[PATCH] Add support for gain in softvol plugin

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Hi all,

What do you think of this patch? I tested this using a stac9205 HDA codec with all sample formats supported by softvol (s32, s16, s24, big and little endian).

The potential drawback I see is that softvol now has to use a intermediate long long multiply and add when computing amplified 32-bit samples. This wasn't an issue with the system I tested on, but possibly could be on slower machines. Also, are their any portability issues with using a long long datatype in alsa-lib? What is the list of embedded architectures that alsa-lib is targeted to build on? I see long long used in many other places in alsa-lib, so I'm hoping this isn't an issue.

Steve

searching for changes
changeset:   2284:b5cecd5c8e13
tag:         tip
user:        stevel@xxxxxxxxxxxxxxxxx
date:        Thu May 03 11:49:45 2007 -0700
summary:     Add support for gain in softvol plugin.

This patch allows for gain in the softvol plugin, in addition to attenuation.
The plugin now has a "max_dB" parameter (up to 50 dB) as well as the
original "min_dB" parameter (down to -51 dB). max_dB defaults to 0 dB, so
unless max_dB is specified in a device conf, the behavior of the plugin will
be the same as before (attenuation only).

The patch also creates a new pcm device "dsnoop_softvol". It's a softvol
device with min_dB set to -30 dB and max_dB set to 40 dB. I came up
with those numbers sort of subjectively by experimenting with record
levels using a digital mic on the stac9205 codec. The softvol plugin allows
a range of -51 to +50 dB, so max_dB could be increased to 50. But
eventually samples are going to get clipped. At 40 dB I was beginning
to get clipping when recording a sample sound at a "reasonably soft"
volume. The dsnoop_softvol device uses dsnoop as its slave device, and
it contains a mixer volume control named "Capture Softvol".

HDA-Intel.conf is also modified to use dsnoop_softvol for its default capture.
So now, capture is filtered through softvol (range -30 to +40 dB) before
being passed on to dsnoop as before.

The motivation for this work is that some HDA codecs have no hardware gain
control for some paths. For instance, the stac9205 has support for digital
mics, but there is no gain control widget for this signal before it is placed
on the Azalia link (only a mute). Therefore gain can only be accomplished
via software.

Signed-off-by: Steve Longerbeam <stevel@xxxxxxxxxxxxxxxxx>

diff -r dd433d4613b4 -r b5cecd5c8e13 src/conf/alsa.conf
--- a/src/conf/alsa.conf	Wed May 02 08:45:45 2007 +0200
+++ b/src/conf/alsa.conf	Thu May 03 11:49:45 2007 -0700
@@ -73,6 +73,8 @@ defaults.pcm.dmix.device defaults.pcm.de
 defaults.pcm.dmix.device defaults.pcm.device
 defaults.pcm.dsnoop.card defaults.pcm.card
 defaults.pcm.dsnoop.device defaults.pcm.device
+defaults.pcm.dsnoop_softvol.card defaults.pcm.card
+defaults.pcm.dsnoop_softvol.device defaults.pcm.device
 defaults.pcm.front.card defaults.pcm.card
 defaults.pcm.front.device defaults.pcm.device
 defaults.pcm.rear.card defaults.pcm.card
@@ -127,6 +129,7 @@ pcm.spdif iec958
 pcm.spdif iec958
 pcm.dmix cards.pcm.dmix
 pcm.dsnoop cards.pcm.dsnoop
+pcm.dsnoop_softvol cards.pcm.dsnoop_softvol
 pcm.modem cards.pcm.modem
 pcm.phoneline cards.pcm.phoneline
 
diff -r dd433d4613b4 -r b5cecd5c8e13 src/conf/cards/HDA-Intel.conf
--- a/src/conf/cards/HDA-Intel.conf	Wed May 02 08:45:45 2007 +0200
+++ b/src/conf/cards/HDA-Intel.conf	Thu May 03 11:49:45 2007 -0700
@@ -19,6 +19,8 @@ HDA-Intel.pcm.front.0 {
 		card $CARD
 	}
 }	
+
+<confdir:pcm/dsnoop_softvol.conf>
 
 # default with dmix+softvol & dsnoop
 HDA-Intel.pcm.default {
@@ -45,7 +47,7 @@ HDA-Intel.pcm.default {
 		type plug
 		slave.pcm {
 			@func concat
-			strings [ "dsnoop:" $CARD ]
+			strings [ "dsnoop_softvol:" $CARD ]
 		}
 	}
 }
diff -r dd433d4613b4 -r b5cecd5c8e13 src/conf/pcm/Makefile.am
--- a/src/conf/pcm/Makefile.am	Wed May 02 08:45:45 2007 +0200
+++ b/src/conf/pcm/Makefile.am	Thu May 03 11:49:45 2007 -0700
@@ -2,7 +2,7 @@ cfg_files = default.conf front.conf rear
 	    surround40.conf surround41.conf \
 	    surround50.conf surround51.conf \
 	    surround71.conf iec958.conf modem.conf \
-	    dmix.conf dsnoop.conf \
+	    dmix.conf dsnoop.conf dsnoop_softvol.conf \
 	    dpl.conf
 
 EXTRA_DIST = $(cfg_files)
diff -r dd433d4613b4 -r b5cecd5c8e13 src/conf/pcm/dsnoop_softvol.conf
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/conf/pcm/dsnoop_softvol.conf	Thu May 03 11:49:45 2007 -0700
@@ -0,0 +1,17 @@
+pcm.!dsnoop_softvol {
+        @args [ CARD ]
+        @args.CARD {
+                type string
+        }
+        type softvol
+	slave.pcm {
+		@func concat
+		strings [ "dsnoop:" $CARD ]
+	}
+        control {
+                name "Capture SoftVol"
+                card $CARD
+        }
+        min_dB -30.0
+        max_dB 40.0
+}
diff -r dd433d4613b4 -r b5cecd5c8e13 src/pcm/pcm_softvol.c
--- a/src/pcm/pcm_softvol.c	Wed May 02 08:45:45 2007 +0200
+++ b/src/pcm/pcm_softvol.c	Thu May 03 11:49:45 2007 -0700
@@ -25,7 +25,7 @@
  *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  *
  */
-  
+
 #include <byteswap.h>
 #include <math.h>
 #include "pcm_local.h"
@@ -46,17 +46,22 @@ typedef struct {
 	snd_ctl_t *ctl;
 	snd_ctl_elem_value_t elem;
 	unsigned int cur_vol[2];
-	unsigned int max_val;
+	unsigned int max_val;     /* max index */
+	unsigned int zero_dB_val; /* index at 0 dB */
 	double min_dB;
-	unsigned short *dB_value;
+	double max_dB;
+	unsigned int *dB_value;
 } snd_pcm_softvol_t;
 
 #define VOL_SCALE_SHIFT		16
+#define VOL_SCALE_MASK          ((1 << VOL_SCALE_SHIFT) - 1)
 
 #define PRESET_RESOLUTION	256
 #define PRESET_MIN_DB		-51.0
-
-static unsigned short preset_dB_value[PRESET_RESOLUTION] = {
+#define ZERO_DB                  0.0
+#define MAX_DB_UPPER_LIMIT      50
+
+static unsigned int preset_dB_value[PRESET_RESOLUTION] = {
 	0x00b8, 0x00bd, 0x00c1, 0x00c5, 0x00ca, 0x00cf, 0x00d4, 0x00d9,
 	0x00de, 0x00e3, 0x00e8, 0x00ed, 0x00f3, 0x00f9, 0x00fe, 0x0104,
 	0x010a, 0x0111, 0x0117, 0x011e, 0x0124, 0x012b, 0x0132, 0x0139,
@@ -96,10 +101,10 @@ typedef union {
 	int i;
 	short s[2];
 } val_t;
-static inline int MULTI_DIV_int(int a, unsigned short b, int swap)
+static inline int MULTI_DIV_32x16(int a, unsigned short b, int swap)
 {
 	val_t v, x, y;
-	v.i = swap ? (int)bswap_32(a) : a;
+	v.i = a;
 	y.i = 0;
 #if __BYTE_ORDER == __LITTLE_ENDIAN
 	x.i = (unsigned int)v.s[0] * b;
@@ -110,14 +115,40 @@ static inline int MULTI_DIV_int(int a, u
 	y.s[1] = x.s[0];
 	y.i += (int)v.s[0] * b;
 #endif
-	return swap ? (int)bswap_32(y.i) : y.i;
-}
-
-/* (16bit x 16bit) >> 16 */
-#define MULTI_DIV_short(src, scale, swap)				  \
-(swap									  \
- ? bswap_16(((short) bswap_16(src) * (scale)) >> VOL_SCALE_SHIFT)   \
- : (((int) (src) * (scale)) >> VOL_SCALE_SHIFT))
+	return y.i;
+}
+
+static inline int MULTI_DIV_int(int a, unsigned int b, int swap)
+{
+	unsigned int gain = (b >> VOL_SCALE_SHIFT);
+	int fraction;
+	a = swap ? (int)bswap_32(a) : a;
+	fraction = MULTI_DIV_32x16(a, b & VOL_SCALE_MASK, swap);
+	if (gain) {
+		long long amp = (long long)a * gain + fraction;
+		if (amp > (int)0x7fffffff)
+			amp = (int)0x7fffffff;
+		else if (amp < (int)0x80000000)
+			amp = (int)0x80000000;
+		return swap ? (int)bswap_32((int)amp) : (int)amp;
+	}
+	return swap ? (int)bswap_32(fraction) : fraction;
+}
+
+static inline short MULTI_DIV_short(short a, unsigned int b, int swap)
+{
+	unsigned int gain = b >> VOL_SCALE_SHIFT;
+	int fraction;
+	a = swap ? (short)bswap_16(a) : a;
+	fraction = (int)(a * (b & VOL_SCALE_MASK)) >> VOL_SCALE_SHIFT;
+	if (gain) {
+		int amp = a * gain + fraction;
+		if (abs(amp) > 0x7fff)
+			amp = (a<0) ? (short)0x8000 : (short)0x7fff;
+		return swap ? (short)bswap_16((short)amp) : (short)amp;
+	}
+	return swap ? (short)bswap_16((short)fraction) : (short)fraction;
+}
 
 #endif /* DOC_HIDDEN */
 
@@ -237,8 +268,8 @@ static void softvol_convert_stereo_vol(s
 		snd_pcm_areas_silence(dst_areas, dst_offset, channels, frames,
 				      svol->sformat);
 		return;
-	} else if (svol->cur_vol[0] == svol->max_val &&
-		   svol->cur_vol[1] == svol->max_val) {
+	} else if (svol->zero_dB_val && svol->cur_vol[0] == svol->zero_dB_val &&
+		   svol->cur_vol[1] == svol->zero_dB_val) {
 		snd_pcm_areas_copy(dst_areas, dst_offset, src_areas, src_offset,
 				   channels, frames, svol->sformat);
 		return;
@@ -288,7 +319,7 @@ static void softvol_convert_mono_vol(snd
 		snd_pcm_areas_silence(dst_areas, dst_offset, channels, frames,
 				      svol->sformat);
 		return;
-	} else if (svol->cur_vol[0] == svol->max_val) {
+	} else if (svol->zero_dB_val && svol->cur_vol[0] == svol->zero_dB_val) {
 		snd_pcm_areas_copy(dst_areas, dst_offset, src_areas, src_offset,
 				   channels, frames, svol->sformat);
 		return;
@@ -539,6 +570,7 @@ static void snd_pcm_softvol_dump(snd_pcm
 	snd_output_printf(out, "Soft volume PCM\n");
 	snd_output_printf(out, "Control: %s\n", svol->elem.id.name);
 	snd_output_printf(out, "min_dB: %g\n", svol->min_dB);
+	snd_output_printf(out, "max_dB: %g\n", svol->max_dB);
 	snd_output_printf(out, "resolution: %d\n", svol->max_val + 1);
 	if (pcm->setup) {
 		snd_output_printf(out, "Its setup is:\n");
@@ -554,7 +586,7 @@ static int add_tlv_info(snd_pcm_softvol_
 	tlv[0] = SND_CTL_TLVT_DB_SCALE;
 	tlv[1] = 2 * sizeof(int);
 	tlv[2] = svol->min_dB * 100;
-	tlv[3] = -svol->min_dB * 100 / svol->max_val;
+	tlv[3] = (svol->max_dB - svol->min_dB) * 100 / svol->max_val;
 	return snd_ctl_elem_tlv_write(svol->ctl, &cinfo->id, tlv);
 }
 
@@ -567,9 +599,9 @@ static int add_user_ctl(snd_pcm_softvol_
 	if (err < 0)
 		return err;
 	add_tlv_info(svol, cinfo);
-	/* set max value as default */
+	/* set zero dB value as default */
 	for (i = 0; i < count; i++)
-		svol->elem.value.integer.value[i] = svol->max_val;
+		svol->elem.value.integer.value[i] = svol->zero_dB_val;
 	return snd_ctl_elem_write(svol->ctl, &svol->elem);
 }
 
@@ -581,7 +613,8 @@ static int add_user_ctl(snd_pcm_softvol_
  */
 static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol,
 				int ctl_card, snd_ctl_elem_id_t *ctl_id,
-				int cchannels, double min_dB, int resolution)
+				int cchannels, double min_dB, double max_dB,
+				int resolution)
 {
 	char tmp_name[32];
 	snd_pcm_info_t *info;
@@ -610,7 +643,14 @@ static int softvol_load_control(snd_pcm_
 	svol->elem.id = *ctl_id;
 	svol->max_val = resolution - 1;
 	svol->min_dB = min_dB;
-
+	svol->max_dB = max_dB;
+	if (svol->max_dB == ZERO_DB)
+		svol->zero_dB_val = svol->max_val;
+	else if (svol->max_dB < 0)
+		svol->zero_dB_val = 0; /* there is no 0 dB setting */
+	else
+		svol->zero_dB_val = (min_dB / (min_dB - max_dB)) * svol->max_val;
+		
 	snd_ctl_elem_info_alloca(&cinfo);
 	snd_ctl_elem_info_set_id(cinfo, ctl_id);
 	if ((err = snd_ctl_elem_info(svol->ctl, cinfo)) < 0) {
@@ -650,24 +690,25 @@ static int softvol_load_control(snd_pcm_
 		}
 	}
 
-	if (min_dB == PRESET_MIN_DB && resolution == PRESET_RESOLUTION)
+	if (min_dB == PRESET_MIN_DB && max_dB == ZERO_DB && resolution == PRESET_RESOLUTION)
 		svol->dB_value = preset_dB_value;
 	else {
 #ifndef HAVE_SOFT_FLOAT
-		svol->dB_value = calloc(resolution, sizeof(unsigned short));
+		svol->dB_value = calloc(resolution, sizeof(unsigned int));
 		if (! svol->dB_value) {
 			SNDERR("cannot allocate dB table");
 			return -ENOMEM;
 		}
 		svol->min_dB = min_dB;
-		for (i = 1; i < svol->max_val; i++) {
-			double db = svol->min_dB - ((i - 1) * svol->min_dB) / (svol->max_val - 1);
+		svol->max_dB = max_dB;
+		for (i = 0; i <= svol->max_val; i++) {
+			double db = svol->min_dB + (i * (svol->max_dB - svol->min_dB)) / svol->max_val;
 			double v = (pow(10.0, db / 20.0) * (double)(1 << VOL_SCALE_SHIFT));
-			svol->dB_value[i] = (unsigned short)v;
-		}
-		svol->dB_value[svol->max_val] = 65535;
+			svol->dB_value[i] = (unsigned int)v;
+		}
+		svol->dB_value[svol->zero_dB_val] = 65535;
 #else
-		SNDERR("Cannot handle the given min_dB and resolution");
+		SNDERR("Cannot handle the given dB range and resolution");
 		return -EINVAL;
 #endif
 	}
@@ -698,6 +739,7 @@ static snd_pcm_ops_t snd_pcm_softvol_ops
  * \param ctl_id The control element
  * \param cchannels PCM channels
  * \param min_dB minimal dB value
+ * \param max_dB maximal dB value
  * \param resolution resolution of control
  * \param slave Slave PCM handle
  * \param close_slave When set, the slave PCM handle is closed with copy PCM
@@ -710,7 +752,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcm
 			 snd_pcm_format_t sformat,
 			 int ctl_card, snd_ctl_elem_id_t *ctl_id,
 			 int cchannels,
-			 double min_dB, int resolution,
+			 double min_dB, double max_dB, int resolution,
 			 snd_pcm_t *slave, int close_slave)
 {
 	snd_pcm_t *pcm;
@@ -728,7 +770,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcm
 	if (! svol)
 		return -ENOMEM;
 	err = softvol_load_control(slave, svol, ctl_card, ctl_id, cchannels,
-				   min_dB, resolution);
+				   min_dB, max_dB, resolution);
 	if (err < 0) {
 		softvol_free(svol);
 		return err;
@@ -812,6 +854,7 @@ pcm.name {
 		[count INT]     # control channels 1 or 2 (default: 2)
 	}
 	[min_dB REAL]           # minimal dB value (default: -51.0)
+	[max_dB REAL]           # maximal dB value (default:   0.0)
 	[resolution INT]        # resolution (default: 256)
 }
 \endcode
@@ -851,6 +894,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pc
 	snd_ctl_elem_id_t *ctl_id;
 	int resolution = PRESET_RESOLUTION;
 	double min_dB = PRESET_MIN_DB;
+	double max_dB = ZERO_DB;
 	int card = -1, cchannels = 2;
 
 	snd_config_for_each(i, next, conf) {
@@ -886,6 +930,14 @@ int _snd_pcm_softvol_open(snd_pcm_t **pc
 			}
 			continue;
 		}
+		if (strcmp(id, "max_dB") == 0) {
+			err = snd_config_get_real(n, &max_dB);
+			if (err < 0) {
+				SNDERR("Invalid max_dB value");
+				return err;
+			}
+			continue;
+		}
 		SNDERR("Unknown field %s", id);
 		return -EINVAL;
 	}
@@ -899,6 +951,11 @@ int _snd_pcm_softvol_open(snd_pcm_t **pc
 	}
 	if (min_dB >= 0) {
 		SNDERR("min_dB must be a negative value");
+		return -EINVAL;
+	}
+	if (max_dB <= min_dB || max_dB > MAX_DB_UPPER_LIMIT) {
+		SNDERR("max_dB must be larger than min_dB and less than %d dB",
+		       MAX_DB_UPPER_LIMIT);
 		return -EINVAL;
 	}
 	if (resolution < 0 || resolution > 1024) {
@@ -930,7 +987,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pc
 		return err;
 	}
 	err = snd_pcm_softvol_open(pcmp, name, sformat, card, ctl_id, cchannels,
-				   min_dB, resolution, spcm, 1);
+				   min_dB, max_dB, resolution, spcm, 1);
 	if (err < 0)
 		snd_pcm_close(spcm);
 	return err;

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