/* 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; Paula(bool stereo = false, int rate = 44100, int interruptFreq = 0); ~Paula(); bool playing() const { return _playing; } void setInterruptFreq(int freq) { _curInt = _intFreq = freq; } void clearVoice(byte voice); void clearVoices() { for (int i = 0; i < NUM_VOICES; ++i) clearVoice(i); } void startPlay(void) { _playing = true; } void stopPlay(void) { _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 }; bool _end; Common::Mutex _mutex; virtual void interrupt(void) = 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 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 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.data = data; ch.dataRepeat = dataRepeat; ch.length = length; ch.lengthRepeat = lengthRepeat; ch.offset = intToFrac(offset); } 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; } private: Channel _voice[NUM_VOICES]; const bool _stereo; const int _rate; int _intFreq; int _curInt; bool _playing; template int readBufferIntern(int16 *buffer, const int numSamples); }; } // End of namespace Audio #endif