October 2008 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] alsa: Emagic Audiowerk8 low-level ALSA driver
by matthias.nyffenegger＠bluewin.ch 13 Oct '08

13 Oct '08

Hi Takashi, Thank you for your quick answer, I am not that fast ;-) >At the next time, it'd be appreciate if you post the patch itself, >then we can review and give comments on the code in a reply. >Even if it's too big, you can split to several pieces. Ok. >Just checking it now, and soon found out that many components look >similar as AW2. We have already AW2 driver (whose authors cc'ed). >Can you reuse the code? Hm, I'm afraid not at his point. The existing AW2 driver is IMHO a very straight-forward implementation - with focus on AW2 - and does not provide generic code parts (apart saa7146 register addr. definitions) for reuse. Maybe that's why it has its own directory not named after the chipset? Besides that I have put much emphasis on providing an abstraction-layer for saa7146 chipset audio functionality, on top of which the audiowerk specific stuff is built. Hence it does not make much sense to me trying to reuse code of the existing AW2 driver. I hope you see my point and would appreciate your opinion on this. regards, Matthias ----Ursprüngliche Nachricht---- Von: tiwai(a)suse.de Datum: 19.09.2008 17:55 An: <matthias.nyffenegger(a)bluewin.ch> Kopie: "Cedric Bregardis"<cedric.bregardis(a)free.fr>, "Jean-Christian Hassler"<jhassler(a)free.fr>, <alsa-devel@alsa- project.org> Betreff: Re: [PATCH] alsa: Emagic Audiowerk8 low-level ALSA driver At Fri, 19 Sep 2008 09:10:00 +0000 (GMT), matthias.nyffenegger(a)bluewin.ch wrote: > > From: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> > > Low-level ALSA driver for Emagic Audiowerk8 sound card. > Project page: http://sourceforge.net/projects/aw8-alsa First off, thanks for your patch. At the next time, it'd be appreciate if you post the patch itself, then we can review and give comments on the code in a reply. Even if it's too big, you can split to several pieces. > Built and tested with Vanilla 2.6.25.16, ALSA 1.0.14a. > > Signed-off-by: Matthias Nyffenegger <matthias.nyffenegger(a)bluewin.ch> > --- > This is a request for submission to ALSA-tree. > URL: http://downloads.sourceforge.net/aw8-alsa/aw-alsa-patch-0.2 Just checking it now, and soon found out that many components look similar as AW2. We have already AW2 driver (whose authors cc'ed). Can you reuse the code? thanks, Takashi

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Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 13 Oct '08

13 Oct '08

At Fri, 10 Oct 2008 20:39:48 +0400, The Source wrote: > > 2008/10/10 Takashi Iwai <tiwai(a)suse.de> > > At Fri, 10 Oct 2008 01:26:15 -0500, > Ted T. Logian wrote: > > > > > > > > On Fri, 2008-10-10 at 08:01 +0200, Takashi Iwai wrote: > > > At Thu, 09 Oct 2008 15:17:58 -0500, > > > Ted T. Logian wrote: > > > > > > > > Sorry for the multiple posts, but once I rmmod snd-sbxfi, running > mixer does > > > > fine. It seems if I have snd-sbxfi loaded and I run alsamixer, it > locks the > > > > entire system. > > > > > > So, do you mean loading the driver itself doesn't lock up? If so, > > > it's better than I expected. > > > > > > Did you run ever PCM playback/capture before that? This is more > > > dangerous :) > > > > > > Also, please give your hardware details, as there are different models > > > for X-Fi, and some are handled pretty differently. > > > > > > [BTW, please stop top-posting, and avoid HTML mails for ML. It's easy > > > to switch to normal (plain) mail mode on Gmail, just a click.] > > > > > > > > > thanks, > > > > > > Takashi > > > > > > I think perhaps from uninstalling pulseaudio I got further. I can use > > mixer now, and it even has a "Master" device, but nothing else... > > > > however, I do get a lock up later. > > > > I get this from running aplay, too, if this helps... > > Thanks! That's rather a stupid bug. > The patch is below. > > Takashi > > diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c > index 8066bf4..8a0eea0 100644 > --- a/sound/pci/sbxfi/sbxfi.c > +++ b/sound/pci/sbxfi/sbxfi.c > @@ -831,7 +831,7 @@ static struct sbxfi_port *sbxfi_port_alloc(struct > sbxfi *chip, > port->src[0] = src; > port->src[1] = src + 1; > spin_lock_irq(&chip->port_lock); > - list_add(&chip->port_list, &port->list); > + list_add(&port->list, &chip->port_list); > spin_unlock_irq(&chip->port_lock); > return port; > } > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Hello, I have tried the driver with the fix you provided, but I couldn't get > anything. Kernel panic on any attempt to use sound or mixer. Also kernel panic > several seconds after Xorg starts. Each crash causes either hang or reboot so > I couldn't get any error messages. If there are some logs left, please tell me > where they are and I'll send them to you. > My card is X-Fi Platinum Fatality Champion Series. For testing, first try without X but just use Linux console, so that you have a better chance to see Oops messages. It's pretty important to see whether it's an Oops or a panic. Also, double-check whether you really installed the corrected drivers. Often it's missing in the right place by stupid reasons. For first tests, don't use any sound server. Try with aplay like: % aplay -Dhw foo.wav Try first 96kHz samples, 48kHz samples, and then others. thanks, Takashi

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[alsa-devel] Creative Sound Blaster 5.1vx
by Zhi Xiong 13 Oct '08

13 Oct '08

Hi, I own a creative sound blaster 5.1vx card, its model is sb1070, and its based on CA0106 and an ADC I believed to be a WM8772S(74AAWR2) chip, for the rest, my weak eyes can't really see it. I tested the driver on 2.6.20 and 2.6.26, both of them didn't work, so I believed this model is not yet "supported" by the driver, I added the lines below into ca0106_chip_details inside ca0106_main.c and the card works. { .serial = 0x10041102, .name = "Sound Blaster 5.1vx [SB1070]", .gpio_type = 1, .i2c_adc = 1, .spi_dac = 1 } , But I don't really know what gpio_type, .i2c_adc, .spi_dac . I do noticed that setting gpio_type to 1 or 2 will work and not setting it will have some performance issue. I wish can write the code more accurately to the specs and hopefully this few lines of code will get added to the next alsa release for others to use. So, can anyone please explain to me what the 3 parameters do? Thanks in advanced. -- Cheers, Zhi Xiong

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[alsa-devel] Issue with onboard sound on Toshiba A110-225
by Travis Place 13 Oct '08

13 Oct '08

Hey all, When working out an issue for a person in #alsa, i found that the onboard sound in their laptop (A110-225) has the Subsystem IDs 1179:ff01 . The card has an ALC861 codec.. the issue (non-fatal thankfully) is that the above mentioned Subsystem IDs match in patch_realtek.c to 'dallas' which is for an ALC861VD codec.. SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), The user was getting the non-fatal: hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... in dmesg.. I had the user force the 'toshiba' model (which is valid for ALC861) and the warning went away, and sound continues to work as it should.. Just wanted to make sure this problem (even though non-fatal) is documented, and we are aware of it. Perhaps this laptop, and the 'Dallas' need to be matched on some other level. Thanks, Travis Place (wishie)

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Re: [alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 13 Oct '08

13 Oct '08

At Sun, 12 Oct 2008 20:16:10 -0500, Christopher Lemire wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Sun, Oct 12, 2008 at 2:34 PM, Takashi Iwai wrote: > > At Sat, 11 Oct 2008 13:04:02 -0500, > > William Pitcock wrote: > >> > >> Hi, > >> > >> On Sat, 2008-10-11 at 19:41 +0200, Takashi Iwai wrote: > >> > Could you build with XXX_CONST_TIMER defined? This will disable the > >> > adaptive timer-resolution handling I added to the driver, which > >> > doesn't > >> > exist in OSS sbxfi code at all. > >> > >> No change in behaviour. It survives a little longer, but then the system > >> hardlocks... without me even touching the keyboard even. > > > > This can be a spin deadlock I fixed today. > > Could you try the latest sound-unstable GIT or snapshot tarball? > > > > Also, now the driver has "debug" module option. As default, it's 1, > > and the driver shows some debug messages. When it's 2, it shows more > > often, e.g. at each IRQ. When it's 3, it shows all register accesses. > > > > The debug option can be changed even dynamically via sysfs. Write > > /sys/modules/snd_sbxfi/parameters/debug file as root, > > # echo 3 > /sys/modules/..../debug > > > > Just a note before I go to bed :) > > > > Takashi > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > Would someone mind telling me how I can put my card in hda-intel mode > and use the snd-hda-intel driver with it? I read it was possible to do > that with the creative windows software but didn't see that anywhere. > Is there any reason I shouldn't do that? Do I really need any Xfi > driver when I can do that? My Xfi card is a newer one that should have > the hda-intel mode. It's Vista compatible (though I use it dual boot > with xp) and a newer PCIe model called Titanium. It's newer than the > Creative Linux beta driver, so it will not work with that or OSS4. I > just removed OSS4 and going back to Alsa to test that snd-sbxfi > because OSS4 would crash my computer with this sound card. I think the > OSS4 was written for older Xfi models, not this newer PCIe model I > have. Chances of me getting it to work in Linux aren't good, so I'll > probably be doing a RMA on it in the next few days before I run out of > time to do that and going back to using onboard audio. The sound-unstable tree contains the HD-audio X-Fi support, but right now it's disabled due to possible conflicts of drivers. Just uncomment the corresponding entries in PCI entries of sound/pci/hda_intel.c. Takashi

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[alsa-devel] sound and sound-unstable git tree rebased
by Takashi Iwai 13 Oct '08

13 Oct '08

Hi, I rebased the whole sound GIT tree right now due to major conflicts with the upstream. Some commits order were changed due to manual pick-ups, but I think I picked all old commits. Please check whether anything missing. Also, the master branch is now pointing to the latest upstream, that is, ahead from 2.6.27. I revived the "devel" branch again for tracking the 2.6.27 tree. If you use 2.6.27 kernel but would like to track ALSA development, use this branch instead of master. Many topic branches vanished now because of the successful merges. There will be still topic/hda and topic/asoc branches, and occasionally more. The sound-unstable GIT tree is rebased as well. This is a usual thing, though... Sorry for inconvenience. Takashi

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[alsa-devel] ALSA: ASoC codec: remove unused #include <version.h>
by Huang Weiyi 13 Oct '08

13 Oct '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/ad1980.c sound/soc/codecs/wm8580.c sound/soc/codecs/wm8900.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f..1397b8e 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe..a10edda 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d3..3b326c9 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

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[alsa-devel] [PATCH] ASoC: update email address for Liam Girdwood
by Liam Girdwood 13 Oct '08

13 Oct '08

Update the contact information for Liam Girdwood in ASoC core and drivers as my old email address is no longer valid. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/oss/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/soc/at91/at91-ssc.c | 2 +- sound/soc/at91/eti_b1_wm8731.c | 2 +- sound/soc/codecs/ac97.c | 3 +-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8753.c | 3 +-- sound/soc/codecs/wm8753.h | 3 +-- sound/soc/codecs/wm9712.c | 3 +-- sound/soc/codecs/wm9713.c | 3 +-- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/soc-core.c | 5 ++--- sound/soc/soc-dapm.c | 5 ++--- 18 files changed, 21 insertions(+), 28 deletions(-) diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e..456a1b4 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * May 02, 2003 Liam Girdwood <lrg(a)slimlogic.co.uk> * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe..6e831af 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * May 2, 2003 Liam Girdwood <lrg(a)slimlogic.co.uk> * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index a5b1a79..1b61cc4 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -5,7 +5,7 @@ * Endrelia Technologies Inc. * * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 684781e..a22f349 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -10,7 +10,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96c..bd1ebdc 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d..d596f70 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44..be817b0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde0..f96c04b 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b..ffb471e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d058..aba402b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1a8373d..fe95384 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b..e6ff692 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8..33851cd 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2fb5829..e758034 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * lrg(a)slimlogic.co.uk * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -405,6 +405,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 9a70b00..8375bb5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa75..1300cb5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> * Richard Purdie <richard(a)openedhand.com> * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..462e635 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * with code, comments and ideas from :- * Richard Purdie <richard(a)openedhand.com> * @@ -1886,7 +1885,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 83fa9c4..efbd0b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -1541,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); -- 1.5.4.3

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[alsa-devel] [PATCH] hda: corrected invalid mixer values
by Matthew Ranostay 13 Oct '08

13 Oct '08

Corrected invalid mixer index values on the 92hd71bxxx codec branch. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 1e7b6c1..c590655 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1114,11 +1114,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),

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[alsa-devel] [PATCH] hda: add mixers for analog mixer on 92hd75xx codecs
by Matthew Ranostay 11 Oct '08

11 Oct '08

Add support for mixers on the analog mixer on some 92hd75xx codecs, along with adding a 'Mixer' entry for it's connection on the dmux. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..1f18d32 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -320,8 +320,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[1] = { - 0x1c, +static hda_nid_t stac92hd71bxx_dmux_nids[2] = { + 0x1c, 0x1d, }; static hda_nid_t stac92hd71bxx_smux_nids[2] = { @@ -837,20 +837,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; -#define HD_DISABLE_PORTF 3 +#define HD_DISABLE_PORTF 2 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for node 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ @@ -859,10 +857,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect port 0d to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute dac0 input in audio mixer */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1083,6 +1077,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1095,8 +1090,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -1616,7 +1620,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, + 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, }; @@ -2961,7 +2965,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) /* labels for amp mux outputs */ static const char *stac92xx_amp_labels[3] = { - "Front Microphone", "Microphone", "Line In" + "Front Microphone", "Microphone", "Line In", }; /* create amp out controls mux on capable codecs */ @@ -4275,6 +4279,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = { #endif }; +static struct hda_input_mux stac92hd71bxx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x00 }, + { "Mixer", 0x01 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4289,6 +4303,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, + sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -4329,6 +4345,7 @@ again: /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; + spec->dinput_mux = &spec->private_dimux; /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; @@ -4346,12 +4363,13 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } - spec->aloopback_mask = 0x20; + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ @@ -4370,7 +4388,6 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); switch (spec->board_config) { case STAC_HP_M4: @@ -4391,6 +4408,10 @@ again: spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + if (spec->dinput_mux) + spec->private_dimux.num_items += + spec->num_dmics - + (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) {

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