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/* ScummVM - Graphic Adventure Engine
*
* ScummVM is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*
*/
#include "sound/mods/paula.h"
namespace Audio {
Paula::Paula(bool stereo, int rate, uint interruptFreq) :
_stereo(stereo), _rate(rate), _periodScale((kPalSystemClock / 2.0) / rate), _intFreq(interruptFreq) {
clearVoices();
_voice[0].panning = 63;
_voice[1].panning = 191;
_voice[2].panning = 191;
_voice[3].panning = 63;
if (_intFreq == 0)
_intFreq = _rate;
_curInt = 0;
_timerBase = 1;
_playing = false;
_end = true;
}
Paula::~Paula() {
}
void Paula::clearVoice(byte voice) {
assert(voice < NUM_VOICES);
_voice[voice].data = 0;
_voice[voice].dataRepeat = 0;
_voice[voice].length = 0;
_voice[voice].lengthRepeat = 0;
_voice[voice].period = 0;
_voice[voice].periodRepeat = 0;
_voice[voice].volume = 0;
_voice[voice].offset = 0;
_voice[voice].dmaCount = 0;
}
int Paula::readBuffer(int16 *buffer, const int numSamples) {
Common::StackLock lock(_mutex);
memset(buffer, 0, numSamples * 2);
if (!_playing) {
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, frac_t &offset, frac_t rate, int end, byte volume, byte panning) {
for (int i = 0; i < end; i++) {
const int32 tmp = ((int32) data[fracToInt(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 samples = _stereo ? numSamples / 2 : numSamples;
while (samples > 0) {
// Handle 'interrupts'. This gives subclasses the chance to adjust the channel data
// (e.g. insert new samples, do pitch bending, whatever).
if (_curInt == 0) {
_curInt = _intFreq;
interrupt();
}
// Compute how many samples to generate: at most the requested number of samples,
// of course, but we may stop earlier when an 'interrupt' is expected.
const uint nSamples = MIN((uint)samples, _curInt);
// Loop over the four channels of the emulated Paula chip
for (int voice = 0; voice < NUM_VOICES; voice++) {
// No data, or paused -> skip channel
if (!_voice[voice].data || (_voice[voice].period <= 0))
continue;
// The Paula chip apparently run at 7.0937892 MHz. We combine this with
// the requested output sampling rate (typicall 44.1 kHz or 22.05 kHz)
// as well as the "period" of the channel we are processing right now,
// to compute the correct output 'rate'.
frac_t rate = doubleToFrac(_periodScale / _voice[voice].period);
// Cap the volume
_voice[voice].volume = MIN((byte) 0x40, _voice[voice].volume);
// Cache some data (helps the compiler to optimize the code, by
// indirectly telling it that no data aliasing can occur).
frac_t offset = _voice[voice].offset;
frac_t sLen = intToFrac(_voice[voice].length);
const int8 *data = _voice[voice].data;
int dmaCount = _voice[voice].dmaCount;
int16 *p = buffer;
int end = 0;
int neededSamples = nSamples;
// Compute the number of samples to generate; that is, either generate
// just as many as were requested, or until the buffer is used up.
// Note that dividing two frac_t yields an integer (as the denominators
// cancel out each other).
// Note that 'end' could be 0 here. No harm in that :-).
end = MIN(neededSamples, (int)((sLen - offset + rate - 1) / rate));
mixBuffer<stereo>(p, data, offset, rate, end, _voice[voice].volume, _voice[voice].panning);
neededSamples -= end;
// If we have not yet generated enough samples, and looping is active: loop!
if (neededSamples > 0 && _voice[voice].lengthRepeat > 2) {
// At this point we know that we have used up all samples in the buffer, so reset it.
_voice[voice].data = data = _voice[voice].dataRepeat;
_voice[voice].length = _voice[voice].lengthRepeat;
sLen = intToFrac(_voice[voice].length);
// TODO: the value in offset shouldnt be dropped but scaled to new rate
if (_voice[voice].period != _voice[voice].periodRepeat) {
_voice[voice].period = _voice[voice].periodRepeat;
rate = doubleToFrac(_periodScale / _voice[voice].period);
}
// If the "rate" exceeds the sample rate, we would have to perform constant
// wrap arounds. So, apply the first step of the euclidean algorithm to
// achieve the same more efficiently: Take rate modulo sLen
// TODO: This messes up dmaCount
if (sLen < rate)
rate %= sLen;
// Repeat as long as necessary.
while (neededSamples > 0) {
// TODO offset -= sLen, but only if same rate otherwise need to scale first
offset &= FRAC_LO_MASK;
dmaCount++;
// Compute the number of samples to generate (see above) and mix 'em.
end = MIN(neededSamples, (int)((sLen - offset + rate - 1) / rate));
mixBuffer<stereo>(p, data, offset, rate, end, _voice[voice].volume, _voice[voice].panning);
neededSamples -= end;
}
}
// Write back the cached data
_voice[voice].offset = offset;
_voice[voice].dmaCount = dmaCount;
}
buffer += _stereo ? nSamples * 2 : nSamples;
_curInt -= nSamples;
samples -= nSamples;
}
return numSamples;
}
} // End of namespace Audio
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