January 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: Merge the soc_value_enum to soc_enum struct
by Peter Ujfalusi 08 Jan '09

08 Jan '09

Merge the recently introduced soc_value_enum structure to the soc_enum. The value based enums are still handled separately from the normal enum types, but with the merge some of the newly introduced functions can be removed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 15 +-------- sound/soc/soc-core.c | 37 +-------------------- sound/soc/soc-dapm.c | 80 +++------------------------------------------- 4 files changed, 9 insertions(+), 125 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 4af1083..93a4edb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -178,7 +178,7 @@ .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_value_enum_double, \ + .info = snd_soc_info_enum_double, \ .get = snd_soc_dapm_get_value_enum_double, \ .put = snd_soc_dapm_put_value_enum_double, \ .private_value = (unsigned long)&xenum } diff --git a/include/sound/soc.h b/include/sound/soc.h index 9b930d3..24593ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -106,7 +106,7 @@ .private_value = (unsigned long)&xenum } #define SOC_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ - .info = snd_soc_info_value_enum_double, \ + .info = snd_soc_info_enum_double, \ .get = snd_soc_get_value_enum_double, \ .put = snd_soc_put_value_enum_double, \ .private_value = (unsigned long)&xenum } @@ -211,8 +211,6 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_info_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, @@ -419,17 +417,6 @@ struct soc_enum { unsigned char shift_l; unsigned char shift_r; unsigned int max; - const char **texts; - void *dapm; -}; - -/* semi enumerated kcontrol */ -struct soc_value_enum { - unsigned short reg; - unsigned short reg2; - unsigned char shift_l; - unsigned char shift_r; - unsigned int max; unsigned int mask; const char **texts; const unsigned int *values; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cbe7e8..55fdb4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1585,37 +1585,6 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** - * snd_soc_info_value_enum_double - semi enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double semi enumerated - * mixer control. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. - */ -int snd_soc_info_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->max; - - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_value_enum_double); - -/** * snd_soc_get_value_enum_double - semi enumerated double mixer get callback * @kcontrol: mixer control * @ucontrol: control element information @@ -1631,8 +1600,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); @@ -1671,8 +1639,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; unsigned short mask; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ad0d801..493a4e8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -137,7 +137,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } break; case snd_soc_dapm_value_mux: { - struct soc_value_enum *e = (struct soc_value_enum *) + struct soc_enum *e = (struct soc_enum *) w->kcontrols[i].private_value; int val, item; @@ -200,30 +200,6 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, return -ENODEV; } -/* connect value_mux widget to it's interconnecting audio paths */ -static int dapm_connect_value_mux(struct snd_soc_codec *codec, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, - struct snd_soc_dapm_path *path, const char *control_name, - const struct snd_kcontrol_new *kcontrol) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - int i; - - for (i = 0; i < e->max; i++) { - if (!(strcmp(control_name, e->texts[i]))) { - list_add(&path->list, &codec->dapm_paths); - list_add(&path->list_sink, &dest->sources); - list_add(&path->list_source, &src->sinks); - path->name = (char *)e->texts[i]; - dapm_set_path_status(dest, path, 0); - return 0; - } - } - - return -ENODEV; -} - /* connect mixer widget to it's interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, @@ -774,45 +750,6 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } -/* test and update the power status of a value_mux widget */ -static int dapm_value_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_value_enum *e) -{ - struct snd_soc_dapm_path *path; - int found = 0; - - if (widget->id != snd_soc_dapm_value_mux) - return -ENODEV; - - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) - return 0; - - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { - if (path->kcontrol != kcontrol) - continue; - - if (!path->name || !e->texts[mux]) - continue; - - found = 1; - /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) - path->connect = 1; /* new connection */ - else - path->connect = 0; /* old connection must be - powered down */ - } - - if (found) { - dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mux power update"); - } - - return 0; -} - /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int reg, @@ -1045,17 +982,12 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->connect = 1; return 0; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: ret = dapm_connect_mux(codec, wsource, wsink, path, control, &wsink->kcontrols[0]); if (ret != 0) goto err; break; - case snd_soc_dapm_value_mux: - ret = dapm_connect_value_mux(codec, wsource, wsink, path, - control, &wsink->kcontrols[0]); - if (ret != 0) - goto err; - break; case snd_soc_dapm_switch: case snd_soc_dapm_mixer: ret = dapm_connect_mixer(codec, wsource, wsink, path, control); @@ -1382,8 +1314,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); @@ -1423,8 +1354,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, mux; unsigned short mask; int ret = 0; @@ -1443,7 +1373,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_value_mux_update_power(widget, kcontrol, mask, mux, val, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.6.1

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[alsa-devel] Autodetection for Acer Aspire 5930G and 6930G
by Łukasz Wojniłowicz 08 Jan '09

08 Jan '09

This is a patch which adds correct auto detection of model for snd-hda-intel for Acer Aspire 5930G and 6930G. Tested on my 5930G. It finally adds hp jack sense and 5.1 speaker system sliders. Information about 5930G Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x1025013f Revision Id: 0x100202 Information about 6930G Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x1025015e Revision Id: 0x100202 It is first patch prepared by me and it is against commit daead538e9d8efe61d7d2bd12993c8d961b0abd6 from "http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=daead538e9d8efe61…" --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8461,5 +8461,6 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", - ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */

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Re: [alsa-devel] Driver for TI/BB PCM3006 codec
by Mark Brown 08 Jan '09

08 Jan '09

On Thu, Jan 08, 2009 at 11:16:00AM +0100, Davide Rizzo wrote: > This patch implements the driver for the TI (ex Burr-Brown) PCM3006 codec > > Signed-off-by: Davide Rizzo <elpa-rizzo(a)gmail.com> This looks good but needs a refresh against the current ASoC code - there have been some API changes around codec registration for 2.6.29. You should make this a platform driver, then register the codec and DAIs with the ASoC core when that driver is probed. Current code is in the topic/asoc branc of: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git A few other comments below. > +++ linux-2.6.28.elpa/sound/soc/codecs/Kconfig 2009-01-04 > 08:58:40.000000000 +0100 > @@ -60,6 +60,10 @@ config SND_SOC_CS4270_VD33_ERRATA > bool > depends on SND_SOC_CS4270 > > +config SND_SOC_PCM3006 > + tristate > + depends on SND_SOC > + The depends isn't needed and this should also be added to SND_SOC_ALL_CODECS. > +/* > + * pcm3006.c -- ALSA Soc PCM3006 codec support > + * > + * Revision history > + * 25th Apr 2008 Initial version. Please drop the changelog from the code - git will keep track of this. > +static unsigned int pcm3006_read(struct snd_soc_codec *codec, > + unsigned int reg) > +{ > + return 0; > +} > + > +static int pcm3006_write(struct snd_soc_codec *codec, unsigned int reg, > + unsigned int val) > +{ > + return 0; > +} Just remove these if the codec doesn't have any registers, they are only needed if that is the case. > 12:30:42.000000000 +0200 > @@ -0,0 +1,19 @@ > +/* > + * linux/sound/codecs/i2s.h -- ALSA SoC Layer > + * Looks like bitrot here...

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[alsa-devel] 1/7 unstable does not compile: error: ‘SW_VIDEOOUT_INSERT’ undeclared here (not in a function)
by Susan Cragin 08 Jan '09

08 Jan '09

Here is last few lines of output. make[1]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable' make -C /lib/modules/2.6.28-4-generic/build SUBDIRS=/home/susan/Desktop/alsa-driver-unstable CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux-headers-2.6.28-4-generic' CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/hrtimer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/hwdep.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memory_wrapper.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memalloc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sgbuf.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_native.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_lib.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_timer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_misc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/pcm_memory.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/rawmidi.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/timer.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/wrappers.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/misc_driver.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sound.o /home/susan/Desktop/alsa-driver-unstable/acore/sound.c: In function ‘snd_request_other’: /home/susan/Desktop/alsa-driver-unstable/acore/sound.c:99: warning: format not a string literal and no format arguments CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/init.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/memory.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/info.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/control.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/misc.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/device.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/isadma.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/sound_oss.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/info_oss.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/vmaster.o CC [M] /home/susan/Desktop/alsa-driver-unstable/acore/jack.o In file included from /home/susan/Desktop/alsa-driver-unstable/acore/jack.c:13: /home/susan/Desktop/alsa-driver-unstable/acore/../alsa-kernel/core/jack.c:31: error: ‘SW_VIDEOOUT_INSERT’ undeclared here (not in a function) make[3]: *** [/home/susan/Desktop/alsa-driver-unstable/acore/jack.o] Error 1 make[2]: *** [/home/susan/Desktop/alsa-driver-unstable/acore] Error 2 make[1]: *** [_module_/home/susan/Desktop/alsa-driver-unstable] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.28-4-generic' make: *** [compile] Error 2 susan@ubuntu:~/Desktop/alsa-driver-unstable$

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[alsa-devel] [PATCH 0/2] ALSA jack updates for 2.6.30
by Mark Brown 07 Jan '09

07 Jan '09

The following changes since commit 6a94cb73064c952255336cc57731904174b2c58f: Linus Torvalds (1): Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git topic/jack-core Jani Nikula (1): ALSA: Add support for video out to the jack reporting API Mark Brown (1): ALSA: Reduce boilerplate for new jack types include/linux/input.h | 1 + include/sound/jack.h | 5 +++++ sound/core/jack.c | 45 +++++++++++++++++++++------------------------ 3 files changed, 27 insertions(+), 24 deletions(-)

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Re: [alsa-devel] Bug tracking help wanted (Re: alsa driver #2234 SoundScape Elite not support)
by Krzysztof Helt 07 Jan '09

07 Jan '09

On Tue, 6 Jan 2009 13:41:35 -0500 "Theodore J. Allen" <tjallen(a)hws.edu> wrote: > sscape: hardware detected at 0x330, using IRQ 5, DMA 1 > ALSA sound/isa/cs423x/cs4231_lib.c:226: codec out - reg 0xc = 0x40 > ALSA sound/isa/cs423x/cs4231_lib.c:1038: cs4231: port = 0x534, id = 0xa > ALSA sound/isa/cs423x/cs4231_lib.c:1044: CS4231: VERSION (I25) = 0x0 > ALSA sound/isa/cs423x/cs4231_lib.c:1063: unknown CS chip with version 0x0 Hi, There is one problem here. The codec is not detected. The codec on supported SoundScape boards is AD1845 which is compatible (more or less) with the CS4231 codec (hence cs4231 library). According to the wikipedia, the SS Elite uses AD1848KP codec which is older and not supported by the cs4231 library. Could you confirm that your card had AD1848KP codec (big chip with "Analog Devices" and "SoundPort" captions)? The simplest solution is to try the 2.6.28 or later kernel as the support for CS4231 and AD1848 codecs were unified. The new kernel may just work for you out of the box due to this change. If it does not please send the dmesg output again. Thank for your findings, Krzysztof ---------------------------------------------------------------------- Szef przynudza? Zagraj sobie! Kliknij >> http://link.interia.pl/f200e

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[alsa-devel] [PATCH] ASoC: add new DAPM control
by Ian Molton 07 Jan '09

07 Jan '09

Hi. The attached patch adds a new mixer control that allows cleaner support for the wm9705 codec Im working on. I think it probably has wider uses though. It saves a few bytes too in shorter names :-) If you prefer I have it available in: git://git.mnementh.co.uk/linux-2.6-im.git asoc and on gitweb at: http://git.mnementh.co.uk/cgi-bin/gitweb.cgi?p=linux-2.6-im.git;a=shortlog;…

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Re: [alsa-devel] Asus w5f. ALC660-VD codec not detected
by Takashi Iwai 07 Jan '09

07 Jan '09

At Wed, 7 Jan 2009 12:57:57 +0100, Salvatore Filippone wrote: > > On Wed, Jan 7, 2009 at 12:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 29 Dec 2008 19:45:25 +0100, > Nigel Henry wrote: > > > > UPDATE: Salvatore has found a workaround, so that both codecs are > detected > > correctly. Any help from alsa developers for a permanent fix would be > > appreciated. See workaround at bottom of post. > > > (snip) > > <Quoting Salvatore> > > Saturday 16:05:58 > > > > Ok, I finally found a workaround. > > According to the thread starting at > > http://thread.gmane.org/gmane.linux.alsa.devel/58849 > > I grabbed the latest source code > alsa-driver-1.0.18a.17.g27a32.139.g6e583/ > > compiled with --with-debug=verbose and got the message > > Dec 27 15:13:06 localhost kernel: ALSA > > /mnt/travel/ALSA/alsa-driver-1.0.18a.17.g27a32.139.g6e583/pci/hda/../../ > alsa-kernel/pci/hda/hda_intel.c:779: > > codec_mask = 0x2 > > > > I then figure out that the sound device was at codec#0 when it worked, > > therefore I changed snd_hda_intel.c by forcing codec_mask > > if (!chip->codec_mask) { > > chip->codec_mask = azx_readw(chip, STATESTS); > > chip->codec_mask=0x03; > > snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); > > } > > > > And now the sound works correctly. > > Of course this opens the question of why azx_readw does not return a > > correct mask; > > Well, this implies a BIOS bug. This value is set by BIOS to indicate > which codec slot is available. > > > maybe the developers will be interested and/or will suggest a > > fix that is a little more elegant/widely applicable? And why did the > base > > code work just once, for no clear reason ? (I did not have debug=verbose > at > > that time, so I don't have detailed messages in the system log). > > Not sure about this as I didn't track this bug. > When did it work once? Is the device enabled by BIOS properly? > > To summarize: > > 1. It always did work with FC7. I guess it worked likely just casually. The old driver didn't clear the register bit before probing, so it got uninitialized bits. Takashi

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[alsa-devel] [PATCH 0/1] ALSA: Add support for video out to the jack reporting API
by Jani Nikula 07 Jan '09

07 Jan '09

The following patch is generated against topic/jack-core branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Jani.

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[alsa-devel] How to handle multiple sound cards?
by cocala 07 Jan '09

07 Jan '09

Hi, Our platform has multiple sound cards: Stereo, Surround51, spdif and Bluetooth PCM. These sound cards are registered in random order as some sound card are registered as module or not registered sometimes. So the application can't use directly use the card number for specific sound card. If using alias, the asound.conf also should be changed according to the registered sound card after boot up. What's the best way to handle multiple sound cards?

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