Any missing patches?

Any missing patches?

[Takashi Iwai]

Hi,

if someone has patches intended for merging to ALSA tree, please
submit them ASAP.

I think the current ALSA CVS tree is forming to a relatively good
shape now, and it's good time to push Linux kernel tree, too.
Let's be not too late to ride a bus.


Takashi


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Re: Any missing patches?

[Juergen Kreileder]

Takashi Iwai <tiwai@suse.de> writes:

> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.
>
> I think the current ALSA CVS tree is forming to a relatively good
> shape now, and it's good time to push Linux kernel tree, too.
> Let's be not too late to ride a bus.

Here's a patch for generic dmix which fixes S16 byte swapping.

Tested on powerpc with snd-usb-audio.  (Without the patch I get crackling.)


BTW: Is it normal that the dmix plug-in consumes 100% CPU?


        Juergen


Signed-off-by: Juergen Kreileder <jk@blackdown.de>

--- o/alsa-lib-1.0.11rc3/src/pcm/pcm_dmix_generic.c	2005-12-19 08:39:04.000000000 +0100
+++ alsa-lib-1.0.11rc3/src/pcm/pcm_dmix_generic.c	2006-03-10 02:04:56.000000000 +0100
@@ -194,7 +194,7 @@ static void mix_areas1_swap(unsigned int
 	register signed int sample;
 
 	for (;;) {
-		sample = bswap_16(*src);
+		sample = (signed short) bswap_16(*src);
 		if (! *dst) {
 			*sum = sample;
 			*dst = *src;
@@ -205,7 +205,7 @@ static void mix_areas1_swap(unsigned int
 				sample = 0x7fff;
 			else if (sample < -0x8000)
 				sample = -0x8000;
-			*dst = bswap_16((signed short)sample);
+			*dst = (signed short) bswap_16((signed short) sample);
 		}
 		if (!--size)
 			return;
=


-- 
Juergen Kreileder, Blackdown Java-Linux Team
http://blog.blackdown.de/


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Re: Any missing patches?

[Juergen Kreileder]

Takashi Iwai <tiwai@suse.de> writes:

> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.
>
> I think the current ALSA CVS tree is forming to a relatively good
> shape now, and it's good time to push Linux kernel tree, too.
> Let's be not too late to ride a bus.

Here's a patch which adds support for S24_3LE and byte-swapped S16 and
S32 to softvol.  I've tested S24_3LE and byte-swapped S16 on powerpc
with snd-usb-audio.  All other cases are untested so far.
(Config at http://blog.blackdown.de/static/alsa/USB-Audio.conf)


        Juergen

Signed-off-by: Juergen Kreileder <jk@blackdown.de>

--- o/alsa-lib-1.0.11rc3/src/pcm/pcm_softvol.c	2005-09-02 18:36:40.000000000 +0200
+++ alsa-lib-1.0.11rc3/src/pcm/pcm_softvol.c	2006-03-10 02:46:39.000000000 +0100
@@ -96,10 +96,10 @@ typedef union {
 	int i;
 	short s[2];
 } val_t;
-static inline int MULTI_DIV_int(int a, unsigned short b)
+static inline int MULTI_DIV_int(int a, unsigned short b, int swap)
 {
 	val_t v, x, y;
-	v.i = a;
+	v.i = swap ? bswap_32(a) : a;
 	y.i = 0;
 #if __BYTE_ORDER == __LITTLE_ENDIAN
 	x.i = (unsigned int)v.s[0] * b;
@@ -110,11 +110,14 @@ static inline int MULTI_DIV_int(int a, u
 	y.s[1] = x.s[0];
 	y.i += (int)v.s[0] * b;
 #endif
-	return y.i;
+	return swap ? bswap_32(y.i) : y.i;
 }
 
 /* (16bit x 16bit) >> 16 */
-#define MULTI_DIV_short(src,scale) (((int)(src) * (scale)) >> VOL_SCALE_SHIFT)
+#define MULTI_DIV_short(src, scale, swap)				  \
+(swap									  \
+ ? bswap_16(((short) bswap_16(src) * (scale)) >> VOL_SCALE_SHIFT)   \
+ : (((int) (src) * (scale)) >> VOL_SCALE_SHIFT))
 
 #endif /* DOC_HIDDEN */
 
@@ -125,7 +128,7 @@ static inline int MULTI_DIV_int(int a, u
  */
 
 #ifndef DOC_HIDDEN
-#define CONVERT_AREA(TYPE) do {	\
+#define CONVERT_AREA(TYPE, swap) do {	\
 	unsigned int ch, fr; \
 	TYPE *src, *dst; \
 	for (ch = 0; ch < channels; ch++) { \
@@ -150,15 +153,55 @@ static inline int MULTI_DIV_int(int a, u
 			} \
 		} else { \
 			while (fr--) { \
-				*dst = MULTI_DIV_##TYPE(*src, vol_scale);\
+				*dst = (TYPE) MULTI_DIV_##TYPE(*src, vol_scale, swap); \
 				src += src_step; \
 				dst += dst_step; \
 			} \
 		} \
 	} \
 } while (0)
-	
 
+#define CONVERT_AREA_S24_3LE() do {					\
+	unsigned int ch, fr;						\
+	unsigned char *src, *dst;					\
+	int tmp;							\
+	for (ch = 0; ch < channels; ch++) {				\
+		src_area = &src_areas[ch];				\
+		dst_area = &dst_areas[ch];				\
+		src = snd_pcm_channel_area_addr(src_area, src_offset);	\
+		dst = snd_pcm_channel_area_addr(dst_area, dst_offset);	\
+		src_step = snd_pcm_channel_area_step(src_area);		\
+		dst_step = snd_pcm_channel_area_step(dst_area);		\
+		GET_VOL_SCALE;						\
+		fr = frames;						\
+		if (! vol_scale) {					\
+			while (fr--) {					\
+				dst[0] = dst[1] = dst[2] = 0;		\
+				dst += dst_step;			\
+			}						\
+		} else if (vol_scale == 0xffff) {			\
+			while (fr--) {					\
+				dst[0] = src[0];			\
+				dst[1] = src[1];			\
+				dst[2] = src[2];			\
+				src += dst_step;			\
+				dst += src_step;			\
+			}						\
+		} else {						\
+			while (fr--) {					\
+				tmp = src[0] |				\
+				      (src[1] << 8) |			\
+				      (((signed char *) src)[2] << 16);	\
+				tmp = MULTI_DIV_int(tmp, vol_scale, 0);	\
+				dst[0] = tmp;				\
+				dst[1] = tmp >> 8;			\
+				dst[2] = tmp >> 16;			\
+				src += dst_step;			\
+				dst += src_step;			\
+			}						\
+		}							\
+	}								\
+} while (0)
 		
 #define GET_VOL_SCALE \
 	switch (ch) { \
@@ -204,12 +247,24 @@ static void softvol_convert_stereo_vol(s
 	vol[0] = svol->dB_value[svol->cur_vol[0]];
 	vol[1] = svol->dB_value[svol->cur_vol[1]];
 	vol_c = svol->dB_value[(svol->cur_vol[0] + svol->cur_vol[1]) / 2];
-	if (svol->sformat == SND_PCM_FORMAT_S16) {
+	switch (svol->sformat) {
+	case SND_PCM_FORMAT_S16_LE:
+	case SND_PCM_FORMAT_S16_BE:
 		/* 16bit samples */
-		CONVERT_AREA(short);
-	} else {
+		CONVERT_AREA(short, 
+			     !snd_pcm_format_cpu_endian(svol->sformat));
+		break;
+	case SND_PCM_FORMAT_S32_LE:
+	case SND_PCM_FORMAT_S32_BE:
 		/* 32bit samples */
-		CONVERT_AREA(int);
+		CONVERT_AREA(int,
+			     !snd_pcm_format_cpu_endian(svol->sformat));
+		break;
+	case SND_PCM_FORMAT_S24_3LE:
+		CONVERT_AREA_S24_3LE();
+		break;
+	default:
+		break;
 	}
 }
 
@@ -240,12 +295,24 @@ static void softvol_convert_mono_vol(snd
 	}
 
 	vol_scale = svol->dB_value[svol->cur_vol[0]];
-	if (svol->sformat == SND_PCM_FORMAT_S16) {
+	switch (svol->sformat) {
+	case SND_PCM_FORMAT_S16_LE:
+	case SND_PCM_FORMAT_S16_BE:
 		/* 16bit samples */
-		CONVERT_AREA(short);
-	} else {
+		CONVERT_AREA(short, 
+			     !snd_pcm_format_cpu_endian(svol->sformat));
+		break;
+	case SND_PCM_FORMAT_S32_LE:
+	case SND_PCM_FORMAT_S32_BE:
 		/* 32bit samples */
-		CONVERT_AREA(int);
+		CONVERT_AREA(int,
+			     !snd_pcm_format_cpu_endian(svol->sformat));
+		break;
+	case SND_PCM_FORMAT_S24_3LE:
+		CONVERT_AREA_S24_3LE();
+		break;
+	default:
+		break;
 	}
 }
 
@@ -287,14 +354,25 @@ static int snd_pcm_softvol_close(snd_pcm
 	return 0;
 }
 
-static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm ATTRIBUTE_UNUSED,
+static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm,
 					      snd_pcm_hw_params_t *params)
 {
 	int err;
+	snd_pcm_softvol_t *svol = pcm->private_data;
 	snd_pcm_access_mask_t access_mask = { SND_PCM_ACCBIT_SHM };
 	snd_pcm_format_mask_t format_mask = {
-		{ (1U << SND_PCM_FORMAT_S16) | (1U << SND_PCM_FORMAT_S32) }
+		{
+			(1ULL << SND_PCM_FORMAT_S16_LE) |
+			(1ULL << SND_PCM_FORMAT_S16_BE) |
+			(1ULL << SND_PCM_FORMAT_S32_LE) |
+ 			(1ULL << SND_PCM_FORMAT_S32_BE),
+			(1ULL << (SND_PCM_FORMAT_S24_3LE - 32))
+		}
 	};
+	if (svol->sformat != SND_PCM_FORMAT_UNKNOWN) {
+		snd_pcm_format_mask_none(&format_mask);
+		snd_pcm_format_mask_set(&format_mask, svol->sformat);
+	}
 	err = _snd_pcm_hw_param_set_mask(params, SND_PCM_HW_PARAM_ACCESS,
 					 &access_mask);
 	if (err < 0)
@@ -396,9 +474,13 @@ static int snd_pcm_softvol_hw_params(snd
 					  snd_pcm_generic_hw_params);
 	if (err < 0)
 		return err;
-	if (slave->format != SND_PCM_FORMAT_S16 &&
-	    slave->format != SND_PCM_FORMAT_S32) {
-		SNDERR("softvol supports only S16 or S32");
+	if (slave->format != SND_PCM_FORMAT_S16_LE &&
+	    slave->format != SND_PCM_FORMAT_S16_BE &&
+	    slave->format != SND_PCM_FORMAT_S24_3LE && 
+	    slave->format != SND_PCM_FORMAT_S32_LE &&
+	    slave->format != SND_PCM_FORMAT_S32_BE) {
+		SNDERR("softvol supports only S16_LE, S16_BE, S24_3LE, S32_LE "
+		       " or S32_BE");
 		return -EINVAL;
 	}
 	svol->sformat = slave->format;
@@ -619,8 +701,11 @@ int snd_pcm_softvol_open(snd_pcm_t **pcm
 	int err;
 	assert(pcmp && slave);
 	if (sformat != SND_PCM_FORMAT_UNKNOWN &&
-	    sformat != SND_PCM_FORMAT_S16 &&
-	    sformat != SND_PCM_FORMAT_S32)
+	    sformat != SND_PCM_FORMAT_S16_LE &&
+	    sformat != SND_PCM_FORMAT_S16_BE &&
+	    sformat != SND_PCM_FORMAT_S24_3LE && 
+	    sformat != SND_PCM_FORMAT_S32_LE &&
+	    sformat != SND_PCM_FORMAT_S32_BE)
 		return -EINVAL;
 	svol = calloc(1, sizeof(*svol));
 	if (! svol)
@@ -808,9 +893,13 @@ int _snd_pcm_softvol_open(snd_pcm_t **pc
 	if (err < 0)
 		return err;
 	if (sformat != SND_PCM_FORMAT_UNKNOWN &&
-	    sformat != SND_PCM_FORMAT_S16 &&
-	    sformat != SND_PCM_FORMAT_S32) {
-		SNDERR("only S16 or S32 format is supported");
+	    sformat != SND_PCM_FORMAT_S16_LE &&
+	    sformat != SND_PCM_FORMAT_S16_BE &&
+	    sformat != SND_PCM_FORMAT_S24_3LE && 
+	    sformat != SND_PCM_FORMAT_S32_LE &&
+	    sformat != SND_PCM_FORMAT_S32_BE) {
+		SNDERR("only S16_LE, S16_BE, S24_3LE, S32_LE or S32_BE format "
+		       "is supported");
 		snd_config_delete(sconf);
 		return -EINVAL;
 	}
=

-- 
Juergen Kreileder, Blackdown Java-Linux Team
http://blog.blackdown.de/


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Re: Any missing patches?

[Takashi Iwai]

[-- Attachment #1: Type: text/plain, Size: 838 bytes --]

At Mon, 20 Mar 2006 21:18:33 +0100,
Juergen Kreileder wrote:
> 
> Takashi Iwai <tiwai@suse.de> writes:
> 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> >
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> 
> Here's a patch for generic dmix which fixes S16 byte swapping.
> 
> Tested on powerpc with snd-usb-audio.  (Without the patch I get crackling.)

Thanks.  Applied now.


> BTW: Is it normal that the dmix plug-in consumes 100% CPU?

No, it's a new problem.  Looks like happening only on ppc.
Did it happen with the older kernel, too?

Anyway, the below are two different patches as workarounds.
Could you test if either of them has any influence?


Takashi

[-- Attachment #2: alsa-lib-pcm-timer-tread-for-ppc.diff --]
[-- Type: application/octet-stream, Size: 1345 bytes --]

Index: alsa-lib/src/pcm/pcm_direct.c
===================================================================
RCS file: /home/iwai/cvs/alsa/alsa-lib/src/pcm/pcm_direct.c,v
retrieving revision 1.55
diff -u -r1.55 pcm_direct.c
--- alsa-lib/src/pcm/pcm_direct.c	20 Mar 2006 18:21:15 -0000	1.55
+++ alsa-lib/src/pcm/pcm_direct.c	20 Mar 2006 20:08:31 -0000
@@ -1046,7 +1046,11 @@
 	char name[128];
 	int capture = dmix->type == SND_PCM_TYPE_DSNOOP ? 1 : 0;
 
+#ifdef __ppc__
+	dmix->tread = 0;
+#else
 	dmix->tread = 1;
+#endif
 	dmix->timer_need_poll = 0;
 	snd_pcm_info_alloca(&info);
 	ret = snd_pcm_info(dmix->spcm, info);
@@ -1059,14 +1063,18 @@
 				snd_pcm_info_get_card(info),
 				snd_pcm_info_get_device(info),
 				snd_pcm_info_get_subdevice(info) * 2 + capture);
-	ret = snd_timer_open(&dmix->timer, name, SND_TIMER_OPEN_NONBLOCK | SND_TIMER_OPEN_TREAD);
-	if (ret < 0) {
-		dmix->tread = 1;
-		ret = snd_timer_open(&dmix->timer, name, SND_TIMER_OPEN_NONBLOCK);
-		if (ret < 0) {
+	for (;;) {
+		int mode = SND_TIMER_OPEN_NONBLOCK;
+		if (dmix->tread)
+			mode |= SND_TIMER_OPEN_TREAD;
+		ret = snd_timer_open(&dmix->timer, name, mode);
+		if (ret >= 0)
+			break;
+		if (! dmix->tread) {
 			SNDERR("unable to open timer '%s'", name);
 			return ret;
 		}
+		dmix->tread = 0;
 	}
 
 	if (snd_timer_poll_descriptors_count(dmix->timer) != 1) {

[-- Attachment #3: alsa-lib-pcm-timer-size-hack.diff --]
[-- Type: application/octet-stream, Size: 831 bytes --]

Index: alsa-lib/src/pcm/pcm_direct.c
===================================================================
RCS file: /home/iwai/cvs/alsa/alsa-lib/src/pcm/pcm_direct.c,v
retrieving revision 1.55
diff -u -r1.55 pcm_direct.c
--- alsa-lib/src/pcm/pcm_direct.c	20 Mar 2006 18:21:15 -0000	1.55
+++ alsa-lib/src/pcm/pcm_direct.c	20 Mar 2006 19:33:49 -0000
@@ -522,9 +522,10 @@
 		}
 	} else {
 		if (dmix->tread) {
-			snd_timer_tread_t rbuf;
-			while (snd_timer_read(dmix->timer, &rbuf, sizeof(rbuf)) > 0)
-				process_timer_event(dmix, &rbuf);
+			char rbuf[sizeof(snd_timer_tread_t) * 2 - 1];
+			// snd_timer_tread_t rbuf;
+			while (snd_timer_read(dmix->timer, rbuf, sizeof(rbuf)) > 0)
+				; // process_timer_event(dmix, &rbuf);
 		} else {
 			snd_timer_read_t rbuf;
 			while (snd_timer_read(dmix->timer, &rbuf, sizeof(rbuf)) > 0)

Re: Any missing patches?

[Takashi Iwai]

At Mon, 20 Mar 2006 21:26:47 +0100,
Juergen Kreileder wrote:
> 
> Takashi Iwai <tiwai@suse.de> writes:
> 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> >
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> 
> Here's a patch which adds support for S24_3LE and byte-swapped S16 and
> S32 to softvol.  I've tested S24_3LE and byte-swapped S16 on powerpc
> with snd-usb-audio.  All other cases are untested so far.
> (Config at http://blog.blackdown.de/static/alsa/USB-Audio.conf)

Ok, it's a bit bigger to apply too quickly.  I'll check and apply it
tomorrow.

Thanks,

Takashi


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Re: Any missing patches?

[Juergen Kreileder]

Takashi Iwai <tiwai@suse.de> writes:

> At Mon, 20 Mar 2006 21:18:33 +0100,
> Juergen Kreileder wrote:
>> BTW: Is it normal that the dmix plug-in consumes 100% CPU?
>
> No, it's a new problem.  Looks like happening only on ppc.
> Did it happen with the older kernel, too?

I'm not sure if it started with a kernel update or with the
introduction of dmix in PMac.conf.

> Anyway, the below are two different patches as workarounds.
> Could you test if either of them has any influence?

alsa-lib-pcm-timer-tread-for-ppc.diff works.  The other patch had no
effect.


        Juergen

-- 
Juergen Kreileder, Blackdown Java-Linux Team
http://blog.blackdown.de/


-------------------------------------------------------
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RE : Any missing patches?

[Thibault LE MEUR]

[-- Attachment #1: Type: text/plain, Size: 327 bytes --]

Hi,

> if someone has patches intended for merging to ALSA tree, 
> please submit them ASAP.

Here is a small and really not urgent patch (but since it is "time to push
Linux kernel tree" :).

Summary: Fixes typos in alsa-kernel/Documentation/Audiophile-USB.txt

Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>

 

[-- Attachment #2: Audiophile-Usb.txt.diff.txt --]
[-- Type: text/plain, Size: 3138 bytes --]

--- alsa-kernel/Documentation/Audiophile-Usb.txt	2006-03-21 09:49:13.000000000 +0100
+++ alsa-kernel/Documentation/Audiophile-Usb.txt.new	2006-03-21 09:47:55.000000000 +0100
@@ -1,4 +1,4 @@
-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.1
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.2
 	========================================================
 
 	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -13,6 +13,9 @@
 
 The device has 4 audio interfaces, and 2 MIDI ports:
  * Analog Stereo Input (Ai)
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
  * Analog Stereo Output (Ao)
  * Digital Stereo Input (Di)
  * Digital Stereo Output (Do)
@@ -42,7 +45,7 @@
 though I haven't tested it under linux
    - Note that in this setup only the Do interface can be enabled
  * Apart from recording an audio digital stream, enabling the Di port is a way 
-to syncrhonize the device to an external sample clock
+to synchronize the device to an external sample clock
    - As a consequence, the Di port must be enable only if an active Digital 
 source is connected
    - Enabling Di when no digital source is connected can result in a 
@@ -180,7 +183,7 @@
      or changing modprobe.conf)
    - turn on the device
 
-2.2.2.3 - Setting and switching configurations with the device_setup parameter
+2.2.2.3 - Audiophile USB's device_setup structure
 
 If you want to understand the device_setup magic numbers for the Audiophile 
 USB, you need some very basic understanding of binary computation. However, 
@@ -226,7 +229,7 @@
    - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
      only be able to use one at the same time
 
-2.2.3 -  Technical Details for Audiophile Usb 
+2.2.3 -  USB implementation details for this device
 
 You may safely skip this section if you're not interrested in driver 
 development.
@@ -234,20 +237,20 @@
 This section describes some internals aspect of the device and summarize the 
 data I got by usb-snooping the windows and linux drivers.
 
-The M-Audio Audiophile USB has 7 Usb Interfaces:
+The M-Audio Audiophile USB has 7 USB Interfaces:
 a "USB interface":
- * Usb Interface nb.0
- * Usb Interface nb.1
+ * USB Interface nb.0
+ * USB Interface nb.1
    - Audio Control function
- * Usb Interface nb.2
+ * USB Interface nb.2
    - Analog Output
- * Usb Interface nb.3
+ * USB Interface nb.3
    - Digital Output
- * Usb Interface nb.4
+ * USB Interface nb.4
    - Analog Input
- * Usb Interface nb.5
+ * USB Interface nb.5
    - Digital Input
- * Usb Interface nb.6
+ * USB Interface nb.6
    - MIDI interface compliant with the MIDIMAN quirk 
 
 Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
@@ -326,5 +329,5 @@
 
 After having applied the patch you can run jackd with the following command 
 line:
-# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
 

Re: Any missing patches?

[Rimas Kudelis]

Hi Takashi,

> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.
>
> I think the current ALSA CVS tree is forming to a relatively good
> shape now, and it's good time to push Linux kernel tree, too.
> Let's be not too late to ride a bus.
>   

if you don't mind, I'd like to polish ALC260 acer model a bit. I'll 
hopefully send a small untested patch for it this evening.

Rimas



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Re: Any missing patches?

[Takashi Iwai]

At Mon, 20 Mar 2006 22:15:03 +0100,
Juergen Kreileder wrote:
> 
> Takashi Iwai <tiwai@suse.de> writes:
> 
> > At Mon, 20 Mar 2006 21:18:33 +0100,
> > Juergen Kreileder wrote:
> >> BTW: Is it normal that the dmix plug-in consumes 100% CPU?
> >
> > No, it's a new problem.  Looks like happening only on ppc.
> > Did it happen with the older kernel, too?
> 
> I'm not sure if it started with a kernel update or with the
> introduction of dmix in PMac.conf.
> 
> > Anyway, the below are two different patches as workarounds.
> > Could you test if either of them has any influence?
> 
> alsa-lib-pcm-timer-tread-for-ppc.diff works.  The other patch had no
> effect.

OK, I fixed now this problem on CVS.  It's because of the difference
of a struct read from 32bit app on 64bit kernel.


Takashi


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Re: Any missing patches?

[Takashi Iwai]

At Tue, 21 Mar 2006 12:18:17 +0200,
Rimas Kudelis wrote:
> 
> Hi Takashi,
> 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> >
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> >   
> 
> if you don't mind, I'd like to polish ALC260 acer model a bit. I'll 
> hopefully send a small untested patch for it this evening.

Yes, I'll sync up all patches in today, so hopefully it will get into
the tree in the right time.

We'll have also a bit time after rc4 until the final release (but not
long this time).  If the patch isn't too destructive, it can be merged
after rc4, too.


Takashi


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Re: RE : Any missing patches?

[Takashi Iwai]

At Tue, 21 Mar 2006 10:15:33 +0100,
Thibault LE MEUR wrote:
> 
> Hi,
> 
> > if someone has patches intended for merging to ALSA tree, 
> > please submit them ASAP.
> 
> Here is a small and really not urgent patch (but since it is "time to push
> Linux kernel tree" :).
> 
> Summary: Fixes typos in alsa-kernel/Documentation/Audiophile-USB.txt
> 
> Signed-off-by: Thibault LE MEUR <Thibault.LeMeur@supelec.fr>

Applied to CVS.  Thanks.


Takashi



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Re: Any missing patches?

[Takashi Iwai]

At Mon, 20 Mar 2006 21:26:47 +0100,
Juergen Kreileder wrote:
> 
> Takashi Iwai <tiwai@suse.de> writes:
> 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> >
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> 
> Here's a patch which adds support for S24_3LE and byte-swapped S16 and
> S32 to softvol.  I've tested S24_3LE and byte-swapped S16 on powerpc
> with snd-usb-audio.  All other cases are untested so far.
> (Config at http://blog.blackdown.de/static/alsa/USB-Audio.conf)
> 
> 
>         Juergen
> 
> Signed-off-by: Juergen Kreileder <jk@blackdown.de>

Applied to CVS now.  I couldn't test all cases, too, but it's no
regression at least :)


Thanks.

Takashi


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Re: Any missing patches?

[Rimas Kudelis]

Hello,

> Yes, I'll sync up all patches in today, so hopefully it will get into
> the tree in the right time.
>
> We'll have also a bit time after rc4 until the final release (but not
> long this time).  If the patch isn't too destructive, it can be merged
> after rc4, too.
The patch just removes some controls from the mixer.

BTW i heard they released Linux 2.6.16 today. Does it have what is in 
CVS (acer and other fixes)?

Rimas



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Re: Any missing patches?

[Takashi Iwai]

At Tue, 21 Mar 2006 13:30:36 +0200,
Rimas Kudelis wrote:
> 
> Hello,
> 
> > Yes, I'll sync up all patches in today, so hopefully it will get into
> > the tree in the right time.
> >
> > We'll have also a bit time after rc4 until the final release (but not
> > long this time).  If the patch isn't too destructive, it can be merged
> > after rc4, too.
> The patch just removes some controls from the mixer.
> 
> BTW i heard they released Linux 2.6.16 today. Does it have what is in 
> CVS (acer and other fixes)?

No.  It'll be in 2.6.17.


Takashi


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Re: Any missing patches?

[Rimas Kudelis]

[-- Attachment #1.1: Type: text/plain, Size: 305 bytes --]

Hi,

this patch cleans up the mixer interface by removing
unpractical/non-working features. Applies for ALC260 "acer" model in
hda-intel module. It also adds/changes a few comments in the source.

The patch is tested and works with an Acer Travelmate 4061NLMi.

Rimas Kudelis <rq@akl.lt>





[-- Attachment #1.2: patch_realtek.c.diff --]
[-- Type: text/plain, Size: 4971 bytes --]

--- alsa-kernel/pci/hda/patch_realtek.c.orig	2006-03-12 18:19:50.000000000 +0200
+++ alsa-kernel/pci/hda/patch_realtek.c	2006-03-21 21:11:29.000000000 +0200
@@ -2738,16 +2738,14 @@
 	},
 };
 
-/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configutation to
- * the Fujitsu S702x, but jacks are marked differently. We won't allow
- * retasking the Headphone jack, so it won't be available here.
+/* Capture sources suitable for Acer TravelMate 4060, Aspire 1640 and similar
+ * laptops.
  */
 static struct hda_input_mux alc260_acer_capture_source = {
-	.num_items = 3,
+	.num_items = 2,
 	.items = {
 		{ "Mic", 0x0 },
 		{ "Line", 0x2 },
-		{ "CD", 0x4 },
 	},
 };
 
@@ -2830,16 +2828,18 @@
 static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
-	ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+	/* Mic jack works as output too, actually, but it only outputs both stereo
+	 * channels mixed to left channel then. Furthermore, the same pin is
+	 * connected to the internal microphone. So, let's forget it.
+	 */
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	/* Line jack doesn't actually preamplify the mic even if set to one of the
+	 * Mic modes. However, it doesn't seem like these can be disabled ATM.
+	 */
 	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -3128,18 +3128,18 @@
  * similar laptops (adapted from Fujitsu init verbs).
  */
 static struct hda_verb alc260_acer_init_verbs[] = {
-	/* On TravelMate laptops, GPIO 0 enables the internal speaker and
-	 * the headphone jack.  Turn this on and rely on the standard mute
-	 * methods whenever the user wants to turn these outputs off.
+	/* On TravelMate 4060, Aspire 1640 and similar laptops, GPIO 0 enables the
+	 * internal speaker and the headphone jack. Turn this on and rely on the
+	 * standard mute methods whenever the user wants to turn these outputs off.
 	 */
 	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
 	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
 	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
-	/* Internal speaker/Headphone jack is connected to Line-out pin */
+	/* Internal speaker and Headphone jack are connected to line-out pin */
 	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Internal microphone/Mic jack is connected to Mic1 pin */
+	/* Internal microphone and Mic jack are connected to mic1 pin */
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
-	/* Line In jack is connected to Line1 pin */
+	/* Line In jack is connected to line1 pin */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
 	/* Ensure all other unused pins are disabled and muted. */
 	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
@@ -3154,7 +3154,7 @@
 	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
 	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
-	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum 
+	/* Ensure mic1 and line1 pin widgets take input from the OUT1 sum 
 	 * bus when acting as outputs.
 	 */
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
@@ -3173,10 +3173,10 @@
 
 	/* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Unmute Mic1 and Line1 pin widget input buffers since they start as
+	/* Unmute mic1 and line1 pin widget input buffers since they start as
 	 * inputs. If the pin mode is changed by the user the pin mode control
 	 * will take care of enabling the pin's input/output buffers as needed.
-	 * Therefore there's no need to enable the input buffer at this
+	 * Therefore there's no need to enable the output buffer at this
 	 * stage.
 	 */
 	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -3184,13 +3184,14 @@
 
 	/* Mute capture amp left and right */
 	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	/* Set ADC connection select to match default mixer setting - mic
+	/* Set ADC connection select to match default mixer setting - Mic
 	 * (on mic1 pin)
 	 */
 	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	/* Do similar with the second ADC: mute capture input amp and
-	 * set ADC connection to line (on line1 pin)
+	 * set ADC connection to Line (on line1 pin)
+	 * FIXME: the connection is still set to Mic for some reason
 	 */
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},

[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 254 bytes --]

Re: Any missing patches?

[Lee Revell]

On Tue, 2006-03-21 at 23:02 +0200, Rimas Kudelis wrote:
> Hi,
> 
> this patch cleans up the mixer interface by removing
> unpractical/non-working features. Applies for ALC260 "acer" model in
> hda-intel module. It also adds/changes a few comments in the source.
> 
> The patch is tested and works with an Acer Travelmate 4061NLMi.
> 
> Rimas Kudelis <rq@akl.lt>

Did you verify (eg with Windows) that CD is not connected to anything or
is this just speculation?

Lee



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Re: Any missing patches?

[Lee Revell]

On Mon, 2006-03-20 at 19:45 +0100, Takashi Iwai wrote:
> Hi,
> 
> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.
> 
> I think the current ALSA CVS tree is forming to a relatively good
> shape now, and it's good time to push Linux kernel tree, too.
> Let's be not too late to ride a bus.
> 

Can you look at bug #1952 (speakers not disabled when headphones plugged
with hda-intel + STAC92xx)?  amixer output shows no "Jack sense"
controls present.  It looks like it just needs an HP_MUTE quirk (or
whatever the hda-intel equivalent is)

Lee




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Re: Any missing patches?

[Jonathan Woithe]

Hi guys

> this patch cleans up the mixer interface by removing
> unpractical/non-working features. Applies for ALC260 "acer" model in
> hda-intel module. It also adds/changes a few comments in the source.

I have a few comments regarding this patch.

> +	/* Line jack doesn't actually preamplify the mic even if set to one of the
> +	 * Mic modes. However, it doesn't seem like these can be disabled ATM.
> +	 */
>  	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),

This comment is actually slightly misleading - see my posts earlier today.
It's not that this jack "doesn't actually preamplify the mic".  At this
stage, on the strength of a sample of one, it appears that at least some of
the Acer laptops have a DC blocking capacitor between the ALC260 chip and
the physical output jack.  This blocks the DC bias supplied by the ALC260
but it doesn't stop "preamplification" - it simply stops unpowered condensor
microphones from working.  If a self-powered mic was plugged in you'll find
that it will work with either setting.

It is true that ALC_PIN_MODE() can't disable the mic modes and leave the
"line in" mode active at present.  I might do a patch at some stage to add
this flexibility.  However, I would probably argue at this stage that it
might not be desireable to omit the mic modes from this pin in the acer
model, since the issue might simply go away with later/other acer models. 
We would then need to define a new Acer model just to cope with this tiny
variation.

> -	/* Internal speaker/Headphone jack is connected to Line-out pin */
> +	/* Internal speaker and Headphone jack are connected to line-out pin */=
>  	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},

Just a comment: this confirms that NID 0x0f is used for the headphone jack.
This explains why the mic controls automagically revert to "line in" mode
for this jack.  See my earlier post for the explanation.

> -	/* Set ADC connection select to match default mixer setting - mic
> +	/* Set ADC connection select to match default mixer setting - Mic
>  	 * (on mic1 pin)
>  	 */
>  	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
>
>  	/* Do similar with the second ADC: mute capture input amp and
> -	 * set ADC connection to line (on line1 pin)
> +	 * set ADC connection to Line (on line1 pin)
> +	 * FIXME: the connection is still set to Mic for some reason
>  	 */

The point isn't that the init verbs don't work - it's that elsewhere during
device initialisation the capture sources are set to the first one in the
source list (or so it seems).  This is why the "to match default mixer
setting" comment is there.  I haven't had time to track this down and I
really didn't think it was important enough at this stage to worry about. I
think it is perfectly acceptable to have all inputs default to "mic" and let
the user work out what they want.  Most distros configure things so the alsa
mixer is restored to its last state during a boot, so I don't think it's that
big a deal.

Regards
  jonathan


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Re: Any missing patches?

[Lee Revell]

On Tue, 2006-03-21 at 20:37 -0500, Lee Revell wrote:
> On Mon, 2006-03-20 at 19:45 +0100, Takashi Iwai wrote:
> > Hi,
> > 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> > 
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> > 
> 
> Can you look at bug #1952 (speakers not disabled when headphones plugged
> with hda-intel + STAC92xx)?  amixer output shows no "Jack sense"
> controls present.  It looks like it just needs an HP_MUTE quirk (or
> whatever the hda-intel equivalent is)

Never mind - user reports fixed in CVS.

Lee



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Re: Any missing patches?

[Takashi Iwai]

At Wed, 22 Mar 2006 13:07:33 +1030 (CST),
Jonathan Woithe wrote:
> 
> > -	/* Set ADC connection select to match default mixer setting - mic
> > +	/* Set ADC connection select to match default mixer setting - Mic
> >  	 * (on mic1 pin)
> >  	 */
> >  	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
> >
> >  	/* Do similar with the second ADC: mute capture input amp and
> > -	 * set ADC connection to line (on line1 pin)
> > +	 * set ADC connection to Line (on line1 pin)
> > +	 * FIXME: the connection is still set to Mic for some reason
> >  	 */
> 
> The point isn't that the init verbs don't work - it's that elsewhere during
> device initialisation the capture sources are set to the first one in the
> source list (or so it seems).

The mixer reads/modifies the values spec->cur_mux[idx] as caches.
The initial values are 0 = Mic.


Takashi


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Re: Any missing patches?

[Takashi Iwai]

At Tue, 21 Mar 2006 17:06:43 -0500,
Lee Revell wrote:
> 
> On Tue, 2006-03-21 at 23:02 +0200, Rimas Kudelis wrote:
> > Hi,
> > 
> > this patch cleans up the mixer interface by removing
> > unpractical/non-working features. Applies for ALC260 "acer" model in
> > hda-intel module. It also adds/changes a few comments in the source.
> > 
> > The patch is tested and works with an Acer Travelmate 4061NLMi.
> > 
> > Rimas Kudelis <rq@akl.lt>
> 
> Did you verify (eg with Windows) that CD is not connected to anything or
> is this just speculation?

If CD doesn't work despite Mic/Line works with the latest code, it
implies that the CD analog doesn't exist.  Of course, you should check
the codec proc file and the corresponding PIN setup to make sure.

I wouldn't be surprised at all if CD analog doesn't exist.  Most of
recent laptops have really no CD analog input.  Disabling unfunctional
things is good for usability.


Takashi


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Re: Any missing patches?

[Lee Revell]

On Mon, 2006-03-20 at 19:45 +0100, Takashi Iwai wrote:
> Hi,
> 
> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.
> 
> I think the current ALSA CVS tree is forming to a relatively good
> shape now, and it's good time to push Linux kernel tree, too.
> Let's be not too late to ride a bus.

It would also be nice if everyone with developer access to the bug
tracker could take a few minutes to look at some old bugs and clear the
invalid or fixed ones, ask for feedback, mark some relationships, etc.
I spent an hour or two on this yesterday and was able to close ~50 bugs
(out of about 700 open).  At this rate we could clean up the bug tracker
nicely with just a few hours developer time.

Lee 



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Re: Any missing patches?

[Takashi Iwai]

At Wed, 22 Mar 2006 13:32:41 -0500,
Lee Revell wrote:
> 
> On Mon, 2006-03-20 at 19:45 +0100, Takashi Iwai wrote:
> > Hi,
> > 
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> > 
> > I think the current ALSA CVS tree is forming to a relatively good
> > shape now, and it's good time to push Linux kernel tree, too.
> > Let's be not too late to ride a bus.
> 
> It would also be nice if everyone with developer access to the bug
> tracker could take a few minutes to look at some old bugs and clear the
> invalid or fixed ones, ask for feedback, mark some relationships, etc.
> I spent an hour or two on this yesterday and was able to close ~50 bugs
> (out of about 700 open).  At this rate we could clean up the bug tracker
> nicely with just a few hours developer time.

Thanks for doing that!
Yeah, it's a thing like a homework for a summer vacation ;)


Takashi


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Re: Any missing patches?

[Jonathan Woithe]

> > > -	/* Set ADC connection select to match default mixer setting - mic
> > > +	/* Set ADC connection select to match default mixer setting - Mic
> > >  	 * (on mic1 pin)
> > >  	 */
> > >  	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
> > >
> > >  	/* Do similar with the second ADC: mute capture input amp and
> > > -	 * set ADC connection to line (on line1 pin)
> > > +	 * set ADC connection to Line (on line1 pin)
> > > +	 * FIXME: the connection is still set to Mic for some reason
> > >  	 */
> > 
> > The point isn't that the init verbs don't work - it's that elsewhere during
> > device initialisation the capture sources are set to the first one in the
> > source list (or so it seems).
> 
> The mixer reads/modifies the values spec->cur_mux[idx] as caches.
> The initial values are 0 = Mic.

Yep, I figured it was something like that - no surprises there.  IMHO the
current state should probably remain as is - that is, default to mic for all
ADCs since not every laptop with an ALC260 has a "line" jack they can use
for input.  That would then require the above patch to be tweaked so the
second ADC is initialised to use the mic channel (if only for internal
consistency).

Regards
  jonathan


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[PATCH] asihpi unbalanced spinlocks

[Eliot Blennerhassett]

[-- Attachment #1: Type: text/plain, Size: 1061 bytes --]

On Tue, 21 Mar 2006 06:45, Takashi Iwai wrote:
> Hi,
>
> if someone has patches intended for merging to ALSA tree, please
> submit them ASAP.

Fix unbalanced spin_lock/unlocks
(Note we have huge changes in the pipeline,  but lets get this hopefully 
uncontroversial fix in first)

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>

--- asihpi_orig.c	2006-04-05 15:44:17.000000000 +1200
+++ asihpi.c	2006-04-05 15:45:49.000000000 +1200
@@ -1825,6 +1825,7 @@
 				    &wSourceType, &wSourceIndex);
 	HPI_Multiplexer_SetSource(phSubSys, hControl, wSourceType,
 				  wSourceIndex);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 
@@ -1902,6 +1903,7 @@
 
 	HPI_ChannelModeSet(phSubSys, hControl,
 			   ucontrol->value.enumerated.item[0] + 1);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 
@@ -1979,6 +1981,7 @@
 
 	HPI_SampleClock_SetSource(phSubSys, hControl,
 				  ucontrol->value.enumerated.item[0] + 1);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 

[-- Attachment #2: asihpi.c.patch --]
[-- Type: text/x-diff, Size: 708 bytes --]

--- asihpi_orig.c	2006-04-05 15:44:17.000000000 +1200
+++ asihpi.c	2006-04-05 15:45:49.000000000 +1200
@@ -1825,6 +1825,7 @@
 				    &wSourceType, &wSourceIndex);
 	HPI_Multiplexer_SetSource(phSubSys, hControl, wSourceType,
 				  wSourceIndex);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 
@@ -1902,6 +1903,7 @@
 
 	HPI_ChannelModeSet(phSubSys, hControl,
 			   ucontrol->value.enumerated.item[0] + 1);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 
@@ -1979,6 +1981,7 @@
 
 	HPI_SampleClock_SetSource(phSubSys, hControl,
 				  ucontrol->value.enumerated.item[0] + 1);
+	spin_unlock_irqrestore(&asihpi->mixer_lock, flags);
 	return change;
 }
 

Re: [PATCH] asihpi unbalanced spinlocks

[Takashi Iwai]

At Wed, 5 Apr 2006 15:58:12 +1200,
Eliot Blennerhassett wrote:
> 
> On Tue, 21 Mar 2006 06:45, Takashi Iwai wrote:
> > Hi,
> >
> > if someone has patches intended for merging to ALSA tree, please
> > submit them ASAP.
> 
> Fix unbalanced spin_lock/unlocks
> (Note we have huge changes in the pipeline,  but lets get this hopefully 
> uncontroversial fix in first)
> 
> Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>

Thanks, now committed to ALSA tree.


Takashi


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