ref: d9f3b8e36558b0825787c14c6512d975afa4cf5b
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 "scopes/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"
// --------------------------------
// hide POSIX warnings
#ifdef _MSC_VER
#pragma warning(disable: 4996)
#endif
#define INITIAL_DITHER_SEED 0x12345000
static int8_t pmpCountDiv, pmpChannels = 2;
static uint16_t smpBuffSize;
static uint32_t oldAudioFreq, tickTimeLen, tickTimeLenFrac, randSeed = INITIAL_DITHER_SEED;
static float fAudioNormalizeMul, fPanningTab[256+1];
static double dAudioNormalizeMul, dPrngStateL, dPrngStateR;
static voice_t voice[MAX_CHANNELS * 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)
{
channel_t *ch = channel;
for (int32_t i = 0; i < MAX_CHANNELS; i++, ch++)
ch->oldFinalPeriod = -1;
voice_t *v = voice;
for (int32_t i = 0; i < MAX_CHANNELS*2; i++, v++)
v->oldDelta = 0;
}
void stopVoice(int32_t i)
{
voice_t *v;
v = &voice[i];
memset(v, 0, sizeof (voice_t));
v->panning = 128;
// clear "fade out" voice too
v = &voice[MAX_CHANNELS + i];
memset(v, 0, sizeof (voice_t));
v->panning = 128;
}
bool setNewAudioSettings(void) // only call this from the main input/video thread
{
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;
}
audio.currOutputDevice = strdup(audio.lastWorkingAudioDeviceName);
}
// 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;
}
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;
fAudioNormalizeMul = (float)dAmp;
}
void decreaseMasterVol(void)
{
if (config.masterVol >= 16)
config.masterVol -= 16;
else
config.masterVol = 0;
setAudioAmp(config.boostLevel, config.masterVol, !!(config.specialFlags & BITDEPTH_32));
// if Config -> I/O Devices is open, update master volume scrollbar
if (ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)
drawScrollBar(SB_MASTERVOL_SCROLL);
}
void increaseMasterVol(void)
{
if (config.masterVol < (256-16))
config.masterVol += 16;
else
config.masterVol = 256;
setAudioAmp(config.boostLevel, config.masterVol, !!(config.specialFlags & BITDEPTH_32));
// if Config -> I/O Devices is open, update master volume scrollbar
if (ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)
drawScrollBar(SB_MASTERVOL_SCROLL);
}
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)
calcReplayerVars(audio.freq);
}
void setBackOldAudioFreq(void) // for song-to-WAV rendering
{
const bool mustRecalcTables = audio.freq != oldAudioFreq;
audio.freq = oldAudioFreq;
if (mustRecalcTables)
calcReplayerVars(audio.freq);
}
void setMixerBPM(int32_t bpm)
{
if (bpm < MIN_BPM || bpm > MAX_BPM)
return;
int32_t i = bpm - MIN_BPM;
audio.samplesPerTick64 = audio.samplesPerTick64Tab[i]; // fixed-point
audio.samplesPerTick = (audio.samplesPerTick64 + (1LL << 31)) >> 32; // rounded
// for audio/video sync timestamp
tickTimeLen = audio.tickTimeTab[i];
tickTimeLenFrac = audio.tickTimeFracTab[i];
// for calculating volume ramp length for tick-length ramps
audio.fRampTickMul = audio.fRampTickMulTab[i];
}
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++)
fPanningTab[i] = sqrtf(i / 256.0f);
}
static void voiceUpdateVolumes(int32_t i, uint8_t status)
{
voice_t *v = &voice[i];
const float fVolumeL = v->fVolume * fPanningTab[256-v->panning];
const float fVolumeR = v->fVolume * fPanningTab[ v->panning];
if (!audio.volumeRampingFlag)
{
// volume ramping is disabled
v->fVolumeL = fVolumeL;
v->fVolumeR = fVolumeR;
v->volumeRampLength = 0;
return;
}
v->fVolumeLTarget = fVolumeL;
v->fVolumeRTarget = fVolumeR;
if (status & IS_Trigger)
{
// sample is about to start, ramp out/in at the same time
// setup "fade out" voice (only if current voice volume > 0)
if (v->fVolumeL > 0.0f || v->fVolumeR > 0.0f)
{
voice_t *f = &voice[MAX_CHANNELS+i];
*f = *v; // copy voice
f->volumeRampLength = audio.quickVolRampSamples;
const float fVolumeLTarget = -f->fVolumeL;
const float fVolumeRTarget = -f->fVolumeR;
f->fVolumeLDelta = fVolumeLTarget * audio.fRampQuickVolMul;
f->fVolumeRDelta = fVolumeRTarget * audio.fRampQuickVolMul;
f->isFadeOutVoice = true;
}
// make current voice fade in from zero when it starts
v->fVolumeL = 0.0f;
v->fVolumeR = 0.0f;
}
// 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 (fVolumeL == v->fVolumeL && fVolumeR == v->fVolumeR && !(status & IS_Trigger))
{
// there is no volume change
v->volumeRampLength = 0;
}
else
{
const float fVolumeLTarget = fVolumeL - v->fVolumeL;
const float fVolumeRTarget = fVolumeR - v->fVolumeR;
if (status & IS_QuickVol)
{
v->fVolumeLDelta = fVolumeLTarget * audio.fRampQuickVolMul;
v->fVolumeRDelta = fVolumeRTarget * audio.fRampQuickVolMul;
v->volumeRampLength = audio.quickVolRampSamples;
}
else
{
v->fVolumeLDelta = fVolumeLTarget * audio.fRampTickMul;
v->fVolumeRDelta = fVolumeRTarget * audio.fRampTickMul;
v->volumeRampLength = audio.samplesPerTick;
}
}
}
static void voiceTrigger(int32_t ch, sample_t *s, int32_t position)
{
voice_t *v = &voice[ch];
int32_t length = s->length;
int32_t loopStart = s->loopStart;
int32_t loopLength = s->loopLength;
int32_t loopEnd = s->loopStart + s->loopLength;
uint8_t loopType = GET_LOOPTYPE(s->flags);
bool sample16Bit = !!(s->flags & SAMPLE_16BIT);
if (s->dataPtr == 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 (sample16Bit)
{
v->base16 = (const int16_t *)s->dataPtr;
v->revBase16 = &v->base16[loopStart + loopEnd]; // for pingpong loops
v->leftEdgeTaps16 = s->leftEdgeTapSamples16 + SINC_LEFT_TAPS;
}
else
{
v->base8 = s->dataPtr;
v->revBase8 = &v->base8[loopStart + loopEnd]; // for pingpong loops
v->leftEdgeTaps8 = s->leftEdgeTapSamples8 + SINC_LEFT_TAPS;
}
v->hasLooped = false; // for sinc interpolation special case
v->samplingBackwards = false;
v->loopType = loopType;
v->sampleEnd = (loopType == LOOP_OFF) ? length : loopEnd;
v->loopStart = loopStart;
v->loopLength = loopLength;
v->position = position;
v->positionFrac = 0;
// if position overflows, shut down voice (f.ex. through 9xx command)
if (v->position >= v->sampleEnd)
{
v->active = false;
return;
}
v->mixFuncOffset = (sample16Bit * 9) + (audio.interpolationType * 3) + loopType;
v->active = true;
}
void resetRampVolumes(void)
{
voice_t *v = voice;
for (int32_t i = 0; i < song.numChannels; i++, v++)
{
v->fVolumeL = v->fVolumeLTarget;
v->fVolumeR = v->fVolumeRTarget;
v->volumeRampLength = 0;
}
}
void updateVoices(void)
{
channel_t *ch = channel;
voice_t *v = voice;
for (int32_t i = 0; i < song.numChannels; i++, ch++, v++)
{
const uint8_t status = ch->tmpStatus = ch->status; // (tmpStatus is used for audio/video sync queue)
if (status == 0)
continue;
ch->status = 0;
if (status & IS_Vol)
{
v->fVolume = ch->fFinalVol;
const int32_t scopeVolume = (int32_t)((SCOPE_HEIGHT * ch->fFinalVol) + 0.5f); // rounded
v->scopeVolume = (uint8_t)scopeVolume;
}
if (status & IS_Pan)
v->panning = ch->finalPan;
if (status & (IS_Vol + IS_Pan))
voiceUpdateVolumes(i, status);
if (status & IS_Period)
{
// use cached values when possible
if (ch->finalPeriod != ch->oldFinalPeriod)
{
ch->oldFinalPeriod = ch->finalPeriod;
if (ch->finalPeriod == 0) // in FT2, period 0 -> delta 0
{
v->scopeDelta = 0;
v->oldDelta = 0;
v->fSincLUT = fKaiserSinc;
}
else
{
const double dHz = dPeriod2Hz(ch->finalPeriod);
const uintCPUWord_t delta = v->oldDelta = (intCPUWord_t)((dHz * audio.dHz2MixDeltaMul) + 0.5); // Hz -> fixed-point delta (rounded)
// decide which polyphase sinc LUT to use according to resampling ratio
if (delta <= (uintCPUWord_t)(1.1875 * MIXER_FRAC_SCALE))
v->fSincLUT = fKaiserSinc;
else if (delta <= (uintCPUWord_t)(1.5 * MIXER_FRAC_SCALE))
v->fSincLUT = fDownSample1;
else
v->fSincLUT = fDownSample2;
// set scope delta
const double dHz2ScopeDeltaMul = SCOPE_FRAC_SCALE / (double)SCOPE_HZ;
v->scopeDelta = (intCPUWord_t)((dHz * dHz2ScopeDeltaMul) + 0.5); // Hz -> fixed-point delta (rounded)
}
}
v->delta = v->oldDelta;
}
if (status & IS_Trigger)
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 = (double)audio.fMixBufferL[i] * dAudioNormalizeMul;
dOut = (dOut + 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 = (double)audio.fMixBufferR[i] * dAudioNormalizeMul;
dOut = (dOut + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = 0.0f;
audio.fMixBufferR[i] = 0.0f;
}
(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 = (double)audio.fMixBufferL[i] * dAudioNormalizeMul;
dOut = (dOut + 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 = (double)audio.fMixBufferR[i] * dAudioNormalizeMul;
dOut = (dOut + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
out32 = (int32_t)dOut;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = 0.0f;
audio.fMixBufferR[i] = 0.0f;
// 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)
{
float fOut, *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
fOut = audio.fMixBufferL[i] * fAudioNormalizeMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer32++ = fOut;
// right channel
fOut = audio.fMixBufferR[i] * fAudioNormalizeMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer32++ = fOut;
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = 0.0f;
audio.fMixBufferR[i] = 0.0f;
}
(void)numAudioChannels;
}
static void sendSamples32BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
float fOut, *fStreamPointer32 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
fOut = audio.fMixBufferL[i] * fAudioNormalizeMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer32++ = fOut;
// right channel
fOut = audio.fMixBufferR[i] * fAudioNormalizeMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer32++ = fOut;
// clear what we read from the mixing buffer
audio.fMixBufferL[i] = 0.0f;
audio.fMixBufferR[i] = 0.0f;
// send zeroes to the rest of the channels
for (uint32_t j = 2; j < numAudioChannels; j++)
*fStreamPointer32++ = 0.0f;
}
}
static void doChannelMixing(int32_t bufferPosition, int32_t samplesToMix)
{
voice_t *v = voice; // normal voices
voice_t *r = &voice[MAX_CHANNELS]; // volume ramp fadeout-voices
for (int32_t i = 0; i < song.numChannels; i++, v++, r++)
{
if (v->active)
{
bool centerMixFlag;
const bool volRampFlag = (v->volumeRampLength > 0);
if (volRampFlag)
{
centerMixFlag = (v->fVolumeLTarget == v->fVolumeRTarget) && (v->fVolumeLDelta == v->fVolumeRDelta);
}
else // no volume ramping active
{
if (v->fVolumeL == 0.0f && v->fVolumeR == 0.0f)
{
silenceMixRoutine(v, samplesToMix);
continue;
}
centerMixFlag = (v->fVolumeL == v->fVolumeR);
}
mixFuncTab[((int32_t)centerMixFlag * 36) + ((int32_t)volRampFlag * 18) + v->mixFuncOffset](v, bufferPosition, samplesToMix);
}
if (r->active) // volume ramp fadeout-voice
{
const bool centerMixFlag = (r->fVolumeLTarget == r->fVolumeRTarget) && (r->fVolumeLDelta == r->fVolumeRDelta);
mixFuncTab[((int32_t)centerMixFlag * 36) + 18 + r->mixFuncOffset](r, bufferPosition, samplesToMix);
}
}
}
// 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);
doChannelMixing(0, 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;
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.tick = song.curReplayerTick;
pattSyncData.row = song.curReplayerRow;
pattSyncData.pattNum = song.curReplayerPattNum;
pattSyncData.songPos = song.curReplayerSongPos;
pattSyncData.BPM = song.BPM;
pattSyncData.speed = (uint8_t)song.speed;
pattSyncData.globalVolume = (uint8_t)song.globalVolume;
pattSyncData.timestamp = audio.tickTime64;
pattQueuePush(pattSyncData);
}
// push channel variables to sync queue
syncedChannel_t *c = chSyncData.channels;
channel_t *s = channel;
voice_t *v = voice;
for (int32_t i = 0; i < song.numChannels; i++, c++, s++, v++)
{
c->scopeVolume = v->scopeVolume;
c->scopeDelta = v->scopeDelta;
c->instrNum = s->instrNum;
c->smpNum = s->smpNum;
c->status = s->tmpStatus;
c->smpStartPos = s->smpStartPos;
c->pianoNoteNum = 255; // no piano key
if (songPlaying && (c->status & IS_Period) && s->envSustainActive)
{
const int32_t note = getPianoKey(s->finalPeriod, s->finetune, s->relativeNote);
if (note >= 0 && note <= 95)
c->pianoNoteNum = (uint8_t)note;
}
}
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;
len /= pmpCountDiv; // bytes -> samples
if (len <= 0)
return;
assert(len <= MAX_WAV_RENDER_SAMPLES_PER_TICK);
int32_t bufferPosition = 0;
int32_t samplesLeft = len;
while (samplesLeft > 0)
{
if (audio.tickSampleCounter64 <= 0) // new replayer tick
{
replayerBusy = true;
if (audio.volumeRampingFlag)
resetRampVolumes();
tickReplayer();
updateVoices();
fillVisualsSyncBuffer();
audio.tickSampleCounter64 += audio.samplesPerTick64;
replayerBusy = false;
}
const int32_t remainingTick = (audio.tickSampleCounter64 + UINT32_MAX) >> 32; // ceil (rounded upwards)
int32_t samplesToMix = samplesLeft;
if (samplesToMix > remainingTick)
samplesToMix = remainingTick;
doChannelMixing(bufferPosition, samplesToMix);
bufferPosition += samplesToMix;
samplesLeft -= samplesToMix;
audio.tickSampleCounter64 -= (int64_t)samplesToMix << 32;
}
// normalize mix buffer and send to audio stream
sendAudSamplesFunc(stream, len, pmpChannels);
(void)userdata;
}
static bool setupAudioBuffers(void)
{
const uint32_t sampleSize = sizeof (float);
audio.fMixBufferLUnaligned = (float *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
audio.fMixBufferRUnaligned = (float *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
if (audio.fMixBufferLUnaligned == NULL || audio.fMixBufferRUnaligned == NULL)
return false;
// make aligned main pointers
audio.fMixBufferL = (float *)ALIGN_PTR(audio.fMixBufferLUnaligned, 256);
audio.fMixBufferR = (float *)ALIGN_PTR(audio.fMixBufferRUnaligned, 256);
// clear buffers
memset(audio.fMixBufferL, 0, MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize);
memset(audio.fMixBufferR, 0, MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize);
return true;
}
static void freeAudioBuffers(void)
{
if (audio.fMixBufferLUnaligned != NULL)
{
free(audio.fMixBufferLUnaligned);
audio.fMixBufferLUnaligned = NULL;
}
if (audio.fMixBufferRUnaligned != NULL)
{
free(audio.fMixBufferRUnaligned);
audio.fMixBufferRUnaligned = NULL;
}
audio.fMixBufferL = NULL;
audio.fMixBufferR = 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;
if (audioFreq == 0)
return;
const double dAudioLatencySecs = audioBufferSize / (double)audioFreq;
double 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)
{
if (audio.lastWorkingAudioDeviceName != NULL)
{
free(audio.lastWorkingAudioDeviceName);
audio.lastWorkingAudioDeviceName = NULL;
}
if (audio.currOutputDevice != NULL)
audio.lastWorkingAudioDeviceName = strdup(audio.currOutputDevice);
}
bool setupAudio(bool showErrorMsg)
{
SDL_AudioSpec want, have;
closeAudio();
if (config.audioFreq < MIN_AUDIO_FREQ || config.audioFreq > MAX_AUDIO_FREQ)
config.audioFreq = DEFAULT_AUDIO_FREQ;
// get audio buffer size from config special flags
uint16_t 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 CPU_64BIT
if (have.freq != 44100 && have.freq != 48000 && have.freq != 96000 && have.freq != 192000)
#else
if (have.freq != 44100 && have.freq != 48000)
#endif
{
if (showErrorMsg)
showErrorMsgBox("Couldn't open audio device:\nThis 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
int8_t 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));
// don't call stopVoices() in this routine
for (int32_t i = 0; i < MAX_CHANNELS; i++)
stopVoice(i);
stopAllScopes();
audio.tickSampleCounter64 = 0; // zero tick sample counter so that it will instantly initiate a tick
calcReplayerVars(audio.freq);
if (song.BPM == 0)
song.BPM = 125;
setMixerBPM(song.BPM); // 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();
}