JANITORIAL: Remove trailing whitespaces.

Powered by:
git ls-files "*.cpp" "*.h" "*.m" "*.mm" | xargs sed -i -e 's/[ \t]*$//'

6 files changed, 7 insertions, 7 deletions

diff --git a/audio/softsynth/mt32/Part.cpp b/audio/softsynth/mt32/Part.cpp
index c9bd86b54a..75912f38a8 100644
--- a/audio/softsynth/mt32/Part.cpp
+++ b/audio/softsynth/mt32/Part.cpp
@@ -411,7 +411,7 @@ void RhythmPart::noteOn(unsigned int midiKey, unsigned int velocity) {
 	// According to info from Mok, keyShift does not appear to affect anything on rhythm part on LAPC-I, but may do on MT-32 - needs investigation
 	synth->printDebug(" Patch: (timbreGroup %u), (timbreNum %u), (keyShift %u), fineTune %u, benderRange %u, assignMode %u, (reverbSwitch %u)", patchTemp->patch.timbreGroup, patchTemp->patch.timbreNum, patchTemp->patch.keyShift, patchTemp->patch.fineTune, patchTemp->patch.benderRange, patchTemp->patch.assignMode, patchTemp->patch.reverbSwitch);
 	synth->printDebug(" PatchTemp: outputLevel %u, (panpot %u)", patchTemp->outputLevel, patchTemp->panpot);
-	synth->printDebug(" RhythmTemp: timbre %u, outputLevel %u, panpot %u, reverbSwitch %u", rhythmTemp[drumNum].timbre, rhythmTemp[drumNum].outputLevel, rhythmTemp[drumNum].panpot, rhythmTemp[drumNum].reverbSwitch); 
+	synth->printDebug(" RhythmTemp: timbre %u, outputLevel %u, panpot %u, reverbSwitch %u", rhythmTemp[drumNum].timbre, rhythmTemp[drumNum].outputLevel, rhythmTemp[drumNum].panpot, rhythmTemp[drumNum].reverbSwitch);
 #endif
 #endif
 	playPoly(drumCache[drumNum], &rhythmTemp[drumNum], midiKey, key, velocity);
diff --git a/audio/softsynth/mt32/Partial.h b/audio/softsynth/mt32/Partial.h
index 95218c858c..5e250769ec 100644
--- a/audio/softsynth/mt32/Partial.h
+++ b/audio/softsynth/mt32/Partial.h
@@ -37,7 +37,7 @@ private:
 	const int debugPartialNum; // Only used for debugging
 	// Number of the sample currently being rendered by generateSamples(), or 0 if no run is in progress
 	// This is only kept available for debugging purposes.
-	unsigned long sampleNum; 
+	unsigned long sampleNum;
 
 	int ownerPart; // -1 if unassigned
 	int mixType;
diff --git a/audio/softsynth/mt32/PartialManager.cpp b/audio/softsynth/mt32/PartialManager.cpp
index 42a3eaa179..0a6be826d6 100644
--- a/audio/softsynth/mt32/PartialManager.cpp
+++ b/audio/softsynth/mt32/PartialManager.cpp
@@ -148,7 +148,7 @@ bool PartialManager::abortFirstPolyPreferHeldWhereReserveExceeded(int minPart) {
 bool PartialManager::freePartials(unsigned int needed, int partNum) {
 	// CONFIRMED: Barring bugs, this matches the real LAPC-I according to information from Mok.
 
-	// BUG: There's a bug in the LAPC-I implementation: 
+	// BUG: There's a bug in the LAPC-I implementation:
 	// When allocating for rhythm part, or when allocating for a part that is using fewer partials than it has reserved,
 	// held and playing polys on the rhythm part can potentially be aborted before releasing polys on the rhythm part.
 	// This bug isn't present on MT-32.
diff --git a/audio/softsynth/mt32/Synth.cpp b/audio/softsynth/mt32/Synth.cpp
index 0861053b5c..7a1b5c2275 100644
--- a/audio/softsynth/mt32/Synth.cpp
+++ b/audio/softsynth/mt32/Synth.cpp
@@ -1156,7 +1156,7 @@ void Synth::writeMemoryRegion(const MemoryRegion *region, Bit32u addr, Bit32u le
 			DT(partial[x].tva.envLevel[0]); \
 			DT(partial[x].tva.envLevel[1]); \
 			DT(partial[x].tva.envLevel[2]); \
-			DT(partial[x].tva.envLevel[3]); 
+			DT(partial[x].tva.envLevel[3]);
 
 			DTP(0);
 			DTP(1);
diff --git a/audio/softsynth/mt32/TVF.cpp b/audio/softsynth/mt32/TVF.cpp
index 58f72e5a9b..80b592ea67 100644
--- a/audio/softsynth/mt32/TVF.cpp
+++ b/audio/softsynth/mt32/TVF.cpp
@@ -64,11 +64,11 @@ static int calcBaseCutoff(const TimbreParam::PartialParam *partialParam, Bit32u
 	int biasPoint = partialParam->tvf.biasPoint;
 	if ((biasPoint & 0x40) == 0) {
 		// biasPoint range here: 0 to 63
-		int bias = biasPoint + 33 - key; // bias range here: -75 to 84 
+		int bias = biasPoint + 33 - key; // bias range here: -75 to 84
 		if (bias > 0) {
 			bias = -bias; // bias range here: -1 to -84
 			baseCutoff += bias * biasLevelToBiasMult[partialParam->tvf.biasLevel]; // Calculation range: -7140 to 7140
-			// baseCutoff range now: -10164 to 10164 
+			// baseCutoff range now: -10164 to 10164
 		}
 	} else {
 		// biasPoint range here: 64 to 127
diff --git a/audio/softsynth/mt32/freeverb.cpp b/audio/softsynth/mt32/freeverb.cpp
index de8f2632cb..181b878596 100644
--- a/audio/softsynth/mt32/freeverb.cpp
+++ b/audio/softsynth/mt32/freeverb.cpp
@@ -202,7 +202,7 @@ void revmodel::process(const float *inputL, const float *inputR, float *outputL,
 		// Calculate output REPLACING anything already there
 		*outputL = outL*wet1 + outR*wet2;
 		*outputR = outR*wet1 + outL*wet2;
-		
+
 		inputL++;
 		inputR++;
 		outputL++;