ref: 6c8bd4efbb4713c2c1e83ded672244219596f4d2
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_mix.h"
#include "ft2_tables.h"
#define INITIAL_DITHER_SEED 0x12345000
static int8_t pmpCountDiv, pmpChannels = 2;
static uint16_t smpBuffSize;
static int32_t masterVol, oldAudioFreq, speedVal, pmpLeft, randSeed = INITIAL_DITHER_SEED;
static int32_t prngStateL, prngStateR;
static uint32_t tickTimeLen, tickTimeLenFrac;
static float fAudioAmpMul;
static voice_t voice[MAX_VOICES * 2];
static void (*sendAudSamplesFunc)(uint8_t *, uint32_t, uint8_t); // "send mixed samples" routines
#if !defined __amd64__ && !defined _WIN64
static int32_t oldPeriod;
static uint32_t oldSFrq, oldSFrqRev;
#endif
pattSyncData_t *pattSyncEntry;
chSyncData_t *chSyncEntry;
volatile bool pattQueueReading, pattQueueClearing, chQueueReading, chQueueClearing;
#if !defined __amd64__ && !defined _WIN64
void resetCachedMixerVars(void)
{
oldPeriod = -1;
oldSFrq = 0;
oldSFrqRev = 0xFFFFFFFF;
}
#endif
void stopVoice(uint8_t i)
{
voice_t *v;
v = &voice[i];
memset(v, 0, sizeof (voice_t));
v->SPan = 128;
// clear "fade out" voice too
v = &voice[MAX_VOICES + i];
memset(v, 0, sizeof (voice_t));
v->SPan = 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_24 + 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 (editor.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();
return true;
}
// ampFactor = 1..32, masterVol = 0..256
void setAudioAmp(int16_t ampFactor, int16_t master, bool bitDepth32Flag)
{
ampFactor = CLAMP(ampFactor, 1, 32);
master = CLAMP(master, 0, 256);
// voiceVolume = (vol(0..65535) * pan(0..65536)) >> 4
const float fAudioNorm = 1.0f / (((65535UL * 65536) / 16) / MAX_VOICES);
if (bitDepth32Flag)
{
// 32-bit floating point (24-bit)
fAudioAmpMul = fAudioNorm * (master / 256.0f) * (ampFactor / 32.0f);
}
else
{
// 16-bit integer
masterVol = master * ampFactor * 2048;
}
}
void setNewAudioFreq(uint32_t freq) // for song to WAV rendering
{
if (freq == 0)
return;
oldAudioFreq = audio.freq;
audio.freq = freq;
calcReplayRate(audio.freq);
}
void setBackOldAudioFreq(void) // for song to WAV rendering
{
audio.freq = oldAudioFreq;
calcReplayRate(audio.freq);
}
void setSpeed(uint16_t bpm)
{
double dInt, dFrac;
if (bpm == 0)
return;
speedVal = ((audio.freq + audio.freq) + (audio.freq >> 1)) / bpm; // (audio.freq * 2.5) / BPM
if (speedVal > 0) // calculate tick time length for audio/video sync timestamp
{
// number of samples per tick -> tick length for performance counter
dFrac = modf(speedVal * audio.dSpeedValMul, &dInt);
/* - integer part -
** Cast to int32_t so that the compiler will use fast SSE2 float->int instructions.
** This result won't be above 2^31, so this is safe.
*/
tickTimeLen = (int32_t)dInt;
// - fractional part (scaled to 0..2^32-1) -
dFrac *= UINT32_MAX;
tickTimeLenFrac = (uint32_t)(dFrac + 0.5);
audio.rampSpeedValMul = 0xFFFFFFFF / speedVal;
}
}
void audioSetVolRamp(bool volRamp)
{
lockMixerCallback();
audio.volumeRampingFlag = volRamp;
unlockMixerCallback();
}
void audioSetInterpolation(bool interpolation)
{
lockMixerCallback();
audio.interpolationFlag = interpolation;
unlockMixerCallback();
}
static void voiceUpdateVolumes(int32_t i, uint8_t status)
{
int32_t volL, volR, destVolL, destVolR;
uint32_t vol;
voice_t *v, *f;
v = &voice[i];
vol = v->SVol; // 0..65535
// (0..65535 * 0..65536) >> 4 = 0..268431360
volR = (vol * panningTab[ v->SPan]) >> 4;
volL = (vol * panningTab[256-v->SPan]) >> 4;
if (!audio.volumeRampingFlag)
{
v->SLVol2 = volL;
v->SRVol2 = volR;
}
else
{
v->SLVol1 = volL;
v->SRVol1 = volR;
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->SLVol2 > 0 || v->SRVol2 > 0)
{
f = &voice[MAX_VOICES + i];
*f = *v; // copy voice
f->SVolIPLen = audio.quickVolSizeVal;
destVolL = -f->SLVol2;
destVolR = -f->SRVol2;
f->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32;
f->SRVolIP = ((int64_t)destVolR * audio.rampQuickVolMul) >> 32;
f->isFadeOutVoice = true;
}
// make current voice fade in when it starts
v->SLVol2 = 0;
v->SRVol2 = 0;
}
// ramp volume changes
/* FT2 has two internal volume ramping lengths:
** IS_QuickVol: 5ms (audioFreq / 200)
** Normal: The duration of a tick (speedVal)
*/
if (volL == v->SLVol2 && volR == v->SRVol2)
{
v->SVolIPLen = 0; // there is no volume change
}
else
{
destVolL = volL - v->SLVol2;
destVolR = volR - v->SRVol2;
if (status & IS_QuickVol)
{
v->SVolIPLen = audio.quickVolSizeVal;
v->SLVolIP = ((int64_t)destVolL * audio.rampQuickVolMul) >> 32;
v->SRVolIP = ((int64_t)destVolR * audio.rampQuickVolMul) >> 32;
}
else
{
v->SVolIPLen = speedVal;
v->SLVolIP = ((int64_t)destVolL * audio.rampSpeedValMul) >> 32;
v->SRVolIP = ((int64_t)destVolR * audio.rampSpeedValMul) >> 32;
}
}
}
}
static void voiceTrigger(int32_t i, sampleTyp *s, int32_t position)
{
bool sampleIs16Bit;
uint8_t loopType;
int32_t oldSLen, length, loopBegin, loopLength;
voice_t *v;
v = &voice[i];
length = s->len;
loopBegin = s->repS;
loopLength = s->repL;
loopType = s->typ & 3;
sampleIs16Bit = (s->typ >> 4) & 1;
if (sampleIs16Bit)
{
assert(!(length & 1));
assert(!(loopBegin & 1));
assert(!(loopLength & 1));
length >>= 1;
loopBegin >>= 1;
loopLength >>= 1;
}
if (s->pek == NULL || length < 1)
{
v->mixRoutine = NULL; // shut down voice (illegal parameters)
return;
}
if (loopLength < 1) // disable loop if loopLength is below 1
loopType = 0;
if (sampleIs16Bit)
{
v->SBase16 = (const int16_t *)s->pek;
v->SRevBase16 = &v->SBase16[loopBegin + (loopBegin + loopLength)]; // for pingpong loops
}
else
{
v->SBase8 = s->pek;
v->SRevBase8 = &v->SBase8[loopBegin + (loopBegin + loopLength)]; // for pingpong loops
}
v->backwards = false;
v->SLen = (loopType > 0) ? (loopBegin + loopLength) : length;
v->SRepS = loopBegin;
v->SRepL = loopLength;
v->SPos = position;
v->SPosDec = 0; // position fraction
// if 9xx position overflows, shut down voice
oldSLen = (loopType > 0) ? (loopBegin + loopLength) : length;
if (v->SPos >= oldSLen)
{
v->mixRoutine = NULL;
return;
}
v->mixRoutine = mixRoutineTable[(sampleIs16Bit * 12) + (audio.volumeRampingFlag * 6) + (audio.interpolationFlag * 3) + loopType];
}
void mix_SaveIPVolumes(void) // for volume ramping
{
voice_t *v = voice;
for (uint32_t i = 0; i < MAX_VOICES; i++, v++)
{
v->SLVol2 = v->SLVol1;
v->SRVol2 = v->SRVol1;
v->SVolIPLen = 0;
}
}
void mix_UpdateChannelVolPanFrq(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; // ch->tmpStatus is used for audio/video sync queue
if (status == 0)
continue;
ch->status = 0;
// volume change
if (status & IS_Vol)
v->SVol = ch->finalVol;
// panning change
if (status & IS_Pan)
v->SPan = ch->finalPan;
// update mixing volumes if vol/pan change
if (status & (IS_Vol + IS_Pan))
voiceUpdateVolumes(i, status);
// frequency change (received even if the period didn't change!)
if (status & IS_Period)
{
#if defined __amd64__ || defined _WIN64
v->SFrq = getFrequenceValue(ch->finalPeriod);
#else
// use cached values to prevent a 32-bit divsion all the time
if (ch->finalPeriod != oldPeriod)
{
oldPeriod = ch->finalPeriod;
oldSFrq = getFrequenceValue(ch->finalPeriod);
oldSFrqRev = 0xFFFFFFFF;
if (oldSFrq != 0)
oldSFrqRev /= oldSFrq;
}
v->SFrq = oldSFrq;
v->SFrqRev = oldSFrqRev;
#endif
}
// sample trigger (note)
if (status & IS_NyTon)
voiceTrigger(i, ch->smpPtr, ch->smpStartPos);
}
}
void resetAudioDither(void)
{
randSeed = INITIAL_DITHER_SEED;
prngStateL = 0;
prngStateR = 0;
}
static inline uint32_t random32(void)
{
// LCG 32-bit random
randSeed *= 134775813;
randSeed++;
return randSeed;
}
static void sendSamples16BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int16_t *streamPointer16;
int32_t out32;
(void)numAudioChannels;
streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
out32 = ((int64_t)audio.mixBufferL[i] * masterVol) >> 32;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel
out32 = ((int64_t)audio.mixBufferR[i] * masterVol) >> 32;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
}
}
static void sendSamples16BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int16_t *streamPointer16;
int32_t out32;
uint32_t i, j;
streamPointer16 = (int16_t *)stream;
for (i = 0; i < sampleBlockLength; i++)
{
// left channel
out32 = ((int64_t)audio.mixBufferL[i] * masterVol) >> 32;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel
out32 = ((int64_t)audio.mixBufferR[i] * masterVol) >> 32;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
for (j = 2; j < numAudioChannels; j++)
*streamPointer16++ = 0;
}
}
static void sendSamples16BitDitherStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int16_t *streamPointer16;
int32_t prng, out32;
(void)numAudioChannels;
streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel - 1-bit triangular dithering
prng = random32();
out32 = ((((int64_t)audio.mixBufferL[i] * masterVol) + prng) - prngStateL) >> 32;
prngStateL = prng;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel - 1-bit triangular dithering
prng = random32();
out32 = ((((int64_t)audio.mixBufferR[i] * masterVol) + prng) - prngStateR) >> 32;
prngStateR = prng;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
}
}
static void sendSamples16BitDitherMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
int16_t *streamPointer16;
int32_t prng, out32;
streamPointer16 = (int16_t *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel - 1-bit triangular dithering
prng = random32();
out32 = ((((int64_t)audio.mixBufferL[i] * masterVol) + prng) - prngStateL) >> 32;
prngStateL = prng;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
// right channel - 1-bit triangular dithering
prng = random32();
out32 = ((((int64_t)audio.mixBufferR[i] * masterVol) + prng) - prngStateR) >> 32;
prngStateR = prng;
CLAMP16(out32);
*streamPointer16++ = (int16_t)out32;
for (uint32_t j = 2; j < numAudioChannels; j++)
*streamPointer16++ = 0;
}
}
static void sendSamples24BitStereo(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
float fOut, *fStreamPointer24;
(void)numAudioChannels;
fStreamPointer24 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
fOut = audio.mixBufferL[i] * fAudioAmpMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer24++ = fOut;
// right channel
fOut = audio.mixBufferR[i] * fAudioAmpMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer24++ = fOut;
}
}
static void sendSamples24BitMultiChan(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
float fOut, *fStreamPointer24;
fStreamPointer24 = (float *)stream;
for (uint32_t i = 0; i < sampleBlockLength; i++)
{
// left channel
fOut = audio.mixBufferL[i] * fAudioAmpMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer24++ = fOut;
// right channel
fOut = audio.mixBufferR[i] * fAudioAmpMul;
fOut = CLAMP(fOut, -1.0f, 1.0f);
*fStreamPointer24++ = fOut;
for (uint32_t j = 2; j < numAudioChannels; j++)
*fStreamPointer24++ = 0.0f;
}
}
static void mixAudio(uint8_t *stream, uint32_t sampleBlockLength, uint8_t numAudioChannels)
{
voice_t *v, *r;
assert(sampleBlockLength <= MAX_WAV_RENDER_SAMPLES_PER_TICK);
memset(audio.mixBufferL, 0, sampleBlockLength * sizeof (int32_t));
memset(audio.mixBufferR, 0, sampleBlockLength * sizeof (int32_t));
// mix channels
v = voice; // normal voices
r = &voice[MAX_VOICES]; // volume ramp voices
for (int32_t i = 0; i < song.antChn; i++, v++, r++)
{
// call the mixing routine currently set for the voice
if (v->mixRoutine != NULL) v->mixRoutine(v, sampleBlockLength); // mix normal voice
if (r->mixRoutine != NULL) r->mixRoutine(r, sampleBlockLength); // mix volume ramp voice
}
// normalize mix buffer and send to audio stream
sendAudSamplesFunc(stream, sampleBlockLength, numAudioChannels);
}
// used for song-to-WAV renderer
uint32_t mixReplayerTickToBuffer(uint8_t *stream, uint8_t bitDepth)
{
voice_t *v, *r;
assert(speedVal <= MAX_WAV_RENDER_SAMPLES_PER_TICK);
memset(audio.mixBufferL, 0, speedVal * sizeof (int32_t));
memset(audio.mixBufferR, 0, speedVal * sizeof (int32_t));
// mix channels
v = voice; // normal voices
r = &voice[MAX_VOICES]; // volume ramp voices
for (int32_t i = 0; i < song.antChn; i++, v++, r++)
{
// call the mixing routine currently set for the voice
if (v->mixRoutine != NULL) v->mixRoutine(v, speedVal); // mix normal voice
if (r->mixRoutine != NULL) r->mixRoutine(r, speedVal); // mix volume ramp voice
}
// normalize mix buffer and send to audio stream
if (bitDepth == 16)
{
if (config.specialFlags2 & DITHERED_AUDIO)
sendSamples16BitDitherStereo(stream, speedVal, 2);
else
sendSamples16BitStereo(stream, speedVal, 2);
}
else
{
sendSamples24BitStereo(stream, speedVal, 2);
}
return speedVal;
}
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 several
** minutes between each time (when queue size is default, 16384) */
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;
}
static 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 SDLCALL mixCallback(void *userdata, Uint8 *stream, int len)
{
int32_t a, b;
pattSyncData_t pattSyncData;
chSyncData_t chSyncData;
syncedChannel_t *c;
stmTyp *s;
(void)userdata;
assert(len < 65536); // limitation in mixer
assert(pmpCountDiv > 0);
a = len / pmpCountDiv;
if (a <= 0)
return;
while (a > 0)
{
if (pmpLeft == 0)
{
// replayer tick
replayerBusy = true;
if (audio.volumeRampingFlag)
mix_SaveIPVolumes();
mainPlayer();
mix_UpdateChannelVolPanFrq();
// AUDIO/VIDEO SYNC
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 = (uint8_t)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
for (int32_t i = 0; i < song.antChn; i++)
{
c = &chSyncData.channels[i];
s = &stm[i];
c->voiceDelta = voice[i].SFrq;
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->finalVol = s->finalVol;
c->smpStartPos = s->smpStartPos;
}
chSyncData.timestamp = audio.tickTime64;
chQueuePush(chSyncData);
audio.tickTime64 += tickTimeLen;
audio.tickTime64Frac += tickTimeLenFrac;
if (audio.tickTime64Frac > 0xFFFFFFFF)
{
audio.tickTime64Frac &= 0xFFFFFFFF;
audio.tickTime64++;
}
pmpLeft = speedVal;
replayerBusy = false;
}
b = a;
if (b > pmpLeft)
b = pmpLeft;
mixAudio(stream, b, pmpChannels);
stream += b * pmpCountDiv;
a -= b;
pmpLeft -= b;
}
}
static bool setupAudioBuffers(void)
{
const uint32_t sampleSize = sizeof (int32_t);
audio.mixBufferLUnaligned = (int32_t *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
audio.mixBufferRUnaligned = (int32_t *)MALLOC_PAD(MAX_WAV_RENDER_SAMPLES_PER_TICK * sampleSize, 256);
if (audio.mixBufferLUnaligned == NULL || audio.mixBufferRUnaligned == NULL)
return false;
// make aligned main pointers
audio.mixBufferL = (int32_t *)ALIGN_PTR(audio.mixBufferLUnaligned, 256);
audio.mixBufferR = (int32_t *)ALIGN_PTR(audio.mixBufferRUnaligned, 256);
return true;
}
static void freeAudioBuffers(void)
{
if (audio.mixBufferLUnaligned != NULL)
{
free(audio.mixBufferLUnaligned);
audio.mixBufferLUnaligned = NULL;
}
if (audio.mixBufferRUnaligned != NULL)
{
free(audio.mixBufferRUnaligned);
audio.mixBufferRUnaligned = NULL;
}
audio.mixBufferL = NULL;
audio.mixBufferR = NULL;
}
void updateSendAudSamplesRoutine(bool lockMixer)
{
if (lockMixer)
lockMixerCallback();
// force dither off if somehow set with 24-bit float (illegal)
if ((config.specialFlags2 & DITHERED_AUDIO) && (config.specialFlags & BITDEPTH_24))
config.specialFlags2 &= ~DITHERED_AUDIO;
if (config.specialFlags2 & DITHERED_AUDIO)
{
if (config.specialFlags & BITDEPTH_16)
{
if (pmpChannels > 2)
sendAudSamplesFunc = sendSamples16BitDitherMultiChan;
else
sendAudSamplesFunc = sendSamples16BitDitherStereo;
}
}
else
{
if (config.specialFlags & BITDEPTH_16)
{
if (pmpChannels > 2)
sendAudSamplesFunc = sendSamples16BitMultiChan;
else
sendAudSamplesFunc = sendSamples16BitStereo;
}
else
{
if (pmpChannels > 2)
sendAudSamplesFunc = sendSamples24BitMultiChan;
else
sendAudSamplesFunc = sendSamples24BitStereo;
}
}
if (lockMixer)
unlockMixerCallback();
}
static void calcAudioLatencyVars(uint16_t haveSamples, int32_t haveFreq)
{
double dHaveFreq, dAudioLatencySecs, dInt, dFrac;
dHaveFreq = haveFreq;
if (dHaveFreq == 0.0)
return; // panic!
dAudioLatencySecs = haveSamples / dHaveFreq;
// XXX: haveSamples and haveFreq better not be bogus values...
dFrac = modf(dAudioLatencySecs * editor.dPerfFreq, &dInt);
// integer part
audio.audLatencyPerfValInt = (uint32_t)dInt;
// fractional part (scaled to 0..2^32-1)
dFrac *= UINT32_MAX;
audio.audLatencyPerfValFrac = (uint32_t)(dFrac + 0.5);
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)
{
// set default rate
config.audioFreq = 48000;
}
// 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_24) ? AUDIO_F32 : AUDIO_S16;
want.channels = 2;
// -------------------------------------------------------------------------------
want.callback = mixCallback;
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 24-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)
{
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_24;
config.specialFlags |= BITDEPTH_16;
if (have.format == AUDIO_F32)
{
newBitDepth = 24;
config.specialFlags &= ~BITDEPTH_16;
config.specialFlags |= BITDEPTH_24;
}
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);
if ((config.specialFlags2 & DITHERED_AUDIO) && newBitDepth == 24)
config.specialFlags2 &= ~DITHERED_AUDIO;
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_24 + BUFFSIZE_512 + BUFFSIZE_1024 + BUFFSIZE_2048);
setLastWorkingAudioDevName();
// update config audio radio buttons if we're on that screen at the moment
if (editor.ui.configScreenShown && editor.currConfigScreen == CONFIG_SCREEN_IO_DEVICES)
showConfigScreen();
updateWavRendererSettings();
setAudioAmp(config.boostLevel, config.masterVol, (config.specialFlags & BITDEPTH_24) ? true : false);
// don't call stopVoices() in this routine
for (uint8_t i = 0; i < MAX_VOICES; i++)
stopVoice(i);
stopAllScopes();
pmpLeft = 0; // reset sample counter
calcReplayRate(audio.freq);
if (song.speed == 0)
song.speed = 125;
setSpeed(song.speed); // this is important
updateSendAudSamplesRoutine(false);
audio.resetSyncTickTimeFlag = true;
return true;
}
void closeAudio(void)
{
if (audio.dev > 0)
{
SDL_PauseAudioDevice(audio.dev, true);
SDL_CloseAudioDevice(audio.dev);
audio.dev = 0;
}
freeAudioBuffers();
}