March 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4] mc13783 sound support
by Philippe Rétornaz 12 Mar '12

12 Mar '12

Hello This is patchset enable sound support on boards with mc13783 codec. The first patch adds the codec driver. It's Sascha's code ported to the current kernel with some minors additions. The second patch adds a generic SoC platform driver for imx3 boards with mc13783 codecs. The last two patches enable the sound on the mx31moboard boards and on mx31pdk_3ds boards. This patchset has been tested on mx31moboard and mx31pdk_3ds. It is based on a 3.3-rc6 kernel. I know there are some patches which change how audmux is handled on i.Mx processor but they are not yet merged so I prefer to post this patchset based on a known tree than to add another dependency. If the audmux change is merged I will rebase this patchset. Small note: If you want to test this, make sure you use redboot as bootloader or that you have the patch i.MX 35/5 AIPS Setup applied. Philippe Rétornaz (3): Add imx3sound sound support. mx31moboard: Add ssi configuration imx31pdk: Add sound support Sascha Hauer (1): add a mc13783 codec driver arch/arm/mach-imx/Kconfig | 2 + arch/arm/mach-imx/mach-mx31_3ds.c | 40 ++- arch/arm/mach-imx/mach-mx31moboard.c | 37 ++- drivers/mfd/mc13xxx-core.c | 3 +- include/linux/mfd/mc13xxx.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 1036 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 30 + sound/soc/imx/Kconfig | 10 + sound/soc/imx/Makefile | 2 + sound/soc/imx/imx3sound-mc13783.c | 141 +++++ 12 files changed, 1315 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/imx3sound-mc13783.c

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[alsa-devel] [PATCH 1/2] ASoC: core - Add platform component mutex.
by Liam Girdwood 12 Mar '12

12 Mar '12

Add mutex support for platform IO operations. e.g. can be used for platform DAPM widget IO ops. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 82bd773..2ebf787 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,7 @@ struct snd_soc_platform { int id; struct device *dev; struct snd_soc_platform_driver *driver; + struct mutex mutex; unsigned int suspended:1; /* platform is suspended */ unsigned int probed:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7978f6c..c90bb01 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3382,6 +3382,7 @@ int snd_soc_register_platform(struct device *dev, platform->dapm.dev = dev; platform->dapm.platform = platform; platform->dapm.stream_event = platform_drv->stream_event; + mutex_init(&platform->mutex); mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); -- 1.7.5.4

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[alsa-devel] [RFC] Initialize volumes of HD-audio slave ctls
by Takashi Iwai 12 Mar '12

12 Mar '12

Hi, the patch below is an attempt to initialize the volume / mutes of slave controls (such as "Headphone", "Speaker") with vmaster in HD-audio, so that the sound can come out only by changing the master volume/mute. We have thought that such initializations could be done well in alsactl init, but it seems that not everyone installs the latest and greatest alsactl, and there is always a risk that any new controls may be added before alsactl is updated and released. Since the master volume is set muted, the risk by this change should be low. patch_cirrus.c still doesn't support this because it's handling vmaster by itself, but it can be fixed later, too. If anyone has a concern by this, please let me know. thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0527ae1..d4736b9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -19,6 +19,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/mm.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/slab.h> @@ -2340,6 +2341,52 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) return 1; } +static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) +{ + mm_segment_t fs; + int _tlv[4]; + const int *tlv = NULL; + int val = -1; + + fs = get_fs(); + set_fs(get_ds()); + if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv)) + tlv = _tlv; + } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) + tlv = kctl->tlv.p; + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) + val = -tlv[2] / tlv[3]; + set_fs(fs); + return val; +} + +static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) +{ + struct snd_ctl_elem_value *ucontrol; + ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL); + if (!ucontrol) + return -ENOMEM; + ucontrol->value.integer.value[0] = val; + ucontrol->value.integer.value[1] = val; + kctl->put(kctl, ucontrol); + kfree(ucontrol); + return 0; +} + +static int init_slave_0dB(void *data, struct snd_kcontrol *slave) +{ + int offset = get_kctl_0dB_offset(slave); + if (offset > 0) + put_kctl_with_value(slave, offset); + return 0; +} + +static int init_slave_unmute(void *data, struct snd_kcontrol *slave) +{ + return put_kctl_with_value(slave, 1); +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2347,6 +2394,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * @tlv: TLV data (optional) * @slaves: slave control names (optional) * @suffix: suffix string to each slave name (optional) + * @init_slave_vol: initialize slaves to unmute/0dB * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2357,9 +2405,9 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * * This function returns zero if successful or a negative error code. */ -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix) + const char *suffix, bool init_slave_vol) { struct snd_kcontrol *kctl; int err; @@ -2380,9 +2428,16 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, (map_slave_func_t)snd_ctl_add_slave, kctl); if (err < 0) return err; + + /* init with master mute & zero volume */ + put_kctl_with_value(kctl, 0); + if (init_slave_vol) + map_slaves(codec, slaves, suffix, + tlv ? init_slave_0dB : init_slave_unmute, kctl); + return 0; } -EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); +EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6094dea8..caa6468 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -139,9 +139,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv); struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix); + const char *suffix, bool init_slave_vol); +#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ + __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true) int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9771b07..f450f2a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -82,6 +82,7 @@ struct ad198x_spec { unsigned int inv_jack_detect: 1;/* inverted jack-detection */ unsigned int inv_eapd: 1; /* inverted EAPD implementation */ unsigned int analog_beep: 1; /* analog beep input present */ + unsigned int avoid_init_slave_vol:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -223,11 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", + err = __snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, (spec->slave_vols ? spec->slave_vols : ad_slave_pfxs), - "Playback Volume"); + "Playback Volume", + !spec->avoid_init_slave_vol); if (err < 0) return err; } @@ -3604,6 +3606,7 @@ static int patch_ad1884(struct hda_codec *codec) spec->vmaster_nid = 0x04; /* we need to cover all playback volumes */ spec->slave_vols = ad1884_slave_vols; + spec->avoid_init_slave_vol = 1; codec->patch_ops = ad198x_patch_ops;

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[alsa-devel] [PATCH] ASoC: mx27vis-aic32x4: Convert it to platform driver
by Fabio Estevam 11 Mar '12

11 Mar '12

Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 2 + sound/soc/imx/mx27vis-aic32x4.c | 40 ++++++++++++++------------- 2 files changed, 23 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index 0a1643c..ed5dd41 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -397,6 +397,8 @@ static void __init visstrim_m10_board_init(void) &iclink_tvp5150, sizeof(iclink_tvp5150)); gpio_led_register_device(0, &visstrim_m10_led_data); visstrim_camera_init(); + imx_add_platform_device("mx27vis_aic32x4", 0, NULL, 0, + NULL, 0); } static void __init visstrim_m10_timer_init(void) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..09448b4 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,15 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +214,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis_aic32x4", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("mx27vis_aic32x4"); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: mx27vis-aic32x4: Make sure we are running on the correct machine
by Fabio Estevam 11 Mar '12

11 Mar '12

Only register mx27vis-aic32x4 if we are running on a imx27_visstrim_m10. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/mx27vis-aic32x4.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..c982bc8 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -194,6 +194,9 @@ static int __init mx27vis_aic32x4_init(void) { int ret; + if (!machine_is_imx27_visstrim_m10()) + return 0; + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); if (!mx27vis_aic32x4_snd_device) return -ENOMEM; -- 1.7.1

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[alsa-devel] [PATCH] ASoC: dapm: Add regulator member to struct dapm_widget.
by Liam Girdwood 11 Mar '12

11 Mar '12

Currently DAPM widgets use the private data for their regulator. Add a regulator * for widgets to use instead of private data. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1beb28d..bde6a52 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -323,6 +323,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct regulator; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -486,6 +487,7 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_context *dapm; void *priv; /* widget specific data */ + struct regulator *regulator; /* attached regulator */ /* dapm control */ short reg; /* negative reg = no direct dapm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e5aa65a..27132db 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,9 +883,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - return regulator_enable(w->priv); + return regulator_enable(w->regulator); else - return regulator_disable_deferred(w->priv, w->shift); + return regulator_disable_deferred(w->regulator, w->shift); } EXPORT_SYMBOL_GPL(dapm_regulator_event); @@ -2790,9 +2790,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->priv = devm_regulator_get(dapm->dev, w->name); - if (IS_ERR(w->priv)) { - ret = PTR_ERR(w->priv); + w->regulator = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->regulator)) { + ret = PTR_ERR(w->regulator); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); return NULL; -- 1.7.5.4

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Re: [alsa-devel] how to enumerate useful ALSA devices?
by Andrew Eikum 11 Mar '12

11 Mar '12

> The 2008 audio guide recommends > http://0pointer.de/blog/projects/guide-to-sound-apis.html > "Do not use the snd_card_xxx() APIs. For enumerating use > snd_device_name_hint() (and related functions). > snd_card_xxx() is obsolete... Bluetooth audio are not included" > > Is that correct advice? > > Wine still uses snd_card, so it knows only about "default" and > "plughw:x,y". I believe that's wrong but don't know what should be > used instead in order not to come up with a long list of artificial > names mentioned in /usr/share/alsa/alsa.conf such as surround51 when > the poor on-board card knows nothing but 16 bit stereo. > > I'm looking for a list of working devices on this machine, which likely > should contain "default" (usually leading to PA on a recent Ubuntu machine), > "plug:dmix" (PA getting out of the way when trying to use other devices), > "hw:0,0" and presumably not much more. Bump, still hoping for an answer to this. I tried experimenting with snd_device_name_hint(). It gave me what looked like a promising list, consisting of: null front:CARD=SB,DEV=0 surround40:CARD=SB,DEV=0 surround41:CARD=SB,DEV=0 surround50:CARD=SB,DEV=0 surround51:CARD=SB,DEV=0 surround71:CARD=SB,DEV=0 iec958:CARD=SB,DEV=0 This is identical to the list from "aplay -L". But, passing these to aplay fails if I have PulseAudio running ("device in use"). If I kill PA, then they don't fail, but give warnings about rate conversion and suggest the plug plugin. So I stick "plug:" in front of the device name, and they fail again ("unknown parameter front:CARD"). Clearly I'm doing something very wrong. But I don't know what! Then it occurred to me that this list also contains neither "default" nor the "pulse" device in my asound.conf[1]. Both of these work when passed to aplay. Looking to other projects for help, I find that VLC uses the snd_card_ functions, just like Wine currently does. Of course, that fails to find the "pulse" device from asound.conf. And mplayer's libao2 uses a couple of hard-coded device names, which is even worse (right?). So I'm really stuck here. Where am I supposed to get a list of devices that I can present to the user, including their Bluetooth devices and virtual devices from asound.conf? [1] asound.conf from Arch Linux pulseaudio-alsa package: http://projects.archlinux.org/svntogit/packages.git/tree/trunk/asound.conf?… Andrew

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Re: [alsa-devel] Port C Out selector , Port E Out selector of ad1988 and ad1988_3stack_dac_nids
by Raymond Yau 10 Mar '12

10 Mar '12

2011/9/16 Takashi Iwai <tiwai(a)suse.de>: > At Fri, 16 Sep 2011 11:59:54 +0800, > Raymond Yau wrote: >> >> 2011/9/12 Takashi Iwai <tiwai(a)suse.de>: >> > At Wed, 31 Aug 2011 11:21:00 +0800, >> > Raymond Yau wrote: >> >> >> >> HDA - Add Headphone Playback Volume control for ad1988/ad1989 >> >> >> >> - use DAC0 instead of DAC1 for Port-A Headphone >> >> - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp" >> > >> > Thanks, applied now. >> > >> > >> > Takashi >> > >> >> while testing the headphone volume control , >> >> I discover that the "Port C Out selector" and "Port E Out Selector" >> (i.e. node 0x31 and 0x32 ) by commit in patch_anlog.c >> >> fd66e0d0591dd12eb0bea1e9f3aa194bb93cebbd >> >> >> >> * AD1988 >> * >> * Output pins and routes >> * >> * Pin Mix Sel DAC >> * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06 >> * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06 >> * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a >> * port-D 0x12 (mute/hp) <- 0x29 <- 04 >> * port-E 0x17 (mute/hp) <- 0x26 <- 0x32 <- 05/0a >> >> >> >> The connection list of node 0x31 and 0x32 are different from datasheet >> >> >> Table 10. Audio Widget >> >> 31 Audio selector Selects DAC_2 and DAC_4 for PORT-C >> PORT-E Out Selector >> 32 Audio selector Selects DAC_2 and DAC_4 for PORT-E >> PORT-C In Selector >> >> >> >> Codec: Analog Devices AD1988 >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x11d41988 >> Subsystem Id: 0x104381e1 >> Revision Id: 0x100400 >> >> >> Node 0x31 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x04* 0x0a >> Node 0x32 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x05* 0x04 >> >> >> >> is the following patch based on real hardware ? >> >> >> commit 8c7c7b8dd2828b42c1230c6b0235e7d1dcf57e5 >> >> hda-codec - Fix surrounds on 3stack mode of AD1988 > > I don't see this commit id in my tree... > > > Takashi > http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=fd66e0d0591dd12eb… Thu, 17 Nov 2005 14:31:34 +0000 (15:31 +0100) Add AD1988 codec support to hda-codec driver. http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=f8c7c7b8dd282… Thu, 24 Nov 2005 15:17:20 +0000 (16:17 +0100)] Fixed the swapped surround/CLFE on 3stack mode of AD1988. hda-emu codecs/ad1988a -m "3stack" route -a 0x15 Out[0a] -x Sel[31] -- Mix[2c] -> Pin[15] Out[04] -> Sel[31] -- Mix[2c] -> Pin[15] route -a 0x17 Out[04] -> Sel[32] -- Mix[26] -> Pin[17] Out[05] -x Sel[32] -- Mix[26] -> Pin[17]

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[alsa-devel] [PATCH] USB-Audio: Fixup card name for Logitech web camera
by David Henningsson 10 Mar '12

10 Mar '12

Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index d7c121e..966172f 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -39,7 +39,7 @@ USB-Audio.pcm.iec958_device { # The below don't have digital in/out, so prevent them from being opened. "Logitech USB Headset" 999 - "Logitech Web Camera" 999 + "USB Device 0x46d:0x992" 999 "Blue Snowball" 999 } -- 1.7.9

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[alsa-devel] [PATCH 0/2] ARM/ASoC: OMAP: McBSP config cleanup
by Peter Ujfalusi 09 Mar '12

09 Mar '12

Hi, Since the McBSP driver has been moved out from arch/arm/plat-omap/ and it is combined as a single driver with the sound/soc/omap/omap-mcbsp.c there is no longer need to have CONFIG_OMAP_MCBSP. This series cleans up the Kconfigs, and Makefiles. Also it is fixes the issue that the OMAP audio drivers were not buildable as modules (the ones using McBSP) reported by Grazvydas Ignotas <notasas(a)gmail.com> I have generated this series on top of Liam's for-3.4 branch. The arch/arm/mach-omap* patch applies cleanly on top of todays linux-next, so if this series goes via ASoC we are not going to have conflicts with stuff coming via l-o. Tested on BeagleBoard, and SDP4430, compile tested for OMAP1. This series should go via ASoC along with the rest of the McBSP changes. Regards, Peter --- Peter Ujfalusi (2): ARM: OMAP: Remove CONFIG_OMAP_MCBSP references ASoC: OMAP: Build config cleanup for McBSP arch/arm/mach-omap1/Kconfig | 3 --- arch/arm/mach-omap1/Makefile | 4 +++- arch/arm/mach-omap2/Makefile | 4 +++- sound/soc/omap/Kconfig | 6 ------ sound/soc/omap/Makefile | 3 +-- 5 files changed, 7 insertions(+), 13 deletions(-) -- 1.7.8.5

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