void SoundEngine::initStreaming(int timerChannel)
{
memset(buffer, 0, sizeof buffer);
bufferIndex = 0;
/*
* Next timer tick will populate the first half of the buffer
* (bufferIndex=0) but by the time we get there, the audio
* hardware will already be started on the second half. We want
* the sound hardware and Timer 0 to stay synchronized, so that
* they're always operating on opposite halves of 'buffer'.
*/
timerStart(timerChannel,
ClockDivider_256,
timerFreqToTicks_256(SAMPLE_RATE / SAMPLES_PER_FRAME),
timerCallback);
/*
* Start playing a circular sound buffer that holds 2 frames.
*/
SCHANNEL_SOURCE(CHANNEL_PCSPEAKER) = (uint32_t) &buffer[0];
SCHANNEL_REPEAT_POINT(CHANNEL_PCSPEAKER) = 0;
SCHANNEL_LENGTH(CHANNEL_PCSPEAKER) = BUFFER_SIZE / sizeof(uint32_t);
SCHANNEL_TIMER(CHANNEL_PCSPEAKER) = SOUND_FREQ(SAMPLE_RATE);
SCHANNEL_CR(CHANNEL_PCSPEAKER) = SCHANNEL_ENABLE |
SOUND_VOL(127) |
SOUND_PAN(64) |
SOUND_REPEAT |
SOUND_FORMAT_8BIT;
}
//---------------------------------------------------------------------------------
void startSound(int sampleRate, const void* data, u32 bytes, u8 channel, u8 vol, u8 pan, u8 format) {
//---------------------------------------------------------------------------------
SCHANNEL_TIMER(channel) = SOUND_FREQ(sampleRate);
SCHANNEL_SOURCE(channel) = (u32)data;
SCHANNEL_LENGTH(channel) = bytes >> 2 ;
SCHANNEL_CR(channel) = SCHANNEL_ENABLE | SOUND_ONE_SHOT | SOUND_VOL(vol) | SOUND_PAN(pan) | (format==1?SOUND_8BIT:SOUND_16BIT);
}
void startSound(int sampleRate, const void* data, uint32 bytes, u8 channel, u8 vol, u8 pan, u8 format)
{
SCHANNEL_TIMER(channel) = SOUND_FREQ(sampleRate);
SCHANNEL_SOURCE(channel) = (u32)data;
SCHANNEL_LENGTH(channel) = bytes >> 2;
u32 form = 0;
switch(format)
{
case 0:
form = SOUND_FORMAT_16BIT;
break;
case 1:
form = SOUND_FORMAT_8BIT;
break;
case 2:
form = SOUND_FORMAT_ADPCM;
break;
}
SCHANNEL_CR(channel) = SCHANNEL_ENABLE | SOUND_ONE_SHOT | SOUND_VOL(vol) | SOUND_PAN(pan) | form;
}
// Original FSS Function: Note_On
int Track::NoteOn(int key, int vel, int len)
{
auto sbnk = this->ply->sseq->bank;
if (this->patch >= sbnk->instruments.size())
return -1;
bool bIsPCM = true;
Channel *chn = nullptr;
int nCh = -1;
auto &instrument = sbnk->instruments[this->patch];
const SBNKInstrumentRange *noteDef = nullptr;
int fRecord = instrument.record;
if (fRecord == 16)
{
if (!(instrument.ranges[0].lowNote <= key && key <= instrument.ranges[instrument.ranges.size() - 1].highNote))
return -1;
int rn = key - instrument.ranges[0].lowNote;
noteDef = &instrument.ranges[rn];
fRecord = noteDef->record;
}
else if (fRecord == 17)
{
size_t reg, ranges;
for (reg = 0, ranges = instrument.ranges.size(); reg < ranges; ++reg)
if (key <= instrument.ranges[reg].highNote)
break;
if (reg == ranges)
return -1;
noteDef = &instrument.ranges[reg];
fRecord = noteDef->record;
}
if (!fRecord)
return -1;
else if (fRecord == 1)
{
if (!noteDef)
noteDef = &instrument.ranges[0];
}
else if (fRecord < 4)
{
// PSG
// fRecord = 2 -> PSG tone, pNoteDef->wavid -> PSG duty
// fRecord = 3 -> PSG noise
bIsPCM = false;
if (!noteDef)
noteDef = &instrument.ranges[0];
if (fRecord == 3)
{
nCh = this->ply->ChannelAlloc(TYPE_NOISE, this->prio);
if (nCh < 0)
return -1;
chn = &this->ply->channels[nCh];
chn->tempReg.CR = SOUND_FORMAT_PSG | SCHANNEL_ENABLE;
}
else
{
nCh = this->ply->ChannelAlloc(TYPE_PSG, this->prio);
if (nCh < 0)
return -1;
chn = &this->ply->channels[nCh];
chn->tempReg.CR = SOUND_FORMAT_PSG | SCHANNEL_ENABLE | SOUND_DUTY(noteDef->swav & 0x7);
}
// TODO: figure out what pNoteDef->tnote means for PSG channels
chn->tempReg.TIMER = -SOUND_FREQ(440 * 8); // key #69 (A4)
chn->reg.samplePosition = -1;
chn->reg.psgX = 0x7FFF;
}
if (bIsPCM)
{
nCh = this->ply->ChannelAlloc(TYPE_PCM, this->prio);
if (nCh < 0)
return -1;
chn = &this->ply->channels[nCh];
auto swav = &sbnk->waveArc[noteDef->swar]->swavs.find(noteDef->swav)->second;
chn->tempReg.CR = SOUND_FORMAT(swav->waveType & 3) | SOUND_LOOP(!!swav->loop) | SCHANNEL_ENABLE;
chn->tempReg.SOURCE = swav;
chn->tempReg.TIMER = swav->time;
chn->tempReg.REPEAT_POINT = swav->loopOffset;
chn->tempReg.LENGTH = swav->nonLoopLength;
chn->reg.samplePosition = -3;
}
chn->state = CS_START;
chn->trackId = this->trackId;
chn->flags.reset();
chn->prio = this->prio;
chn->key = key;
chn->orgKey = bIsPCM ? noteDef->noteNumber : 69;
chn->velocity = Cnv_Sust(vel);
chn->pan = static_cast<int>(noteDef->pan) - 64;
chn->modDelayCnt = 0;
chn->modCounter = 0;
chn->noteLength = len;
chn->reg.sampleIncrease = 0;
chn->attackLvl = Cnv_Attack(this->a == 0xFF ? noteDef->attackRate : this->a);
chn->decayRate = Cnv_Fall(this->d == 0xFF ? noteDef->decayRate : this->d);
chn->sustainLvl = this->s == 0xFF ? noteDef->sustainLevel : this->s;
chn->releaseRate = Cnv_Fall(this->r == 0xFF ? noteDef->releaseRate : this->r);
chn->UpdateVol(*this);
chn->UpdatePan(*this);
chn->UpdateTune(*this);
chn->UpdateMod(*this);
chn->UpdatePorta(*this);
this->portaKey = key;
return nCh;
}