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

[alsa-devel] [RESUBMIT PATCH] ASoC: SAMSUNG: Change Kconfig to support all SAMSUNG ASoC
by Sangsu Park 28 Aug '12

28 Aug '12

All SAMSUNG ASoC needs SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all SAMSUNG ASoC. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..e7b8317 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on PLAT_SAMSUNG select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1 On 08/28/12 8:00, Kukjin Kim wrote: > On 08/27/12 11:08, Mark Brown wrote: > > On Mon, Aug 27, 2012 at 11:06:47AM -0700, Kukjin Kim wrote: > > > >> Hmm, all of Samsung stuff is available...so how about? > >> > >> + depends on PLAT_SAMSUNG > > > > Yes, that's a very good idea! > > Mark, thanks. > > Sangsu, could you please re-submit as per my suggestion? > > Thanks. > Kukjin, thanks. This is re-submit patch. In fact, I'd wanted to submit your idea :) Thanks. > Best regards, > Kgene. > -- > Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung > Electronics Co., Ltd.

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[alsa-devel] [PATCH] sound: tegra_alc5632: remove HP detect GPIO inversion
by Stephen Warren 28 Aug '12

28 Aug '12

From: Stephen Warren <swarren(a)nvidia.com> Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Leon Romanovsky <leon(a)leon.nu> Cc: <stable(a)vger.kernel.org> # v3.4 v3.5 --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e463529..76cb1b3 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { -- 1.7.9.5

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Re: [alsa-devel] [PATCH v2 02/14] MFD: twl4030-audio: Convert to use devm_kzalloc
by Mark Brown 28 Aug '12

28 Aug '12

On Tue, Aug 28, 2012 at 12:50:10PM +0300, Tero Kristo wrote: > On Tue, 2012-08-14 at 17:22 +0300, Peter Ujfalusi wrote: > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > I think this one could use a short commit message, also about why > kfree():s are dropped (handled internally by devm_* etc.) Honestly I think people should be able to figure this out, it's not like it's an obscure API nobody ever used before.

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[alsa-devel] [PATCH] ALSA: hda: Make core jack usage unconditional
by Mark Brown 28 Aug '12

28 Aug '12

Currently the HDA drivers have a Kconfig variable SND_HDA_INPUT_JACK which disables jack reporting via the ALSA core jack API. This is bad since (modulo the current problems with the HDA-specific reporting API) it means that all reporting mechanisms will get disabled, not just input. It's also not good that this is open coded in an individual driver - if it's useful to have this control it's going to be equally useful for any driver with jack detection support so should be handled in the core. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/hda/Kconfig | 9 +-------- sound/pci/hda/hda_jack.c | 8 -------- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 194d625..2c7f626 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -2,6 +2,7 @@ menuconfig SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND select SND_KCTL_JACK help Say Y here to include support for Intel "High Definition @@ -62,14 +63,6 @@ config SND_HDA_INPUT_BEEP_MODE Set 1 to always enable the digital beep interface for HD-audio by default. -config SND_HDA_INPUT_JACK - bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND - select SND_JACK - help - Say Y here to enable the jack plugging notification via - input layer. - config SND_HDA_PATCH_LOADER bool "Support initialization patch loading for HD-audio" depends on EXPERIMENTAL diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc02..0760d75 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -109,7 +109,6 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); void snd_hda_jack_tbl_clear(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && codec->jacktbl.list) { struct hda_jack_tbl *jack = codec->jacktbl.list; @@ -119,7 +118,6 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) snd_device_free(codec->bus->card, jack->jack); } } -#endif snd_array_free(&codec->jacktbl); } @@ -225,16 +223,13 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, state ? jack->type : 0); -#endif } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); -#ifdef CONFIG_SND_HDA_INPUT_JACK /* guess the jack type from the pin-config */ static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { @@ -261,7 +256,6 @@ static void hda_free_jack_priv(struct snd_jack *jack) jacks->nid = 0; jacks->jack = NULL; } -#endif /** * snd_hda_jack_add_kctl - Add a kctl for the given pin @@ -292,7 +286,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); err = snd_jack_new(codec->bus->card, name, jack->type, @@ -303,7 +296,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->jack->private_free = hda_free_jack_priv; snd_jack_report(jack->jack, state ? jack->type : 0); } -#endif return 0; } -- 1.7.10.4

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[alsa-devel] ASoC: SAMSUNG: Change Kconfig to support all EXYNOSs
by 박상수 27 Aug '12

27 Aug '12

All EXYNOSs use SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all EXYNOSs like EXYNOS4 and EXYNOS5. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..9e64ae6 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1

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[alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 27 Aug '12

27 Aug '12

Hello! I have been working on prototypes for the ASoC OMAP HDMI audio driver to propagate events from the HDMI output (e.g., display getting enabled/disabled/suspended). This for the users of the driver to react to such events. For instance, if the display is disabled or disconected, audio could be stopped, rerouted or whatever other decision the user makes. This is needed because, if, for instance, the HDMI IP goes off, audio will stall and the audio users will only see a "playback write error (DMA or IRQ trouble?)" In my prototypes I have used snd_soc_jack for this purpose and I have some questions: *I see snd_soc_jack is used mostly for headsets and microphones with actual external mechanical connections. Strictly, in my case I propagate events originated by the OMAP display driver (changes in the power state), and not from external events. Some of these events are generated from an actual HDMI cable connection/disconnection, though. *Maybe the event should be propagated by omapdss/omapdrm/drm and the entity in charge of the audio policy should listen those events instead. *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is feasible for an audio driver to report events from an AV output. I was wondering about how much sense does it make to you guys use a snd_soc_jack in this case? Thanks in advance for your comments! Ricardo

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[alsa-devel] Kernel module build workflow (Re: Logitech USB headset not working in 3.6-rc3)
by Takashi Iwai 27 Aug '12

27 Aug '12

At Sat, 25 Aug 2012 07:16:21 -0500, Bruno Wolff III wrote: > > On Sat, Aug 25, 2012 at 08:13:27 -0400, > Josh Boyer <jwboyer(a)redhat.com> wrote: > >On Sat, Aug 25, 2012 at 07:07:40AM -0500, Bruno Wolff III wrote: > >> On Sat, Aug 25, 2012 at 14:02:51 +0200, > >> Daniel Mack <zonque(a)gmail.com> wrote: > >> > > >> >Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix > >> >scheduling-while-atomic bug in PCM capture stream") and see if that > >> > >> I can try that, but it takes a long time to build a new kernel on my > >> old hardware. > > > >I started a scratch build of a kernel with that patch reverted (via > >patch -R). You'll find it here when it finishes: > > > >http://koji.fedoraproject.org/koji/taskinfo?taskID=4421817 > > > >josh > > I'll test it when it finishes. Thanks. It takes me a long time to build > kernels locally, and I wasn't sure how to do a scratch build on koji without > committing stuff remotely. Actually you don't have to rebuild the whole kernel at all just for debugging this usb-audio bug. The workflow below is what I do often for debugging a driver problem on a distro kernel. 1. Prepare the compile-ready kernel source tree corresponding to your running kernel. This may depend on distro; in the case of SUSE, kernel-default-devel.rpm must be installed for kernel-default.rpm, for example. 2. Copy sound/usb subdirectory of the kernel source tree locally anywhere else, say, $HOME/usb. It can be a normal user directory. 3. Apply the patch to that local directory. 4. Build modules as a normal user like % make -C /lib/modules/$(uname -r)/build M=$HOME/usb modules 5. Make an extra update directory in the module directory once, e.g. # mkdir /lib/modules/$(uname -r)/updates 6. Copy *.ko files to there # cp $HOME/usb/*.ko /lib/modules/$(uname -r)/updates 7. Run depmod once # /sbin/depmod -a 8. Reload snd-usb-audio module. If you want to take back to the original module, just remove updates module directory and run depmod again. Pretty convenient, eh? HTH, Takashi

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Re: [alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Use module_spi_driver
by Mark Brown 27 Aug '12

27 Aug '12

On Mon, Aug 27, 2012 at 05:00:26PM +0530, Sachin Kamat wrote: > module_spi_driver makes the code simpler by eliminating > module_init and module_exit calls. Applied, thanks.

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Re: [alsa-devel] [PATCH - pcxhr 1/1] ALSA: pcxhr: Add 8 new sound cards
by Takashi Iwai 27 Aug '12

27 Aug '12

At Mon, 27 Aug 2012 09:03:55 +0200, Markus Bollinger wrote: > > Hi Takashi, > > sorry, I had problems to get git-send-email working, so I tried it on > another way, but it seems that was not the right one.... > Please find the patch attached. Generated from alsa.kmirror.git. Thanks, applied now. Takashi

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Re: [alsa-devel] [pulseaudio-discuss] Documentation for how rewinding works
by Raymond Yau 27 Aug '12

27 Aug '12

2012-8-19 下午7:17 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > Hi, > > While reviewing a rewind related patch, I noticed that I don't > understand the rewinding code thoroughly, even though I have studied it > a few times in the past. I decided that I should write a wiki page for > the code so that working with it would be easier in the future for me > and everybody else. > > So, here's the result: > http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer… > > While doing that, I found quite a few bugs, which are noted on that > page. I don't plan fixing them any time soon (help with that would be > very welcome), so I will just file bugs about the issues so that they > are not forgotten. > it seem that the alsa sink is more complicated than your figure. Refer to figure 14 HD Audio DMA and buffering , section 4.6 Energy Efficient HA audio mechanism most sound cards have fixed size FIFO buffer instead of dynamic FIFO and minimum transfer size(brust) similar to hda controller which snd_pcm_rewind cannot rewind any audio data in those first in first out buffer. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa?i… 1) do the server really need to rewind when there is only one pulse client ? 2) does pulseaudio add this rewind safeguard delay to the latency ?

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