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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.
*
* $URL$
* $Id$
*
*/
#ifndef SOUND_MODS_PAULA_H
#define SOUND_MODS_PAULA_H
#include "sound/audiostream.h"
#include "common/frac.h"
#include "common/mutex.h"
namespace Audio {
/**
* Emulation of the "Paula" Amiga music chip
* The interrupt frequency specifies the number of mixed wavesamples between
* calls of the interrupt method
*/
class Paula : public AudioStream {
public:
static const int NUM_VOICES = 4;
enum {
kPalSystemClock = 7093790,
kNtscSystemClock = 7159090,
kPalCiaClock = kPalSystemClock / 10,
kNtscCiaClock = kNtscSystemClock / 10,
kPalPaulaClock = kPalSystemClock / 2,
kNtscPauleClock = kNtscSystemClock / 2
};
Paula(bool stereo = false, int rate = 44100, uint interruptFreq = 0);
~Paula();
bool playing() const { return _playing; }
void setTimerBaseValue( uint32 ticksPerSecond ) { _timerBase = ticksPerSecond; }
uint32 getTimerBaseValue() { return _timerBase; }
void setSingleInterrupt(uint sampleDelay) { assert(sampleDelay < _intFreq); _curInt = sampleDelay; }
void setSingleInterruptUnscaled(uint timerDelay) {
setSingleInterrupt((uint)(((double)timerDelay * getRate()) / _timerBase));
}
void setInterruptFreq(uint sampleDelay) { _intFreq = sampleDelay; _curInt = 0; }
void setInterruptFreqUnscaled(uint timerDelay) {
setInterruptFreq((uint)(((double)timerDelay * getRate()) / _timerBase));
}
void clearVoice(byte voice);
void clearVoices() { for (int i = 0; i < NUM_VOICES; ++i) clearVoice(i); }
void startPlay() { _playing = true; }
void stopPlay() { _playing = false; }
void pausePlay(bool pause) { _playing = !pause; }
// AudioStream API
int readBuffer(int16 *buffer, const int numSamples);
bool isStereo() const { return _stereo; }
bool endOfData() const { return _end; }
int getRate() const { return _rate; }
protected:
struct Channel {
const int8 *data;
const int8 *dataRepeat;
uint32 length;
uint32 lengthRepeat;
int16 period;
byte volume;
frac_t offset;
byte panning; // For stereo mixing: 0 = far left, 255 = far right
int dmaCount;
};
bool _end;
Common::Mutex _mutex;
virtual void interrupt() = 0;
void startPaula() {
_playing = true;
_end = false;
}
void stopPaula() {
_playing = false;
_end = true;
}
void setChannelPanning(byte channel, byte panning) {
assert(channel < NUM_VOICES);
_voice[channel].panning = panning;
}
void disableChannel(byte channel) {
assert(channel < NUM_VOICES);
_voice[channel].data = 0;
}
void enableChannel(byte channel) {
assert(channel < NUM_VOICES);
Channel &ch = _voice[channel];
ch.data = ch.dataRepeat;
ch.length = ch.lengthRepeat;
// actually first 2 bytes are dropped?
ch.offset = intToFrac(0);
// ch.period = ch.periodRepeat;
}
void setChannelPeriod(byte channel, int16 period) {
assert(channel < NUM_VOICES);
_voice[channel].period = period;
}
void setChannelVolume(byte channel, byte volume) {
assert(channel < NUM_VOICES);
_voice[channel].volume = volume;
}
void setChannelSampleStart(byte channel, const int8 *data) {
assert(channel < NUM_VOICES);
_voice[channel].dataRepeat = data;
}
void setChannelSampleLen(byte channel, uint32 length) {
assert(channel < NUM_VOICES);
assert(length < 32768/2);
_voice[channel].lengthRepeat = 2 * length;
}
void setChannelData(uint8 channel, const int8 *data, const int8 *dataRepeat, uint32 length, uint32 lengthRepeat, int32 offset = 0) {
assert(channel < NUM_VOICES);
// For now, we only support 32k samples, as we use 16bit fixed point arithmetics.
// If this ever turns out to be a problem, we can still enhance this code.
assert(0 <= offset && offset < 32768);
assert(length < 32768);
assert(lengthRepeat < 32768);
Channel &ch = _voice[channel];
ch.dataRepeat = data;
ch.lengthRepeat = length;
enableChannel(channel);
ch.offset = intToFrac(offset);
ch.dataRepeat = dataRepeat;
ch.lengthRepeat = lengthRepeat;
}
void setChannelOffset(byte channel, frac_t offset) {
assert(channel < NUM_VOICES);
assert(0 <= offset);
_voice[channel].offset = offset;
}
frac_t getChannelOffset(byte channel) {
assert(channel < NUM_VOICES);
return _voice[channel].offset;
}
int getChannelDmaCount(byte channel) {
assert(channel < NUM_VOICES);
return _voice[channel].dmaCount;
}
void setChannelDmaCount(byte channel, int dmaVal = 0) {
assert(channel < NUM_VOICES);
_voice[channel].dmaCount = dmaVal;
}
void setAudioFilter(bool enable) {
// TODO: implement
}
private:
Channel _voice[NUM_VOICES];
const bool _stereo;
const int _rate;
const double _periodScale;
uint _intFreq;
uint _curInt;
uint32 _timerBase;
bool _playing;
template<bool stereo>
int readBufferIntern(int16 *buffer, const int numSamples);
};
} // End of namespace Audio
#endif
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