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

Re: [alsa-devel] [PATCH 05/25] ASoC: pxa: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 10:26 PM, Bill Pemberton <wfp5p(a)virginia.edu> wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> > Cc: linux-arm-kernel(a)lists.infradead.org > --- > sound/soc/pxa/brownstone.c | 6 +++--- > sound/soc/pxa/corgi.c | 6 +++--- > sound/soc/pxa/e740_wm9705.c | 6 +++--- > sound/soc/pxa/e750_wm9705.c | 6 +++--- > sound/soc/pxa/e800_wm9712.c | 6 +++--- > sound/soc/pxa/imote2.c | 6 +++--- > sound/soc/pxa/mioa701_wm9713.c | 6 +++--- > sound/soc/pxa/mmp-pcm.c | 6 +++--- > sound/soc/pxa/mmp-sspa.c | 6 +++--- > sound/soc/pxa/palm27x.c | 4 ++-- > sound/soc/pxa/poodle.c | 6 +++--- > sound/soc/pxa/pxa-ssp.c | 6 +++--- > sound/soc/pxa/pxa2xx-ac97.c | 8 ++++---- > sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- > sound/soc/pxa/pxa2xx-pcm.c | 6 +++--- > sound/soc/pxa/tosa.c | 6 +++--- > sound/soc/pxa/ttc-dkb.c | 6 +++--- > 17 files changed, 50 insertions(+), 50 deletions(-) > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

1 0

[alsa-devel] [patch] usb6fire: prevent driver panic state when stopping
by Jurgen Kramer 07 Dec '12

07 Dec '12

Hi, The patch below prevents the 6fire usb driver going into panic state when stopping playing. On some systems the urb in handler (usb6fire_pcm_in_urb_handler) is being called while urbs are being killed off, this causes the driver to set panic state and can result in the kernel warning 'URB %p submitted while active'. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> diff -uNrp linux/sound/usb/6fire/pcm.c linux.new/sound/usb/6fire/pcm.c --- linux/sound/usb/6fire/pcm.c 2012-11-23 17:01:41.754673597 +0100 +++ linux.new/sound/usb/6fire/pcm.c 2012-12-07 10:09:11.386666744 +0100 @@ -135,6 +135,9 @@ static void usb6fire_pcm_stream_stop(str struct control_runtime *ctrl_rt = rt->chip->control; if (rt->stream_state != STREAM_DISABLED) { + + rt->stream_state = STREAM_STOPPING; + for (i = 0; i < PCM_N_URBS; i++) { usb_kill_urb(&rt->in_urbs[i].instance); usb_kill_urb(&rt->out_urbs[i].instance); Jurgen

2 1

[alsa-devel] M-Audio Audiophile 24/96 (snd-ice1712) lacks ADCs
by Pils Bert 07 Dec '12

07 Dec '12

Hi guys, I have a problem with my M-Audio Audiophile 24/96 soundcard: There are no ADC volume sliders present neither in alsamixer nor in envy24control. But instead there are also DAC sliders on the capture page in alsamixer?! But changing them doesn't have an effect on the H/W input gain. I read on website that some other users had the same problem (but it seams to be a rare problem). First of all my card has been correctly detected as Audiophile 24/96. But in /sys/module/snd_ice1712/parameters/model are only null values. I loaded the module with the model=audiophile parameter, but no ADCs appear. Second I skimmed over the module source code and found the lines case ICE1712_SUBDEVICE_AUDIOPHILE: ice->num_total_dacs = 2; ice->num_total_adcs = 2; break; in __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) in delta.c So the module says there have to be 2 ADCs... I also disabled my onboard soundcard (HDA Intel), but no ADCs appear. Maybe its some hardware configuration problem? Moreover, I had to edit my .asoundrc file in order to play and capture audio anyway (so no 'out of the box' experience here...): pcm.audiophile { type hw card 1 device 0 } pcm.analog { type plug slave.pcm audiophile ttable.0.0 1 # H/W 1 ttable.1.1 1 # H/W 2 } I'm using plain ALSA 1.0.25 (without pulseaudio) on Debian Testing (Linux Rowdy 3.2.0-4-amd64 #1 SMP Debian 3.2.32-1 x86_64 GNU/Linux) Maybe some of you guys have an good idea ;) Best regards, Jannik

4 3

Re: [alsa-devel] [PATCH RESEND V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Fri, Dec 07, 2012 at 01:59:21PM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 07 Dec '12

07 Dec '12

At Thu, 06 Dec 2012 12:45:56 -0500, Bill Pemberton wrote: > > Takashi Iwai writes: > > > > At Thu, 6 Dec 2012 12:35:09 -0500, > > Bill Pemberton wrote: > > > > > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > > > and __devexit so that HOTPLUG can be retired. > > > > > > Bill Pemberton (20): > > > ALSA: pci: remove __dev* attributes > > > ALSA: AACI: remove __dev* attributes > > > ALSA: pxa2xx: remove __dev* attributes > > > ALSA: atmel: remove __dev* attributes > > > ALSA: firewire-speakers: remove __dev* attributes > > > ALSA: mips: remove __dev* attributes > > > ALSA: parisc/harmony: remove __dev* attributes > > > ALSA: oss: remove __dev* attributes > > > ALSA: ad1889: remove __dev* attributes > > > ALSA: bt87X: remove __dev* attributes > > > ALSA: oxygen: remove __dev* attributes > > > > Any reason the three above to be separate patches? > > > > They are separately listed in MAINTAINERS, so I broke them out for the > appropriate Cc:'s. Ah, understood. In general, split isn't needed for a case like this. Most of such trivial cleanup patches are applied at once to the relevant codes in the whole subsystem. In anyway, I applied all patches now. thanks, Takashi

1 0

[alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

2 1

Re: [alsa-devel] [PATCH V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 09:20:36AM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Looks good however this doesn't apply against my topic/samsung branch - can you please check if I need anything from Kukjin's tree? Though I guess the easiest thing is just to wait until the merge window is over, we're near to the next release.

2 1

Re: [alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

Forgot to mention the version 1.0.26 From: Pankaj DEV Sent: Friday, December 07, 2012 12:01 PM To: alsa-devel(a)alsa-project.org Subject: Compilation error for static build of alsa-lib When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

1 0

[alsa-devel] Compilation error for static build of alsa-utils 1.0.26
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/configure +++ b/configure @@ -6247,7 +6247,7 @@ if test "$alsa_prefix" != "" ; then LDFLAGS="$LDFLAGS $ALSA_LIBS" fi -ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread" +ALSA_LIBS="$ALSA_LIBS -lasound -lm -lpthread" LIBS="$ALSA_LIBS $LIBS" { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ALSA_LIBS" >&5 $as_echo "$ALSA_LIBS" >&6; }

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 07 Dec '12

07 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

2 3