[Scummvm-cvs-logs] CVS: scummvm/sound audiocd.cpp,1.21,1.22 audiostream.cpp,1.74,1.75 audiostream.h,1.47,1.48 flac.cpp,1.10,1.11 fmopl.cpp,1.31,1.32 fmopl.h,1.16,1.17 mididrv.cpp,1.64,1.65 mididrv.h,1.48,1.49 midiparser_xmidi.cpp,1.21,1.22 mixer.cpp,1.193,1.194 mixer.h,1.107,1.108 mp3.cpp,1.23,1.24 mpu401.cpp,1.31,1.32 rate.cpp,1.40,1.41 vorbis.cpp,1.26,1.27 wave.cpp,1.7,1.8

Eugene Sandulenko sev at users.sourceforge.net
Sat Jul 30 14:15:03 CEST 2005


Update of /cvsroot/scummvm/scummvm/sound
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv9428/sound

Modified Files:
	audiocd.cpp audiostream.cpp audiostream.h flac.cpp fmopl.cpp 
	fmopl.h mididrv.cpp mididrv.h midiparser_xmidi.cpp mixer.cpp 
	mixer.h mp3.cpp mpu401.cpp rate.cpp vorbis.cpp wave.cpp 
Log Message:
Remove trailing whitespaces.


Index: audiocd.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/audiocd.cpp,v
retrieving revision 1.21
retrieving revision 1.22
diff -u -d -r1.21 -r1.22
--- audiocd.cpp	24 Jun 2005 15:23:46 -0000	1.21
+++ audiocd.cpp	30 Jul 2005 21:11:42 -0000	1.22
@@ -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 @@
 };
 
 static const TrackFormat TRACK_FORMATS[] = {
-	/* decoderName,		openTrackFunction */ 
+	/* decoderName,		openTrackFunction */
 #ifdef USE_FLAC
 	{ "Flac",			getFlacTrack },
 #endif // #ifdef USE_FLAC

Index: audiostream.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/audiostream.cpp,v
retrieving revision 1.74
retrieving revision 1.75
diff -u -d -r1.74 -r1.75
--- audiostream.cpp	24 Jun 2005 15:23:46 -0000	1.74
+++ audiostream.cpp	30 Jul 2005 21:11:42 -0000	1.75
@@ -32,15 +32,15 @@
 	/** 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 @@
 	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 @@
 		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 @@
 
 /**
  * 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 @@
 		}
 		if (stereo)	// Stereo requires even sized data
 			assert(len % 2 == 0);
-		
+
 		_origPtr = autoFreeMemory ? ptr : 0;
 	}
 	~LinearMemoryStream() {
@@ -191,7 +191,7 @@
 	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);

Index: audiostream.h
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/audiostream.h,v
retrieving revision 1.47
retrieving revision 1.48
diff -u -d -r1.47 -r1.48
--- audiostream.h	24 Jun 2005 15:23:46 -0000	1.47
+++ audiostream.h	30 Jul 2005 21:11:42 -0000	1.48
@@ -47,7 +47,7 @@
 
 	/** 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 @@
 	 * 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 @@
 	 * 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 @@
 	}
 	bool isStereo() const { return false; }
 	bool eos() const { return _len <= 0; }
-	
+
 	int getRate() const { return -1; }
 };
 

Index: flac.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/flac.cpp,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -d -r1.10 -r1.11
--- flac.cpp	11 May 2005 00:01:34 -0000	1.10
+++ flac.cpp	30 Jul 2005 21:11:42 -0000	1.11
@@ -86,7 +86,7 @@
 	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 @@
 	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 @@
 		/** 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 @@
 };
 
 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 @@
 	_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 @@
 	_fileInfo.fileStartPos = fileStart;
 	_fileInfo.filePos = fileStart;
 	_fileInfo.fileEndPos = fileStop;
-	
+
 	_fileInfo.fileHandle->incRef();
 }
 
@@ -203,7 +203,7 @@
 	}
 	if (_preBuffer.bufData != NULL)
 		delete[] _preBuffer.bufData;
-	
+
 	_fileInfo.fileHandle->decRef();
 }
 
@@ -246,7 +246,7 @@
 	}
 
 	warning("FlacInputStream: could not create an Audiostream from File %s", _fileInfo.fileHandle->name());
-	return false; 
+	return false;
 }
 
 bool FlacInputStream::finish() {
@@ -304,7 +304,7 @@
 
 		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 @@
 
 	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 @@
 	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 @@
 
 	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 @@
 	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 @@
 	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::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 @@
 		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()

Index: fmopl.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/fmopl.cpp,v
retrieving revision 1.31
retrieving revision 1.32
diff -u -d -r1.31 -r1.32
--- fmopl.cpp	24 Jun 2005 16:16:46 -0000	1.31
+++ fmopl.cpp	30 Jul 2005 21:11:42 -0000	1.32
@@ -451,7 +451,7 @@
 	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_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 @@
 		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 @@
 		ARM_CALL(ARM_COMMON, PNO_DATA())
 	ARM_END();
 #endif
-	
+
 	int i;
 	int data;
 	int16 *buf = buffer;

Index: fmopl.h
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/fmopl.h,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- fmopl.h	28 Jan 2005 20:46:36 -0000	1.16
+++ fmopl.h	30 Jul 2005 21:11:43 -0000	1.17
@@ -59,7 +59,7 @@
 	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 @@
 
 	/* Rythm sention */
 	uint8 rythm;		/* Rythm mode , key flag */
-	
+
 	/* time tables */
 	int AR_TABLE[75];	/* atttack rate tables				*/
 	int DR_TABLE[75];	/* decay rate tables				*/

Index: mididrv.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mididrv.cpp,v
retrieving revision 1.64
retrieving revision 1.65
diff -u -d -r1.64 -r1.65
--- mididrv.cpp	24 Jun 2005 15:23:46 -0000	1.64
+++ mididrv.cpp	30 Jul 2005 21:11:43 -0000	1.65
@@ -136,7 +136,7 @@
 					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 @@
 	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__)

Index: mididrv.h
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mididrv.h,v
retrieving revision 1.48
retrieving revision 1.49
diff -u -d -r1.48 -r1.49
--- mididrv.h	20 Jun 2005 18:27:33 -0000	1.48
+++ mididrv.h	30 Jul 2005 21:11:43 -0000	1.49
@@ -149,7 +149,7 @@
 
 	// 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;
 

Index: midiparser_xmidi.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/midiparser_xmidi.cpp,v
retrieving revision 1.21
retrieving revision 1.22
diff -u -d -r1.21 -r1.22
--- midiparser_xmidi.cpp	24 Jun 2005 16:16:46 -0000	1.21
+++ midiparser_xmidi.cpp	30 Jul 2005 21:11:43 -0000	1.22
@@ -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 @@
 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 @@
 	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;

Index: mixer.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mixer.cpp,v
retrieving revision 1.193
retrieving revision 1.194
diff -u -d -r1.193 -r1.194
--- mixer.cpp	24 Jun 2005 15:23:47 -0000	1.193
+++ mixer.cpp	30 Jul 2005 21:11:43 -0000	1.194
@@ -114,7 +114,7 @@
 		_volumeForSoundType[i] = kMaxMixerVolume;
 
 	_paused = false;
-	
+
 	for (i = 0; i != NUM_CHANNELS; i++)
 		_channels[i] = 0;
 
@@ -144,7 +144,7 @@
 
 	delete _premixChannel;
 	_premixChannel = 0;
-	
+
 	if (stream == 0)
 		return;
 
@@ -399,7 +399,7 @@
 		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 @@
 
 int Mixer::getVolumeForSoundType(SoundType type) const {
 	assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
-	
+
 	return _volumeForSoundType[type];
 }
 
@@ -462,7 +462,7 @@
 		// 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;
 

Index: mixer.h
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mixer.h,v
retrieving revision 1.107
retrieving revision 1.108
diff -u -d -r1.107 -r1.108
--- mixer.h	24 Jun 2005 15:23:47 -0000	1.107
+++ mixer.h	30 Jul 2005 21:11:43 -0000	1.108
@@ -69,7 +69,7 @@
 		/** loop the audio */
 		FLAG_LOOP = 1 << 6
 	};
-	
+
 	enum SoundType {
 		kPlainSoundType = 0,
 
@@ -77,7 +77,7 @@
 		kSFXSoundType = 2,
 		kSpeechSoundType = 3
 	};
-	
+
 	enum {
 		kMaxChannelVolume = 255,
 		kMaxMixerVolume = 256
@@ -98,7 +98,7 @@
 	int _volumeForSoundType[4];
 
 	bool _paused;
-	
+
 	uint32 _handleSeed;
 	Channel *_channels[NUM_CHANNELS];
 

Index: mp3.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mp3.cpp,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -d -r1.23 -r1.24
--- mp3.cpp	26 Jun 2005 23:47:19 -0000	1.23
+++ mp3.cpp	30 Jul 2005 21:11:43 -0000	1.24
@@ -64,7 +64,7 @@
 
 	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 @@
 			warning("MP3InputStream: Cannot determine number of channels");
 			return false;
 	}
-	
+
 	return true;
 }
 
@@ -256,7 +256,7 @@
 		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
 	}

Index: mpu401.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/mpu401.cpp,v
retrieving revision 1.31
retrieving revision 1.32
diff -u -d -r1.31 -r1.32
--- mpu401.cpp	24 Jun 2005 16:16:46 -0000	1.31
+++ mpu401.cpp	30 Jul 2005 21:11:43 -0000	1.32
@@ -90,7 +90,7 @@
 	_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);

Index: rate.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/rate.cpp,v
retrieving revision 1.40
retrieving revision 1.41
diff -u -d -r1.40 -r1.41
--- rate.cpp	24 Jun 2005 15:23:47 -0000	1.40
+++ rate.cpp	30 Jul 2005 21:11:43 -0000	1.41
@@ -209,10 +209,10 @@
 
 	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 @@
 
 		// 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 @@
 
 			// output left channel
 			clampedAdd(*obuf++, (tmp0 * (int)vol_l) / Audio::Mixer::kMaxMixerVolume);
-	
+
 			// output right channel
 			clampedAdd(*obuf++, (tmp1 * (int)vol_r) / Audio::Mixer::kMaxMixerVolume);
 		}

Index: vorbis.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/vorbis.cpp,v
retrieving revision 1.26
retrieving revision 1.27
diff -u -d -r1.26 -r1.27
--- vorbis.cpp	30 Jun 2005 09:14:36 -0000	1.26
+++ vorbis.cpp	30 Jul 2005 21:11:43 -0000	1.27
@@ -126,7 +126,7 @@
 
 
 VorbisTrackInfo::VorbisTrackInfo(File *file) {
-	
+
 	_file = file;
 	if (openTrack()) {
 		warning("Invalid file format");
@@ -149,9 +149,9 @@
 	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 @@
 	const int16 *_bufferEnd;
 	const int16 *_pos;
 	bool _deleteFileAfterUse;
-	
+
 	void refill();
 	inline bool eosIntern() const;
 public:
@@ -229,7 +229,7 @@
 
 	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 @@
 #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) {

Index: wave.cpp
===================================================================
RCS file: /cvsroot/scummvm/scummvm/sound/wave.cpp,v
retrieving revision 1.7
retrieving revision 1.8
diff -u -d -r1.7 -r1.8
--- wave.cpp	24 Jun 2005 15:23:48 -0000	1.7
+++ wave.cpp	30 Jul 2005 21:11:43 -0000	1.8
@@ -53,7 +53,7 @@
 		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 @@
 
 	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 @@
 		warning("getWavInfo: unsupported bitsPerSample %d", bitsPerSample);
 		return false;
 	}
-	
+
 	if (numChannels == 2)
 		flags |= Audio::Mixer::FLAG_STEREO;
 	else if (numChannels != 1) {
@@ -145,7 +145,7 @@
 		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 @@
 	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);
 





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