scummvm/audio/mods/paula.h
Sven Hesse b49c7fa644 AUDIO: Implement low-pass filtering for Paula
Paula low-pass filtering, as implemented by UAE.

The Amiga has two filtering circuits: a static RC filter
(only) on the A500, and an LED filter that can be enabled
or disabled dynamically.

By default, the Paula now doesn't apply the static RC
filter, but allows for enabling the LED filter (with
setAudioFilter()).

NOTE: At the moment, this code still uses floating point
arithmetics! It also calls tan() three times per
instantiation.
2019-06-20 16:00:59 +02:00

228 lines
5.9 KiB
C++

/* 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.
*
*/
#ifndef AUDIO_MODS_PAULA_H
#define AUDIO_MODS_PAULA_H
#include "audio/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,
kNtscPaulaClock = kNtscSystemClock / 2
};
enum FilterMode {
kFilterModeNone = 0,
kFilterModeA500,
kFilterModeA1200
};
/* TODO: Document this */
struct Offset {
uint int_off; // integral part of the offset
frac_t rem_off; // fractional part of the offset, at least 0 and less than 1
explicit Offset(int off = 0) : int_off(off), rem_off(0) {}
};
struct FilterState {
FilterMode mode;
bool ledFilter;
float a0[3];
float rc[NUM_VOICES][5];
};
Paula(bool stereo = false, int rate = 44100, uint interruptFreq = 0,
FilterMode filterMode = kFilterModeA1200);
~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() { filterResetState(); _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;
Offset 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 = Offset(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);
Channel &ch = _voice[channel];
ch.dataRepeat = data;
ch.lengthRepeat = length;
enableChannel(channel);
ch.offset = Offset(offset);
ch.dataRepeat = dataRepeat;
ch.lengthRepeat = lengthRepeat;
}
void setChannelOffset(byte channel, Offset offset) {
assert(channel < NUM_VOICES);
_voice[channel].offset = offset;
}
Offset 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) {
_filterState.ledFilter = enable;
}
private:
Channel _voice[NUM_VOICES];
const bool _stereo;
const int _rate;
const double _periodScale;
uint _intFreq;
uint _curInt;
uint32 _timerBase;
bool _playing;
FilterState _filterState;
template<bool stereo>
int readBufferIntern(int16 *buffer, const int numSamples);
void filterResetState();
float filterCalculateA0(int rate, int cutoff);
};
} // End of namespace Audio
#endif