January 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Takashi Iwai 19 Jan '13

19 Jan '13

At Thu, 17 Jan 2013 21:40:43 +0100, Manolo Díaz wrote: > > El jue, 17 ene 2013 a las 21:16 horas > Takashi Iwai escribió: > > >At Thu, 17 Jan 2013 19:12:24 +0100, > >Manolo Díaz wrote: > >> > >> El jue, 17 ene 2013 a las 16:32 horas > >> Takashi Iwai escribió: > >> > >> >At Thu, 17 Jan 2013 09:59:20 -0500, > >> >Miro Hodak wrote: > >> >> > >> >> On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> >> > >> >> > > >> >> > > > >> >> > > OK, now as far as Auto-Mute goes, should it do what expected when > >> >> > enabled - i.e. muting the speakers when headphones are connected to the > >> >> > front panel? I have tried that, does not seem to work. > >> >> > > > >> >> > > >> >> > Do your computer chassis have a hda front audio panel with Jack detection ? > >> >> > > >> >> > The front headphone Jack should return true when headphone Jack is plugged > >> >> > > >> >> > control.35 { > >> >> > iface CARD > >> >> > name 'Front Headphone Jack' > >> >> > > >> >> > value false > >> >> > comment { > >> >> > access read > >> >> > type BOOLEAN > >> >> > count 1 > >> >> > } > >> >> > } > >> >> > > >> >> > All ADI codecs support impedance measurement > >> >> > > >> >> > Try the modified hda-jack-sense-test which need hda-analyzer > >> >> > > >> >> > http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115 > >> >> > > >> >> > >> >> Hmmm, ran into some python issues. Even hda-analyzer itself is not working: > >> >> ... > >> >> File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile > >> >> raise ValueError, "wrong proc file format (%s)" % msg > >> >> ValueError: wrong proc file format (integer decode 'State of AFG node > >> >> 0x01:' ('Node ')) > >> > > >> >Ah, it's a new proc file entry that has been added recently. > >> >hda-analyzer needs to be fixed. > >> > > >> >In anyway, only for checking the presence unsolicited events, you can > >> >do without extra program. Suppose you built your kernel with the > >> >tracing support. See "Tracepoints" option in > >> >Documentation/sound/alsa/HD-Audio.txt. > >> > > >> > # echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable > >> > > >> > (plug and unplug your headphone jacks) > >> > > >> > # cat /sys/kernel/debug/tracing/trace > >> > > >> >If you see some hda_unsol_event lines there, it implies that the jack > >> >detection basically works in hardware level. > >> > > >> > > >> >Takashi > >> > >> I Suppose that when auto-mute is enabled an HP independent is disabled, > >> inserting the HP jack should disable sound from front speakers. If so > >> then auto-mute doesn't work. Jack insertion/extraction events are well > >> detected for front speakers and HP. > >> > >> #~ cat /sys/kernel/debug/tracing/trace > >> # tracer: nop > >> # > >> # entries-in-buffer/entries-written: 4/4 #P:2 > >> # > >> # _-----=> irqs-off > >> # / _----=> need-resched > >> # | / _---=> hardirq/softirq > >> # || / _--=> preempt-depth > >> # ||| / delay > >> # TASK-PID CPU# |||| TIMESTAMP FUNCTION > >> # | | | |||| | | > >> <idle>-0 [001] d.h. 137.881919: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 162.511102: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 246.529813: hda_unsol_event: [0] res=4000000, res_ex=10 > >> <idle>-0 [001] d.h. 272.381719: hda_unsol_event: [0] res=4000000, res_ex=10 > >> > >> branch tested: test/hda-gen-parser > > > >Please test with either master or test/hda-migrate branch. > >test/hda-gen-parser branch doesn't contain the patch for AD codecs. > > > >In anyway, please attach alsa-info.sh outputs before and after > >plugging the headphone with the latest sound-unstable tree (and a > >proper branch). > > > > > >thanks, > > > >Takashi > > branch tested: master > > commit d89012bef4ad6d33c1050dd2924f010f5072ff52 > Merge: ad92fbe 36c9db7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Jan 17 17:47:19 2013 +0100 > > Just the same: auto-mute doesn't work. > > alsa-info output show no differences (attached). Thanks. It turned out that simply missing unsol_event handlers in the codec driver code. The patch below should fix the issue. It's already merged in test/hda-migrate and master branches. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix missing unsol event handler in some codec drivers This resulted in non-working auto-mute behavior, of course... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_ca0110.c | 1 + sound/pci/hda/patch_cmedia.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 02fe0d1..6feaec4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -595,6 +595,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = ad198x_free, + .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, .suspend = ad198x_suspend, diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 8d09325..db7635c 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -36,6 +36,7 @@ static const struct hda_codec_ops ca0110_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int ca0110_parse_auto_config(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 04dd3b6..087cabb 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -576,6 +576,7 @@ static const struct hda_codec_ops cmi_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int cmi_parse_auto_config(struct hda_codec *codec) -- 1.8.1

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Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Raymond Yau 19 Jan '13

19 Jan '13

> > > Testing latest updates in sound-unstable: > > commit f741c8a1f649379f8f3eb81119df2676c46075cf > Merge: d821c1e 164a7ad > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Fri Jan 18 18:30:00 2013 +0100 > > Playback and capture appear to work correctly with one possible caveats: > > Item labeled "Front" controls the level of sound through the speakers. I would expect that it would only control sound from front HP. This happens whether Independent HP is on and off. > Is it feasible to add a hint which disable the creation of virtual master playback volume and switch ?

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[alsa-devel] [PATCH 0/3] You guessed it, it's generic parser patches
by David Henningsson 18 Jan '13

18 Jan '13

The third one seems to apply to the regular sound tree too. David Henningsson (3): ALSA: hda - make sure there are enough input labels and paths ALSA: hda - don't compare with yourself in fill_input_pin_labels ALSA: hda - Don't add unnecessary indices on HDMI and SPDIF sound/pci/hda/hda_auto_parser.c | 22 ++++++++++++---------- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_generic.h | 6 +++--- 3 files changed, 17 insertions(+), 15 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Fix mute led for another HP machine
by David Henningsson 18 Jan '13

18 Jan '13

This machine also has the "HP_Mute_LED_0_A" string in DMI information. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1096789 Tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f519627..6369b23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6251,6 +6251,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1977, "HP Pavilion 14", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

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[alsa-devel] [PATCH 0/4] Fixes for missing boost controls in generic parser
by Takashi Iwai 18 Jan '13

18 Jan '13

Hi, this is a series of patches to fix the missing boost controls in the generic parser. I found the problem not only on AD1988 but some others, and while debugging this issue, more issues have been revealed, thus it became more volumes than expected. All patches are found in sound-unstable tree test/hda-gen-parser branch. And, now both test/hda-gen-parser branch and test/hda-migrate branch become identical. It doesn't matter which branch you take. Also, it's already merged in master branch of sound-unstable tree. So.. at this point, it'd be helpful if anyone can test this tree, and let me know any issues. thanks, Takashi

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Re: [alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Hebbar, Gururaja 18 Jan '13

18 Jan '13

On Fri, Jan 18, 2013 at 14:13:40, Michal Bachraty wrote: > On Friday, January 18, 2013 08:22:53 Hebbar, Gururaja wrote: > > On Tue, Jan 15, 2013 at 20:11:00, Michal Bachraty wrote: > > > Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> > > > > Even simple patches need a simple changelog. > > Add a brief description about what & why it is done? > > > > Hi Gurujaja, > > > > - u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. > (rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in > 4 bits (max value 0x7)and then masks them . That data are sended to i2s bus. > For 24 bit or 20 bit or other data formats, this code rotates data badly and > you hear somethink like noise. You need to use > > > + u32 rotate = (word_length / 4) & 0x7; > to proper data rotation. Correct. Now can you add the above info into the patch and re-submit. That way when users sees this patch, they will get all the info needed. > > Regards, > Michal > > Regards, Gururaja

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
by Chris Rattray 18 Jan '13

18 Jan '13

Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9a3f220..56bd6ba 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -103,6 +103,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -817,12 +824,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CONTROL_31, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d7..cb8871a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Michal Bachraty 18 Jan '13

18 Jan '13

Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 13f8d73..a03bece 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -626,7 +626,7 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (32 - word_length) / 4; + u32 rotate = (word_length / 4) & 0x7; u32 mask = (1ULL << word_length) - 1; /* -- 1.7.9.5

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[alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Miro Hodak 18 Jan '13

18 Jan '13

Commit identified in the subject (ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989) causes problem with audio on at least two Asus boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop). The symptoms: 1. XFCE and Gnome2 audio mixers crash whenever sound is started or stopped. The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed. 2. On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are misconfigured since muting the Master channel does not turn the sound off. Either Master is mislabeled or it does not work correctly. Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit fixes these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box. We do not understand enough of alsa in kernel to propose a patch, but hope that we have provided enough information that it will be easy to write. We are happy to provide more information and test any proposed patches. Alsa-info for both boards can be found below. Best regards, Miro Hodak and Manolo Diaz Results of alsa-info Asus Rampage II Gene: Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW Asus M3A79-T DELUXE Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM

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[alsa-devel] Problems capturing and playback via the loopback device
by Erik Kazandjian 18 Jan '13

18 Jan '13

Hi, I'm new to alsa and I'm trying to do the following. I have configured my rythm box to use the loopback device. I've written a little program to capture everything that comes on the other side of the loopback and save that into a file. If I play a song on rythm box I see actual packets on the couterside of the loopback however I save these packets in a file and try to play them back using my sound card I hear no sound. I use the following .asoundrc pcm.loopplayback { type hw card 1 device 1 subdevice 0 } pcm.loopcapture { type hw card 1 device 0 subdevice 0 } pcm.play48 { type plug slave { pcm loopplayback rate 44100 channels 2 } } attached is my test program to do the capturing. Anyone has any idea what I'm doing wrong ? Kind regards Erik

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