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

Re: [alsa-devel] [PATCH] Add missing modules aliases to get sound working on omap devices
by Peter Ujfalusi 06 Jul '12

06 Jul '12

On 07/04/2012 12:16 PM, Guillaume Gardet wrote: > This patch add missing modules aliases to get sound working on omap devices. > > Tested on Beagleboard xM rev. B. > > This patch is against 3.5-rc5 vanilla. > > Signed-off-by: Guillaume GARDET <guillaume.gardet(a)free.fr> > > add-missing-modules-aliases-to-fix-audio-on-omap-devices.patch Can you send the patch generated by git format-patch? Since this is sound related patch, please send it to alsa-devel. Please also CC the maintainers for the patches to be sure they are reaching the right audience. > --- ./sound/soc/omap/omap-pcm.c.orig 2012-07-04 10:30:13.611641186 +0200 > +++ ./sound/soc/omap/omap-pcm.c 2012-07-04 10:30:38.894034111 +0200 > @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); > MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); > MODULE_DESCRIPTION("OMAP PCM DMA module"); > MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:omap-pcm-audio"); > --- ./sound/soc/omap/omap-mcbsp.c.orig 2012-07-04 10:29:14.240066817 +0200 > +++ ./sound/soc/omap/omap-mcbsp.c 2012-07-04 12:03:28.020756978 +0200 > @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver > MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); > MODULE_DESCRIPTION("OMAP I2S SoC Interface"); > MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:omap-mcbsp-dai"); This should be: +MODULE_ALIAS("platform:omap-mcbsp"); -- Péter

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Enable ASYNCCLK domain for audio interfaces
by Mark Brown 05 Jul '12

05 Jul '12

If an audio interface is configured to use ASYNCCLK then update the asynchronous sample rate rather than one of our primary sample rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8e066eb..d0bcca9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -507,6 +507,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; int i; @@ -530,10 +532,6 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* - * We will need to be more flexible than this in future, - * currently we use a single sample rate for the chip. - */ for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) if (arizona_sr_vals[i] == params_rate(params)) break; @@ -552,8 +550,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, -- 1.7.10

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Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 05 Jul '12

05 Jul '12

On 05.07.2012 17:32, Philipp Dreimann wrote: > On 5 July 2012 16:41, Takashi Iwai <tiwai(a)suse.de> wrote: >>> Try the patch below. If it still doesn't work, give the kernel >>> message again and lsusb -v output, too. >> >> Did it make any difference? > > No, it unfortunately did not. > > lsusb is attached and the parts of the kernel log are in the mail: > > [ 41.243211] usb 2-1.2: new high-speed USB device number 3 using ehci_hcd > [ 41.605666] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) > [ 42.002275] input: UVC Camera (046d:0821) as > /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input12 > [ 42.085880] usbcore: registered new interface driver snd-usb-audio > > [ 52.790139] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.794363] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.798375] uvcvideo: Failed to query (GET_DEF) UVC control 3 on > unit 2: -71 (exp. 2). > [ 52.802499] uvcvideo: Failed to query (GET_DEF) UVC control 7 on > unit 2: -71 (exp. 2). > [ 52.806617] uvcvideo: Failed to query (GET_DEF) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.810734] uvcvideo: Failed to query (GET_DEF) UVC control 4 on > unit 2: -71 (exp. 2). > [ 52.814855] uvcvideo: Failed to query (GET_DEF) UVC control 5 on > unit 2: -71 (exp. 1). > [ 52.818976] uvcvideo: Failed to query (GET_CUR) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.823096] uvcvideo: Failed to query (GET_DEF) UVC control 8 on > unit 2: -71 (exp. 2). > [ 52.827215] uvcvideo: Failed to query (GET_DEF) UVC control 1 on > unit 2: -71 (exp. 2). > [ 52.831331] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > [ 52.835459] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > The last line repeats a few times, then the Failed to query block > repeats again, and so on. But only if I try to use the webcam. > > [ 82.363599] 3:1:3: cannot set freq 32000 to ep 0x86 > Is showing up without using the webcam. > What really puzzles me is that the uvcvideo code fails as well, and even before the sound driver bails out. If you wouldn't have bisected the problem, I would clearly blame some other change for that. It seems something confuses the device quite badly so requests to it return -EPROTO. From a quick grep, this error can only be originated from low-level EHCI functions. I need to ponder on that a little ... Daniel

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Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 05 Jul '12

05 Jul '12

At Tue, 3 Jul 2012 10:40:51 +0200, Philipp Dreimann wrote: > > On 3 July 2012 10:27, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 3 Jul 2012 10:20:13 +0200, > > Philipp Dreimann wrote: > >> > >> On 3 July 2012 10:00, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 3 Jul 2012 09:41:09 +0200, > >> > Philipp Dreimann wrote: > >> >> > >> >> On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > >> >> > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> >> >> Hello, > >> >> >> > >> >> >> my usb webcam, which happens to have a microphone too, stopped working > >> >> >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > >> >> > > >> >> > Please try 3.5-rc5. A number of regressions were fixed just recently. > >> >> > > >> >> > Let me know if that version still causes problems. > >> >> > >> >> It still does. > >> > > >> > Try to pass ignore_ctl_error=1 to snd-usb-audio module. > >> > If this works and no other problem happens, we can add a similar quirk > >> > like some Logitech webcams. > >> > >> This does not seem to help. > > > > And webcam itself is working with our without snd-usb-audio module? > Yes, if I blacklist the module, the webcam works fine. > > > Which device at all? > logitech c910 > > > Too little information to analyze. > You guys tell me what you need to know. Try the patch below. If it still doesn't work, give the kernel message again and lsusb -v output, too. thanks, Takashi --- diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 54607f8..f0ede13 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -387,7 +387,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; } - if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) + if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) return err; subs->cur_audiofmt = fmt; @@ -450,7 +450,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct usb_interface *iface; iface = usb_ifnum_to_if(subs->dev, fmt->iface); alts = &iface->altsetting[fmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, subs->interface, alts, fmt, rate); + ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate); if (ret < 0) return ret; subs->cur_rate = rate;

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[alsa-devel] [PATCH] ASoC: wm1250-ev1: Flag all supported rates in the DAI
by Mark Brown 05 Jul '12

05 Jul '12

Not previously noticed due to normal usage being with CODEC<->CODEC links. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index e0b51e9..951d7b4 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = { .hw_params = wm1250_ev1_hw_params, }; +#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000) + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &wm1250_ev1_ops, -- 1.7.10

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Re: [alsa-devel] [PATCH 2/2] ASoC: imx-mc13783: Add audmux settings for mx27pdk
by Mark Brown 05 Jul '12

05 Jul '12

On Tue, Jul 03, 2012 at 03:44:58PM -0300, Fabio Estevam wrote: > mx27pdk board also has a mc13783 codec. > > Add support for it and do a run-time machine type check to perform the correct > audiomux settings. Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000
by Mark Brown 05 Jul '12

05 Jul '12

On Wed, Jul 04, 2012 at 03:38:27PM +0800, Hui Wang wrote: > If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR > register (means Tx clock pin is input), we don't need to set > IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and > i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL > settings are ignored". Applied, thanks.

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[alsa-devel] [PATCH] ALSA: pcm: Make constraints lists const
by Mark Brown 05 Jul '12

05 Jul '12

They aren't modified by the core so the drivers can declare them const. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 2 +- sound/core/pcm_lib.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e7afcc9..c75c0d1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l); + const struct snd_pcm_hw_constraint_list *l); int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa6..8659bc5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1253,7 +1253,7 @@ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_list, l, + snd_pcm_hw_rule_list, (void *)l, var, -1); } -- 1.7.10

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[alsa-devel] [PATCH] ALSA: pcm: Mostly constify constraints
by Mark Brown 05 Jul '12

05 Jul '12

There is no need for the constraints to be modified while being applied so they can be declared const, allowing drivers to declare constraint lists const if they wish. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 6 +++--- sound/core/pcm_lib.c | 10 +++++----- sound/core/pcm_native.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e7afcc9..3b8f6ca 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -217,7 +217,7 @@ struct snd_pcm_hw_rule { snd_pcm_hw_rule_func_t func; int var; int deps[4]; - void *private; + const void *private; }; struct snd_pcm_hw_constraints { @@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l); + const struct snd_pcm_hw_constraint_list *l); int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, @@ -835,7 +835,7 @@ int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, - snd_pcm_hw_rule_func_t func, void *private, + snd_pcm_hw_rule_func_t func, const void *private, int dep, ...); int snd_pcm_format_signed(snd_pcm_format_t format); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa6..4372de5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1104,7 +1104,7 @@ static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned */ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, - snd_pcm_hw_rule_func_t func, void *private, + snd_pcm_hw_rule_func_t func, const void *private, int dep, ...) { struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; @@ -1233,7 +1233,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_list *list = rule->private; + const struct snd_pcm_hw_constraint_list *list = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask); } @@ -1250,7 +1250,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l) + const struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, snd_pcm_hw_rule_list, l, @@ -1262,7 +1262,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_list); static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratnums *r = rule->private; + const struct snd_pcm_hw_constraint_ratnums *r = rule->private; unsigned int num = 0, den = 0; int err; err = snd_interval_ratnum(hw_param_interval(params, rule->var), @@ -1296,7 +1296,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratdens *r = rule->private; + const struct snd_pcm_hw_constraint_ratdens *r = rule->private; unsigned int num = 0, den = 0; int err = snd_interval_ratden(hw_param_interval(params, rule->var), r->nrats, r->rats, &num, &den); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..5cc8c4f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1775,7 +1775,7 @@ const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + const struct snd_pcm_hardware *hw = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), snd_pcm_known_rates.count, snd_pcm_known_rates.list, hw->rates); @@ -1785,7 +1785,7 @@ static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - struct snd_pcm_substream *substream = rule->private; + const struct snd_pcm_substream *substream = rule->private; t.min = 0; t.max = substream->buffer_bytes_max; t.openmin = 0; -- 1.7.10

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[alsa-devel] [PATCH] ALSA: hda - Always call standard unsolicited event
by David Henningsson 05 Jul '12

05 Jul '12

With the model parsers out of the way, we have no custom unsol events to worry about, we can therefore simplify the code. In addition, this fixes a bug on the ASUS TC710, which has only a headphone jack and a mic jack, but no internal mic or speakers. Therefore the unsol_event pointer was not set, and as a result, the jack kcontrols were not correctly updated. BugLink: https://bugs.launchpad.net/bugs/1021192 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Alsa-info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1021192/+attachment/32… sound/pci/hda/patch_realtek.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c2ff945..04e9243 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -174,7 +174,6 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif @@ -688,7 +687,7 @@ static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) } /* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) { int action; @@ -1024,11 +1023,9 @@ static void alc_init_automute(struct hda_codec *codec) spec->automute_lo = spec->automute_lo_possible; spec->automute_speaker = spec->automute_speaker_possible; - if (spec->automute_speaker_possible || spec->automute_lo_possible) { + if (spec->automute_speaker_possible || spec->automute_lo_possible) /* create a control for automute mode */ alc_add_automute_mode_enum(codec); - spec->unsol_event = alc_sku_unsol_event; - } } /* return the position of NID in the list, or -1 if not found */ @@ -1191,7 +1188,6 @@ static void alc_init_auto_mic(struct hda_codec *codec) snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n", ext, fixed, dock); - spec->unsol_event = alc_sku_unsol_event; } /* check the availabilities of auto-mute and auto-mic switches */ @@ -2062,14 +2058,6 @@ static int alc_init(struct hda_codec *codec) return 0; } -static void alc_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct alc_spec *spec = codec->spec; - - if (spec->unsol_event) - spec->unsol_event(codec, res); -} - #ifdef CONFIG_SND_HDA_POWER_SAVE static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -4271,14 +4259,12 @@ static void set_capture_mixer(struct hda_codec *codec) */ static void alc_auto_init_std(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); alc_auto_init_input_src(codec); alc_auto_init_digital(codec); - if (spec->unsol_event) - alc_inithook(codec); + alc_inithook(codec); } /* @@ -4879,7 +4865,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->automute_speaker = 1; spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); - spec->unsol_event = alc_sku_unsol_event; snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs); } } -- 1.7.9.5

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