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audio/mods/paula.h

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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 3 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, see <http://www.gnu.org/licenses/>.
+ *
+ */
+
+#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;
+	// Default panning value for left channels.
+	static const int PANNING_LEFT = 63;
+	// Default panning value for right channels.
+	static const int PANNING_RIGHT = 191;
+	enum {
+		kPalSystemClock  = 7093790,
+		kNtscSystemClock = 7159090,
+		kPalCiaClock     = kPalSystemClock / 10,
+		kNtscCiaClock    = kNtscSystemClock / 10,
+		kPalPaulaClock   = kPalSystemClock / 2,
+		kNtscPaulaClock  = kNtscSystemClock / 2
+	};
+
+	enum FilterMode {
+		kFilterModeNone = 0,
+		kFilterModeA500,
+		kFilterModeA1200,
+
+#if defined(__DS__)
+		kFilterModeDefault = kFilterModeNone
+#else
+		kFilterModeDefault = kFilterModeA1200
+#endif
+	};
+
+	/* 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 = kFilterModeDefault, int periodScaleDivisor = 1);
+	~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 interrupt;
+	};
+
+	bool _end;
+	Common::Mutex &_mutex;
+
+	virtual void interrupt() = 0;
+
+	virtual void interruptChannel(byte channel) { }
+
+	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 setChannelInterrupt(byte channel, bool enable) {
+		assert(channel < NUM_VOICES);
+		_voice[channel].interrupt = enable;
+	}
+
+	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