Remove trailing whitespaces.

svn-id: r18604

26 files changed, 123 insertions, 123 deletions

diff --git a/sound/audiocd.cpp b/sound/audiocd.cpp
index ab894724d4..7be50dc4d6 100644
--- a/sound/audiocd.cpp
+++ b/sound/audiocd.cpp
@@ -33,7 +33,7 @@
 struct TrackFormat {
 	/** Decodername */
 	const char* decoderName;
-	/** 
+	/**
 	 * Pointer to a function which tries to open the specified track - the only argument
 	 * is the number of the track to be played.
 	 * Returns either the DigitalTrackInfo object representing the requested track or null
@@ -43,7 +43,7 @@ struct TrackFormat {
 };
 
 static const TrackFormat TRACK_FORMATS[] = {
-	/* decoderName,		openTrackFunction */ 
+	/* decoderName,		openTrackFunction */
 #ifdef USE_FLAC
 	{ "Flac",			getFlacTrack },
 #endif // #ifdef USE_FLAC
diff --git a/sound/audiostream.cpp b/sound/audiostream.cpp
index e8ac3a4ab3..9cab081f69 100644
--- a/sound/audiostream.cpp
+++ b/sound/audiostream.cpp
@@ -32,15 +32,15 @@ struct StreamFileFormat {
 	/** Decodername */
 	const char* decoderName;
 	const char* fileExtension;
-	/** 
+	/**
 	 * Pointer to a function which tries to open a file of type StreamFormat.
-	 * Return NULL in case of an error (invalid/nonexisting file). 
+	 * Return NULL in case of an error (invalid/nonexisting file).
 	 */
 	AudioStream* (*openStreamFile)(Common::File *file, uint32 size);
 };
 
-static const StreamFileFormat STREAM_FILEFORMATS[] = {	
-	/* decoderName,		fileExt, openStreamFuntion */ 
+static const StreamFileFormat STREAM_FILEFORMATS[] = {
+	/* decoderName,		fileExt, openStreamFuntion */
 #ifdef USE_FLAC
 	{ "Flac",			"flac", makeFlacStream },
 	{ "Flac",			"fla",  makeFlacStream },
@@ -60,7 +60,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename)
 	char buffer[1024];
 	const uint len = strlen(filename);
 	assert(len+6 < sizeof(buffer)); // we need a bigger buffer if wrong
-	
+
 	memcpy(buffer, filename, len);
 	buffer[len] = '.';
 	char *ext = &buffer[len+1];
@@ -74,7 +74,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename)
 		if (fileHandle->isOpen())
 			stream = STREAM_FILEFORMATS[i].openStreamFile(fileHandle, fileHandle->size());
 	}
-	
+
 	// Do not reference the file anymore. If the stream didn't incRef the file,
 	// the object will be deleted (and the file be closed).
 	fileHandle->decRef();
@@ -93,7 +93,7 @@ AudioStream* AudioStream::openStreamFile(const char *filename)
 
 /**
  * A simple raw audio stream, purely memory based. It operates on a single
- * block of data, which is passed to it upon creation. 
+ * block of data, which is passed to it upon creation.
  * Optionally supports looping the sound.
  *
  * Design note: This code tries to be as optimized as possible (without
@@ -129,7 +129,7 @@ public:
 		}
 		if (stereo)	// Stereo requires even sized data
 			assert(len % 2 == 0);
-		
+
 		_origPtr = autoFreeMemory ? ptr : 0;
 	}
 	~LinearMemoryStream() {
@@ -191,7 +191,7 @@ AudioStream *makeLinearInputStream(int rate, byte flags, const byte *ptr, uint32
 	const bool isUnsigned = (flags & Audio::Mixer::FLAG_UNSIGNED) != 0;
 	const bool isLE       = (flags & Audio::Mixer::FLAG_LITTLE_ENDIAN) != 0;
 	const bool autoFree   = (flags & Audio::Mixer::FLAG_AUTOFREE) != 0;
-	
+
 	if (isStereo) {
 		if (isUnsigned) {
 			MAKE_LINEAR(true, true);
diff --git a/sound/audiostream.h b/sound/audiostream.h
index 13a70afa65..798c77e6c7 100644
--- a/sound/audiostream.h
+++ b/sound/audiostream.h
@@ -47,7 +47,7 @@ public:
 
 	/** Is this a stereo stream? */
 	virtual bool isStereo() const = 0;
-	
+
 	/**
 	 * End of data reached? If this returns true, it means that at this
 	 * time there is no data available in the stream. However there may be
@@ -56,7 +56,7 @@ public:
 	 * converting data or stop.
 	 */
 	virtual bool endOfData() const = 0;
-	
+
 	/**
 	 * End of stream reached? If this returns true, it means that all data
 	 * in this stream is used up and no additional data will appear in it
@@ -75,7 +75,7 @@ public:
 	 * In case of an error, the file handle will be closed, but deleting
 	 * it is still the responsibilty of the caller.
 	 * @param filename	a filename without an extension
-	 * @return	an Audiostream ready to use in case of success; 
+	 * @return	an Audiostream ready to use in case of success;
 	 *			NULL in case of an error (e.g. invalid/nonexisting file)
 	 */
 	static AudioStream* openStreamFile(const char *filename);
@@ -98,7 +98,7 @@ public:
 	}
 	bool isStereo() const { return false; }
 	bool eos() const { return _len <= 0; }
-	
+
 	int getRate() const { return -1; }
 };
 
diff --git a/sound/flac.cpp b/sound/flac.cpp
index c5840476ed..a6551e544d 100644
--- a/sound/flac.cpp
+++ b/sound/flac.cpp
@@ -86,7 +86,7 @@ protected:
 	inline ::FLAC__StreamDecoderWriteStatus callbackWrite(const ::FLAC__Frame *frame, const FLAC__int32 * const buffer[]);
 	inline void callbackMetadata(const ::FLAC__StreamMetadata *metadata);
 	inline void callbackError(::FLAC__StreamDecoderErrorStatus status);
-	
+
 	::FLAC__SeekableStreamDecoder *_decoder;
 
 private:
@@ -103,14 +103,14 @@ private:
 	void operator=(const FlacInputStream &);
 
 	bool isValid() const { return _decoder != NULL; }
-	
+
 	bool allocateBuffer(uint minSamples);
 	inline void flushBuffer();
 	inline void deleteBuffer();
-	
+
 	/** Header of the Stream */
 	FLAC__StreamMetadata_StreamInfo _streaminfo;
-	
+
 	struct {
 		/** Handle to the File */
 		File *fileHandle;
@@ -121,25 +121,25 @@ private:
 		/** last index of Stream + 1(!) - not necessary end of file */
 		uint32 fileEndPos;
 	} _fileInfo;
-	
+
 	/** index of the first Sample to be played */
 	FLAC__uint64 _firstSample;
 	/** index + 1(!) of the last Sample to be played - 0 is end of Stream*/
 	FLAC__uint64 _lastSample;
-	
+
 	/** true if the last Sample was decoded from the FLAC-API - there might still be data in the buffer */
 	bool _lastSampleWritten;
-	
+
 	typedef int16 bufType;
 	enum { BUFTYPE_BITS = 16 };
-	
+
 	struct {
 		bufType *bufData;
 		bufType *bufReadPos;
 		uint bufSize;
 		uint bufFill;
 	} _preBuffer;
-	
+
 	bufType *_outBuffer;
 	uint _requestedSamples;
 
@@ -154,7 +154,7 @@ private:
 };
 
 FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart)
-			:	_decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), 
+			:	_decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0),
 				_outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true),
 				_methodConvertBuffers(&FlacInputStream::convertBuffersGeneric)
 {
@@ -170,15 +170,15 @@ FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart)
 	_fileInfo.fileStartPos = fileStart;
 	_fileInfo.filePos = fileStart;
 	_fileInfo.fileEndPos = sourceFile->size();
-	
+
 	_fileInfo.fileHandle->incRef();
 }
 
-FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop)	
-			:	_decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0), 
+FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const uint32 fileStop)
+			:	_decoder(::FLAC__seekable_stream_decoder_new()), _firstSample(0), _lastSample(0),
 				_outBuffer(NULL), _requestedSamples(0), _lastSampleWritten(true),
 				_methodConvertBuffers(&FlacInputStream::convertBuffersGeneric)
-{ 
+{
 	assert(sourceFile != NULL && sourceFile->isOpen());
 	assert(fileStop <= 0 || (fileStart < fileStop && fileStop <= sourceFile->size()));
 
@@ -192,7 +192,7 @@ FlacInputStream::FlacInputStream(File *sourceFile, const uint32 fileStart, const
 	_fileInfo.fileStartPos = fileStart;
 	_fileInfo.filePos = fileStart;
 	_fileInfo.fileEndPos = fileStop;
-	
+
 	_fileInfo.fileHandle->incRef();
 }
 
@@ -203,7 +203,7 @@ FlacInputStream::~FlacInputStream() {
 	}
 	if (_preBuffer.bufData != NULL)
 		delete[] _preBuffer.bufData;
-	
+
 	_fileInfo.fileHandle->decRef();
 }
 
@@ -246,7 +246,7 @@ bool FlacInputStream::init() {
 	}
 
 	warning("FlacInputStream: could not create an Audiostream from File %s", _fileInfo.fileHandle->name());
-	return false; 
+	return false;
 }
 
 bool FlacInputStream::finish() {
@@ -304,7 +304,7 @@ int FlacInputStream::readBuffer(int16 *buffer, const int numSamples) {
 
 		const uint copySamples = MIN((uint)numSamples, _preBuffer.bufFill);
 		memcpy(buffer, _preBuffer.bufReadPos, copySamples*sizeof(buffer[0]));
-		
+
 		_outBuffer = buffer + copySamples;
 		_requestedSamples = numSamples - copySamples;
 		_preBuffer.bufReadPos += copySamples;
@@ -351,9 +351,9 @@ inline ::FLAC__SeekableStreamDecoderReadStatus FlacInputStream::callbackRead(FLA
 
 	if (*bytes == 0)
 		return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR; /* abort to avoid a deadlock */
-	
+
 	const uint32 length = MIN(_fileInfo.fileEndPos - _fileInfo.filePos, static_cast<uint32>(*bytes));
-	
+
 	_fileInfo.fileHandle->seek(_fileInfo.filePos);
 	const uint32 bytesRead = _fileInfo.fileHandle->read(buffer, length);
 
@@ -365,7 +365,7 @@ inline ::FLAC__SeekableStreamDecoderReadStatus FlacInputStream::callbackRead(FLA
 	return FLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
 }
 
-inline void FlacInputStream::setLastSample(FLAC__uint64 absoluteSample) { 
+inline void FlacInputStream::setLastSample(FLAC__uint64 absoluteSample) {
 	if (_lastSampleWritten && absoluteSample > _lastSample)
 		_lastSampleWritten = false;
 	_lastSample = absoluteSample;
@@ -555,14 +555,14 @@ void FlacInputStream::convertBuffersGeneric(bufType* bufDestination, const FLAC_
 
 	if (numBits < BUFTYPE_BITS) {
 		const uint8 kPower = (uint8)(BUFTYPE_BITS - numBits);
-		
+
 		for (; numSamples > 0; numSamples -= numChannels) {
 			for (uint i = 0; i < numChannels; ++i)
 				*bufDestination++ = static_cast<bufType>(*(inChannels[i]++)) << kPower;
 		}
 	} else if (numBits > BUFTYPE_BITS) {
 		const uint8 kPower = (uint8)(numBits - BUFTYPE_BITS);
-		
+
 		for (; numSamples > 0; numSamples -= numChannels) {
 			for (uint i = 0; i < numChannels; ++i)
 				*bufDestination++ = static_cast<bufType>(*(inChannels[i]++) >> kPower) ;
@@ -582,7 +582,7 @@ inline ::FLAC__StreamDecoderWriteStatus FlacInputStream::callbackWrite(const ::F
 	assert(frame->header.sample_rate == _streaminfo.sample_rate);
 	assert(frame->header.bits_per_sample == _streaminfo.bits_per_sample);
 	assert(frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER || _streaminfo.min_blocksize == _streaminfo.max_blocksize);
-	
+
 	assert(_preBuffer.bufFill == 0); // we dont append data
 
 	uint nSamples = frame->header.blocksize;
@@ -611,7 +611,7 @@ inline ::FLAC__StreamDecoderWriteStatus FlacInputStream::callbackWrite(const ::F
 	if (_requestedSamples > 0) {
 		assert(_requestedSamples % kNumChannels == 0); // must be integral multiply of channels
 		assert(_outBuffer != NULL);
-			
+
 		const uint copySamples = MIN(_requestedSamples,nSamples);
 		(*_methodConvertBuffers)(_outBuffer, inChannels, copySamples, kNumChannels, kNumBits);
 
@@ -674,7 +674,7 @@ inline void FlacInputStream::callbackMetadata(const ::FLAC__StreamMetadata *meta
 }
 inline void FlacInputStream::callbackError(::FLAC__StreamDecoderErrorStatus status) {
 	// some of these are non-critical-Errors
-	debug(1, "FlacInputStream: An error occured while decoding. DecoderState is: %s", 
+	debug(1, "FlacInputStream: An error occured while decoding. DecoderState is: %s",
 			FLAC__StreamDecoderErrorStatusString[status]);
 }
 
@@ -794,7 +794,7 @@ void FlacTrackInfo::play(Audio::Mixer *mixer, Audio::SoundHandle *handle, int st
 		debug(1, "FlacTrackInfo: Audiostream %s could not seek to frame %d (ca %d secs)", _file->name(), startFrame, startFrame/75);
 		flac->finish();
 	}
-	delete flac; 
+	delete flac;
 }
 
 FlacTrackInfo::~FlacTrackInfo()
diff --git a/sound/fmopl.cpp b/sound/fmopl.cpp
index ddba9dbfef..056459ceea 100644
--- a/sound/fmopl.cpp
+++ b/sound/fmopl.cpp
@@ -451,7 +451,7 @@ inline void OPL_CALC_CH(OPL_CH *CH) {
 	env_out=OPL_CALC_SLOT(SLOT);
 	if(env_out < (uint)(EG_ENT - 1)) {
 		/* PG */
-		if(SLOT->vib) 
+		if(SLOT->vib)
 			SLOT->Cnt += (SLOT->Incr * vib / VIB_RATE);
 		else
 			SLOT->Cnt += SLOT->Incr;
@@ -487,7 +487,7 @@ inline void OPL_CALC_CH(OPL_CH *CH) {
 inline void OPL_CALC_RH(OPL_CH *CH) {
 	uint env_tam, env_sd, env_top, env_hh;
 	int whitenoise = int(oplRnd.getRandomNumber(1) * (WHITE_NOISE_db / EG_STEP));
-	
+
 	int tone8;
 
 	OPL_SLOT *SLOT;
@@ -584,7 +584,7 @@ static void init_timetables(FM_OPL *OPL, int ARRATE, int DRRATE) {
 		OPL->AR_TABLE[i] = OPL->DR_TABLE[i] = 0;
 	for (i = 4; i <= 60; i++){
 		rate = OPL->freqbase;						/* frequency rate */
-		if(i < 60) 
+		if(i < 60)
 			rate *= 1.0 + (i & 3) * 0.25;		/* b0-1 : x1 , x1.25 , x1.5 , x1.75 */
 		rate *= 1 << ((i >> 2) - 1);						/* b2-5 : shift bit */
 		rate *= (double)(EG_ENT << ENV_BITS);
@@ -973,7 +973,7 @@ void YM3812UpdateOne(FM_OPL *OPL, int16 *buffer, int length) {
 		ARM_CALL(ARM_COMMON, PNO_DATA())
 	ARM_END();
 #endif
-	
+
 	int i;
 	int data;
 	int16 *buf = buffer;
diff --git a/sound/fmopl.h b/sound/fmopl.h
index c65fd553ce..b80a4eb2d0 100644
--- a/sound/fmopl.h
+++ b/sound/fmopl.h
@@ -59,7 +59,7 @@ typedef struct fm_opl_slot {
 	uint mul;	/* multiple        :ML_TABLE[ML]		*/
 	uint Cnt;	/* frequency count						*/
 	uint Incr;	/* frequency step						*/
-	
+
 	/* envelope generator state */
 	uint8 eg_typ;/* envelope type flag					*/
 	uint8 evm;	/* envelope phase						*/
@@ -116,7 +116,7 @@ typedef struct fm_opl_f {
 
 	/* Rythm sention */
 	uint8 rythm;		/* Rythm mode , key flag */
-	
+
 	/* time tables */
 	int AR_TABLE[75];	/* atttack rate tables				*/
 	int DR_TABLE[75];	/* decay rate tables				*/
diff --git a/sound/mididrv.cpp b/sound/mididrv.cpp
index 45a7031f7a..fe24f51a1e 100644
--- a/sound/mididrv.cpp
+++ b/sound/mididrv.cpp
@@ -136,7 +136,7 @@ int MidiDriver::detectMusicDriver(int midiFlags) {
 					musicDriver = MD_ETUDE;
 				#elif defined(_WIN32_WCE) || defined(UNIX) || defined(X11_BACKEND) || defined (__SYMBIAN32__)
 					// Always use MIDI emulation via adlib driver on CE and UNIX device
-				
+
 					// TODO: We should, for the Unix targets, attempt to detect
 					// whether a sequencer is available, and use it instead.
 					musicDriver = MD_ADLIB;
@@ -195,7 +195,7 @@ MidiDriver *MidiDriver::createMidi(int midiDriver) {
 	case MD_ZODIAC:    return MidiDriver_Zodiac_create();
 #endif
 #endif
-#if defined(WIN32) && !defined(_WIN32_WCE) && !defined(__SYMBIAN32__) 
+#if defined(WIN32) && !defined(_WIN32_WCE) && !defined(__SYMBIAN32__)
 	case MD_WINDOWS:   return MidiDriver_WIN_create();
 #endif
 #if defined(__MORPHOS__)
diff --git a/sound/mididrv.h b/sound/mididrv.h
index 07b6e18611..556c0c2414 100644
--- a/sound/mididrv.h
+++ b/sound/mididrv.h
@@ -149,7 +149,7 @@ public:
 
 	// Timing functions - MidiDriver now operates timers
 	virtual void setTimerCallback(void *timer_param, Common::Timer::TimerProc timer_proc) = 0;
-	
+
 	/** The time in microseconds between invocations of the timer callback. */
 	virtual uint32 getBaseTempo(void) = 0;
 
diff --git a/sound/midiparser_xmidi.cpp b/sound/midiparser_xmidi.cpp
index 8d8c2e9ea1..3ee08ccb7d 100644
--- a/sound/midiparser_xmidi.cpp
+++ b/sound/midiparser_xmidi.cpp
@@ -26,7 +26,7 @@
 
 /**
  * The XMIDI version of MidiParser.
- * 
+ *
  * Much of this code is adapted from the XMIDI implementation from the exult
  * project.
  */
@@ -52,7 +52,7 @@ public:
 uint32 MidiParser_XMIDI::readVLQ2(byte * &pos) {
 	uint32 value = 0;
 	int i;
-	
+
 	for (i = 0; i < 4; ++i) {
 		if (pos[0] & 0x80)
 			break;
@@ -140,12 +140,12 @@ bool MidiParser_XMIDI::loadMusic(byte *data, uint32 size) {
 	if (!memcmp(pos, "FORM", 4)) {
 		pos += 4;
 
-		// Read length of 
+		// Read length of
 		len = read4high(pos);
 		start = pos;
 
 		// XDIRless XMIDI, we can handle them here.
-		if (!memcmp(pos, "XMID", 4)) {	
+		if (!memcmp(pos, "XMID", 4)) {
 			warning("XMIDI doesn't have XDIR");
 			pos += 4;
 			_num_tracks = 1;
diff --git a/sound/mixer.cpp b/sound/mixer.cpp
index f97c563ccf..c32675a85d 100644
--- a/sound/mixer.cpp
+++ b/sound/mixer.cpp
@@ -114,7 +114,7 @@ Mixer::Mixer() {
 		_volumeForSoundType[i] = kMaxMixerVolume;
 
 	_paused = false;
-	
+
 	for (i = 0; i != NUM_CHANNELS; i++)
 		_channels[i] = 0;
 
@@ -144,7 +144,7 @@ void Mixer::setupPremix(AudioStream *stream, SoundType type) {
 
 	delete _premixChannel;
 	_premixChannel = 0;
-	
+
 	if (stream == 0)
 		return;
 
@@ -399,7 +399,7 @@ void Mixer::setVolumeForSoundType(SoundType type, int volume) {
 		volume = kMaxMixerVolume;
 	else if (volume < 0)
 		volume = 0;
-	
+
 	// TODO: Maybe we should do logarithmic (not linear) volume
 	// scaling? See also Player_V2::setMasterVolume
 
@@ -408,7 +408,7 @@ void Mixer::setVolumeForSoundType(SoundType type, int volume) {
 
 int Mixer::getVolumeForSoundType(SoundType type) const {
 	assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
-	
+
 	return _volumeForSoundType[type];
 }
 
@@ -462,7 +462,7 @@ void Channel::mix(int16 *data, uint len) {
 		// balance value ranges from -127 to 127.  The mixer (music/sound)
 		// volume is in the range 0 - kMaxMixerVolume.
 		// Hence, the vol_l/vol_r values will be in that range, too
-		
+
 		int vol = _mixer->getVolumeForSoundType(_type) * _volume;
 		st_volume_t vol_l, vol_r;
 
diff --git a/sound/mixer.h b/sound/mixer.h
index 001753301a..72f881d626 100644
--- a/sound/mixer.h
+++ b/sound/mixer.h
@@ -69,7 +69,7 @@ public:
 		/** loop the audio */
 		FLAG_LOOP = 1 << 6
 	};
-	
+
 	enum SoundType {
 		kPlainSoundType = 0,
 
@@ -77,7 +77,7 @@ public:
 		kSFXSoundType = 2,
 		kSpeechSoundType = 3
 	};
-	
+
 	enum {
 		kMaxChannelVolume = 255,
 		kMaxMixerVolume = 256
@@ -98,7 +98,7 @@ private:
 	int _volumeForSoundType[4];
 
 	bool _paused;
-	
+
 	uint32 _handleSeed;
 	Channel *_channels[NUM_CHANNELS];
 
diff --git a/sound/mp3.cpp b/sound/mp3.cpp
index 792f8f6bf4..4c9f238eed 100644
--- a/sound/mp3.cpp
+++ b/sound/mp3.cpp
@@ -64,7 +64,7 @@ public:
 
 	bool endOfData() const		{ return eosIntern(); }
 	bool isStereo() const		{ return _isStereo; }
-	
+
 	int getRate() const			{ return _frame.header.samplerate; }
 #ifdef __SYMBIAN32__
 	// Used to store the last position stream was read for symbian
@@ -193,7 +193,7 @@ bool MP3InputStream::init() {
 			warning("MP3InputStream: Cannot determine number of channels");
 			return false;
 	}
-	
+
 	return true;
 }
 
@@ -256,7 +256,7 @@ void MP3InputStream::refill(bool first) {
 		mad_timer_t frame_duration = _frame.header.duration;
 		mad_timer_negate(&frame_duration);
 		mad_timer_add(&_duration, frame_duration);
-	
+
 		if (!first && mad_timer_compare(_duration, mad_timer_zero) <= 0)
 			_size = -1;	// Mark for EOF
 	}
diff --git a/sound/mpu401.cpp b/sound/mpu401.cpp
index 460aed5f52..c46c3d60ac 100644
--- a/sound/mpu401.cpp
+++ b/sound/mpu401.cpp
@@ -90,7 +90,7 @@ MidiDriver_MPU401::MidiDriver_MPU401() :
 	_timer_proc (0),
 	_channel_mask (0xFFFF) // Permit all 16 channels by default
 {
-	
+
 	uint i;
 	for (i = 0; i < ARRAYSIZE(_midi_channels); ++i) {
 		_midi_channels [i].init (this, i);
diff --git a/sound/rate.cpp b/sound/rate.cpp
index eedc283cc6..997aa7e7cf 100644
--- a/sound/rate.cpp
+++ b/sound/rate.cpp
@@ -209,10 +209,10 @@ public:
 
 	virtual int flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
 		assert(input.isStereo() == stereo);
-		
+
 		st_sample_t *ptr;
 		st_size_t len;
-		
+
 		if (stereo)
 			osamp *= 2;
 
@@ -225,7 +225,7 @@ public:
 
 		// Read up to 'osamp' samples into our temporary buffer
 		len = input.readBuffer(_buffer, osamp);
-		
+
 		// Mix the data into the output buffer
 		ptr = _buffer;
 		while (len--) {
@@ -241,7 +241,7 @@ public:
 
 			// output left channel
 			clampedAdd(*obuf++, (tmp0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
-	
+
 			// output right channel
 			clampedAdd(*obuf++, (tmp1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
 		}
diff --git a/sound/softsynth/adlib.cpp b/sound/softsynth/adlib.cpp
index 5f0d0435bd..8fb7aea19e 100644
--- a/sound/softsynth/adlib.cpp
+++ b/sound/softsynth/adlib.cpp
@@ -195,7 +195,7 @@ struct AdlibVoice {
 	Struct11 _s11a;
 	Struct10 _s10b;
 	Struct11 _s11b;
-	
+
 	AdlibVoice() { memset(this, 0, sizeof(AdlibVoice)); }
 };
 
@@ -461,14 +461,14 @@ static byte gm_percussion_to_fm[39][30] = {
 };
 
 static const byte gm_percussion_lookup[128] = {
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 
-	0xFF, 0xFF, 0xFF, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 
-	0x0D, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0xFF, 0xFF, 0x17, 0x18, 0x19, 0x1A, 
-	0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0x21, 0x22, 0x23, 0xFF, 0xFF, 
-	0x24, 0x25, 0xFF, 0xFF, 0xFF, 0x26, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 
-	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 
+	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+	0xFF, 0xFF, 0xFF, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C,
+	0x0D, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0xFF, 0xFF, 0x17, 0x18, 0x19, 0x1A,
+	0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0x21, 0x22, 0x23, 0xFF, 0xFF,
+	0x24, 0x25, 0xFF, 0xFF, 0xFF, 0x26, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
+	0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
 };
 
 static byte lookup_table[64][32];
@@ -551,7 +551,7 @@ public:
 	void send(byte channel, uint32 b); // Supports higher than channel 15
 	uint32 property(int prop, uint32 param);
 
-	void setPitchBendRange(byte channel, uint range); 
+	void setPitchBendRange(byte channel, uint range);
 	void sysEx_customInstrument(byte channel, uint32 type, byte *instr);
 
 	MidiChannel *allocateChannel();
@@ -798,7 +798,7 @@ void AdlibPercussionChannel::noteOn(byte note, byte velocity) {
 
 // MidiDriver method implementations
 
-MidiDriver_ADLIB::MidiDriver_ADLIB(Audio::Mixer *mixer) 
+MidiDriver_ADLIB::MidiDriver_ADLIB(Audio::Mixer *mixer)
 	: MidiDriver_Emulated(mixer) {
 	uint i;
 
@@ -865,7 +865,7 @@ void MidiDriver_ADLIB::close() {
 
 	// Turn off the OPL emulation
 //	YM3812Shutdown();
-	
+
 	free(_adlib_reg_cache);
 }
 
@@ -1308,7 +1308,7 @@ AdlibVoice *MidiDriver_ADLIB::allocate_voice(byte pri) {
 	}
 
 	/* V3 games don't have note priorities, first comes wins. */
-	if (_game_SmallHeader) 
+	if (_game_SmallHeader)
 		return NULL;
 
 	if (best)
diff --git a/sound/softsynth/emumidi.h b/sound/softsynth/emumidi.h
index 579aebeeff..75559530f3 100644
--- a/sound/softsynth/emumidi.h
+++ b/sound/softsynth/emumidi.h
@@ -46,10 +46,10 @@ protected:
 public:
 	MidiDriver_Emulated(Audio::Mixer *mixer) : _mixer(mixer) {
 		_isOpen = false;
-	
+
 		_timerProc = 0;
 		_timerParam = 0;
-	
+
 		_nextTick = 0;
 		_samplesPerTick = 0;
 
@@ -89,7 +89,7 @@ public:
 				step = (_nextTick >> FIXP_SHIFT);
 
 			generateSamples(data, step);
-	
+
 			_nextTick -= step << FIXP_SHIFT;
 			if (!(_nextTick >> FIXP_SHIFT)) {
 				if (_timerProc)
diff --git a/sound/softsynth/fluidsynth.cpp b/sound/softsynth/fluidsynth.cpp
index 5bcf0993af..563dbd84f1 100644
--- a/sound/softsynth/fluidsynth.cpp
+++ b/sound/softsynth/fluidsynth.cpp
@@ -73,7 +73,7 @@ MidiDriver_FluidSynth::MidiDriver_FluidSynth(Audio::Mixer *mixer)
 	}
 
 	// It ought to be possible to get FluidSynth to generate samples at
-	// lower 
+	// lower
 
 	_outputRate = _mixer->getOutputRate();
 	if (_outputRate < 22050)
diff --git a/sound/softsynth/mt32.cpp b/sound/softsynth/mt32.cpp
index 94373b980a..2cef1cbedf 100644
--- a/sound/softsynth/mt32.cpp
+++ b/sound/softsynth/mt32.cpp
@@ -238,7 +238,7 @@ int MidiDriver_MT32::open() {
 		return MERR_ALREADY_OPEN;
 
 	MidiDriver_Emulated::open();
-	
+
 	memset(&prop, 0, sizeof(prop));
 	prop.sampleRate = getRate();
 	prop.useReverb = true;
@@ -253,10 +253,10 @@ int MidiDriver_MT32::open() {
 	_synth = new MT32Emu::Synth();
 	_initialising = true;
 	const byte dummy_palette[] = {
-		0, 0, 0, 0, 
-		0, 0, 171, 0, 
-		0, 171, 0, 0, 
-		0, 171, 171, 0, 
+		0, 0, 0, 0,
+		0, 0, 171, 0,
+		0, 171, 0, 0,
+		0, 171, 171, 0,
 		171, 0, 0, 0
 	};
 
@@ -470,7 +470,7 @@ void MidiDriver_ThreadedMT32::onTimer() {
 
 MidiDriver *MidiDriver_MT32_create(Audio::Mixer *mixer) {
 	// HACK: It will stay here until engine plugin loader overhaul
-	if (ConfMan.hasKey("extrapath"))                                        
+	if (ConfMan.hasKey("extrapath"))
 		Common::File::addDefaultDirectory(ConfMan.get("extrapath"));
 	return new MidiDriver_MT32(mixer);
 }
diff --git a/sound/softsynth/mt32/freeverb.cpp b/sound/softsynth/mt32/freeverb.cpp
index 42865c1e71..0a75e7bc70 100644
--- a/sound/softsynth/mt32/freeverb.cpp
+++ b/sound/softsynth/mt32/freeverb.cpp
@@ -26,7 +26,7 @@
 
 // Comb filter implementation
 //
-// Written by 
+// Written by
 // http://www.dreampoint.co.uk
 // This code is public domain
 
@@ -39,7 +39,7 @@ comb::comb() {
 }
 
 void comb::setbuffer(float *buf, int size) {
-	buffer = buf; 
+	buffer = buf;
 	bufsize = size;
 }
 
@@ -49,7 +49,7 @@ void comb::mute() {
 }
 
 void comb::setdamp(float val) {
-	damp1 = val; 
+	damp1 = val;
 	damp2 = 1 - val;
 }
 
@@ -72,7 +72,7 @@ allpass::allpass() {
 }
 
 void allpass::setbuffer(float *buf, int size) {
-	buffer = buf; 
+	buffer = buf;
 	bufsize = size;
 }
 
diff --git a/sound/softsynth/mt32/freeverb.h b/sound/softsynth/mt32/freeverb.h
index 2a5d662a44..b34413f188 100644
--- a/sound/softsynth/mt32/freeverb.h
+++ b/sound/softsynth/mt32/freeverb.h
@@ -101,10 +101,10 @@ private:
 inline float allpass::process(float input) {
 	float output;
 	float bufout;
-	
+
 	bufout = buffer[bufidx];
 	undenormalise(bufout);
-	
+
 	output = -input + bufout;
 	buffer[bufidx] = input + (bufout * feedback);
 
@@ -196,7 +196,7 @@ private:
 	float width;
 	float mode;
 
-	// The following are all declared inline 
+	// The following are all declared inline
 	// to remove the need for dynamic allocation
 	// with its subsequent error-checking messiness
 
diff --git a/sound/softsynth/mt32/part.h b/sound/softsynth/mt32/part.h
index 8a5612e32d..54c4999653 100644
--- a/sound/softsynth/mt32/part.h
+++ b/sound/softsynth/mt32/part.h
@@ -42,7 +42,7 @@ private:
 
 	PatchCache patchCache[4];
 
-	float bend; // -1.0 .. +1.0 
+	float bend; // -1.0 .. +1.0
 
 	dpoly polyTable[MT32EMU_MAX_POLY];
 
diff --git a/sound/softsynth/mt32/synth.cpp b/sound/softsynth/mt32/synth.cpp
index 66832b0e59..62356dcda9 100644
--- a/sound/softsynth/mt32/synth.cpp
+++ b/sound/softsynth/mt32/synth.cpp
@@ -824,31 +824,31 @@ void Synth::readMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u len
 
 	switch(region->type) {
 	case MR_PatchTemp:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.patchSettings[first])[off + m];
 		break;
 	case MR_RhythmTemp:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.rhythmSettings[first])[off + m];
 		break;
 	case MR_TimbreTemp:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.timbreSettings[first])[off + m];
 		break;
 	case MR_Patches:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.patches[first])[off + m];
 		break;
 	case MR_Timbres:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.timbres[first])[off + m];
 		break;
 	case MR_System:
-		for (m = 0; m < len; m++) 
+		for (m = 0; m < len; m++)
 			data[m] = ((Bit8u *)&mt32ram.system)[m + off];
 		break;
 	default:
-		for (m = 0; m < len; m += 2) { 
+		for (m = 0; m < len; m += 2) {
 			data[m] = 0xff;
 			if (m + 1 < len) {
 				data[m+1] = (Bit8u)region->type;
diff --git a/sound/softsynth/mt32/tables.cpp b/sound/softsynth/mt32/tables.cpp
index e12f306cf9..006f91d0b8 100644
--- a/sound/softsynth/mt32/tables.cpp
+++ b/sound/softsynth/mt32/tables.cpp
@@ -170,7 +170,7 @@ void Tables::initEnvelopes(float samplerate) {
 			}
 			envDeltaMaxTime[lf] = (int)cap;
 		}
-		
+
 
 		// This (approximately) represents the time durations when the target level is 0.
 		// Not sure why this is a special case, but it's seen to be from the real thing.
diff --git a/sound/softsynth/ym2612.cpp b/sound/softsynth/ym2612.cpp
index e37b50170e..80c0cad8e8 100644
--- a/sound/softsynth/ym2612.cpp
+++ b/sound/softsynth/ym2612.cpp
@@ -106,7 +106,7 @@ protected:
 	int _frequencyOffs;
 	int _frequency;
 	int _algorithm;
-	
+
 	int *_buffer;
 	int _buflen;
 
diff --git a/sound/vorbis.cpp b/sound/vorbis.cpp
index 0f771cb5e3..7c18a83a8e 100644
--- a/sound/vorbis.cpp
+++ b/sound/vorbis.cpp
@@ -126,7 +126,7 @@ static ov_callbacks g_File_wrap = {
 
 
 VorbisTrackInfo::VorbisTrackInfo(File *file) {
-	
+
 	_file = file;
 	if (openTrack()) {
 		warning("Invalid file format");
@@ -149,9 +149,9 @@ bool VorbisTrackInfo::openTrack() {
 	f->len = _file->size();
 	f->curr_pos = 0;
 	_file->seek(0);
-	
+
 	bool err = (ov_open_callbacks((void *) f, &_ov_file, NULL, 0, g_File_wrap) < 0);
-	
+
 	if (err) {
 		delete f;
 	} else {
@@ -218,7 +218,7 @@ class VorbisInputStream : public AudioStream {
 	const int16 *_bufferEnd;
 	const int16 *_pos;
 	bool _deleteFileAfterUse;
-	
+
 	void refill();
 	inline bool eosIntern() const;
 public:
@@ -229,7 +229,7 @@ public:
 
 	bool endOfData() const		{ return eosIntern(); }
 	bool isStereo() const		{ return _numChannels >= 2; }
-	
+
 	int getRate() const			{ return ov_info(_ov_file, -1)->rate; }
 
 };
@@ -240,7 +240,7 @@ public:
 #endif
 
 
-VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse) 
+VorbisInputStream::VorbisInputStream(OggVorbis_File *file, int duration, bool deleteFileAfterUse)
 	: _ov_file(file),
 	_bufferEnd(_buffer + ARRAYSIZE(_buffer)),
 	_deleteFileAfterUse(deleteFileAfterUse) {
diff --git a/sound/wave.cpp b/sound/wave.cpp
index 42c1bd34db..97dd030ab8 100644
--- a/sound/wave.cpp
+++ b/sound/wave.cpp
@@ -53,7 +53,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
 		warning("getWavInfo: No 'fmt' header");
 		return false;
 	}
-	
+
 	uint32 fmtLength = stream.readUint32LE();
 	if (fmtLength < 16) {
 		// A valid fmt chunk always contains at least 16 bytes
@@ -79,7 +79,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
 
 	if (wavType != 0)
 		*wavType = type;
-#if 0	
+#if 0
 	printf("WAVE information:\n");
 	printf("  total size: %d\n", wavLength);
 	printf("  fmt size: %d\n", fmtLength);
@@ -118,7 +118,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
 		warning("getWavInfo: unsupported bitsPerSample %d", bitsPerSample);
 		return false;
 	}
-	
+
 	if (numChannels == 2)
 		flags |= Audio::Mixer::FLAG_STEREO;
 	else if (numChannels != 1) {
@@ -145,7 +145,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
 		printf("  found a '%s' tag of size %d\n", buf, offset);
 #endif
 	} while (memcmp(buf, "data", 4) != 0);
-	
+
 	// Stream now points at 'offset' bytes of sample data...
 	size = offset;
 
@@ -156,10 +156,10 @@ AudioStream *makeWAVStream(Common::SeekableReadStream &stream) {
 	int size, rate;
 	byte flags;
 	uint16 type;
-	
+
 	if (!loadWAVFromStream(stream, size, rate, flags, &type))
 		return 0;
-	
+
 	if (type == 17) // IMA ADPCM
 		return makeADPCMStream(stream, size, kADPCMIma);