April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] linux-next - strange audio bug
by Valdis Kletnieks 04 Apr '15

04 Apr '15

Next-20150310 works OK, next-20150324 has the bug. Dell Latitude E6530 laptop, lspci tells me: 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1) Subsystem: Dell Device 0535 ... Kernel driver in use: snd_hda_intel Symptoms: Audio works perfectly at boot, and for some amount of time after. But somewhere around 6-8 hours of uptime, it gets into a strange state where I'll get no audio... and then it will cut in for all of 2 seconds or so (fairly consistent, that part), and then cut back out for anywhere from 10 seconds to a minute, then another 2 seconds of sound.. lather rinse repeat. Before I start the thankless task of bisecting this (which means I'll be able to do only 1, *maybe* 2 steps a day), does this ring any bells?

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[alsa-devel] [PATCH] ALSA: hda: fix possible null dereference
by Sudip Mukherjee 04 Apr '15

04 Apr '15

we are dereferencing pcm first then checking pcm. instead now lets put them in same if condition so that pcm is checked first. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/hda/hda_codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 145cae7..7df18e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3717,9 +3717,8 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (pcm->own_chmap) - continue; - if (!pcm || !hinfo->substreams) + if (!pcm || pcm->own_chmap || + !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: hda - Fix subsystem ID read regression
by David Henningsson 04 Apr '15

04 Apr '15

A regression was introduced in 7639a06c23c7d4cda3: if AC_PAR_SUBSYSTEM_ID reads as zero, one should retry using AC_VERB_GET_SUBSYSTEM_ID. This seems to hit many codecs (my own laptop included), and causes quirks for some machines not to apply correctly. Reported-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6e8ee1d..53b6b95 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -91,7 +91,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->power_caps = snd_hdac_read_parm(codec, fg, AC_PAR_POWER_STATE); /* reread ssid if not set by parameter */ - if (codec->subsystem_id == -1) + if (codec->subsystem_id == -1 || codec->subsystem_id == 0) snd_hdac_read(codec, fg, AC_VERB_GET_SUBSYSTEM_ID, 0, &codec->subsystem_id); -- 1.9.1

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Re: [alsa-devel] [RFC] ALSA: usb-audio: reduce "cannot get freq at ep" spew
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:36:28 +0000, Eric Wong wrote: > > Eric Wong <normalperson(a)yhbt.net> wrote: > > I also had another generic patch prepared before I noticed Joe's > > quirk addition for the MS Lifecam HD-5000. > > This is the generic patch I prepared before I noticed Joe's quirk > addition (applies cleanly to 3.19 and 4.0-rc6): > > ------------------------------8<--------------------------- > Subject: [PATCH] ALSA: usb-audio: reduce "cannot get freq at ep" spew > > If a device fails to support reading the sample rate, it will likely > fail again and there is no point in logging the message every time the > sample rate is set. > > This reduces dmesg noise when using the Benchmark DAC1 PRE as a USB > sound card. Hm, this shuts up all the rest for the device, so it also hides if the message comes from multiple EPs. Takashi > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > --- > sound/usb/clock.c | 7 +++++-- > sound/usb/usbaudio.h | 3 ++- > 2 files changed, 7 insertions(+), 3 deletions(-) > > diff --git a/sound/usb/clock.c b/sound/usb/clock.c > index 03fed66..512fe12 100644 > --- a/sound/usb/clock.c > +++ b/sound/usb/clock.c > @@ -307,8 +307,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, > USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, > UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, > data, sizeof(data))) < 0) { > - dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > - iface, fmt->altsetting, ep); > + if (!chip->no_get_freq) { > + dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > + iface, fmt->altsetting, ep); > + chip->no_get_freq = 1; > + } > return 0; /* some devices don't support reading */ > } > > diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h > index 91d0380..be65671 100644 > --- a/sound/usb/usbaudio.h > +++ b/sound/usb/usbaudio.h > @@ -43,7 +43,8 @@ struct snd_usb_audio { > unsigned int in_pm:1; > unsigned int autosuspended:1; > unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ > - > + unsigned int no_get_freq:1; > + > int num_interfaces; > int num_suspended_intf; > > -- > EW >

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Re: [alsa-devel] [PATCH] ALSA: usb-audio: don't try to get Benchmark DAC1 sample rate
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:34:05 +0000, Eric Wong wrote: > > Adding this quirk allows us to avoid the noisy > "cannot get freq at ep 0x1" message in dmesg output every time > playback starts. > > This ought to affect other Benchmark DAC1 variations using the same > "Microchip Technology, Inc." chip as well, but I have only tested > with the "Pre" variant. > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > Cc: Joe Turner <joe(a)oampo.co.uk> > Cc: Takashi Iwai <tiwai(a)suse.de> Applied, thanks. Takashi > --- > I also had another generic patch prepared before I noticed Joe's > quirk addition for the MS Lifecam HD-5000. > > sound/usb/quirks.c | 9 +++++++-- > 1 file changed, 7 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 753a47d..9a28365 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > { > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > - return chip->usb_id == USB_ID(0x045E, 0x076D); > + /* devices which do not support reading the sample rate. */ > + switch (chip->usb_id) { > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > + return true; > + } > + return false; > } > > /* Marantz/Denon USB DACs need a vendor cmd to switch > -- > EW >

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[alsa-devel] Support Dell headset mode for ALC256
by Kailang 04 Apr '15

04 Apr '15

Hi Takashi, Dell has new platform with ALC256. Attach patch is support headset mode for ALC256. Thanks. BR, Kailang

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Re: [alsa-devel] [PATCH 76/86] sound/pci: use uapi/linux/pci_ids.h directly
by Takashi Iwai 04 Apr '15

04 Apr '15

At Sun, 29 Mar 2015 15:42:59 +0200, Michael S. Tsirkin wrote: > > Header moved from linux/pci_ids.h to uapi/linux/pci_ids.h, > use the new header directly so we can drop > the wrapper in include/linux/pci_ids.h. > > Signed-off-by: Michael S. Tsirkin <mst(a)redhat.com> Do we need this change in the driver code at all...? I thought include/uapi is already included in the include path. Takashi > --- > sound/pci/ctxfi/xfi.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c > index f2f3277..7e26876 100644 > --- a/sound/pci/ctxfi/xfi.c > +++ b/sound/pci/ctxfi/xfi.c > @@ -11,7 +11,7 @@ > #include <linux/init.h> > #include <linux/pci.h> > #include <linux/moduleparam.h> > -#include <linux/pci_ids.h> > +#include <uapi/linux/pci_ids.h> > #include <linux/module.h> > #include <sound/core.h> > #include <sound/initval.h> > -- > MST >

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[alsa-devel] [PATCH v1.1 07/14] media: omap3isp: remove unused clkdev
by Laurent Pinchart 03 Apr '15

03 Apr '15

From: Russell King <rmk+kernel(a)arm.linux.org.uk> No merged platform supplies xclks via platform data. As we want to slightly change the clkdev interface, rather than fixing this unused code, remove it instead. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Signed-off-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> --- drivers/media/platform/omap3isp/isp.c | 24 ------------------------ drivers/media/platform/omap3isp/isp.h | 1 - include/media/omap3isp.h | 6 ------ 3 files changed, 31 deletions(-) In case this would be helpful, I've rebased the original patch on top of the linuxtv master branch scheduled for merge in v4.1. diff --git a/drivers/media/platform/omap3isp/isp.c b/drivers/media/platform/omap3isp/isp.c index ff51c4f..18d0a87 100644 --- a/drivers/media/platform/omap3isp/isp.c +++ b/drivers/media/platform/omap3isp/isp.c @@ -304,7 +304,6 @@ static struct clk *isp_xclk_src_get(struct of_phandle_args *clkspec, void *data) static int isp_xclk_init(struct isp_device *isp) { - struct isp_platform_data *pdata = isp->pdata; struct device_node *np = isp->dev->of_node; struct clk_init_data init; unsigned int i; @@ -335,26 +334,6 @@ static int isp_xclk_init(struct isp_device *isp) xclk->clk = clk_register(NULL, &xclk->hw); if (IS_ERR(xclk->clk)) return PTR_ERR(xclk->clk); - - /* When instantiated from DT we don't need to register clock - * aliases. - */ - if (np) - continue; - - if (!pdata || (pdata->xclks[i].con_id == NULL && - pdata->xclks[i].dev_id == NULL)) - continue; - - xclk->lookup = kzalloc(sizeof(*xclk->lookup), GFP_KERNEL); - if (xclk->lookup == NULL) - return -ENOMEM; - - xclk->lookup->con_id = pdata->xclks[i].con_id; - xclk->lookup->dev_id = pdata->xclks[i].dev_id; - xclk->lookup->clk = xclk->clk; - - clkdev_add(xclk->lookup); } if (np) @@ -376,9 +355,6 @@ static void isp_xclk_cleanup(struct isp_device *isp) if (!IS_ERR(xclk->clk)) clk_unregister(xclk->clk); - - if (xclk->lookup) - clkdev_drop(xclk->lookup); } } diff --git a/drivers/media/platform/omap3isp/isp.h b/drivers/media/platform/omap3isp/isp.h index 431224e..e579943 100644 --- a/drivers/media/platform/omap3isp/isp.h +++ b/drivers/media/platform/omap3isp/isp.h @@ -132,7 +132,6 @@ enum isp_xclk_id { struct isp_xclk { struct isp_device *isp; struct clk_hw hw; - struct clk_lookup *lookup; struct clk *clk; enum isp_xclk_id id; diff --git a/include/media/omap3isp.h b/include/media/omap3isp.h index 0f0c08b..048f8f9 100644 --- a/include/media/omap3isp.h +++ b/include/media/omap3isp.h @@ -150,13 +150,7 @@ struct isp_platform_subdev { struct isp_bus_cfg *bus; }; -struct isp_platform_xclk { - const char *dev_id; - const char *con_id; -}; - struct isp_platform_data { - struct isp_platform_xclk xclks[2]; struct isp_platform_subdev *subdevs; void (*set_constraints)(struct isp_device *isp, bool enable); }; -- Regards, Laurent Pinchart

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Re: [alsa-devel] [PATCH 1/6] ASoC: jz4740: Remove Makefile entry for removed file
by Greg Kroah-Hartman 03 Apr '15

03 Apr '15

On Fri, Mar 27, 2015 at 10:51:43AM +0000, Markos Chandras wrote: > On 04/22/2014 09:46 PM, Lars-Peter Clausen wrote: > > Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > jz4740-pcm.c file, but neglected to remove the Makefile entries. > > > > Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > Reported-by: kbuild test robot <fengguang.wu(a)intel.com> > > Reported-by: Ralf Baechle <ralf(a)linux-mips.org> > > Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > --- > > sound/soc/jz4740/Makefile | 2 -- > > 1 file changed, 2 deletions(-) > > > > diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile > > index be873c1..d32c540 100644 > > --- a/sound/soc/jz4740/Makefile > > +++ b/sound/soc/jz4740/Makefile > > @@ -1,10 +1,8 @@ > > # > > # Jz4740 Platform Support > > # > > -snd-soc-jz4740-objs := jz4740-pcm.o > > snd-soc-jz4740-i2s-objs := jz4740-i2s.o > > > > -obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o > > obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o > > > > # Jz4740 Machine Support > > > Hi, > > This patch (eebdec044e82) is present in 3.15-rc1 but the build failure > that was introduced by 0406a40a0 ("ASoC: jz4740: Use the generic > dmaengine PCM driver") is present in 3.14-rc1 so 3.14 is still broken. > > Greg, would it be possible to cherry pick eebdec044e82 ("ASoC: jz4740: > Remove Makefile entry for removed file") to 3.14 stable branch? It seems > to apply without conflicts. Now applied, thanks. greg k-h

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[alsa-devel] dummy codec + simple card combination
by Belisko Marek 03 Apr '15

03 Apr '15

Hi, I'm trying to add audio support for gta04 (for gsm + bt and fm audio). All audio sources are clock masters so basically connection is dummy codecs -> cpu dai (mcbsp ports). For implementing such thing I can see 2 options: 1. simple card driver which can easily describe cpu dai interface but dummy codec cannot because there are no DT bindings (there was sever attempts to add DT bindings for dummy-codec but AFAIK none was accepted) This turns it to only solution to create gta04 soundcard which will get from DT cpu dai links and use in card driver dummy codecs directly. Is my assumption correct? I would like to get some feedback before wasting my time with implementing something which cannot be pushed mainline. Thanks for all suggestions. BR, marek -- as simple and primitive as possible ------------------------------------------------- Marek Belisko - OPEN-NANDRA Freelance Developer Ruska Nova Ves 219 | Presov, 08005 Slovak Republic Tel: +421 915 052 184 skype: marekwhite twitter: #opennandra web: http://open-nandra.com

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