Re: [hatari-devel] Gain too low in DmaSnd_LowPassFilter for STE ?

[David Savinkoff <dsavnkff@xxxxxxxxx>]

----- Nicolas Pomarède wrote:
> Hi
> 
> as discussed here 
> http://atari-forum.com/viewtopic.php?f=16&t=29615&p=299270#p299270 the 
> STE DMA sound samples can't be heard when using an output freq < 50066 
> Hz in Hatari (for example, try "salamander : planet latis")
> 
> At 50066 Hz you hear the samples, but at 48 kHz you don't hear them 
> anymore. This is because 50066 explicitely turn DMA low pass filter OFF 
> in the code path.
> 
> David, do you think it would be possible to increase the gain in 
> DmaSnd_LowPassFilter to hear the samples even when using < 50 kHz output 
> freq ?
> 
> Note that in this music demo, samples already have a low volume (if you 
> just listen to the wav file with Audacity for example), so maybe the 
> fact that the sample don't use the full 8 bit range is fooling the 
> filter and you get an even lower volume in the end ?
> 
> Nicolas

Hi,
I've made another patch. This one was tested with the STE Pacemaker
demo for sound quality. Please test it to your specifications.

--- hatari/src/dmaSnd.c	2016-08-06 10:07:17.000000000 -0700
+++ hatari/src/dmaSnd.c	2016-08-07 18:33:07.142586393 -0700
@@ -510,6 +510,12 @@
 		return;
 	}
 
+	/* DMA Anti-alias filter */
+	if (DmaSnd_DetectSampleRate() >  nAudioFrequency)
+		DmaSnd_LowPass = true;
+	else
+		DmaSnd_LowPass = false;
+
 
 	/* DMA Audio ON or FIFO not empty yet */
 
@@ -1259,16 +1265,14 @@
 	static	Sint16	out = 0;
 
 	if (DmaSnd_LowPass)
-	{
 		out = lowPassFilter[0] + (lowPassFilter[1]<<1) + in;
-		lowPassFilter[0] = lowPassFilter[1];
-		lowPassFilter[1] = in;
+	else
+		out = lowPassFilter[1] << 2;
 
-		return out; /* Filter Gain = 4 */
-	}else
-	{
-		return in << 2;
-	}
+	lowPassFilter[0] = lowPassFilter[1];
+	lowPassFilter[1] = in;
+
+	return out; /* Filter Gain = 4 */
 }
 
 /**
@@ -1280,16 +1284,14 @@
 	static	Sint16	out = 0;
 
 	if (DmaSnd_LowPass)
-	{
 		out = lowPassFilter[0] + (lowPassFilter[1]<<1) + in;
-		lowPassFilter[0] = lowPassFilter[1];
-		lowPassFilter[1] = in;
+	else
+		out = lowPassFilter[1] << 2;
 
-		return out; /* Filter Gain = 4 */
-	}else
-	{
-		return in << 2;
-	}
+	lowPassFilter[0] = lowPassFilter[1];
+	lowPassFilter[1] = in;
+
+	return out; /* Filter Gain = 4 */
 }
 
 /**
@@ -1404,10 +1406,4 @@
 	/* Initialize IIR Filter Gain and use as a Volume Control */
 	lmc1992.left_gain = (microwire.leftVolume * (Uint32)microwire.masterVolume) * (2.0/(65536.0*65536.0));
 	lmc1992.right_gain = (microwire.rightVolume * (Uint32)microwire.masterVolume) * (2.0/(65536.0*65536.0));
-
-	/* Anti-alias filter is not required when nAudioFrequency == 50066 Hz */
-	if (nAudioFrequency>50000 && nAudioFrequency<50100)
-		DmaSnd_LowPass = false;
-	else
-		DmaSnd_LowPass = true;
 }