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

[alsa-devel] [PATCH] ASoC: sigmadsp: Fix endianness conversion issue
by Lars-Peter Clausen 10 Dec '12

10 Dec '12

The 'addr' field of the sigma_action struct is stored as big endian in the firmware file. Cc: stable(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sigmadsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf..4068f24 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware); static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { - return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), sa->payload, len - 2); } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 10 Dec '12

10 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

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[alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj 09 Dec '12

09 Dec '12

can anyone help me to figure out the concepts between period_size ,buffer_size and boundary size? n * period_size = buffer_size? m * buffer_size = boundary_size? and how the dma buffer mapped to the alsa buffer? thank you . br

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Re: [alsa-devel] [PATCH 00/25] ASoC: remove __dev* attributes for retirement of HOTPLUG
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 09:26:12AM -0500, Bill Pemberton wrote: > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. Applied all, thanks.

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[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by Joe Cooper 09 Dec '12

09 Dec '12

On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash. This effects every kernel I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the console: ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22 ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 Blacklisting this module allows the system to function, but with most of its speakers not working. It also prevents use of other USB audio devices, making multitrack recording impossible. I have a ticket about it in the Fedora bug tracker, but was referred to alsa-devel. That ticket links to other discussions of this bug, as well. Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 Please let me know if you need further details, or if there's anything else I can do to help troubleshoot this.

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[alsa-devel] [FT C400,PATCH RFC,v4 00/10] M-Audio Fast Track C400
by Eldad Zack 09 Dec '12

09 Dec '12

Hi, The following patches adds support for the M-Audio Fast Track C400. This is version 4 of the patch series. * Incoroprated the feedback from Clemens Ladisch <clemens(a)ladisch.de>: Moved the quirks structure to the right place (also in other places, though not strictly required) and removed the "set quirks" log messages. Thanks, Clemens! * Added a patch to address the regression introduced by commit 947d299686aa9cc8aecf749d54e8475c6e498956 , "ALSA: snd-usb: properly initialize the sync endpoint". I believe there are no other devices that affected by this issue. Opening only the playback stream without first opening the capture stream now works with the C400. v3: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/057477.h… v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/056750.h… This series was tested with the latest mainline tree, 3.7-rc7 (HEAD e23739b4ade80a3a7f87198f008f6c44a7cbc9fd ) Also applies against current sound-unstable (not tested). (HEAD 900b0690ee7b2a227c5db5f1642ceb67d90c49fc ) * Everything works: mixer controls, effects control, playback and capture (with the correct sampling frequency set). Note that the device doesn't really have a master control or left/right pan. It's just "emulated" in the software. * To deal with the mixer channel IDs, limited by cmask bit length, I introduced an offset value. This allows to get rid of the shorehorning of the IDs into cmask and device-referencing in the low-level functions. Is this a good approach? * Audio in/out seems to work good in this series, mainly because the playback is an implicit feedback endpoint, and it is now configured with the capture endpoint as its sync-buddy. Also needed to correct the calculation, because of the different channel numbers between the two endpoints on the C400 (6 playback - implicit feedback, 4 capture). * The clock sources are still named "Unit 129" (0x81, internal) and "Unit 130" (0x82, SPDIF). I'll fix this later. I also plan to fix the channel names (i.e., instead of "AIn3", use "SPDIF In L"). * I've touched one FTU mixer creation function. I'd appreciate if someone can test this with the FTU to make sure I didn't break something. (I haven't gotten any feedback on that one yet) Cheers, Eldad Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> Eldad Zack (10): usb-audio: replace hardcoded value with const usb-audio: correct sync endpoint initialization usb-audio: use sender stride for implicit feedback usb-audio: add control index offset usb-audio: skip UAC2 EFFECT_UNIT usb-audio: parameterize FTU effect unit control usb-audio: M-Audio Fast Track C400 quirks table usb-audio: Fast Track C400 mixer ranges usb-audio: Fast Track C400 mixer controls usb-audio: FT C400 sync playback EP to capture EP sound/usb/endpoint.c | 9 ++- sound/usb/mixer.c | 44 +++++++++- sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 216 ++++++++++++++++++++++++++++++++++++++++++++-- sound/usb/pcm.c | 94 ++++++++++++++++++-- sound/usb/quirks-table.h | 71 +++++++++++++++ 6 files changed, 414 insertions(+), 21 deletions(-) -- 1.7.8.6

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[alsa-devel] [asoc:topic/arizona 34/34] ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined!
by kbuild test robot 09 Dec '12

09 Dec '12

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 commit: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 [34/34] ASoC: wm5110: Enable volume ramp control config: x86_64-randconfig-h424 (attached as .config) All error/warnings: ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_out_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add volume ramp controls
by Mark Brown 09 Dec '12

09 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 ++++++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d5385dc..a377d1f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -274,6 +274,31 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +static const char *arizona_vol_ramp_text[] = { + "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", + "15ms/6dB", "30ms/6dB", +}; + +const struct soc_enum arizona_in_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vd_ramp); + +const struct soc_enum arizona_in_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vi_ramp); + +const struct soc_enum arizona_out_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vd_ramp); + +const struct soc_enum arizona_out_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); + static const char *arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 24b9692..4f09bb5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -165,6 +165,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name, name "L"), \ ARIZONA_MIXER_ROUTES(name, name "R") +extern const struct soc_enum arizona_in_vi_ramp; +extern const struct soc_enum arizona_in_vd_ramp; + +extern const struct soc_enum arizona_out_vi_ramp; +extern const struct soc_enum arizona_out_vd_ramp; + extern const struct soc_enum arizona_lhpf1_mode; extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; -- 1.7.10.4

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Re: [alsa-devel] [PATCH] ASoC: adsp: Set DSP clock less than SYSCLK at initialisation
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 01:55:22PM +0000, Chris Rattray wrote: > In some configurations it is necessary to set the ADSP core clock > to less than the SYSCLK rate in order for the ADSP RAM to initialise > successfully. Once the firmware image is downloaded, the ADSP core > clock is then brought back up to the SYSCLK rate. As followed up offline this is a system-specific issue you were running into that this happened to work around. > Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> > Change-Id: I14412a301dcf1c0a6a728fa8c12d23715e944cd4 Don't include noise like Change-id that's specific to internal systems in upstream submissions.

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Re: [alsa-devel] [PATCH 03/20] ALSA: pxa2xx: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 1:35 AM, 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/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +++--- > 2 files changed, 4 insertions(+), 4 deletions(-) > > diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c > index 48d7c0a..6fc0ae9 100644 > --- a/sound/arm/pxa2xx-ac97-lib.c > +++ b/sound/arm/pxa2xx-ac97-lib.c > @@ -314,7 +314,7 @@ int pxa2xx_ac97_hw_resume(void) > EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); > #endif > > -int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) > +int pxa2xx_ac97_hw_probe(struct platform_device *dev) > { > int ret; > pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; > diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c > index 4e1fda7..ec54be4 100644 > --- a/sound/arm/pxa2xx-ac97.c > +++ b/sound/arm/pxa2xx-ac97.c > @@ -163,7 +163,7 @@ static int pxa2xx_ac97_resume(struct device *dev) > static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); > #endif > > -static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) > +static int pxa2xx_ac97_probe(struct platform_device *dev) > { > struct snd_card *card; > struct snd_ac97_bus *ac97_bus; > @@ -224,7 +224,7 @@ err_dev: > return ret; > } > > -static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > +static int pxa2xx_ac97_remove(struct platform_device *dev) > { > struct snd_card *card = platform_get_drvdata(dev); > > @@ -239,7 +239,7 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > > static struct platform_driver pxa2xx_ac97_driver = { > .probe = pxa2xx_ac97_probe, > - .remove = __devexit_p(pxa2xx_ac97_remove), > + .remove = pxa2xx_ac97_remove, > .driver = { > .name = "pxa2xx-ac97", > .owner = THIS_MODULE, > -- > 1.8.0.1 > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

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