ref: 8457c7dcbb4115460ae4d882982763fbeb80595c
dir: /src/ft2_audio.c/
// for finding memory leaks in debug mode with Visual Studio
#if defined _DEBUG && defined _MSC_VER
#include <crtdbg.h>
#endif
#include <stdio.h>
#include <stdint.h>
#include "ft2_header.h"
#include "ft2_config.h"
#include "ft2_scopes.h"
#include "ft2_video.h"
#include "ft2_gui.h"
#include "ft2_midi.h"
#include "ft2_wav_renderer.h"
#include "ft2_tables.h"
#include "ft2_structs.h"
// --------------------------------
#include "mixer/ft2_mix.h"
#include "mixer/ft2_center_mix.h"
#include "mixer/ft2_silence_mix.h"
// --------------------------------
#define INITIAL_DITHER_SEED 0x12345000
static int8_t pmpCountDiv, pmpChannels = 2;
static uint16_t smpBuffSize;
static int32_t randSeed = INITIAL_DITHER_SEED;
static uint32_t oldAudioFreq, tickTimeLen, tickTimeLenFrac;
static double dAudioNormalizeMul, dPrngStateL, dPrngStateR, dPanningTab[256+1];
static voice_t voice[MAX_VOICES * 2];
static void (*sendAudSamplesFunc)(uint8_t *, uint32_t, uint8_t); // "send mixed samples" routines
// globalized
audio_t audio;
pattSyncData_t *pattSyncEntry;
chSyncData_t *chSyncEntry;
chSync_t chSync;
pattSync_t pattSync;
volatile bool pattQueueClearing, chQueueClearing;
void resetCachedMixerVars(void)
{
stmTyp *ch = stm;
for (int32_t i = 0; i < MAX_VOICES; i++, ch++)
ch->oldFinalPeriod = -1;
voice_t *v = voice;
for (int32_t i = 0; i < MAX_VOICES*2; i++, v++)
{
v->oldDelta = 0;
v->oldRevDelta = UINT32_MAX;
}
}
void stopVoice(int32_t i)
{
voice_t *v;
v = &voice[i];
memset(v, 0, sizeof (voice_t));
v->pan = 128;
// clear "fade out" voice too
v = &voice[MAX_VOICES + i];
memset(v, 0, sizeof (voice_t));
v->pan = 128;
}
bool setNewAudioSettings(void) // only call this from the main input/video thread
{
uint32_t stringLen;
pauseAudio();
if (!setupAudio(CONFIG_HIDE_ERRORS))
{
// set back old known working settings
config.audioFreq = audio.lastWorkingAudioFreq;
config.specialFlags &= ~(BITDEPTH_16 + BITDEPTH_32 + BUFFSIZE_512 + BUFFSIZE_1024 + BUFFSIZE_2048);
config.specialFlags |= audio.lastWorkingAudioBits;
if (audio.lastWorkingAudioDeviceName != NULL)
{
if (audio.currOutputDevice != NULL)
{
free(audio.currOutputDevice);
audio.currOutputDevice = NULL;
}
stringLen = (uint32_t)strlen(audio.lastWorkingAudioDeviceName);
audio.currOutputDevice = (char *)malloc(stringLen + 2);
if (audio.currOutputDevice != NULL)
{
strcpy(audio.currOutputDevice, audio.lastWorkingAudioDeviceName);
audio.currOutputDevice[stringLen + 1] = '\0'; // UTF-8 needs double null termination
}
}
// also update config audio radio buttons if we're on that screen at the moment
if (ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)
setConfigIORadioButtonStates();
// if it didn't work to use the old settings again, then something is seriously wrong...
if (!setupAudio(CONFIG_HIDE_ERRORS))
okBox(0, "System message", "Couldn't find a working audio mode... You'll get no sound / replayer timer!");
resumeAudio();
return false;
}
calcRevMixDeltaTable();
resumeAudio();
setWavRenderFrequency(audio.freq);
setWavRenderBitDepth((config.specialFlags & BITDEPTH_32) ? 32 : 16);
return true;
}
// amp = 1..32, masterVol = 0..256
void setAudioAmp(int16_t amp, int16_t masterVol, bool bitDepth32Flag)
{
amp = CLAMP(amp, 1, 32);
masterVol = CLAMP(masterVol, 0, 256);
double dAmp = (amp * masterVol) / (32.0 * 256.0);
if (!bitDepth32Flag)
dAmp *= 32768.0;
dAudioNormalizeMul = dAmp;
}
void setNewAudioFreq(uint32_t freq) // for song-to-WAV rendering
{
if (freq == 0)
return;
oldAudioFreq = audio.freq;
audio.freq = freq;
const bool mustRecalcTables = audio.freq != oldAudioFreq;
if (mustRecalcTables)
{
calcReplayRate(audio.freq);
calcRevMixDeltaTable();
}
}
void setBackOldAudioFreq(void) // for song-to-WAV rendering
{
const bool mustRecalcTables = audio.freq != oldAudioFreq;
audio.freq = oldAudioFreq;
if (mustRecalcTables)
{
calcReplayRate(audio.freq);
calcRevMixDeltaTable();
}
}
void P_SetSpeed(uint16_t bpm)
{
if (bpm == 0)
return;
// non-FT2 check for security
if (bpm > MAX_BPM)
return;
audio.dSamplesPerTick = audio.dSamplesPerTickTab[bpm];
audio.samplesPerTick = (int32_t)(audio.dSamplesPerTick + 0.5);
// get tick time length for audio/video sync timestamp
const uint64_t tickTimeLen64 = audio.tickTimeTab[bpm];
tickTimeLen = tickTimeLen64 >> 32;
tickTimeLenFrac = tickTimeLen64 & UINT32_MAX;
// used for calculating volume ramp length for "ticks" ramps
audio.dRampTickMul = audio.dRampTickMulTab[bpm];
}
void audioSetVolRamp(bool volRamp)
{
lockMixerCallback();
audio.volumeRampingFlag = volRamp;
unlockMixerCallback();
}
void audioSetInterpolationType(uint8_t interpolationType)
{
lockMixerCallback();
audio.interpolationType = interpolationType;
unlockMixerCallback();
}
void calcPanningTable(void)
{
// same formula as FT2's panning table (with 0.0f..1.0f range)
for (int32_t i = 0; i <= 256; i++)
dPanningTab[i] = sqrt(i / 256.0);
}
static void voiceUpdateVolumes(int32_t i, uint8_t status)
{
double dDestVolL, dDestVolR;
voice_t *v = &voice[i];
const double dVolL = v->dVol * dPanningTab[256-v->pan];
const double dVolR = v->dVol * dPanningTab[ v->pan];
if (!audio.volumeRampingFlag)
{
// volume ramping is disabled
v->dVolL = dVolL;
v->dVolR = dVolR;
v->volRampSamples = 0;
return;
}
v->dDestVolL = dVolL;
v->dDestVolR = dVolR;
if (status & IS_NyTon)
{
// sample is about to start, ramp out/in at the same time
// setup "fade out" voice (only if current voice volume > 0)
if (v->dVolL > 0.0 || v->dVolR > 0.0)
{
voice_t *f = &voice[MAX_VOICES+i];
*f = *v; // copy voice
f->volRampSamples = audio.quickVolRampSamples;
dDestVolL = -f->dVolL;
dDestVolR = -f->dVolR;
f->dVolDeltaL = dDestVolL * audio.dRampQuickVolMul;
f->dVolDeltaR = dDestVolR * audio.dRampQuickVolMul;
f->isFadeOutVoice = true;
}
// make current voice fade in from zero when it starts
v->dVolL = 0.0;
v->dVolR = 0.0;
}
// ramp volume changes
/* FT2 has two internal volume ramping lengths:
** IS_QuickVol: 5ms
** Normal: The duration of a tick (samplesPerTick)
*/
// if destination volume and current volume is the same (and we have no sample trigger), don't do ramp
if (dVolL == v->dVolL && dVolR == v->dVolR && !(status & IS_NyTon))
{
// there is no volume change
v->volRampSamples = 0;
}
else
{
dDestVolL = dVolL - v->dVolL;
dDestVolR = dVolR - v->dVolR;
if (status & IS_QuickVol)
{
v->volRampSamples = audio.quickVolRampSamples;
v->dVolDeltaL = dDestVolL * audio.dRampQuickVolMul;
v->dVolDeltaR = dDestVolR * audio.dRampQuickVolMul;
}
else
{
v->volRampSamples = audio.samplesPerTick;
v->dVolDeltaL = dDestVolL * audio.dRampTickMul;
v->dVolDeltaR = dDestVolR * audio.dRampTickMul;
}
}
}
static void voiceTrigger(int32_t ch, sampleTyp *s, int32_t position)
{
bool sampleIs16Bit;
uint8_t loopType;
int32_t length, loopStart, loopLength, loopEnd;
voice_t *v = &voice[ch];
length = s->len;
loopStart = s->repS;
loopLength = s->repL;
loopEnd = s->repS + s->repL;
loopType = s->typ & 3;
sampleIs16Bit = (s->typ >> 4) & 1;
if (sampleIs16Bit)
{
assert(!(length & 1));
assert(!(loopStart & 1));
assert(!(loopLength & 1));
assert(!(loopEnd & 1));
length >>= 1;
loopStart >>= 1;
loopLength >>= 1;
loopEnd >>= 1;
}
if (s->pek == NULL || length < 1)
{
v->active = false; // shut down voice (illegal parameters)
return;
}
if (loopLength < 1) // disable loop if loopLength is below 1
loopType = 0;
if (sampleIs16Bit)
{
v->base16 = (const int16_t *)s->pek;
v->revBase16 = &v->base16[loopStart + loopEnd]; // for pingpong loops
v->leftEdgeTaps16 = s->leftEdgeTapSamples16 + 3;
}
else
{
v->base8 = s->pek;
v->revBase8 = &v->base8[loopStart + loopEnd]; // for pingpong loops
v->leftEdgeTaps8 = s->leftEdgeTapSamples8 + 3;
}
v->hasLooped = false; // for sinc interpolation special case
v->backwards = false;
v->loopType = loopType;
v->end = (loopType > 0) ? loopEnd : length;
v->loopStart = loopStart;
v->loopLength = loopLength;
v->pos = position;
v->posFrac = 0;
// if position overflows, shut down voice (f.ex. through 9xx command)
if (v->pos >= v->end)
{
v->active = false;
return;
}
v->mixFuncOffset = (sampleIs16Bit * 9) + (audio.interpolationType * 3) + loopType;
v->active = true;
}
void resetRampVolumes(void)
{
voice_t *v = voice;
for (int32_t i = 0; i < song.antChn; i++, v++)
{
v->dVolL = v->dDestVolL;
v->dVolR = v->dDestVolR;
v->volRampSamples = 0;
}
}
void updateVoices(void)
{
uint8_t status;
stmTyp *ch;
voice_t *v;
ch = stm;
v = voice;
for (int32_t i = 0; i < song.antChn; i++, ch++, v++)
{
status = ch->tmpStatus = ch->status; // (tmpStatus is used for audio/video sync queue)
if (status == 0) continue; // nothing to do
ch->status = 0;
if (status & IS_Vol)
v->dVol = ch->dFinalVol;
if (status & IS_Pan)
v->pan = ch->finalPan;
if (status & (IS_Vol + IS_Pan))
voiceUpdateVolumes(i, status);
if (status & IS_Period)
{
// use cached values if possible
if (ch->finalPeriod != ch->oldFinalPeriod)
{
ch->oldFinalPeriod = ch->finalPeriod;
if (ch->finalPeriod == 0) // in FT2, period 0 -> delta 0
{
v->oldDelta = 0;
v->oldRevDelta = UINT32_MAX;
v->dSincLUT = gKaiserSinc;
}
else
{
v->oldDelta = getMixerDelta(ch->finalPeriod);
v->oldRevDelta = getRevMixerDelta(ch->finalPeriod); // for calc'ing max samples in inner mix loop before sample end/loop end
// decide which sinc LUT to use according to resampling ratio
if (v->oldDelta <= 0x130000000) // 1.1875 (32.32 fixed-point)
v->dSincLUT = gKaiserSinc;
else if (v->oldDelta <= 0x180000000) // 1.5 (32.32 fixed-point)
v->dSincLUT = gDownSample1;
else
v->dSincLUT = gDownSample2;
}
}
v->delta = v->oldDelta;
v->revDelta = v->oldRevDelta;
}
if (status & IS_NyTon)
voiceTrigger(i, ch->smpPtr, ch->smpStartPos);
}
}
void resetAudioDither(void)
{
randSeed = INITIAL_DITHER_SEED;
dPrngStateL = 0.0;
dPrngStateR = 0.0;
}
static inline int32_t random32(void)
{
// LCG 32-bit random
randSeed *= 134775813;
randSeed++;
return (int32_t)randSeed;
}
static void sendSamples16BitDitherStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int32_t out32;
double dOut, dPrng;
int16_t *streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel - 1-bit triangular dithering
dPrng = random32() * (0.5 / INT32_MAX); // -0.5 .. 0.5
dOut = ((audio.dMixBufferL[i] * dAudioNormalizeMul) + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel - 1-bit triangular dithering
dPrng = random32() * (0.5 / INT32_MAX); // -0.5 .. 0.5
dOut = ((audio.dMixBufferR[i] * dAudioNormalizeMul) + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
}
(void)numAudioChannels;
}
static void sendSamples16BitDitherMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int32_t out32;
double dOut, dPrng;
int16_t *streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel - 1-bit triangular dithering
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut = ((audio.dMixBufferL[i] * dAudioNormalizeMul) + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel - 1-bit triangular dithering
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dOut = ((audio.dMixBufferR[i] * dAudioNormalizeMul) + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// send zeroes to the rest of the channels
for (uint32_t j = 2; j < numAudioChannels; j++)
*streamPointer16++ = 0;
}
}
static void sendSamples32BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
double dOut;
float *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
dOut = audio.dMixBufferL[i] * dAudioNormalizeMul;
dOut = CLAMP(dOut, -1.0, 1.0);
*fStreamPointer32++ = (float)dOut;
// right channel
dOut = audio.dMixBufferR[i] * dAudioNormalizeMul;
dOut = CLAMP(dOut, -1.0, 1.0);
*fStreamPointer32++ = (float)dOut;
}
(void)numAudioChannels;
}
static void sendSamples32BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
double dOut;
float *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
dOut = audio.dMixBufferL[i] * dAudioNormalizeMul;
dOut = CLAMP(dOut, -1.0, 1.0);
*fStreamPointer32++ = (float)dOut;
// right channel
dOut = audio.dMixBufferR[i] * dAudioNormalizeMul;
dOut = CLAMP(dOut, -1.0, 1.0);
*fStreamPointer32++ = (float)dOut;
// send zeroes to the rest of the channels
for (uint32_t j = 2; j < numAudioChannels; j++)
*fStreamPointer32++ = 0.0f;
}
}
static void doChannelMixing(int32_t samplesToMix)
{
voice_t *v = voice; // normal voices
voice_t *r = &voice[MAX_VOICES]; // volume ramp fadeout-voices
for (int32_t i = 0; i < song.antChn; i++, v++, r++)
{
if (v->active)
{
bool centerMixFlag;
const bool volRampFlag = v->volRampSamples > 0;
if (volRampFlag)
{
centerMixFlag = (v->dDestVolL == v->dDestVolR) && (v->dVolDeltaL == v->dVolDeltaR);
}
else // no volume ramping active
{
if (v->dVolL == 0.0 && v->dVolR == 0.0)
{
silenceMixRoutine(v, samplesToMix);
continue;
}
centerMixFlag = v->dVolL == v->dVolR;
}
mixFuncTab[(centerMixFlag * 36) + (volRampFlag * 18) + v->mixFuncOffset](v, samplesToMix);
}
if (r->active) // volume ramp fadeout-voice
{
const bool centerMixFlag = (r->dDestVolL == r->dDestVolR) && (r->dVolDeltaL == r->dVolDeltaR);
mixFuncTab[(centerMixFlag * 36) + 18 + r->mixFuncOffset](r, samplesToMix);
}
}
}
static void mixAudio(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
assert(sampleBlockLength <= MAX_WAV_RENDER_SAMPLES_PER_TICK);
memset(audio.dMixBufferL, 0, sampleBlockLength * sizeof (double));
memset(audio.dMixBufferR, 0, sampleBlockLength * sizeof (double));
doChannelMixing(sampleBlockLength);
// normalize mix buffer and send to audio stream
sendAudSamplesFunc(stream, sampleBlockLength, numAudioChannels);
}
// used for song-to-WAV renderer
void mixReplayerTickToBuffer(uint32_t samplesToMix, uint8_t *stream, uint8_t bitDepth)
{
assert(samplesToMix <= MAX_WAV_RENDER_SAMPLES_PER_TICK);
memset(audio.dMixBufferL, 0, samplesToMix * sizeof (double));
memset(audio.dMixBufferR, 0, samplesToMix * sizeof (double));
doChannelMixing(samplesToMix);
// normalize mix buffer and send to audio stream
if (bitDepth == 16)
sendSamples16BitDitherStereo(stream, samplesToMix, 2);
else
sendSamples32BitStereo(stream, samplesToMix, 2);
}
int32_t pattQueueReadSize(void)
{
while (pattQueueClearing);
if (pattSync.writePos > pattSync.readPos)
return pattSync.writePos - pattSync.readPos;
else if (pattSync.writePos < pattSync.readPos)
return pattSync.writePos - pattSync.readPos + SYNC_QUEUE_LEN + 1;
else
return 0;
}
int32_t pattQueueWriteSize(void)
{
int32_t size;
if (pattSync.writePos > pattSync.readPos)
{
size = pattSync.readPos - pattSync.writePos + SYNC_QUEUE_LEN;
}
else if (pattSync.writePos < pattSync.readPos)
{
pattQueueClearing = true;
/* Buffer is full, reset the read/write pos. This is actually really nasty since
** read/write are two different threads, but because of timestamp validation it
** shouldn't be that dangerous.
** It will also create a small visual stutter while the buffer is getting filled,
** though that is barely noticable on normal buffer sizes, and it takes a minute
** or two at max BPM between each time (when queue size is default, 4095)
*/
pattSync.data[0].timestamp = 0;
pattSync.readPos = 0;
pattSync.writePos = 0;
size = SYNC_QUEUE_LEN;
pattQueueClearing = false;
}
else
{
size = SYNC_QUEUE_LEN;
}
return size;
}
bool pattQueuePush(pattSyncData_t t)
{
if (!pattQueueWriteSize())
return false;
assert(pattSync.writePos <= SYNC_QUEUE_LEN);
pattSync.data[pattSync.writePos] = t;
pattSync.writePos = (pattSync.writePos + 1) & SYNC_QUEUE_LEN;
return true;
}
bool pattQueuePop(void)
{
if (!pattQueueReadSize())
return false;
pattSync.readPos = (pattSync.readPos + 1) & SYNC_QUEUE_LEN;
assert(pattSync.readPos <= SYNC_QUEUE_LEN);
return true;
}
pattSyncData_t *pattQueuePeek(void)
{
if (!pattQueueReadSize())
return NULL;
assert(pattSync.readPos <= SYNC_QUEUE_LEN);
return &pattSync.data[pattSync.readPos];
}
uint64_t getPattQueueTimestamp(void)
{
if (!pattQueueReadSize())
return 0;
assert(pattSync.readPos <= SYNC_QUEUE_LEN);
return pattSync.data[pattSync.readPos].timestamp;
}
int32_t chQueueReadSize(void)
{
while (chQueueClearing);
if (chSync.writePos > chSync.readPos)
return chSync.writePos - chSync.readPos;
else if (chSync.writePos < chSync.readPos)
return chSync.writePos - chSync.readPos + SYNC_QUEUE_LEN + 1;
else
return 0;
}
int32_t chQueueWriteSize(void)
{
int32_t size;
if (chSync.writePos > chSync.readPos)
{
size = chSync.readPos - chSync.writePos + SYNC_QUEUE_LEN;
}
else if (chSync.writePos < chSync.readPos)
{
chQueueClearing = true;
/* Buffer is full, reset the read/write pos. This is actually really nasty since
** read/write are two different threads, but because of timestamp validation it
** shouldn't be that dangerous.
** It will also create a small visual stutter while the buffer is getting filled,
** though that is barely noticable on normal buffer sizes, and it takes several
** minutes between each time (when queue size is default, 16384)
*/
chSync.data[0].timestamp = 0;
chSync.readPos = 0;
chSync.writePos = 0;
size = SYNC_QUEUE_LEN;
chQueueClearing = false;
}
else
{
size = SYNC_QUEUE_LEN;
}
return size;
}
bool chQueuePush(chSyncData_t t)
{
if (!chQueueWriteSize())
return false;
assert(chSync.writePos <= SYNC_QUEUE_LEN);
chSync.data[chSync.writePos] = t;
chSync.writePos = (chSync.writePos + 1) & SYNC_QUEUE_LEN;
return true;
}
bool chQueuePop(void)
{
if (!chQueueReadSize())
return false;
chSync.readPos = (chSync.readPos + 1) & SYNC_QUEUE_LEN;
assert(chSync.readPos <= SYNC_QUEUE_LEN);
return true;
}
chSyncData_t *chQueuePeek(void)
{
if (!chQueueReadSize())
return NULL;
assert(chSync.readPos <= SYNC_QUEUE_LEN);
return &chSync.data[chSync.readPos];
}
uint64_t getChQueueTimestamp(void)
{
if (!chQueueReadSize())
return 0;
assert(chSync.readPos <= SYNC_QUEUE_LEN);
return chSync.data[chSync.readPos].timestamp;
}
void lockAudio(void)
{
if (audio.dev != 0)
SDL_LockAudioDevice(audio.dev);
audio.locked = true;
}
void unlockAudio(void)
{
if (audio.dev != 0)
SDL_UnlockAudioDevice(audio.dev);
audio.locked = false;
}
void resetSyncQueues(void)
{
pattSync.data[0].timestamp = 0;
pattSync.readPos = 0;
pattSync.writePos = 0;
chSync.data[0].timestamp = 0;
chSync.writePos = 0;
chSync.readPos = 0;
}
void lockMixerCallback(void) // lock audio + clear voices/scopes (for short operations)
{
if (!audio.locked)
lockAudio();
audio.resetSyncTickTimeFlag = true;
stopVoices(); // VERY important! prevents potential crashes by purging pointers
// scopes, mixer and replayer are guaranteed to not be active at this point
resetSyncQueues();
}
void unlockMixerCallback(void)
{
stopVoices(); // VERY important! prevents potential crashes by purging pointers
if (audio.locked)
unlockAudio();
}
void pauseAudio(void) // lock audio + clear voices/scopes + render silence (for long operations)
{
if (audioPaused)
{
stopVoices(); // VERY important! prevents potential crashes by purging pointers
return;
}
if (audio.dev > 0)
SDL_PauseAudioDevice(audio.dev, true);
audio.resetSyncTickTimeFlag = true;
stopVoices(); // VERY important! prevents potential crashes by purging pointers
// scopes, mixer and replayer are guaranteed to not be active at this point
resetSyncQueues();
audioPaused = true;
}
void resumeAudio(void) // unlock audio
{
if (!audioPaused)
return;
if (audio.dev > 0)
SDL_PauseAudioDevice(audio.dev, false);
audioPaused = false;
}
static void fillVisualsSyncBuffer(void)
{
pattSyncData_t pattSyncData;
chSyncData_t chSyncData;
syncedChannel_t *c;
stmTyp *s;
if (audio.resetSyncTickTimeFlag)
{
audio.resetSyncTickTimeFlag = false;
audio.tickTime64 = SDL_GetPerformanceCounter() + audio.audLatencyPerfValInt;
audio.tickTime64Frac = audio.audLatencyPerfValFrac;
}
if (songPlaying)
{
// push pattern variables to sync queue
pattSyncData.timer = song.curReplayerTimer;
pattSyncData.patternPos = song.curReplayerPattPos;
pattSyncData.pattern = song.curReplayerPattNr;
pattSyncData.songPos = song.curReplayerSongPos;
pattSyncData.speed = song.speed;
pattSyncData.tempo = (uint8_t)song.tempo;
pattSyncData.globalVol = (uint8_t)song.globVol;
pattSyncData.timestamp = audio.tickTime64;
pattQueuePush(pattSyncData);
}
// push channel variables to sync queue
c = chSyncData.channels;
s = stm;
for (int32_t i = 0; i < song.antChn; i++, c++, s++)
{
c->finalPeriod = s->finalPeriod;
c->fineTune = s->fineTune;
c->relTonNr = s->relTonNr;
c->instrNr = s->instrNr;
c->sampleNr = s->sampleNr;
c->envSustainActive = s->envSustainActive;
c->status = s->tmpStatus;
c->dFinalVol = s->dFinalVol;
c->smpStartPos = s->smpStartPos;
}
chSyncData.timestamp = audio.tickTime64;
chQueuePush(chSyncData);
audio.tickTime64 += tickTimeLen;
audio.tickTime64Frac += tickTimeLenFrac;
if (audio.tickTime64Frac > UINT32_MAX)
{
audio.tickTime64Frac &= UINT32_MAX;
audio.tickTime64++;
}
}
static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)
{
if (editor.wavIsRendering)
return;
int32_t samplesLeft = len / pmpCountDiv;
if (samplesLeft <= 0)
return;
while (samplesLeft > 0)
{
if (audio.dTickSampleCounter <= 0.0)
{
// new replayer tick
replayerBusy = true;
if (audio.volumeRampingFlag)
resetRampVolumes();
tickReplayer();
updateVoices();
fillVisualsSyncBuffer();
audio.dTickSampleCounter += audio.dSamplesPerTick;
replayerBusy = false;
}
const int32_t remainingTick = (int32_t)ceil(audio.dTickSampleCounter);
int32_t samplesToMix = samplesLeft;
if (samplesToMix > remainingTick)
samplesToMix = remainingTick;
mixAudio(stream, samplesToMix, pmpChannels);
stream += samplesToMix * pmpCountDiv;
samplesLeft -= samplesToMix;
audio.dTickSampleCounter -= samplesToMix;
}
(void)userdata; // make compiler not complain
}
static bool setupAudioBuffers(void)
{
const uint32_t sampleSize = sizeof (double);
audio.dMixBufferLUnaligned = (double *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
audio.dMixBufferRUnaligned = (double *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
if (audio.dMixBufferLUnaligned == NULL || audio.dMixBufferRUnaligned == NULL)
return false;
// make aligned main pointers
audio.dMixBufferL = (double *)ALIGN_PTR(audio.dMixBufferLUnaligned, 256);
audio.dMixBufferR = (double *)ALIGN_PTR(audio.dMixBufferRUnaligned, 256);
return true;
}
static void freeAudioBuffers(void)
{
if (audio.dMixBufferLUnaligned != NULL)
{
free(audio.dMixBufferLUnaligned);
audio.dMixBufferLUnaligned = NULL;
}
if (audio.dMixBufferRUnaligned != NULL)
{
free(audio.dMixBufferRUnaligned);
audio.dMixBufferRUnaligned = NULL;
}
audio.dMixBufferL = NULL;
audio.dMixBufferR = NULL;
}
void updateSendAudSamplesRoutine(bool lockMixer)
{
if (lockMixer)
lockMixerCallback();
if (config.specialFlags & BITDEPTH_16)
{
if (pmpChannels > 2)
sendAudSamplesFunc = sendSamples16BitDitherMultiChan;
else
sendAudSamplesFunc = sendSamples16BitDitherStereo;
}
else
{
if (pmpChannels > 2)
sendAudSamplesFunc = sendSamples32BitMultiChan;
else
sendAudSamplesFunc = sendSamples32BitStereo;
}
if (lockMixer)
unlockMixerCallback();
}
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
audio.audLatencyPerfValInt = (int32_t)dInt;
// fractional part (scaled to 0..2^32-1)
dFrac *= UINT32_MAX+1.0;
audio.audLatencyPerfValFrac = (uint32_t)dFrac;
audio.dAudioLatencyMs = dAudioLatencySecs * 1000.0;
}
static void setLastWorkingAudioDevName(void)
{
uint32_t stringLen;
if (audio.lastWorkingAudioDeviceName != NULL)
{
free(audio.lastWorkingAudioDeviceName);
audio.lastWorkingAudioDeviceName = NULL;
}
if (audio.currOutputDevice != NULL)
{
stringLen = (uint32_t)strlen(audio.currOutputDevice);
audio.lastWorkingAudioDeviceName = (char *)malloc(stringLen + 2);
if (audio.lastWorkingAudioDeviceName != NULL)
{
if (stringLen > 0)
strcpy(audio.lastWorkingAudioDeviceName, audio.currOutputDevice);
audio.lastWorkingAudioDeviceName[stringLen + 1] = '\0'; // UTF-8 needs double null termination
}
}
}
bool setupAudio(bool showErrorMsg)
{
int8_t newBitDepth;
uint16_t configAudioBufSize;
SDL_AudioSpec want, have;
closeAudio();
if (config.audioFreq < MIN_AUDIO_FREQ || config.audioFreq > MAX_AUDIO_FREQ)
config.audioFreq = 48000; // set default rate
// get audio buffer size from config special flags
configAudioBufSize = 1024;
if (config.specialFlags & BUFFSIZE_512)
configAudioBufSize = 512;
else if (config.specialFlags & BUFFSIZE_2048)
configAudioBufSize = 2048;
audio.wantFreq = config.audioFreq;
audio.wantSamples = configAudioBufSize;
audio.wantChannels = 2;
// set up audio device
memset(&want, 0, sizeof (want));
// these three may change after opening a device, but our mixer is dealing with it
want.freq = config.audioFreq;
want.format = (config.specialFlags & BITDEPTH_32) ? AUDIO_F32 : AUDIO_S16;
want.channels = 2;
// -------------------------------------------------------------------------------
want.callback = audioCallback;
want.samples = configAudioBufSize;
audio.dev = SDL_OpenAudioDevice(audio.currOutputDevice, 0, &want, &have, SDL_AUDIO_ALLOW_ANY_CHANGE); // prevent SDL2 from resampling
if (audio.dev == 0)
{
if (showErrorMsg)
showErrorMsgBox("Couldn't open audio device:\n\"%s\"\n\nDo you have any audio device enabled and plugged in?", SDL_GetError());
return false;
}
// test if the received audio format is compatible
if (have.format != AUDIO_S16 && have.format != AUDIO_F32)
{
if (showErrorMsg)
showErrorMsgBox("Couldn't open audio device:\nThe program doesn't support an SDL_AudioFormat of '%d' (not 16-bit or 32-bit float).",
(uint32_t)have.format);
closeAudio();
return false;
}
// test if the received audio rate is compatible
if (have.freq != 44100 && have.freq != 48000 && have.freq != 96000 && have.freq != 192000)
{
if (showErrorMsg)
showErrorMsgBox("Couldn't open audio device:\nThe program doesn't support an audio output rate of %dHz. Sorry!", have.freq);
closeAudio();
return false;
}
if (!setupAudioBuffers())
{
if (showErrorMsg)
showErrorMsgBox("Not enough memory!");
closeAudio();
return false;
}
// set new bit depth flag
newBitDepth = 16;
config.specialFlags &= ~BITDEPTH_32;
config.specialFlags |= BITDEPTH_16;
if (have.format == AUDIO_F32)
{
newBitDepth = 24;
config.specialFlags &= ~BITDEPTH_16;
config.specialFlags |= BITDEPTH_32;
}
audio.haveFreq = have.freq;
audio.haveSamples = have.samples;
audio.haveChannels = have.channels;
// set a few variables
config.audioFreq = have.freq;
audio.freq = have.freq;
smpBuffSize = have.samples;
calcAudioLatencyVars(have.samples, have.freq);
pmpChannels = have.channels;
pmpCountDiv = pmpChannels * ((newBitDepth == 16) ? sizeof (int16_t) : sizeof (float));
// make a copy of the new known working audio settings
audio.lastWorkingAudioFreq = config.audioFreq;
audio.lastWorkingAudioBits = config.specialFlags & (BITDEPTH_16 + BITDEPTH_32 + BUFFSIZE_512 + BUFFSIZE_1024 + BUFFSIZE_2048);
setLastWorkingAudioDevName();
// update config audio radio buttons if we're on that screen at the moment
if (ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)
showConfigScreen();
updateWavRendererSettings();
setAudioAmp(config.boostLevel, config.masterVol, (config.specialFlags & BITDEPTH_32) ? true : false);
// don't call stopVoices() in this routine
for (int32_t i = 0; i < MAX_VOICES; i++)
stopVoice(i);
stopAllScopes();
audio.dTickSampleCounter = 0.0; // zero tick sample counter so that it will instantly initiate a tick
calcReplayRate(audio.freq);
if (song.speed == 0)
song.speed = 125;
P_SetSpeed(song.speed); // this is important
updateSendAudSamplesRoutine(false);
audio.resetSyncTickTimeFlag = true;
setWavRenderFrequency(audio.freq);
setWavRenderBitDepth((config.specialFlags & BITDEPTH_32) ? 32 : 16);
return true;
}
void closeAudio(void)
{
if (audio.dev > 0)
{
SDL_PauseAudioDevice(audio.dev, true);
SDL_CloseAudioDevice(audio.dev);
audio.dev = 0;
}
freeAudioBuffers();
}