[Scummvm-cvs-logs] CVS: residual/mixer audiostream.cpp,NONE,1.1 audiostream.h,NONE,1.1 mixer.cpp,NONE,1.1 mixer.h,NONE,1.1 rate.cpp,NONE,1.1 rate.h,NONE,1.1

[Pawel Kolodziejski]

Update of /cvsroot/scummvm/residual/mixer
In directory sc8-pr-cvs1:/tmp/cvs-serv13367/mixer

Added Files:
	audiostream.cpp audiostream.h mixer.cpp mixer.h rate.cpp 
	rate.h 
Log Message:
added mixer code, handle smush sound, handle smush filenames in lua

--- NEW FILE: audiostream.cpp ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

#include "../stdafx.h"
#include "../debug.h"
#include "mixer.h"
#include "audiostream.h"


// This used to be an inline template function, but
// buggy template function handling in MSVC6 forced
// us to go with the macro approach. So far this is
// the only template function that MSVC6 seemed to
// compile incorrectly. Knock on wood.
#define READSAMPLE(is16Bit, isUnsigned, ptr) \
	((is16Bit ? READ_BE_UINT16(ptr) : (*ptr << 8)) ^ (isUnsigned ? 0x8000 : 0))

template<bool stereo, bool is16Bit, bool isUnsigned>
class LinearMemoryStream : public AudioInputStream {
protected:
	const byte *_ptr;
	const byte *_end;
	const byte *_loopPtr;
	const byte *_loopEnd;

	inline int16 readIntern() {
		//assert(_ptr < _end);
		int16 val = READSAMPLE(is16Bit, isUnsigned, _ptr);
		_ptr += (is16Bit ? 2 : 1);
		if (_loopPtr && eosIntern()) {
			_ptr = _loopPtr;
			_end = _loopEnd;
		}
		return val;
	}
	inline bool eosIntern() const	{ return _ptr >= _end; };
public:
	LinearMemoryStream(const byte *ptr, uint len, uint loopOffset, uint loopLen)
		: _ptr(ptr), _end(ptr+len), _loopPtr(0), _loopEnd(0) {

		// Verify the buffer sizes are sane
		if (is16Bit && stereo)
			assert((len & 3) == 0 && (loopLen & 3) == 0);
		else if (is16Bit || stereo)
			assert((len & 1) == 0 && (loopLen & 1) == 0);

		if (loopLen) {
			_loopPtr = _ptr + loopOffset;
			_loopEnd = _loopPtr + loopLen;
		}
		if (stereo)	// Stereo requires even sized data
			assert(len % 2 == 0);
	}
	int readBuffer(int16 *buffer, const int numSamples);

	int16 read()				{ return readIntern(); }
	bool eos() const			{ return eosIntern(); }
	bool isStereo() const		{ return stereo; }
};

template<bool stereo, bool is16Bit, bool isUnsigned>
int LinearMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
	int samples = 0;
	while (samples < numSamples && !eosIntern()) {
		const int len = MIN(numSamples, samples + (int)(_end - _ptr) / (is16Bit ? 2 : 1));
		while (samples < len) {
			*buffer++ = READSAMPLE(is16Bit, isUnsigned, _ptr);
			_ptr += (is16Bit ? 2 : 1);
			samples++;
		}
		// Loop, if looping was specified
		if (_loopPtr && eosIntern()) {
			_ptr = _loopPtr;
			_end = _loopEnd;
		}
	}
	return samples;
}

// Wrapped memory stream, to be used by the ChannelStream class (and possibly others?)
template<bool stereo, bool is16Bit, bool isUnsigned>
class WrappedMemoryStream : public WrappedAudioInputStream {
protected:
	byte *_bufferStart;
	byte *_bufferEnd;
	byte *_pos;
	byte *_end;

	inline int16 readIntern();
	inline bool eosIntern() const { return _end == _pos; };
public:
	WrappedMemoryStream(uint bufferSize);
	~WrappedMemoryStream()		{ free(_bufferStart); }
	int readBuffer(int16 *buffer, const int numSamples);

	int16 read()				{ return readIntern(); }
	bool eos() const			{ return eosIntern(); }
	bool isStereo() const		{ return stereo; }

	void append(const byte *data, uint32 len);
};


template<bool stereo, bool is16Bit, bool isUnsigned>
WrappedMemoryStream<stereo, is16Bit, isUnsigned>::WrappedMemoryStream(uint bufferSize) {

	// Verify the buffer size is sane
	if (is16Bit && stereo)
		assert((bufferSize & 3) == 0);
	else if (is16Bit || stereo)
		assert((bufferSize & 1) == 0);

	_bufferStart = (byte *)malloc(bufferSize);
	_pos = _end = _bufferStart;
	_bufferEnd = _bufferStart + bufferSize;
}

template<bool stereo, bool is16Bit, bool isUnsigned>
inline int16 WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readIntern() {
	//assert(_pos != _end);
	int16 val = READSAMPLE(is16Bit, isUnsigned, _pos);
	_pos += (is16Bit ? 2 : 1);

	// Wrap around?
	if (_pos >= _bufferEnd)
		_pos = _pos - (_bufferEnd - _bufferStart);

	return val;
}

template<bool stereo, bool is16Bit, bool isUnsigned>
int WrappedMemoryStream<stereo, is16Bit, isUnsigned>::readBuffer(int16 *buffer, const int numSamples) {
	int samples = 0;
	while (samples < numSamples && !eosIntern()) {
		const byte *endMarker = (_pos > _end) ? _bufferEnd : _end;
		const int len = MIN(numSamples, samples + (int)(endMarker - _pos) / (is16Bit ? 2 : 1));
		while (samples < len) {
			*buffer++ = READSAMPLE(is16Bit, isUnsigned, _pos);
			_pos += (is16Bit ? 2 : 1);
			samples++;
		}
		// Wrap around?
		if (_pos >= _bufferEnd)
			_pos = _pos - (_bufferEnd - _bufferStart);
	}
	return samples;
}

template<bool stereo, bool is16Bit, bool isUnsigned>
void WrappedMemoryStream<stereo, is16Bit, isUnsigned>::append(const byte *data, uint32 len) {

	// Verify the buffer size is sane
	if (is16Bit && stereo)
		assert((len & 3) == 0);
	else if (is16Bit || stereo)
		assert((len & 1) == 0);

	if (_end + len > _bufferEnd) {
		// Wrap-around case
		uint32 size_to_end_of_buffer = _bufferEnd - _end;
		len -= size_to_end_of_buffer;
		if ((_end < _pos) || (_bufferStart + len >= _pos)) {
			warning("WrappedMemoryStream: buffer overflow (A)");
			return;
		}
		memcpy(_end, data, size_to_end_of_buffer);
		memcpy(_bufferStart, data + size_to_end_of_buffer, len);
		_end = _bufferStart + len;
	} else {
		if ((_end < _pos) && (_end + len >= _pos)) {
			warning("WrappedMemoryStream: buffer overflow (B)");
			return;
		}
		memcpy(_end, data, len);
		_end += len;
	}
}

template<bool stereo>
static AudioInputStream *makeLinearInputStream(const byte *ptr, uint32 len, bool is16Bit, bool isUnsigned, uint loopOffset, uint loopLen) {
	if (isUnsigned) {
		if (is16Bit)
			return new LinearMemoryStream<stereo, true, true>(ptr, len, loopOffset, loopLen);
		else
			return new LinearMemoryStream<stereo, false, true>(ptr, len, loopOffset, loopLen);
	} else {
		if (is16Bit)
			return new LinearMemoryStream<stereo, true, false>(ptr, len, loopOffset, loopLen);
		else
			return new LinearMemoryStream<stereo, false, false>(ptr, len, loopOffset, loopLen);
	}
}

template<bool stereo>
static WrappedAudioInputStream *makeWrappedInputStream(uint32 len, bool is16Bit, bool isUnsigned) {
	if (isUnsigned) {
		if (is16Bit)
			return new WrappedMemoryStream<stereo, true, true>(len);
		else
			return new WrappedMemoryStream<stereo, false, true>(len);
	} else {
		if (is16Bit)
			return new WrappedMemoryStream<stereo, true, false>(len);
		else
			return new WrappedMemoryStream<stereo, false, false>(len);
	}
}

AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen) {
	const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
	const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
	if (_flags & SoundMixer::FLAG_STEREO) {
		return makeLinearInputStream<true>(ptr, len, is16Bit, isUnsigned, loopOffset, loopLen);
	} else {
		return makeLinearInputStream<false>(ptr, len, is16Bit, isUnsigned, loopOffset, loopLen);
	}
}

WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len) {
	const bool is16Bit = (_flags & SoundMixer::FLAG_16BITS) != 0;
	const bool isUnsigned = (_flags & SoundMixer::FLAG_UNSIGNED) != 0;
	if (_flags & SoundMixer::FLAG_STEREO) {
		return makeWrappedInputStream<true>(len, is16Bit, isUnsigned);
	} else {
		return makeWrappedInputStream<false>(len, is16Bit, isUnsigned);
	}
}

--- NEW FILE: audiostream.h ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

#ifndef AUDIOSTREAM_H
#define AUDIOSTREAM_H

#include "../stdafx.h"
#include "../bits.h"

/**
 * Generic input stream for the resampling code.
 */
class AudioInputStream {
public:
	virtual ~AudioInputStream() {}

	/**
	 * Fill the given buffer with up to numSamples samples.
	 * Returns the actual number of samples read, or -1 if
	 * a critical error occured (note: you *must* check if
	 * this value is less than what you requested, this can
	 * happen when the stream is fully used up).
	 * For stereo stream, buffer will be filled with interleaved
	 * left and right channel samples.
	 *
	 * For maximum efficency, subclasses should always override
	 * the default implementation!
	 */
	virtual int readBuffer(int16 *buffer, const int numSamples) {
		int samples;
		for (samples = 0; samples < numSamples && !eos(); samples++) {
			*buffer++ = read();
		}
		return samples;
	}

	/** Read a singel (16 bit signed) sample from the stream. */
	virtual int16 read() = 0;
	
	/** Is this a stereo stream? */
	virtual bool isStereo() const = 0;
	
	/* End of stream reached? */
	virtual bool eos() const = 0;

	virtual int getRate() const { return -1; }
};

class WrappedAudioInputStream : public AudioInputStream {
public:
	virtual void append(const byte *data, uint32 len) = 0;
};

class ZeroInputStream : public AudioInputStream {
protected:
	int _len;
public:
	ZeroInputStream(uint len) : _len(len) { }
	int readBuffer(int16 *buffer, const int numSamples) {
		int samples = MIN(_len, numSamples);
		memset(buffer, 0, samples * 2);
		_len -= samples;
		return samples;
	}
	int16 read() { assert(_len > 0); _len--; return 0; }
	int size() const { return _len; }
	bool isStereo() const { return false; }
	bool eos() const { return _len <= 0; }
};

class MusicStream : public AudioInputStream {
public:
	virtual int getRate() const = 0;
};


AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len);

#endif

--- NEW FILE: mixer.cpp ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

#include "../stdafx.h"
#include "../bits.h"
#include "../debug.h"

#include "mixer.h"
#include "rate.h"
#include "audiostream.h"

SoundMixer *g_mixer = NULL;

StackLock::StackLock(MutexRef mutex)
	: _mutex(mutex) {
	lock_mutex(_mutex);
}

StackLock::~StackLock() {
	unlock_mutex(_mutex);
}

MutexRef create_mutex() {
	return (MutexRef) SDL_CreateMutex();
}

void lock_mutex(MutexRef mutex) {
	SDL_mutexP((SDL_mutex *) mutex);
}

void unlock_mutex(MutexRef mutex) {
	SDL_mutexV((SDL_mutex *) mutex);
}

void delete_mutex(MutexRef mutex) {
	SDL_DestroyMutex((SDL_mutex *) mutex);
}

/**
 * Channels used by the sound mixer.
 */
class Channel {
protected:
	SoundMixer *_mixer;
	PlayingSoundHandle *_handle;
	RateConverter *_converter;
	AudioInputStream *_input;
	byte _volume;
	int8 _pan;
	bool _paused;

public:
	int _id;

	Channel(SoundMixer *mixer, PlayingSoundHandle *handle)
		: _mixer(mixer), _handle(handle), _converter(0), _input(0), _volume(0), _pan(0), _paused(false), _id(-1) {
		assert(mixer);
	}
	virtual ~Channel();
	void destroy();
	virtual void mix(int16 *data, uint len);
	virtual void pause(bool paused) {
		_paused = paused;
	}
	virtual bool isPaused() {
		return _paused;
	}
	virtual void setChannelVolume(const byte volume) {
		_volume = volume;
	}
	virtual void setChannelPan(const int8 pan) {
		_pan = pan;
	}
	virtual int getVolume() const {
		return _mixer->getVolume();
	}
};

class ChannelRaw : public Channel {
	byte *_ptr;
public:
	ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, byte volume, int8 pan, int id, uint32 loopStart, uint32 loopEnd);
	~ChannelRaw();
};

class ChannelStream : public Channel {
	bool _finished;
public:
	ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size, byte volume, int8 pan);
	void mix(int16 *data, uint len);
	void append(void *sound, uint32 size);
	void finish()					{ _finished = true; }
};

SoundMixer::SoundMixer() {
	_mutex = NULL;

	_premixParam = NULL;
	_premixProc = NULL;
	int i = 0;

	_outputRate = 0;

	_globalVolume = 0;

	_paused = false;

	for (i = 0; i != NUM_CHANNELS; i++)
		_channels[i] = NULL;
}

SoundMixer::~SoundMixer() {
	SDL_CloseAudio();
	for (int i = 0; i != NUM_CHANNELS; i++) {
		delete _channels[i];
	}
	delete_mutex(_mutex);
}

void set_sound_proc(SoundProc proc, void *param) {
	SDL_AudioSpec desired;

	memset(&desired, 0, sizeof(desired));

	/* only one format supported at the moment */
	desired.freq = 22050;
	desired.format = AUDIO_S16SYS;
	desired.channels = 2;
	desired.samples = 2048;
	desired.callback = proc;
	desired.userdata = param;
	if (SDL_OpenAudio(&desired, NULL) != 0) {
		return;
	}
	SDL_PauseAudio(0);
}

void SoundMixer::bindToSystem() {
	_mutex = create_mutex();
	_outputRate = 22050;

	if (_outputRate == 0)
		error("OSystem returned invalid sample rate");

	set_sound_proc(mixCallback, this);
}

void SoundMixer::setupPremix(PremixProc *proc, void *param) {
	StackLock lock(_mutex);
	_premixParam = param;
	_premixProc = proc;
}

int SoundMixer::newStream(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size, byte volume, int8 pan) {
	StackLock lock(_mutex);
	return insertChannel(handle, new ChannelStream(this, handle, sound, size, rate, flags, buffer_size, volume, pan));
}

void SoundMixer::appendStream(PlayingSoundHandle handle, void *sound, uint32 size) {
	StackLock lock(_mutex);
	
	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::appendStream has invalid index %d", index);
		return;
	}

	ChannelStream *chan;
	chan = (ChannelStream *)_channels[index];
	if (!chan) {
		error("Trying to append to nonexistant streamer : %d", index);
	} else {
		chan->append(sound, size);
	}
}

void SoundMixer::endStream(PlayingSoundHandle handle) {
	StackLock lock(_mutex);

	// Simply ignore stop requests for handles of sounds that already terminated
	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::endStream has invalid index %d", index);
		return;
	}

	ChannelStream *chan;
	chan = (ChannelStream *)_channels[index];
	if (!chan) {
		error("Trying to end a nonexistant streamer : %d", index);
	} else {
		chan->finish();
	}
}

int SoundMixer::insertChannel(PlayingSoundHandle *handle, Channel *chan) {
	int index = -1;
	for (int i = 0; i != NUM_CHANNELS; i++) {
		if (_channels[i] == NULL) {
			index = i;
			break;
		}
	}
	if(index == -1) {
		warning("SoundMixer::out of mixer slots");
		delete chan;
		return -1;
	}

	_channels[index] = chan;
	if (handle)
		*handle = index + 1;
	return index;
}

int SoundMixer::playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, int id, byte volume, int8 pan, uint32 loopStart, uint32 loopEnd) {
	StackLock lock(_mutex);

	// Prevent duplicate sounds
	if (id != -1) {
		for (int i = 0; i != NUM_CHANNELS; i++)
			if (_channels[i] != NULL && _channels[i]->_id == id)
				return -1;
	}

	return insertChannel(handle, new ChannelRaw(this, handle, sound, size, rate, flags, volume, pan, id, loopStart, loopEnd));
}

void SoundMixer::mix(int16 *buf, uint len) {
	StackLock lock(_mutex);

	if (_premixProc && !_paused) {
		_premixProc(_premixParam, buf, len);
	} else {
		//  zero the buf out
		memset(buf, 0, 2 * len * sizeof(int16));
	}

	if (!_paused) {
		// now mix all channels
		for (int i = 0; i != NUM_CHANNELS; i++)
			if (_channels[i] && !_channels[i]->isPaused())
				_channels[i]->mix(buf, len);
	}
}

void SoundMixer::mixCallback(void *s, byte *samples, int len) {
	assert(s);
	assert(samples);
	// Len is the number of bytes in the buffer; we divide it by
	// four to get the number of samples (stereo 16 bit).
	((SoundMixer *)s)->mix((int16 *)samples, len >> 2);
}

void SoundMixer::stopAll() {
	StackLock lock(_mutex);
	for (int i = 0; i != NUM_CHANNELS; i++)
		if (_channels[i])
			_channels[i]->destroy();
}

void SoundMixer::stopChannel(int index) {
	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::stop has invalid index %d", index);
		return;
	}

	StackLock lock(_mutex);
	if (_channels[index])
		_channels[index]->destroy();
}

void SoundMixer::stopID(int id) {
	StackLock lock(_mutex);
	for (int i = 0; i != NUM_CHANNELS; i++) {
		if (_channels[i] != NULL && _channels[i]->_id == id) {
			_channels[i]->destroy();
			return;
		}
	}
}

void SoundMixer::stopHandle(PlayingSoundHandle handle) {
	StackLock lock(_mutex);

	// Simply ignore stop requests for handles of sounds that already terminated
	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::stopHandle has invalid index %d", index);
		return;
	}

	if (_channels[index])
		_channels[index]->destroy();
}

void SoundMixer::setChannelVolume(PlayingSoundHandle handle, byte volume) {
	StackLock lock(_mutex);

	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::setChannelVolume has invalid index %d", index);
		return;
	}

	if (_channels[index])
		_channels[index]->setChannelVolume(volume);
}

void SoundMixer::setChannelPan(PlayingSoundHandle handle, int8 pan) {
	StackLock lock(_mutex);

	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::setChannelVolume has invalid index %d", index);
		return;
	}

	if (_channels[index])
		_channels[index]->setChannelPan(pan);
}

void SoundMixer::pauseAll(bool paused) {
	_paused = paused;
}

void SoundMixer::pauseChannel(int index, bool paused) {
	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::pauseChannel has invalid index %d", index);
		return;
	}

	StackLock lock(_mutex);
	if (_channels[index])
		_channels[index]->pause(paused);
}

void SoundMixer::pauseID(int id, bool paused) {
	StackLock lock(_mutex);
	for (int i = 0; i != NUM_CHANNELS; i++) {
		if (_channels[i] != NULL && _channels[i]->_id == id) {
			_channels[i]->pause(paused);
			return;
		}
	}
}

void SoundMixer::pauseHandle(PlayingSoundHandle handle, bool paused) {
	StackLock lock(_mutex);

	// Simply ignore pause/unpause requests for handles of sound that alreayd terminated
	if (handle == 0)
		return;

	int index = handle - 1;

	if ((index < 0) || (index >= NUM_CHANNELS)) {
		warning("soundMixer::pauseHandle has invalid index %d", index);
		return;
	}

	if (_channels[index])
		_channels[index]->pause(paused);
}

void SoundMixer::setVolume(int volume) {
	// Check range
	if (volume > 256)
		volume = 256;
	else if (volume < 0)
		volume = 0;

	_globalVolume = volume;
}

Channel::~Channel() {
	delete _converter;
	delete _input;
	if (_handle)
		*_handle = 0;
}

void Channel::destroy() {
	for (int i = 0; i != SoundMixer::NUM_CHANNELS; i++)
		if (_mixer->_channels[i] == this)
			_mixer->_channels[i] = 0;
	delete this;
}

/* len indicates the number of sample *pairs*. So a value of
   10 means that the buffer contains twice 10 sample, each
   16 bits, for a total of 40 bytes.
 */
void Channel::mix(int16 *data, uint len) {
	assert(_input);
	if (_input->eos()) {
		// TODO: call drain method
		destroy();
	} else {
		assert(_converter);

		// The pan value ranges from -127 to +127. That's 255 different values.
		// From the channel pan/volume and the global volume, we compute the
		// effective volume for the left and right channel.
		// Note the slightly odd divisor: the 255 reflects the fact that
		// the maximal value for _volume is 255, while the 254 is there
		// because the maximal left/right pan value is 2*127 = 254.
		// The value getVolume() returns is in the range 0 - 256.
		// Hence, the vol_l/vol_r values will be in that range, too
		
		int vol = getVolume() * _volume;
		st_volume_t vol_l = (127 - _pan) * vol / (255 * 254);
		st_volume_t vol_r = (127 + _pan) * vol / (255 * 254);

		_converter->flow(*_input, data, len, vol_l, vol_r);
	}
}

/* RAW mixer */
ChannelRaw::ChannelRaw(SoundMixer *mixer, PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, byte volume, int8 pan, int id, uint32 loopStart, uint32 loopEnd)
	: Channel(mixer, handle) {
	_id = id;
	_ptr = (byte *)sound;
	_volume = volume;
	_pan = pan;
	
	// Create the input stream
	if (flags & SoundMixer::FLAG_LOOP) {
		if (loopEnd == 0) {
			_input = makeLinearInputStream(flags, _ptr, size, 0, size);
		} else {
			assert(loopStart < loopEnd && loopEnd <= size);
			_input = makeLinearInputStream(flags, _ptr, size, loopStart, loopEnd - loopStart);
		}
	} else {
		_input = makeLinearInputStream(flags, _ptr, size, 0, 0);
	}

	// Get a rate converter instance
	_converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0);

	if (!(flags & SoundMixer::FLAG_AUTOFREE))
		_ptr = 0;
}

ChannelRaw::~ChannelRaw() {
	free(_ptr);
}

ChannelStream::ChannelStream(SoundMixer *mixer, PlayingSoundHandle *handle,
							void *sound, uint32 size, uint rate,
							byte flags, uint32 buffer_size, byte volume, int8 pan)
	: Channel(mixer, handle) {
	_volume = volume;
	_pan = pan;
	assert(size <= buffer_size);

	// Create the input stream
	_input = makeWrappedInputStream(flags, buffer_size);
	
	// Append the initial data
	((WrappedAudioInputStream *)_input)->append((const byte *)sound, size);

	// Get a rate converter instance
	_converter = makeRateConverter(rate, mixer->getOutputRate(), _input->isStereo(), (flags & SoundMixer::FLAG_REVERSE_STEREO) != 0);

	_finished = false;
}

void ChannelStream::append(void *data, uint32 len) {
	((WrappedAudioInputStream *)_input)->append((const byte *)data, len);
}

void ChannelStream::mix(int16 *data, uint len) {
	assert(_input);
	if (_input->eos()) {
		// TODO: call drain method

		// Normally, the stream stays around even if all its data is used up.
		// This is in case more data is streamed into it. To make the stream
		// go away, one can either stop() it (which takes effect immediately,
		// ignoring any remaining sound data), or finish() it, which means
		// it will finish playing before it terminates itself.
		if (_finished) {
			destroy();
		}
	} else {
		// Invoke the parent implementation.
		Channel::mix(data, len);
	}
}

--- NEW FILE: mixer.h ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

#ifndef SOUND_MIXER_H
#define SOUND_MIXER_H

#include "../stdafx.h"
#include "../bits.h"
#include <SDL.h>

typedef uint32 PlayingSoundHandle;
typedef struct Mutex *MutexRef;
typedef void (*SoundProc)(void *param, byte *buf, int len);
typedef int (*TimerProc)(int interval);

class StackLock {
	MutexRef _mutex;
public:
	StackLock(MutexRef mutex);
	~StackLock();
};

MutexRef create_mutex();
void lock_mutex(MutexRef mutex);
void unlock_mutex(MutexRef mutex);
void delete_mutex(MutexRef mutex);

class Channel;

class SoundMixer {
	friend class Channel;
public:
	typedef void PremixProc (void *param, int16 *data, uint len);

	enum {
		NUM_CHANNELS = 16
	};

	enum {
		FLAG_UNSIGNED = 1 << 0,         // unsigned samples (default: signed)
		FLAG_STEREO = 1 << 1,           // sound is in stereo (default: mono)
		FLAG_16BITS = 1 << 2,           // sound is 16 bits wide (default: 8bit)
		FLAG_AUTOFREE = 1 << 3,         // sound buffer is freed automagically at the end of playing
		FLAG_REVERSE_STEREO = 1 << 4,   // reverse the left and right stereo channel
		FLAG_LOOP = 1 << 5              // loop the audio
	};

private:
	MutexRef _mutex;

	void *_premixParam;
	PremixProc *_premixProc;

	uint _outputRate;

	int _globalVolume;

	bool _paused;

	Channel *_channels[NUM_CHANNELS];

public:
	SoundMixer();
	~SoundMixer();

	void bindToSystem();

	void setupPremix(PremixProc *proc, void *param);

	// start playing a raw sound
	int playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags,
				int id = -1, byte volume = 255, int8 pan = 0, uint32 loopStart = 0, uint32 loopEnd = 0);

	/** Start a new stream. */
	int newStream(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags, uint32 buffer_size, byte volume = 255, int8 pan = 0);

	/** Append to an existing stream. */
	void appendStream(PlayingSoundHandle handle, void *sound, uint32 size);

	/** Mark a stream as finished - it will play all its remaining data, then stop. */
	void endStream(PlayingSoundHandle handle);

	/** stop all currently playing sounds */
	void stopAll();

	/** stop playing the given channel */
	void stopChannel(int channel);

	/** stop playing the sound with given ID  */
	void stopID(int id);

	/** stop playing the channel for the given handle */
	void stopHandle(PlayingSoundHandle handle);

	/** pause/unpause all channels */
	void pauseAll(bool paused);

	/** pause/unpause the given channel */
	void pauseChannel(int index, bool paused);

	/** pause/unpause the sound with the given ID */
	void pauseID(int id, bool paused);

	/** pause/unpause the channel for the given handle */
	void pauseHandle(PlayingSoundHandle handle, bool paused);

	/** set the channel volume for the given handle (0 - 255) */
	void setChannelVolume(PlayingSoundHandle handle, byte volume);

	/** set the channel pan for the given handle (-127 ... 0 ... 127) (left ... center ... right)*/
	void setChannelPan(PlayingSoundHandle handle, int8 pan);

	/** set the global volume, 0-256 */
	void setVolume(int volume);

	/** query the global volume, 0-256 */
	int getVolume() const { return _globalVolume; }

	/** query the output rate in kHz */
	uint getOutputRate() const { return _outputRate; }

private:
	int insertChannel(PlayingSoundHandle *handle, Channel *chan);

	/** main mixer method */
	void mix(int16 *buf, uint len);

	static void mixCallback(void *s, byte *samples, int len);
};

#endif

--- NEW FILE: rate.cpp ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

/*
 * The code in this file is based on code with Copyright 1998 Fabrice Bellard
 * Fabrice original code is part of SoX (http://sox.sourceforge.net).
 * Max Horn adapted that code to the needs of ScummVM and rewrote it partial,
 * in the process removing any use of floating point arithmetic. Various other
 * improvments over the original code were made.
 */

#include "../stdafx.h"
#include "rate.h"
#include "audiostream.h"
#include "../debug.h"

/**
 * The precision of the fractional computations used by the rate converter.
 * Normally you should never have to modify this value.
 */
#define FRAC_BITS 16

/**
 * The size of the intermediate input cache. Bigger values may increase
 * performance, but only until some point (depends largely on cache size,
 * target processor and various other factors), at which it will decrease
 * again.
 */
#define INTERMEDIATE_BUFFER_SIZE 512


/**
 * Audio rate converter based on simple linear Interpolation.
 *
 * The use of fractional increment allows us to use no buffer. It
 * avoid the problems at the end of the buffer we had with the old
 * method which stored a possibly big buffer of size
 * lcm(in_rate,out_rate).
 *
 * Limited to sampling frequency <= 65535 Hz.
 */

template<bool stereo, bool reverseStereo>
class LinearRateConverter : public RateConverter {
protected:
	st_sample_t inBuf[INTERMEDIATE_BUFFER_SIZE];
	const st_sample_t *inPtr;
	int inLen;

	/** fractional position of the output stream in input stream unit */
	unsigned long opos, opos_frac;

	/** fractional position increment in the output stream */
	unsigned long opos_inc, opos_inc_frac;

	/** position in the input stream (integer) */
	unsigned long ipos;

	/** last sample(s) in the input stream (left/right channel) */
	st_sample_t ilast[2];
	/** current sample(s) in the input stream (left/right channel) */
	st_sample_t icur[2];

public:
	LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
	int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
	int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
		return (ST_SUCCESS);
	}
};


/*
 * Prepare processing.
 */
template<bool stereo, bool reverseStereo>
LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
	unsigned long incr;

	if (inrate == outrate) {
		error("Input and Output rates must be different to use rate effect");
	}

	if (inrate >= 65536 || outrate >= 65536) {
		error("rate effect can only handle rates < 65536");
	}

	opos_frac = 0;
	opos = 1;

	/* increment */
	incr = (inrate << FRAC_BITS) / outrate;

	opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1);
	opos_inc = incr >> FRAC_BITS;

	ipos = 0;

	ilast[0] = ilast[1] = 0;
	icur[0] = icur[1] = 0;

	inLen = 0;
}

/*
 * Processed signed long samples from ibuf to obuf.
 * Return number of samples processed.
 */
template<bool stereo, bool reverseStereo>
int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
	st_sample_t *ostart, *oend;
	st_sample_t out[2];

	const int numChannels = stereo ? 2 : 1;
	int i;

	ostart = obuf;
	oend = obuf + osamp * 2;

	while (obuf < oend) {

		// read enough input samples so that ipos > opos
		while (ipos <= opos) {
			// Check if we have to refill the buffer
			if (inLen == 0) {
				inPtr = inBuf;
				inLen = input.readBuffer(inBuf, ARRAYSIZE(inBuf));
				if (inLen <= 0)
					goto the_end;
			}
			for (i = 0; i < numChannels; i++) {
				ilast[i] = icur[i];
				icur[i] = *inPtr++;
				inLen--;
			}
			ipos++;
		}

		// Loop as long as the outpos trails behind, and as long as there is
		// still space in the output buffer.
		while (ipos > opos) {

			// interpolate
			out[0] = out[1] = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));

			if (stereo) {
				// interpolate
				out[reverseStereo ? 0 : 1] = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
			}

			// output left channel
			clampedAdd(*obuf++, (out[0] * (int)vol_l) >> 8);

			// output right channel
			clampedAdd(*obuf++, (out[1] * (int)vol_r) >> 8);

			// Increment output position
			unsigned long tmp = opos_frac + opos_inc_frac;
			opos += opos_inc + (tmp >> FRAC_BITS);
			opos_frac = tmp & ((1UL << FRAC_BITS) - 1);

			// Abort if we reached the end of the output buffer
			if (obuf >= oend)
				goto the_end;
		}
	}

the_end:
	return (ST_SUCCESS);
}

/**
 * Simple audio rate converter for the case that the inrate equals the outrate.
 */
template<bool stereo, bool reverseStereo>
class CopyRateConverter : public RateConverter {
public:
	virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
		int16 tmp[2];
		st_size_t len = osamp;
		assert(input.isStereo() == stereo);
		while (!input.eos() && len--) {
			tmp[0] = tmp[1] = input.read();
			if (stereo)
				tmp[reverseStereo ? 0 : 1] = input.read();

			// output left channel
			clampedAdd(*obuf++, (tmp[0] * (int)vol_l) >> 8);
	
			// output right channel
			clampedAdd(*obuf++, (tmp[1] * (int)vol_r) >> 8);
		}
		return (ST_SUCCESS);
	}
	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
		return (ST_SUCCESS);
	}
};

/**
 * Create and return a RateConverter object for the specified input and output rates.
 */
RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
	if (inrate != outrate) {
		if (stereo) {
			if (reverseStereo)
				return new LinearRateConverter<true, true>(inrate, outrate);
			else
				return new LinearRateConverter<true, false>(inrate, outrate);
		} else
			return new LinearRateConverter<false, false>(inrate, outrate);
	} else {
		if (stereo) {
			if (reverseStereo)
				return new CopyRateConverter<true, true>();
			else
				return new CopyRateConverter<true, false>();
		} else
			return new CopyRateConverter<false, false>();
	}
}

--- NEW FILE: rate.h ---
// Residual - Virtual machine to run LucasArts' 3D adventure games
// Copyright (C) 2003 The ScummVM-Residual Team (www.scummvm.org)
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library 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
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307  USA

#ifndef SOUND_RATE_H
#define SOUND_RATE_H

#include "../bits.h"

class AudioInputStream;

typedef int16 st_sample_t;
typedef uint16 st_volume_t;
typedef uint32 st_size_t;
typedef uint32 st_rate_t;

/* Minimum and maximum values a sample can hold. */
#define ST_SAMPLE_MAX 0x7fffL
#define ST_SAMPLE_MIN (-ST_SAMPLE_MAX - 1L)

#define ST_EOF (-1)
#define ST_SUCCESS (0)

static inline void clampedAdd(int16& a, int b) {
	register int val;
#ifdef OUTPUT_UNSIGNED_AUDIO
	val = (a ^ 0x8000) + b;
#else
	val = a + b;
#endif

	if (val > ST_SAMPLE_MAX)
		val = ST_SAMPLE_MAX;
	else if (val < ST_SAMPLE_MIN)
		val = ST_SAMPLE_MIN;

#ifdef OUTPUT_UNSIGNED_AUDIO
	a = ((int16)val) ^ 0x8000;
#else
	a = val;
#endif
}

// Q&D hack to get this SOX stuff to work
#define st_report warning
#define st_warn warning
#define st_fail error


class RateConverter {
public:
	RateConverter() {}
	virtual ~RateConverter() {}
	virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) = 0;
	virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
};

RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo = false);

#endif