apply astyle

git-svn-id: https://mpc-hc.svn.sourceforge.net/svnroot/mpc-hc/trunk@4039 10f7b99b-c216-0410-bff0-8a66a9350fd8

1 files changed, 24 insertions, 24 deletions

diff --git a/src/filters/switcher/AudioSwitcher/Audio.cpp b/src/filters/switcher/AudioSwitcher/Audio.cpp
index 8f5a905ef..e6c872e28 100644
--- a/src/filters/switcher/AudioSwitcher/Audio.cpp
+++ b/src/filters/switcher/AudioSwitcher/Audio.cpp
@@ -32,7 +32,7 @@ static long audio_pointsample_8(void *dst, void *src, long accum, long samp_frac
 	do {

 		*d++ = s[accum>>19];

 		accum += samp_frac;

-	} while(--cnt);

+	} while (--cnt);

 

 	return accum;

 }

@@ -45,7 +45,7 @@ static long audio_pointsample_16(void *dst, void *src, long accum, long samp_fra
 	do {

 		*d++ = s[accum>>19];

 		accum += samp_frac;

-	} while(--cnt);

+	} while (--cnt);

 

 	return accum;

 }

@@ -58,7 +58,7 @@ static long audio_pointsample_32(void *dst, void *src, long accum, long samp_fra
 	do {

 		*d++ = s[accum>>19];

 		accum += samp_frac;

-	} while(--cnt);

+	} while (--cnt);

 

 	return accum;

 }

@@ -79,7 +79,7 @@ static long audio_downsample_mono8(void *dst, void *src, long *filter_bank, int
 		s_ptr = s + (accum>>19);

 		do {

 			sum += *fb_ptr++ * (int)*s_ptr++;

-		} while(--w);

+		} while (--w);

 

 		if (sum < 0) {

 			*d++ = 0;

@@ -90,7 +90,7 @@ static long audio_downsample_mono8(void *dst, void *src, long *filter_bank, int
 		}

 

 		accum += samp_frac;

-	} while(--cnt);

+	} while (--cnt);

 

 	return accum;

 }

@@ -111,7 +111,7 @@ static long audio_downsample_mono16(void *dst, void *src, long *filter_bank, int
 		s_ptr = s + (accum>>19);

 		do {

 			sum += *fb_ptr++ * (int)*s_ptr++;

-		} while(--w);

+		} while (--w);

 

 		if (sum < -0x20000000) {

 			*d++ = -0x8000;

@@ -122,7 +122,7 @@ static long audio_downsample_mono16(void *dst, void *src, long *filter_bank, int
 		}

 

 		accum += samp_frac;

-	} while(--cnt);

+	} while (--cnt);

 

 	return accum;

 }

@@ -144,7 +144,7 @@ static void make_downsample_filter(long *filter_bank, int filter_width, long sam
 

 	filt_max = (16384.0 * 524288.0) / samp_frac;

 

-	for(i=0; i<128*filter_width; i++) {

+	for (i=0; i<128*filter_width; i++) {

 		int y = 0;

 		double d = i / filtwidth_frac;

 

@@ -153,22 +153,22 @@ static void make_downsample_filter(long *filter_bank, int filter_width, long sam
 		}

 

 		filter_bank[permute_index(128*filter_width + i, filter_width)]

-		= filter_bank[permute_index(128*filter_width - i, filter_width)]

-		  = y;

+			= filter_bank[permute_index(128*filter_width - i, filter_width)]

+			  = y;

 	}

 

 	// Normalize the filter to correct for integer roundoff errors

 

-	for(i=0; i<256*filter_width; i+=filter_width) {

+	for (i=0; i<256*filter_width; i+=filter_width) {

 		v=0;

-		for(j=0; j<filter_width; j++) {

+		for (j=0; j<filter_width; j++) {

 			v += filter_bank[i+j];

 		}

 

 		//		_RPT2(0,"error[%02x] = %04x\n", i/filter_width, 0x4000 - v);

 

 		v = (0x4000 - v)/filter_width;

-		for(j=0; j<filter_width; j++) {

+		for (j=0; j<filter_width; j++) {

 			filter_bank[i+j] += v;

 		}

 	}

@@ -182,10 +182,10 @@ AudioStreamResampler::AudioStreamResampler(int bps, long org_rate, long new_rate
 

 	this->bps = bps;

 

-	if(bps == 1) {

+	if (bps == 1) {

 		ptsampleRout = audio_pointsample_8;

 		dnsampleRout = audio_downsample_mono8;

-	} else if(bps >= 2) {

+	} else if (bps >= 2) {

 		ptsampleRout = audio_pointsample_16;

 		dnsampleRout = audio_downsample_mono16;

 	} else {

@@ -202,14 +202,14 @@ AudioStreamResampler::AudioStreamResampler(int bps, long org_rate, long new_rate
 

 	// If this is a high-quality downsample, allocate memory for the filter bank

 

-	if(fHighQuality) {

-		if(samp_frac>0x80000) {

+	if (fHighQuality) {

+		if (samp_frac>0x80000) {

 			// HQ downsample: allocate filter bank

 

 			filter_width = ((samp_frac + 0x7ffff)>>19)<<1 <<1;

 

 			filter_bank = DNew long[filter_width * 256];

-			if(!filter_bank) {

+			if (!filter_bank) {

 				filter_width = 1;

 				return;

 			}

@@ -239,7 +239,7 @@ long AudioStreamResampler::Downsample(void* input, long samplesin, void* output,
 	//

 	// We need (n/2) points to the left and (n/2-1) points to the right.

 

-	while(samplesin > 0 && samplesout > 0) {

+	while (samplesin > 0 && samplesout > 0) {

 		long srcSamples, dstSamples;

 		int nhold;

 

@@ -253,14 +253,14 @@ long AudioStreamResampler::Downsample(void* input, long samplesin, void* output,
 

 		// Don't exceed the buffer (BUFFER_SIZE - holdover).

 

-		if(srcSamples > BUFFER_SIZE - holdover) {

+		if (srcSamples > BUFFER_SIZE - holdover) {

 			srcSamples = BUFFER_SIZE - holdover;

 		}

 

 		// Read into buffer.

 

 		srcSamples = min(srcSamples, samplesin);

-		if(!srcSamples) {

+		if (!srcSamples) {

 			break;

 		}

 

@@ -274,12 +274,12 @@ long AudioStreamResampler::Downsample(void* input, long samplesin, void* output,
 

 		dstSamples = (((__int64)(srcSamples+holdover-filter_width)<<19) + 0x7ffff - accum) / samp_frac + 1;

 

-		if(dstSamples > samplesout) {

+		if (dstSamples > samplesout) {

 			dstSamples = samplesout;

 		}

 

-		if(dstSamples >= 1) {

-			if(filter_bank) {

+		if (dstSamples >= 1) {

+			if (filter_bank) {

 				accum = dnsampleRout(output, cbuffer, filter_bank, filter_width, accum, samp_frac, dstSamples);

 			} else {

 				accum = ptsampleRout(output, cbuffer, accum, samp_frac, dstSamples);