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-rw-r--r--sound/mods/paula.cpp131
-rw-r--r--sound/mods/paula.h3
2 files changed, 64 insertions, 70 deletions
diff --git a/sound/mods/paula.cpp b/sound/mods/paula.cpp
index 988ca3f74d..9061891c35 100644
--- a/sound/mods/paula.cpp
+++ b/sound/mods/paula.cpp
@@ -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
-
- 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;
- }
+ if (offset < sLen) { // Sample data left?
+ end = MIN(nSamples, (int)((sLen - offset) / rate));
+
+ mixBuffer<stereo>(p, data, offset, rate, end, _voice[voice].volume, _voice[voice].panning);
+ _voice[voice].offset = offset;
}
}
}
diff --git a/sound/mods/paula.h b/sound/mods/paula.h
index 28fed004a7..1e196daf40 100644
--- a/sound/mods/paula.h
+++ b/sound/mods/paula.h
@@ -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
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-16 01:35:45 +0000