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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((double)kPalPaulaClock / rate), _intFreq(interruptFreq) {
clearVoices();
_voice[0].panning = 191;
_voice[1].panning = 63;
_voice[2].panning = 63;
_voice[3].panning = 191;
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].volume = 0;
_voice[voice].offset = 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 int mixBuffer(int16 *&buf, const int8 *data, Paula::Offset &offset, frac_t rate, int neededSamples, uint bufSize, byte volume, byte panning) {
int samples;
for (samples = 0; samples < neededSamples && offset.int_off < bufSize; ++samples) {
const int32 tmp = ((int32) data[offset.int_off]) * volume;
if (stereo) {
*buf++ += (tmp * (255 - panning)) >> 7;
*buf++ += (tmp * (panning)) >> 7;
} else
*buf++ += tmp;
// Step to next source sample
offset.rem_off += rate;
if (offset.rem_off >= FRAC_ONE) {
offset.int_off += fracToInt(offset.rem_off);
offset.rem_off &= FRAC_LO_MASK;
}
}
return samples;
}
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 in the PAL
// version and at 7.1590905 MHz in the NTSC version. We divide this
// by the requested the requested output sampling rate _rate
// (typically 44.1 kHz or 22.05 kHz) obtaining the value _periodScale.
// This is then divided by the "period" of the channel we are
// processing, to obtain the correct output 'rate'.
frac_t rate = doubleToFrac(_periodScale / _voice[voice].period);
// Cap the volume
_voice[voice].volume = MIN((byte) 0x40, _voice[voice].volume);
Channel &ch = _voice[voice];
int16 *p = buffer;
int neededSamples = nSamples;
assert(ch.offset.int_off < ch.length);
// Mix the generated samples into the output buffer
neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning);
// Wrap around if necessary
if (ch.offset.int_off >= ch.length) {
// Important: Wrap around the offset *before* updating the voice length.
// Otherwise, if length != lengthRepeat we would wrap incorrectly.
// Note: If offset >= 2*len ever occurs, the following would be wrong;
// instead of subtracting, we then should compute the modulus using "%=".
// Since that requires a division and is slow, and shouldn't be necessary
// in practice anyway, we only use subtraction.
ch.offset.int_off -= ch.length;
ch.dmaCount++;
ch.data = ch.dataRepeat;
ch.length = ch.lengthRepeat;
}
// If we have not yet generated enough samples, and looping is active: loop!
if (neededSamples > 0 && ch.length > 2) {
// Repeat as long as necessary.
while (neededSamples > 0) {
// Mix the generated samples into the output buffer
neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning);
if (ch.offset.int_off >= ch.length) {
// Wrap around. See also the note above.
ch.offset.int_off -= ch.length;
ch.dmaCount++;
}
}
}
}
buffer += _stereo ? nSamples * 2 : nSamples;
_curInt -= nSamples;
samples -= nSamples;
}
return numSamples;
}
} // End of namespace Audio
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