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Some optimizations to the Paula sound chip emu code (removing common sub expressions and stuff)

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svn-id: r27761
This commit is contained in:
Max Horn 2007-06-28 19:47:47 +00:00
parent 6dfa44dbd5
commit 21aa642e7a
2 changed files with 63 additions and 69 deletions
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129
sound/mods/paula.cpp
View file

@ -59,26 +59,7 @@ void Paula::clearVoice(byte voice) {
_voice[voice].offset = 0;
}
static inline void mix(int16 *&buf, int8 data, byte volume, byte panning, bool stereo) {
const int32 tmp = ((int32) data) * volume;
if (stereo) {
*buf++ += (tmp * (255 - panning)) >> 7;
*buf++ += (tmp * panning) >> 7;
} else
*buf++ += tmp;
}
int Paula::readBuffer(int16 *buffer, const int numSamples) {
int voice;
int samples;
int nSamples;
int sLen;
double frequency;
double rate;
double offset;
int16 *p;
const int8 *data;
Common::StackLock lock(_mutex);
memset(buffer, 0, numSamples * 2);
@ -86,6 +67,38 @@ int Paula::readBuffer(int16 *buffer, const int numSamples) {
return numSamples;
}
if (_stereo)
return readBufferIntern<true>(buffer, numSamples);
else
return readBufferIntern<false>(buffer, numSamples);
}
template<bool stereo>
inline void mixBuffer(int16 *&buf, const int8 *data, double &offset, double rate, int end, byte volume, byte panning) {
for (int i = 0; i < end; i++) {
// FIXME: We should avoid using floating point arithmetic here, since
// FP calculations and int<->FP conversions are very expensive on many
// architectures.
// So consider replacing offset and rate with fixed point values...
const int32 tmp = ((int32) data[(int)offset]) * volume;
if (stereo) {
*buf++ += (tmp * (255 - panning)) >> 7;
*buf++ += (tmp * (panning)) >> 7;
} else
*buf++ += tmp;
offset += rate;
}
}
template<bool stereo>
int Paula::readBufferIntern(int16 *buffer, const int numSamples) {
int voice;
int samples;
int nSamples;
samples = _stereo ? numSamples / 2 : numSamples;
while (samples > 0) {
if (_curInt == _intFreq) {
@ -97,74 +110,52 @@ int Paula::readBuffer(int16 *buffer, const int numSamples) {
if (!_voice[voice].data || (_voice[voice].period <= 0))
continue;
frequency = (7093789.2 / 2.0) / _voice[voice].period;
rate = frequency / _rate;
offset = _voice[voice].offset;
sLen = _voice[voice].length;
data = _voice[voice].data;
p = buffer;
double frequency = (7093789.2 / 2.0) / _voice[voice].period;
double rate = frequency / _rate;
double offset = _voice[voice].offset;
int sLen = _voice[voice].length;
const int8 *data = _voice[voice].data;
int16 *p = buffer;
int end = 0;
_voice[voice].volume = MIN((byte) 0x40, _voice[voice].volume);
// If looping has been enabled and we see that we will have to loop
// to generate enough samples, then use the "loop" branch.
if ((_voice[voice].lengthRepeat > 2) &&
((int)(offset + nSamples * rate) >= sLen)) {
int neededSamples = nSamples;
int end = (int)((sLen - offset) / rate);
for (int i = 0; i < end; i++)
mix(p, data[(int)(offset + rate * i)], _voice[voice].volume, _voice[voice].panning, _stereo);
_voice[voice].length = sLen = _voice[voice].lengthRepeat;
_voice[voice].data = data = _voice[voice].dataRepeat;
_voice[voice].offset = offset = 0;
neededSamples -= end;
while (neededSamples > 0) {
if (neededSamples >= (int) ((sLen - offset) / rate)) {
end = MIN(neededSamples, (int)((sLen - offset) / rate));
if (end == 0) {
// This means that "rate" is too high, bigger than the sample size.
// So we scale it down according to the euclidean algorithm.
while (rate > (sLen - offset))
rate -= (sLen - offset);
end = (int)((sLen - offset) / rate);
end = MIN(neededSamples, (int)((sLen - offset) / rate));
}
for (int i = 0; i < end; i++)
mix(p, data[(int)(offset + rate * i)], _voice[voice].volume, _voice[voice].panning, _stereo);
mixBuffer<stereo>(p, data, offset, rate, end, _voice[voice].volume, _voice[voice].panning);
_voice[voice].offset = offset;
neededSamples -= end;
// If we read beyond the sample end, loop back to the start.
if (ceil(_voice[voice].offset) >= sLen) {
_voice[voice].data = data = _voice[voice].dataRepeat;
_voice[voice].length = sLen =
_voice[voice].lengthRepeat;
_voice[voice].length = sLen = _voice[voice].lengthRepeat;
_voice[voice].offset = offset = 0;
neededSamples -= end;
} else {
for (int i = 0; i < neededSamples; i++)
mix(p, data[(int)(offset + rate * i)], _voice[voice].volume, _voice[voice].panning, _stereo);
_voice[voice].offset += rate * neededSamples;
if (ceil(_voice[voice].offset) >= sLen) {
_voice[voice].data = data = _voice[voice].dataRepeat;
_voice[voice].length = sLen =
_voice[voice].lengthRepeat;
_voice[voice].offset = offset = 0;
}
neededSamples = 0;
}
}
} else {
if (offset < sLen) {
if ((int)(offset + nSamples * rate) >= sLen) {
// The end of the sample is the limiting factor
if (offset < sLen) { // Sample data left?
end = MIN(nSamples, (int)((sLen - offset) / rate));
int end = (int)((sLen - offset) / rate);
for (int i = 0; i < end; i++)
mix(p, data[(int)(offset + rate * i)], _voice[voice].volume, _voice[voice].panning, _stereo);
_voice[voice].offset = sLen;
} else {
// The requested number of samples is the limiting
// factor, not the sample
for (int i = 0; i < nSamples; i++)
mix(p, data[(int)(offset + rate * i)], _voice[voice].volume, _voice[voice].panning, _stereo);
_voice[voice].offset += rate * nSamples;
}
mixBuffer<stereo>(p, data, offset, rate, end, _voice[voice].volume, _voice[voice].panning);
_voice[voice].offset = offset;
}
}
}

3
sound/mods/paula.h
View file

@ -127,6 +127,9 @@ private:
int _intFreq;
int _curInt;
bool _playing;
template<bool stereo>
int readBufferIntern(int16 *buffer, const int numSamples);
};
} // End of namespace Audio