sfx lowpass filter patch

cvar changes:

The "sndspeed" cvar / "-sndspeed" command-line option now control whether the low-pass filter is applied. If it's set to 11025 you get the low-pass filter, otherwise it's not used.

New "snd_mixspeed" cvar (and the "-mixspeed" command-line option); these default to 44100 and just control the sample rate we request from SDL. Not archived.

New “snd_filterquality” cvar, value can be 1-5. Not archived. The “5” setting closely matches the Windows resampler, and the “1” setting closely matches the OS X resampler. The default depends on the OS, “5” is used on windows builds, otherwise “1”, because I wanted the sfx to sound the same as they do with 0.85.9 on each platform. TODO is checking if a setting other than 1 sounds closer to the system resampler on linux (though it probably depends on the distro).

The lowpass filter is only used for sndspeed=11025 and snd_mixspeed=44100, though these are the defaults.


git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@952 af15c1b1-3010-417e-b628-4374ebc0bcbd

Quake/q_sound.h

@@ -168,6 +168,8 @@ extern	vec3_t		listener_right;
 extern	vec3_t		listener_up;
 
 extern	cvar_t		sndspeed;
+extern	cvar_t		snd_mixspeed;
+extern	cvar_t		snd_filterquality;
 extern	cvar_t		sfxvolume;
 extern	cvar_t		loadas8bit;
 

Quake/snd_dma.c

@@ -76,6 +76,16 @@ cvar_t		precache = {"precache", "1", CVAR_NONE};
 cvar_t		loadas8bit = {"loadas8bit", "0", CVAR_NONE};
 
 cvar_t		sndspeed = {"sndspeed", "11025", CVAR_NONE};
+cvar_t		snd_mixspeed = {"snd_mixspeed", "44100", CVAR_NONE};
+
+#if defined(_WIN32)
+#define SND_FILTERQUALITY_DEFAULT "5"
+#else
+#define SND_FILTERQUALITY_DEFAULT "1"
+#endif
+
+cvar_t		snd_filterquality = {"snd_filterquality", SND_FILTERQUALITY_DEFAULT,
+								 CVAR_NONE};
 
 static	cvar_t	nosound = {"nosound", "0", CVAR_NONE};
 static	cvar_t	ambient_level = {"ambient_level", "0.3", CVAR_NONE};
@@ -108,6 +118,14 @@ static void SND_Callback_sfxvolume (cvar_t *var)
 	SND_InitScaletable ();
 }
 
+static void SND_Callback_snd_filterquality (cvar_t *var)
+{
+	if (snd_filterquality.value < 1 || snd_filterquality.value > 5)
+	{
+		Con_Printf ("snd_filterquality must be between 1 and 5\n");
+		Cvar_SetQuick (&snd_filterquality, SND_FILTERQUALITY_DEFAULT);
+	}
+}
 
 /*
 ================
@@ -161,7 +179,9 @@ void S_Init (void)
 	Cvar_RegisterVariable(&snd_show);
 	Cvar_RegisterVariable(&_snd_mixahead);
 	Cvar_RegisterVariable(&sndspeed);
-
+	Cvar_RegisterVariable(&snd_mixspeed);
+	Cvar_RegisterVariable(&snd_filterquality);
+	
 	if (safemode || COM_CheckParm("-nosound"))
 		return;
 
@@ -178,6 +198,12 @@ void S_Init (void)
 	{
 		Cvar_SetQuick (&sndspeed, com_argv[i+1]);
 	}
+	
+	i = COM_CheckParm("-mixspeed");
+	if (i && i < com_argc-1)
+	{
+		Cvar_SetQuick (&snd_mixspeed, com_argv[i+1]);
+	}
 
 	if (host_parms->memsize < 0x800000)
 	{
@@ -186,6 +212,7 @@ void S_Init (void)
 	}
 
 	Cvar_SetCallback(&sfxvolume, SND_Callback_sfxvolume);
+	Cvar_SetCallback(&snd_filterquality, &SND_Callback_snd_filterquality);
 
 	SND_InitScaletable ();
 

Quake/snd_mix.c

@@ -149,6 +149,180 @@ static void S_TransferPaintBuffer (int endtime)
 	}
 }
 
+/*
+==============
+S_MakeBlackmanWindowKernel
+
+Makes a lowpass filter kernel, from equation 16-4 in
+"The Scientist and Engineer's Guide to Digital Signal Processing"
+
+M is the kernel size (not counting the center point), must be even
+kernel has room for M+1 floats
+f_c is the filter cutoff frequency, as a fraction of the samplerate
+==============
+*/
+static void S_MakeBlackmanWindowKernel(float *kernel, int M, float f_c)
+{
+	int i;
+	for (i = 0; i <= M; i++)
+	{
+		if (i == M/2)
+		{
+			kernel[i] = 2 * M_PI * f_c;
+		}
+		else
+		{
+			kernel[i] = ( sin(2 * M_PI * f_c * (i - M/2.0)) / (i - (M/2.0)) )
+				* (0.42 - 0.5*cos(2 * M_PI * i / (double)M)
+				   + 0.08*cos(4 * M_PI * i / (double)M) );
+		}
+	}
+	
+// normalize the kernel so all of the values sum to 1
+	{
+		float sum = 0;
+		for (i = 0; i <= M; i++)
+		{
+			sum += kernel[i];
+		}
+		
+		for (i = 0; i <= M; i++)
+		{
+			kernel[i] /= sum;
+		}
+	}
+}
+
+typedef struct {
+	float *memory;  // kernelsize floats
+	float *kernel;  // kernelsize floats
+	int kernelsize; // M+1, rounded up to be a multiple of 16
+	int M;			// M value used to make kernel, even
+	int parity;		// 0-3
+	float f_c;		// cutoff frequency, [0..1], fraction of sample rate
+} filter_t;
+
+static void S_UpdateFilter(filter_t *filter, int M, float f_c)
+{
+	if (filter->f_c != f_c || filter->M != M)
+	{
+		if (filter->memory != NULL) free(filter->memory);
+		if (filter->kernel != NULL) free(filter->kernel);
+		
+		filter->M = M;
+		filter->f_c = f_c;
+		
+		filter->parity = 0;
+	// M + 1 rounded up to the next multiple of 16
+		filter->kernelsize = (M + 1) + 16 - ((M + 1) % 16);
+		filter->memory = calloc(filter->kernelsize, sizeof(float));
+		filter->kernel = calloc(filter->kernelsize, sizeof(float));
+		
+		S_MakeBlackmanWindowKernel(filter->kernel, M, f_c);
+	}
+}
+
+/*
+==============
+S_ApplyFilter
+
+Lowpass-filter the given buffer containing 44100Hz audio.
+
+As an optimization, it decimates the audio to 11025Hz (setting every sample
+position that's not a multiple of 4 to 0), then convoluting with the filter
+kernel is 4x faster, because we can skip 3/4 of the input samples that are
+known to be 0 and skip 3/4 of the filter kernel.
+==============
+*/
+static void S_ApplyFilter(filter_t *filter, int *data, int stride, int count)
+{
+	int i, j;
+	float *input;
+	const int kernelsize = filter->kernelsize;
+	const float *kernel = filter->kernel;
+	int parity;
+	
+	input = malloc(sizeof(float) * (filter->kernelsize + count));
+	
+// set up the input buffer
+// memory holds the previous filter->kernelsize samples of input.
+	
+	memcpy(input, filter->memory, filter->kernelsize * sizeof(float));
+
+	for (i=0; i<count; i++)
+	{
+		input[filter->kernelsize+i] = data[i * stride] / (32768.0 * 256.0);
+	}
+	
+// copy out the last filter->kernelsize samples to 'memory' for next time
+	
+	memcpy(filter->memory, input + count, filter->kernelsize * sizeof(float));
+	
+// apply the filter
+
+	parity = filter->parity;
+	
+	for (i=0; i<count; i++)
+	{
+		const float *input_plus_i = input + i;
+		float val[4] = {0, 0, 0, 0};
+		
+		for (j = (4 - parity) % 4; j < kernelsize; j+=16)
+		{
+			val[0] += kernel[j] * input_plus_i[j];
+			val[1] += kernel[j+4] * input_plus_i[j+4];
+			val[2] += kernel[j+8] * input_plus_i[j+8];
+			val[3] += kernel[j+12] * input_plus_i[j+12];
+		}
+		
+	// 4.0 factor is to increase volume by 12 dB; this is to make up the
+	// volume drop caused by the zero-filling this filter does.
+		data[i * stride] = (val[0] + val[1] + val[2] + val[3])
+			* (32768.0 * 256.0 * 4.0);
+		
+		parity = (parity + 1) % 4;
+	}
+	
+	filter->parity = parity;
+
+	free(input);
+}
+
+/*
+==============
+S_LowpassFilter
+
+lowpass filters 24-bit integer samples in 'data' (stored in 32-bit ints).
+assumes 44100Hz sample rate, and lowpasses at around 5kHz
+memory should be a zero-filled filter_t struct
+==============
+*/
+static void S_LowpassFilter(int *data, int stride, int count,
+							filter_t *memory)
+{
+	int M;
+	float bw, f_c;
+	
+	switch ((int)snd_filterquality.value)
+	{
+		case 1:
+			M = 126; bw = 0.900; break;
+		case 2:
+			M = 150; bw = 0.915; break;
+		case 3:
+			M = 174; bw = 0.930; break;
+		case 4:
+			M = 198; bw = 0.945; break;
+		case 5:
+		default:
+			M = 222; bw = 0.960; break;
+	}
+
+	f_c = (bw * 11025 / 2.0) / 44100.0;
+	
+	S_UpdateFilter(memory, M, f_c);
+	S_ApplyFilter(memory, data, stride, count);
+}
 
 /*
 ===============================================================================
@@ -178,32 +352,7 @@ void S_PaintChannels (int endtime)
 			end = paintedtime + PAINTBUFFER_SIZE;
 
 	// clear the paint buffer
-		if (s_rawend < paintedtime)
-		{	// clear
-			memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
-		}
-		else
-		{	// copy from the streaming sound source
-			int		s;
-			int		stop;
-
-			stop = (end < s_rawend) ? end : s_rawend;
-
-			for (i = paintedtime; i < stop; i++)
-			{
-				s = i & (MAX_RAW_SAMPLES - 1);
-				paintbuffer[i - paintedtime] = s_rawsamples[s];
-			}
-		//	if (i != end)
-		//		Con_Printf ("partial stream\n");
-		//	else
-		//		Con_Printf ("full stream\n");
-			for ( ; i < end; i++)
-			{
-				paintbuffer[i - paintedtime].left =
-				paintbuffer[i - paintedtime].right = 0;
-			}
-		}
+		memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
 
 	// paint in the channels.
 		ch = snd_channels;
@@ -255,6 +404,44 @@ void S_PaintChannels (int endtime)
 			}
 		}
 
+	// clip each sample to 0dB, then reduce by 6dB (to leave some headroom for
+	// the lowpass filter and the music). the lowpass will smooth out the
+	// clipping
+		for (i=0; i<end-paintedtime; i++)
+		{
+			paintbuffer[i].left = CLAMP(-32768 << 8, paintbuffer[i].left, 32767 << 8) >> 1;
+			paintbuffer[i].right = CLAMP(-32768 << 8, paintbuffer[i].right, 32767 << 8) >> 1;
+		}
+		
+	// apply a lowpass filter
+		if (sndspeed.value == 11025 && shm->speed == 44100)
+		{
+			static filter_t memory_l, memory_r;
+			S_LowpassFilter((int *)paintbuffer,       2, end - paintedtime, &memory_l);
+			S_LowpassFilter(((int *)paintbuffer) + 1, 2, end - paintedtime, &memory_r);
+		}
+		
+	// paint in the music
+		if (s_rawend >= paintedtime)
+		{	// copy from the streaming sound source
+			int		s;
+			int		stop;
+			
+			stop = (end < s_rawend) ? end : s_rawend;
+			
+			for (i = paintedtime; i < stop; i++)
+			{
+				s = i & (MAX_RAW_SAMPLES - 1);
+			// lower music by 6db to match sfx
+				paintbuffer[i - paintedtime].left += s_rawsamples[s].left >> 1;
+				paintbuffer[i - paintedtime].right += s_rawsamples[s].right >> 1;
+			}
+			//	if (i != end)
+			//		Con_Printf ("partial stream\n");
+			//	else
+			//		Con_Printf ("full stream\n");
+		}
+		
 	// transfer out according to DMA format
 		S_TransferPaintBuffer(end);
 		paintedtime = end;

Quake/snd_sdl.c

@@ -86,7 +86,7 @@ qboolean SNDDMA_Init (dma_t *dma)
 	}
 
 	/* Set up the desired format */
-	desired.freq = tmp = sndspeed.value;
+	desired.freq = tmp = snd_mixspeed.value;
 	desired.format = (loadas8bit.value) ? AUDIO_U8 : AUDIO_S16SYS;
 	desired.channels = 2; /* = desired_channels; */
 	if (desired.freq <= 11025)