Strip trailing whitespaces in the whole code base.

svn-id: r40867

8 files changed, 46 insertions, 46 deletions

diff --git a/sound/adpcm.cpp b/sound/adpcm.cpp
index a5f2f7c4eb..37b14140b7 100644
--- a/sound/adpcm.cpp
+++ b/sound/adpcm.cpp
@@ -112,12 +112,12 @@ public:
 
 ADPCMInputStream::ADPCMInputStream(Common::SeekableReadStream *stream, bool disposeAfterUse, uint32 size, typesADPCM type, int rate, int channels, uint32 blockAlign, uint numLoops)
 	: _stream(stream), _disposeAfterUse(disposeAfterUse), _channels(channels), _type(type), _blockAlign(blockAlign), _rate(rate), _numLoops(numLoops) {
-	
+
 	if (type == kADPCMMSIma && blockAlign == 0)
 		error("ADPCMInputStream(): blockAlign isn't specified for MS IMA ADPCM");
 	if (type == kADPCMMS && blockAlign == 0)
 		error("ADPCMInputStream(): blockAlign isn't specified for MS ADPCM");
-	
+
 	if (type == kADPCMTinsel4 && blockAlign == 0)
 		error("ADPCMInputStream(): blockAlign isn't specified for Tinsel 4-bit ADPCM");
 	if (type == kADPCMTinsel6 && blockAlign == 0)
@@ -131,7 +131,7 @@ ADPCMInputStream::ADPCMInputStream(Common::SeekableReadStream *stream, bool disp
 		error("ADPCMInputStream(): Tinsel 6-bit ADPCM only supports mono");
 	if (type == kADPCMTinsel8 && channels != 1)
 		error("ADPCMInputStream(): Tinsel 8-bit ADPCM only supports mono");
-	
+
 	_startpos = stream->pos();
 	_endpos = _startpos + size;
 	_curLoop = 0;
@@ -182,7 +182,7 @@ int ADPCMInputStream::readBuffer(int16 *buffer, const int numSamples) {
 		error("Unsupported ADPCM encoding");
 		break;
 	}
-	
+
 	// Loop if necessary
 	if (samplesDecoded < numSamples || _stream->pos() == _endpos) {
 		_curLoop++;
@@ -192,7 +192,7 @@ int ADPCMInputStream::readBuffer(int16 *buffer, const int numSamples) {
 			return samplesDecoded + readBuffer(buffer + samplesDecoded, numSamples - samplesDecoded);
 		}
 	}
-	
+
 	return samplesDecoded;
 }
 
diff --git a/sound/fmopl.h b/sound/fmopl.h
index 81d9610a44..d0b15368e0 100644
--- a/sound/fmopl.h
+++ b/sound/fmopl.h
@@ -76,7 +76,7 @@ public:
 
 	/**
 	 * Creates the specific driver with a specific type setup.
-	 */ 
+	 */
 	static OPL *create(DriverId driver, OplType type);
 
 	/**
@@ -86,7 +86,7 @@ public:
 	static OPL *create(OplType type = kOpl2) { return create(detect(type), type); }
 
 private:
-	static const EmulatorDescription _drivers[];	
+	static const EmulatorDescription _drivers[];
 };
 
 class OPL {
diff --git a/sound/softsynth/opl/dosbox.cpp b/sound/softsynth/opl/dosbox.cpp
index cbb3090608..25fda7f2e6 100644
--- a/sound/softsynth/opl/dosbox.cpp
+++ b/sound/softsynth/opl/dosbox.cpp
@@ -40,7 +40,7 @@
 #include <string.h>
 
 namespace OPL {
-namespace DOSBox {	
+namespace DOSBox {
 
 Timer::Timer() {
 	masked = false;
@@ -51,7 +51,7 @@ Timer::Timer() {
 }
 
 void Timer::update(double time) {
-	if (!enabled || !delay) 
+	if (!enabled || !delay)
 		return;
 	double deltaStart = time - startTime;
 	// Only set the overflow flag when not masked
@@ -66,7 +66,7 @@ void Timer::reset(double time) {
 	double delta = (time - startTime);
 	double rem = fmod(delta, delay);
 	double next = delay - rem;
-	startTime = time + next;		
+	startTime = time + next;
 }
 
 void Timer::stop() {
@@ -116,7 +116,7 @@ bool Chip::write(uint32 reg, uint8 val) {
 				timer[1].stop();
 
 			timer[1].masked = (val & 0x20) > 0;
-	
+
 			if (timer[1].masked)
 				timer[1].overflow = false;
 		}
@@ -148,7 +148,7 @@ namespace OPL2 {
 #include "opl_impl.h"
 
 struct Handler : public DOSBox::Handler {
-	void writeReg(uint32 reg, uint8 val) {	
+	void writeReg(uint32 reg, uint8 val) {
 		adlib_write(reg, val);
 	}
 
@@ -171,7 +171,7 @@ namespace OPL3 {
 #include "opl_impl.h"
 
 struct Handler : public DOSBox::Handler {
-	void writeReg(uint32 reg, uint8 val) {	
+	void writeReg(uint32 reg, uint8 val) {
 		adlib_write(reg, val);
 	}
 
@@ -217,13 +217,13 @@ bool OPL::init(int rate) {
 	case Config::kOpl3:
 		_handler = new OPL3::Handler();
 		break;
-	
+
 	default:
 		return false;
 	}
 
 	_handler->init(rate);
-	
+
 	if (_type == Config::kDualOpl2) {
 		// Setup opl3 mode in the hander
 		_handler->writeReg(0x105, 1);
@@ -234,7 +234,7 @@ bool OPL::init(int rate) {
 }
 
 void OPL::reset() {
-	init(_rate);	
+	init(_rate);
 }
 
 void OPL::write(int port, int val) {
@@ -325,7 +325,7 @@ void OPL::writeReg(int r, int v) {
 
 void OPL::dualWrite(uint8 index, uint8 reg, uint8 val) {
 	// Make sure you don't use opl3 features
-	// Don't allow write to disable opl3		
+	// Don't allow write to disable opl3
 	if (reg == 5)
 		return;
 
@@ -334,7 +334,7 @@ void OPL::dualWrite(uint8 index, uint8 reg, uint8 val) {
 		val &= 3;
 
 	// Write to the timer?
-	if (_chip[index].write(reg, val)) 
+	if (_chip[index].write(reg, val))
 		return;
 
 	// Enabling panning
diff --git a/sound/softsynth/opl/opl_impl.h b/sound/softsynth/opl/opl_impl.h
index c5ef87ce3c..5c68baa485 100644
--- a/sound/softsynth/opl/opl_impl.h
+++ b/sound/softsynth/opl/opl_impl.h
@@ -6,12 +6,12 @@
  *  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., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
@@ -131,7 +131,7 @@ static fltype decrelconst[4] = {1/39.28064,1/31.41608,1/26.17344,1/22.44608};
 
 void operator_advance(op_type* op_pt, Bit32s vib) {
 	op_pt->wfpos = op_pt->tcount;						// waveform position
-	
+
 	// advance waveform time
 	op_pt->tcount += op_pt->tinc;
 	op_pt->tcount += (Bit32s)(op_pt->tinc)*vib/FIXEDPT;
@@ -311,7 +311,7 @@ void change_attackrate(Bitu regbase, op_type* op_pt) {
 		op_pt->env_step_a = (1<<(steps<=12?12-steps:0))-1;
 
 		Bits step_num = (step_skip<=48)?(4-(step_skip&3)):0;
-		static Bit8u step_skip_mask[5] = {0xff, 0xfe, 0xee, 0xba, 0xaa}; 
+		static Bit8u step_skip_mask[5] = {0xff, 0xfe, 0xee, 0xba, 0xaa};
 		op_pt->env_step_skip_a = step_skip_mask[step_num];
 
 #if defined(OPLTYPE_IS_OPL3)
@@ -1001,7 +1001,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
 						operator_advance(&cptr[9],vibval1[i]);
 						opfuncs[cptr[9].op_state](&cptr[9]);
 						operator_output(&cptr[9],0,tremval1[i]);
-						
+
 						Bit32s chanval = cptr[9].cval*2;
 						CHANVAL_OUT
 					}
@@ -1033,7 +1033,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
 						operator_advance(&cptr[9],vibval2[i]);
 						opfuncs[cptr[9].op_state](&cptr[9]);
 						operator_output(&cptr[9],cptr[0].cval*FIXEDPT,tremval2[i]);
-						
+
 						Bit32s chanval = cptr[9].cval*2;
 						CHANVAL_OUT
 					}
@@ -1347,7 +1347,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
 
 					} else {
 						// FM-FM-style synthesis (op1[fb] * op2 * op3 * op4)
-						if ((cptr[0].op_state != OF_TYPE_OFF) || (cptr[9].op_state != OF_TYPE_OFF) || 
+						if ((cptr[0].op_state != OF_TYPE_OFF) || (cptr[9].op_state != OF_TYPE_OFF) ||
 							(cptr[3].op_state != OF_TYPE_OFF) || (cptr[3+9].op_state != OF_TYPE_OFF)) {
 							if ((cptr[0].vibrato) && (cptr[0].op_state != OF_TYPE_OFF)) {
 								vibval1 = vibval_var1;
diff --git a/sound/softsynth/opl/opl_inc.h b/sound/softsynth/opl/opl_inc.h
index 0798bd9f6a..b7b6ac1d41 100644
--- a/sound/softsynth/opl/opl_inc.h
+++ b/sound/softsynth/opl/opl_inc.h
@@ -6,12 +6,12 @@
  *  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., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
@@ -132,7 +132,7 @@ typedef struct operator_struct {
 	Bit32u act_state;				// activity state (regular, percussion)
 	bool sus_keep;					// keep sustain level when decay finished
 	bool vibrato,tremolo;			// vibrato/tremolo enable bits
-	
+
 	// variables used to provide non-continuous envelopes
 	Bit32u generator_pos;			// for non-standard sample rates we need to determine how many samples have passed
 	Bits cur_env_step;				// current (standardized) sample position
@@ -151,7 +151,7 @@ Bitu chip_num;
 op_type op[MAXOPERATORS];
 
 Bits int_samplerate;
-	
+
 Bit8u status;
 Bit32u index;
 #if defined(OPLTYPE_IS_OPL3)
diff --git a/sound/timestamp.h b/sound/timestamp.h
index af0cf96ff9..deb86bc2d7 100644
--- a/sound/timestamp.h
+++ b/sound/timestamp.h
@@ -45,23 +45,23 @@ protected:
 
 public:
 	Timestamp();
-	
+
 	/**
 	 * Set up a timestamp with a given time and framerate.
 	 * @param msecs		staring time in milliseconds
 	 * @param frameRate	number of frames per second
 	 */
 	Timestamp(uint32 msecs, int frameRate);
-	
+
 	/** Adds a number of frames to a timestamp. */
 	Timestamp addFrames(int frames) const;
-	
+
 	/** Computes the difference (# of frames) between this timestamp and b. */
 	int frameDiff(const Timestamp &b) const;
-	
+
 	/** Computes the difference (# of milliseconds) between this timestamp and b. */
 	int msecsDiff(const Timestamp &b) const;
-	
+
 	/** Determines the time in milliseconds described by this timestamp. */
 	uint32 msecs() const;
 };
diff --git a/sound/vag.cpp b/sound/vag.cpp
index 46e6a64233..a08c1c0a72 100644
--- a/sound/vag.cpp
+++ b/sound/vag.cpp
@@ -22,7 +22,7 @@
  * $Id$
  *
  */
- 
+
 #include "sound/vag.h"
 
 namespace Audio {
@@ -48,10 +48,10 @@ double f[5][2] = { { 0.0, 0.0 },
 
 int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 	int32 samplesDecoded = 0;
-	
+
 	if (_samplesRemaining) {
 		byte i = 0;
-		
+
 		for (i = 28 - _samplesRemaining; i < 28 && samplesDecoded < numSamples; i++) {
 			_samples[i] = _samples[i] + _s1 * f[_predictor][0] + _s2 * f[_predictor][1];
 			_s2 = _s1;
@@ -70,17 +70,17 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 
 		_samplesRemaining = 0;
 	}
-	
+
 	while (samplesDecoded < numSamples) {
 		byte i = 0;
-	
+
 		_predictor = _stream->readByte();
 		byte shift = _predictor & 0xf;
 		_predictor >>= 4;
-	
+
 		if (_stream->readByte() == 7)
 			return samplesDecoded;
-	
+
 		for (i = 0; i < 28; i += 2) {
 			byte d = _stream->readByte();
 			int16 s = (d & 0xf) << 12;
@@ -92,7 +92,7 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 				s |= 0xffff0000;
 			_samples[i + 1] = (double)(s >> shift);
 		}
-	
+
 		for (i = 0; i < 28 && samplesDecoded < numSamples; i++) {
 			_samples[i] = _samples[i] + _s1 * f[_predictor][0] + _s2 * f[_predictor][1];
 			_s2 = _s1;
@@ -101,7 +101,7 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
 			buffer[samplesDecoded] = d;
 			samplesDecoded++;
 		}
-		
+
 		if (i != 27)
 			_samplesRemaining = 28 - i;
 	}
diff --git a/sound/vag.h b/sound/vag.h
index 228f68279e..f6f79d2b81 100644
--- a/sound/vag.h
+++ b/sound/vag.h
@@ -33,19 +33,19 @@
 
 #include "sound/audiostream.h"
 #include "common/stream.h"
- 
+
 namespace Audio {
 
 class VagStream : public Audio::AudioStream {
 public:
 	VagStream(Common::SeekableReadStream *stream, bool loop = false, int rate = 11025);
 	~VagStream();
-	
+
 	bool isStereo() const { return false; }
 	bool endOfData() const { return _stream->pos() == _stream->size(); }
 	int getRate() const { return _rate; }
 	int readBuffer(int16 *buffer, const int numSamples);
-	void rewind();	
+	void rewind();
 
 private:
 	Common::SeekableReadStream *_stream;