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Re: [alsa-devel] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200
by Takashi Iwai 08 Oct '09

08 Oct '09

At Wed, 23 Sep 2009 00:39:45 -0500 (CDT), Manoj Iyer wrote: > > > >From c96629e96fecbe59de7124ec913019ed378d0029 Mon Sep 17 00:00:00 2001 > From: Manoj Iyer <manoj.iyer(a)canonical.com> > Date: Tue, 22 Sep 2009 18:33:29 -0500 > Subject: [PATCH] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200 > > Patch was tested on Toshiba NB200 and is found to enable sound. So, it doesn't work without the model quirk? Could you give alsa-info.sh output (run with --no-upload option)? thanks, Takashi > > Signed-off-by: Manoj Iyer <manoj.iyer(a)canonical.com> > Cc: stable(a)kernel.org > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 30eeb30..713bf69 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16876,6 +16876,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { > SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), > SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", > ALC662_3ST_6ch_DIG), > + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), > SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), > SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", > ALC662_3ST_6ch_DIG), > -- > 1.6.3.3 > > > --- manjo >

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[alsa-devel] [PATCHv3 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Lopez Cruz, Misael 08 Oct '09

08 Oct '09

In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data may be passed: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected However, codec driver can operate if any or none of them are passed. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- drivers/mfd/twl-core.c | 15 +++++++++++++++ include/linux/i2c/twl.h | 7 +++++++ 2 files changed, 22 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index af4cf47..6432af1 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -108,6 +108,12 @@ #define twl_has_mmc() false #endif +#if defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) +#define twl_has_codec() true +#else +#define twl_has_codec() false +#endif + #if defined(CONFIG_TWL4030_USB) || defined(CONFIG_TWL4030_USB_MODULE) #define twl_has_usb() true #else @@ -190,6 +196,7 @@ #define BCI_SUB_CHIP_ID SUB_CHIP_ID1 #define GPIO_SUB_CHIP_ID 0 /* NOT SUPPORTED IN TWL6030 */ #define KEYPAD_SUB_CHIP_ID 0 /* ADDED FOR COMPILATION ONLY */ +#define CODEC_SUB_CHIP_ID SUB_CHIP_ID3 /* subchip/slave 0 0x48 - POWER */ #define TWL6030_BASEADD_RTC 0x0000 @@ -632,6 +639,14 @@ add_children(struct twl_platform_data *pdata, unsigned long features) if (IS_ERR(child)) return PTR_ERR(child); } + + if (twl_has_codec()) { + child = add_child(CODEC_SUB_CHIP_ID, "twl6030_codec", + pdata->codec, sizeof(*pdata->codec), false, + 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } #endif if (twl_has_usb() && pdata->usb) { diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index b687a8b..e76ca9b 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -472,6 +472,12 @@ struct twl_usb_data { enum twl_usb_mode usb_mode; }; +struct twl_codec_data { + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ +}; + struct twl_platform_data { unsigned irq_base, irq_end; struct twl_bci_platform_data *bci; @@ -479,6 +485,7 @@ struct twl_platform_data { struct twl_madc_platform_data *madc; struct twl_keypad_data *keypad; struct twl_usb_data *usb; + struct twl_codec_data *codec; /* LDO regulators common to TWL4030/TWL6030 */ struct regulator_init_data *vdac; -- 1.5.4.3

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[alsa-devel] Possibility to build alsa driver without having exact kernel source?
by Christian Steffen 07 Oct '09

07 Oct '09

Hello list, I need a new driver for a atom cpu driven device. It needs the "snd_intel_hda" driver to work as it has intels Poulsbo chipset. Currently I have a compiled module, but sound has a constant clicking noise with this one. Therefore I'd like to try an updated alsa driver, which hopefully fixes a problem on Poulsbo chipsets with stepping D0 or newer. My problem is that I only have a compiled running kernel, but no source from it. All I can see (trough uname -a) is: Linux joggler 2.6.30 #4 SMP Tue Jun 23 21:55:08 BST 2009 i686 GNU/Linux So I know, the kernel is a 2.6.30, arch i686. I tried to compile the driver now just with the standard kernel 2.6.30 source from kernel. org. But when trying to load these modules, I get "Invalid module format". Is there any chance to get a working compiled driver without having more information? Thanks! Christian Steffen P Bitte drucken Sie diese Mail nur, wenn dies notwendig ist. please don't print this e-mail unless you really need to

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Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Aha, sorry - I misunderstood what was being asked to test. Two channel audio is working correctly with the development snapshot. Thanks, Gus On Wed 7/10/09 10:16 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 7 Oct 2009 03:12:49 +0000, > wrote: > > > > Hello Takashi Iwai, > > > > I tested your latest snapshot and it did not detect the 6 channel > audio. > This is so per design. The default auto-parser doesn't enable > 6-channel since otherwise you get problems with the automatic HP / > mic > jack detection. > Does it work except for 6-channel things? > thanks, > Takashi > _______________________________________________ > Alsa-devel mailing list > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] ascenario: Small updates for new API.
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. [PATCH 1/3] ascenario: Rename SND_SCN_INT_QOS to SND_SCN_INT_QOS_POWER As proposed by Jaroslav. Emphasis the power consumption relationship. [PATCH 2/3] ascenario: Fix snd_scenario_get_control_id() declaration to match implementation Just different names in declaration and implementation. I favoured _control_id(). [PATCH 3/3] ascenario: Implement snd_scenario_get_control_id() Replace the four explicit functions with a generic one. I was not able to find another way to get the id from the given numid then getting the full list and iterate through it. If there is something easier let me know. :) regards Stefan Schmidt

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[alsa-devel] [RFC] Add scenario management
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. This patchset add support for scenario management, aka use cases, in alsa-lib and example programs in alsa-utils. It allows to adjust the current audio kcontrol settings from a prespective of a changed use case like the switch from listening to music on your phone to an incoming phone call and back. To achieve this it offers control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. This is of course not meant to replace gstreamer, PulseAudio or a sound server, but is meant to work in tandem with such audio software. We think alsa-lib is the best place for this as it will be used on small embedded system which only use alsa from the full stack of audio libs in linux. We see potential for desktop usage as well though. There is also the support of QoS and the submission to salsa on the TODO list, but we think it is best to gather soem feedback and get the core merged before drifting away with to much other features. Please let me know what can be improved. Project page is here :- http://www.slimlogic.co.uk/?p=40 regards Stefan Schmidt

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Re: [alsa-devel] sscape: general code improvements
by krzysztof.h1＠poczta.fm 07 Oct '09

07 Oct '09

"Takashi Iwai" <tiwai(a)suse.de> pisze: > At Tue, 6 Oct 2009 19:34:38 +0200, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > General code and coding style improvements. > > > > Add a missing argument for SET_XXX_MIDI_VOL command. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Do you really want to change oss/sscape.c that you wanted to remove? > No, it is a mistake. > > > --- > > > > What is a standard name for switch to select internal or external midi? > It is very easy to > > add such a functionality to the driver. > > What do you mean exactly to switch int/ext midi? > Usually the driver provides the interfaces for both, and let user choose. > If they are exclusive, it's checked in the open. > Or, if preferred, a control element can be used to choose the > interface... > The driver interface (MPU-401 UART) is only one and there is a command to switch it to the on-board synthetizer or external midi (joysitck port on the card). Regards, Krzysztof --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

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Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Hello Takashi Iwai, I tested your latest snapshot and it did not detect the 6 channel audio. I have attached the output of alsa-info for analysis. The original patch that I submitted did not detect the coaxial/optical SPDIF interface on this board. I've been running with the ALC662_3ST_6ch_DIG method instead of ALC662_3ST_6ch which was in the original patch. The _DIG method now shows my spdif interface correctly and my 6 channel analog audio is also working correctly. I do not have a receiver to test the coaxial/optical output, though I suspect it works correctly. I've attached a new patch which uses the _DIG method as I believe its the more proper method to use for this hardware layout. Thanks, Gus Bourg On Mon 28/09/09 9:30 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > Could you check whether the very latest alsa-driver snapshot still > doesn't work without your change? > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna > pshot.tar.gzIf it doesn't work, could you give alsa-info.sh output (run > --no-upload option)? > If your fix is mandatory, I'm going to apply it happily. But, then > please give your sign-off, too. > thanks, > Takashi > > Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] alsa-utils cross compilation error
by Harish Kumar 07 Oct '09

07 Oct '09

Hi, I am trying to cross compile the alsa utils for PPC and as well as arm architecture. the compilation of alsa-libs was successful with their respective toolchains. But configuring the alsa-util is failing. I am using alsa-utils-1.0.11rc2,alsa-lib-1.0.11rc2. I am getting the error checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. I feel i have to provide the path of the include files but not sure which path to be passed and how it needs to be passed to configure. Request your help to resolve the same. I invoked the configure using the following command and the output of the configure is as below. root@localhost alsa-utils-1.0.11rc2]# CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-linux configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ANSI C... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld checking if the linker (/opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking whether NLS is requested... yes checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ANSI C... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. [2]+ Done gedit configure [root@localhost alsa-utils-1.0.11rc2]# Thanks and Regards Harish Kumar.B

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[alsa-devel] ice1724 SPDIF buffer size?
by Robert Hancock 07 Oct '09

07 Oct '09

I noticed that in the ice1724 driver, we set a default and maximum preallocated buffer size of 256K for the "professional" stream, but only 64K for the SPDIF and independent stereo pair streams, even though the hardware supports 256K for all streams. Is there any reason why we can't use 256K for all of them? I made this change here and it seems to work fine (for SPDIF at least, haven't tested stereo). From a power and CPU usage perspective, it makes sense to allow as big a buffer as possible for when latency isn't critical..

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