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

[alsa-devel] [PATCH] Sony Vaio with stac9872
by 107 27 Feb '12

27 Feb '12

Hello, I have no sound on my Sony Vaio SZ4XRN laptop with newest kernels. Sound: hda-intel, sigmatel - STAC9872 To solve this issue I make this patch for sound/pci/hda/patch_sigmatel.c: --- patch_sigmatel.c.bak 2012-02-23 21:01:28.577515317 +0400 +++ patch_sigmatel.c 2012-02-23 21:53:21.035605216 +0400 @@ -6357,7 +6357,16 @@ */ static const struct hda_verb stac9872_core_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2 /*0x1*/ }, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; Constants got from old kernel (2.6.22). With this patch sound works fine. Thank you.

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Re: [alsa-devel] [PATCH] ALSA: azt3328 - Fix NULL ptr dereference on cards without OPL3
by Takashi Iwai 25 Feb '12

25 Feb '12

At Sat, 25 Feb 2012 16:15:57 +0100, Alban Bedel wrote: > > opl3->private_data was set even if opl3 could not be created. > > Signed-off-by: Alban Bedel <albeu(a)free.fr> Thanks, applied now. Takashi > --- > sound/pci/azt3328.c | 3 +-- > 1 files changed, 1 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c > index 95ffa6a..496f14c 100644 > --- a/sound/pci/azt3328.c > +++ b/sound/pci/azt3328.c > @@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) > err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); > if (err < 0) > goto out_err; > + opl3->private_data = chip; > } > > - opl3->private_data = chip; > - > sprintf(card->longname, "%s at 0x%lx, irq %i", > card->shortname, chip->ctrl_io, chip->irq); > > -- > 1.7.1 >

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Re: [alsa-devel] linux-next: Tree for Feb 24 (sound/usb/6fire)
by Randy Dunlap 25 Feb '12

25 Feb '12

On 02/23/2012 10:05 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120223: on x86_64: ERROR: "_snd_ctl_add_slave" [sound/usb/6fire/snd-usb-6fire.ko] undefined! ERROR: "snd_ctl_make_virtual_master" [sound/usb/6fire/snd-usb-6fire.ko] undefined! Full randconfig file is attached. CONFIG_SND_VMASTER is not enabled. Looks like this driver should either depend on or select SND_VMASTER. -- ~Randy

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[alsa-devel] USB Headset, digital profile problem
by David Henningsson 25 Feb '12

25 Feb '12

Hi, I'm researching I problem I'm seeing here. As you might know PulseAudio tries to open devices in different ways, and so it is trying to open my USB headset (at card index 1) with the device string "iec958:1", and I want this to fail (to make sure this device is not shown in the user interface). So far it seems like a cold boot will cause the issue to disappear (i e, the call fails), whereas a suspend/resume can cause it to appear; but I'm not completely sure this is the actual trigger. When the call fails (that's what I want), the PulseAudio log looks like: D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)confmisc.c: Unable to find definition 'cards.USB-Audio.pcm.iec958.0:CARD=1,AES0=4,AES1=130,AES2=0,AES3=2' I: [pulseaudio] (alsa-lib)conf.c: function snd_func_refer returned error: Filen eller katalogen finns inte I: [pulseaudio] (alsa-lib)conf.c: Evaluate error: Filen eller katalogen finns inte I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM iec958:1 I: [pulseaudio] alsa-util.c: Error opening PCM device iec958:1: Filen eller katalogen finns inte ("Filen eller katalogen finns inte" means "File or directory not found") When the call succeeds, the PulseAudio log looks like: D: [pulseaudio] alsa-util.c: Trying iec958:1 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open iec958:1 I have yet to master the alsa-lib configuration language fully, but at least I'm not changing the files between runs. Do you have an idea of what the problem could be, or how I can debug it a little deeper? For reference, all logs are at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145 With alsa-infos at: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145/+attachmen… https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/940145/+attachmen… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] [PATCH 4/4] ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX
by Timur Tabi 25 Feb '12

25 Feb '12

Sergei Shtylyov wrote: > These functions are not equivalent concerning endianness. You should > probably have used readl_be() instead, else it won't work on PPC anymore. readl_be() is not good enough, either. The in/out_be functions have specific syncronization instructions in them, so that the operations conform to the PowerPC instruction ordering architecture. You probably need to do something like this: #ifdef PPC #define read_ssi(x) in_be32(x) #else #if defined ARM #define read_ssi(x) readl(x) #endif -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] [PATCH] sound/soc-core: Don't overwrite .poweroff in snd_soc_pm_ops
by Mark Brown 24 Feb '12

24 Feb '12

On Fri, Feb 24, 2012 at 04:25:49PM +0530, Viresh Kumar wrote: > SET_SYSTEM_SLEEP_PM_OPS writes .poweroff = *_resume once. Then we overwrite it > again explicitly as .poweroff = snd_soc_poweroff. Even though it works, as the Applied but please do try to use subject lines appropriate for the subsystem you're submitting against. I'd also suggest creating a mailing list for ST internal people if you need to CC so many people on your team, the vger lists have a fairly low limit on the number of CCs they will accept which I think is about twenty or so, you're using a fair proportion of that.

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[alsa-devel] [Q] ASoC: What's the real meaning of SND_SOC_DAIFMT_NB_NF?
by javier Martin 24 Feb '12

24 Feb '12

Hi, I am trying to configure a DAI interface between a i.MX27 SSI and a wm8974 codec. The definition of SND_SOC_DAIFMT_NB_NF specifies 'normal' bit clock and frame for the specified format. http://lxr.linux.no/#linux+v3.2.7/include/sound/soc-dai.h#L48 As I wasn't sure about the meaning of 'normal' I've dug into the code of some platforms to see what they do. What I've found, however, is that it doesn't seem to be an agreement about it. Let me expose some examples: 1. Frame clock is active High: i.MX: http://lxr.linux.no/#linux+v3.2.7/sound/soc/imx/imx-ssi.c#L131 + MCIMX27RM.pdf (p 1606) (default) PXA: http://lxr.linux.no/#linux+v3.2.7/sound/soc/pxa/pxa-ssp.c#L498 pxa-ssp.c + PXA3xx_DM_Vol_IV.pdf (p 332) (not default) 2. Frame clock is active Low: OMAP: http://lxr.linux.no/#linux+v3.2.7/sound/soc/omap/omap-mcbsp.c#L478 + sprugn4h.pdf (p 3165) (not default) Some chips like s6000 seem to provide a more original use for this flag: http://lxr.linux.no/#linux+v3.2.7/sound/soc/s6000/s6000-i2s.c#L255 Similar behavior has been found for bit clock. ALso, it seems that there is no possibility to specify different configurations for RX and TX parts. -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[PATCH] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 24 Feb '12

24 Feb '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, mute the right channel and make sure they can't unmute it. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 41 +++++++++++++++++++++++++++++---------- 1 files changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 266e5a6..4a05817 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -140,6 +140,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4044,7 +4045,7 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -4056,7 +4057,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4074,7 +4075,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4152,14 +4153,22 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + int chs = 3; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (spec->fixup_stereo_dmic && (strcmp("Internal Mic", label) == 0)) { + /* Make left channel controllable and right channel muted */ + chs = 1; + snd_hda_codec_write_cache(codec, adc_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT | (idx << AC_AMP_SET_INDEX_SHIFT) | AC_AMP_MUTE); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, chs); } return 0; } @@ -4335,22 +4344,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4360,10 +4377,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4387,7 +4406,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); err = cx_auto_search_adcs(codec); if (err < 0) -- 1.7.9 --------------080406020609030400030200--

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Re: [alsa-devel] Problem with "set_period_size"
by Clemens Ladisch 24 Feb '12

24 Feb '12

Alicia Romero wrote: > I forgot to write that at the end I activate all the params using: > > /* Activate the parameters */ > result = snd_pcm_hw_params (stream->pb_pcm, params); > if (result < 0) { If set_period_size fails, params has not been changed. Regards, Clemens

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[alsa-devel] [PATCH] hda - add id for Atom Cedar Trail HDMI codec
by Wu Fengguang 24 Feb '12

24 Feb '12

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) --- sound.orig/sound/pci/hda/patch_hdmi.c 2012-02-23 14:51:19.562540743 +0800 +++ sound/sound/pci/hda/patch_hdmi.c 2012-02-23 15:05:51.774522844 +0800 @@ -1913,6 +1913,7 @@ static const struct hda_codec_preset snd { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1959,6 +1960,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862804" MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); +MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("HDMI HD-audio codec");

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