Corrected a mono bug and fixed an error in the resampling routine...

git-svn-id: https://svn.musicpd.org/mpd/branches/ancl@3293 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 59 insertions, 70 deletions

diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index 8454c8871..1dbc7b3a0 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -117,13 +117,13 @@ void pcm_convertSampleRate(
 		AudioFormat * inFormat, mpd_fixed_t * inBuffer, int inFrames, 
 		AudioFormat *outFormat, mpd_fixed_t *outBuffer, int outFrames) 
 {
-	static int inRate; 		/* reduced bitRate */
+	static int inRate;
 	static int outRate;
 	static int shift;
-	static int unshift;
+	static int rateShift;
 	static mpd_fixed_t oldSampleL = 0;
 	static mpd_fixed_t oldSampleR = 0;
-	mpd_uint32 curTime;
+	int delay;
 	
 	/* recalculate static data if samplerate has changed */
 	if(inFormat->sampleRate != convSampleRateData.inRate || 
@@ -135,11 +135,11 @@ void pcm_convertSampleRate(
 		/* calculate the rates to use in calculations... */
 		inRate = inFormat->sampleRate;
 		outRate = outFormat->sampleRate;
-		unshift=0;
+		rateShift=0;
 		shift = 16; /* worst case shift */
 		/* ...reduce them to minimize shifting */
 		while(!(inRate & 1) && !(outRate & 1)) {
-			unshift++;
+			rateShift++;
 			shift--;
 			inRate >>= 1;
 			outRate >>= 1;
@@ -149,59 +149,50 @@ void pcm_convertSampleRate(
 	}
 
 	/* compute */
-	curTime = convSampleRateData.delay >> unshift;
-/*
-ERROR("pcm_convertSampleRate2: inFrames=%i, outFrames=%i, inRate=%i, outRate=%i, curTime=%i\n",
-		inFrames, outFrames, inRate, outRate, curTime);*/
+	
+	delay = convSampleRateData.delay >> rateShift;
 	switch(inFormat->channels) {
 	case 1:
 		while(inFrames--) {
-			curTime += outRate;
+			delay += outRate;
 			/* calculate new samples */
-			while(curTime >= inRate) {
-				curTime -= inRate;
+			while(delay >= inRate) {
+				mpd_sint32 raise;
+				delay -= inRate;
+				raise = *inBuffer - oldSampleL;
 				*outBuffer++ = oldSampleL + 
-					(((*inBuffer - oldSampleL) * 
-					  ((outRate - curTime) >> shift)) / 
-					 outRate);
-				outFrames--;
+					((((raise>>shift) * (outRate - delay)) / 
+					 outRate) << shift);
 			}
 			oldSampleL = *inBuffer++;
 		}
+		break;
 	case 2:
 		while(inFrames--) {
-			curTime += outRate;
+			delay += outRate;
 			/* calculate new samples */
-			while(curTime >= inRate) {
-				curTime -= inRate;
+			while(delay >= inRate) {
+				mpd_sint32 raise;
+				delay -= inRate;
+				/* left channel */
+				raise = *inBuffer - oldSampleL;
 				*outBuffer++ = oldSampleL + 
-					(((*inBuffer - oldSampleL) * 
-					  ((outRate - curTime) >> shift)) / 
-					 outRate);
+					((((raise>>shift) * (outRate - delay)) / 
+					 outRate) << shift);
+				/* right channel */
+				raise = inBuffer[1] - oldSampleR;
 				*outBuffer++ = oldSampleR + 
-					(((inBuffer[1] - oldSampleR) *
-					  ((outRate - curTime) >> shift)) /
-					 outRate);
-				outFrames--;
+					((((raise>>shift) * (outRate - delay)) /
+					 outRate) << shift);
 			}
 			oldSampleL = *inBuffer++;
 			oldSampleR = *inBuffer++;
 		}
-	}		
-	convSampleRateData.delay = curTime << unshift;
-/*
-ERROR("pcm_convertSampleRate2: inFrames=%i, outFrames=%i, curTime=%i\n\n",
-		inFrames, outFrames, curTime);*/
-
-	/* some temporary debugging tests */
-	if(inFrames>0) {
-		ERROR("pcm_convertSampleRate produced to few outFrames!\n");
-		exit(EXIT_FAILURE);
-	}
-	if(outFrames>0) {
-		ERROR("pcm_convertSampleRate produced to many outFrames!\n");
-		exit(EXIT_FAILURE);
+		/* exit(EXIT_FAILURE);*/
+		break;
 	}
+
+	convSampleRateData.delay = delay << rateShift;
 }
 	
 
@@ -358,9 +349,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 			inFormat, inSize, outFormat);
 	const int outSamples = (outSize << 3) / outFormat->bits;
 	const int outFrames = outSamples / outFormat->channels;
-/*ERROR("pcm_convertAudioFormat: outSize=%i, outSamples=%i, outFrames=%i," 
-	"outFormat.bits=%i, outFormat.channels=%i\n", 
-	outSize, outSamples, outFrames, outFormat->bits, outFormat->channels);*/
+	
 	/* make sure convBuffer is big enough for 2 channels of 32 bit samples */
 	dataLen = inFrames << 3;
 	if(dataLen > convBufferLength) {
@@ -384,6 +373,27 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 				dataConv, fracBits);
 	}
 	
+	/****** convert sample rate ******/
+
+	if(inFormat->sampleRate != outFormat->sampleRate) {
+		/* check size of buffer */
+		dataLen = outFrames << 3;
+		if(dataLen > convSampleBufferLength) {
+			convSampleBuffer = (char *) 
+				realloc(convSampleBuffer, dataLen);
+			if(!convSampleBuffer) {
+				ERROR("Could not allocate memory for " 
+						"convSampleBuffer!\n");
+				exit(EXIT_FAILURE);
+			}
+			convSampleBufferLength = dataLen;
+		}
+		/* convert samples */
+		pcm_convertSampleRate(inFormat, (mpd_fixed_t*)dataConv, inFrames, 
+			outFormat, (mpd_fixed_t*)convSampleBuffer, outFrames);
+		dataConv = convSampleBuffer;
+	}
+
 	/****** convert between mono and stereo samples ******/
 	
 	if(inFormat->channels != outFormat->channels) {
@@ -394,7 +404,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 			/* in reverse order to allow for same in and out buffer */
 			mpd_fixed_t *in = ((mpd_fixed_t *)dataConv)+inFrames;
 			mpd_fixed_t *out = ((mpd_fixed_t *)convBuffer)+(inFrames<<1);
-			int f = inFrames;
+			int f = outFrames;
 			while(f--) {
 				*out-- = *in;
 				*out-- = *in--;
@@ -406,7 +416,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 			{
 			mpd_fixed_t *in = ((mpd_fixed_t *)dataConv);
 			mpd_fixed_t *out = ((mpd_fixed_t *)convBuffer);
-			int f = inFrames;
+			int f = outFrames;
 			while(f--) {
 				*out = (*in++)>>1;
 				*out++ += (*in++)>>1;
@@ -420,30 +430,10 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 		dataConv = convBuffer;
 	}
 
-	/****** convert sample rate ******/
-
-	if(inFormat->sampleRate != outFormat->sampleRate) {
-		/* check size of buffer */
-		dataLen = outFrames << 3;
-		if(dataLen > convSampleBufferLength) {
-			convSampleBuffer = (char *) 
-				realloc(convSampleBuffer, dataLen);
-			if(!convSampleBuffer) {
-				ERROR("Could not allocate memory for " 
-						"convSampleBuffer!\n");
-				exit(EXIT_FAILURE);
-			}
-			convSampleBufferLength = dataLen;
-		}
-		/* convert samples */
-		pcm_convertSampleRate(inFormat, (mpd_fixed_t*)dataConv, inFrames, 
-			outFormat, (mpd_fixed_t*)convSampleBuffer, outFrames);
-		dataConv = convSampleBuffer;
-	}
-
 	/****** convert to output format ******/
 
-	/* if outformat is mpd_fixed_t then we are done ?! TODO */
+	/* if outformat is mpd_fixed_t then we are done ?! 
+	 * TODO take care of case when in and out have different fracBits */
 	if(outFormat->fracBits==fracBits) {
 		if(outFormat->bits==32) {
 			if(outBuffer != dataConv)
@@ -510,6 +500,8 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
 	if(inFormat->sampleRate == outFormat->sampleRate)
 		outFrames = inFrames;
 	else {
+		/* The previous delay from the sample rate conversion affect 
+		 * the size of the output */
 		mpd_uint32 delay = convSampleRateData.delay;
 		if(inFormat->sampleRate != convSampleRateData.inRate || 
 			outFormat->sampleRate != convSampleRateData.outRate) 
@@ -518,9 +510,6 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
 		}
 		outFrames = (inFrames * (mpd_uint32)(outFormat->sampleRate) 
 				+ delay) / inFormat->sampleRate;
-/*
-ERROR("pcm_size... inSize=%i, inShift=%i, outShift=%i, inFrames=%i, outFrames=%i, delay=%i\n", 
-	inSize, inShift, outShift, inFrames, outFrames, delay);	*/
 	}
 	return outFrames * outShift;
 }