSample)
dSamplePeak = dSample;
}
return dSamplePeak;
}
void normalize16BitTo8Bit(int16_t *sampleData, uint32_t sampleLength)
{
const uint16_t samplePeak = get16BitPeak(sampleData, sampleLength);
if (samplePeak == 0 || samplePeak >= INT16_MAX)
return;
const double dGain = (double)INT16_MAX / samplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
{
const int32_t sample = (const int32_t)(sampleData[i] * dGain);
sampleData[i] = (int16_t)sample;
}
}
void normalize32BitTo8Bit(int32_t *sampleData, uint32_t sampleLength)
{
const uint32_t samplePeak = get32BitPeak(sampleData, sampleLength);
if (samplePeak == 0 || samplePeak >= INT32_MAX)
return;
const double dGain = (double)INT32_MAX / samplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
{
const int32_t sample = (const int32_t)(sampleData[i] * dGain);
sampleData[i] = (int32_t)sample;
}
}
void normalizeFloatTo8Bit(float *fSampleData, uint32_t sampleLength)
{
const float fSamplePeak = getFloatPeak(fSampleData, sampleLength);
if (fSamplePeak <= 0.0f)
return;
const float fGain = INT8_MAX / fSamplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
fSampleData[i] *= fGain;
}
void normalizeDoubleTo8Bit(double *dSampleData, uint32_t sampleLength)
{
const double dSamplePeak = getDoublePeak(dSampleData, sampleLength);
if (dSamplePeak <= 0.0)
return;
const double dGain = INT8_MAX / dSamplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
dSampleData[i] *= dGain;
}