diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/mods/paula.cpp | 131 | ||||
-rw-r--r-- | sound/mods/paula.h | 3 |
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 |