1 files changed, 212 insertions, 0 deletions
diff --git a/audio/mods/paula.cpp b/audio/mods/paula.cpp
new file mode 100644
index 0000000000..ef841ac9bf
--- /dev/null
+++ b/audio/mods/paula.cpp
@@ -0,0 +1,212 @@
+/* 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 "audio/mods/paula.h"
+#include "audio/null.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_t)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
+
+
+//	Plugin interface
+//	(This can only create a null driver since apple II gs support seeems not to be implemented
+//  and also is not part of the midi driver architecture. But we need the plugin for the options
+//  menu in the launcher and for MidiDriver::detectDevice() which is more or less used by all engines.)
+
+class AmigaMusicPlugin : public NullMusicPlugin {
+public:
+	const char *getName() const {
+		return _s("Amiga Audio Emulator");
+	}
+
+	const char *getId() const {
+		return "amiga";
+	}
+
+	MusicDevices getDevices() const;
+};
+
+MusicDevices AmigaMusicPlugin::getDevices() const {
+	MusicDevices devices;
+	devices.push_back(MusicDevice(this, "", MT_AMIGA));
+	return devices;
+}
+
+//#if PLUGIN_ENABLED_DYNAMIC(AMIGA)
+	//REGISTER_PLUGIN_DYNAMIC(AMIGA, PLUGIN_TYPE_MUSIC, AmigaMusicPlugin);
+//#else
+	REGISTER_PLUGIN_STATIC(AMIGA, PLUGIN_TYPE_MUSIC, AmigaMusicPlugin);
+//#endif