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

Re: [alsa-devel] [PATCH] ALSA: hda - Add digital BEEP generator support for Realtek codecs.
by Kusanagi Kouichi 10 Feb '09

10 Feb '09

Sorry for the late reply. On Mon, Nov 17, 2008 at 17:53:34PM +0100, Takashi Iwai wrote: > Thanks for the patch. Do these beep widgets have the mute amp bits? > If not, we'd likely need to add mute switches as well (otherwise it > gets too annoying) controlling on software. Check patch_sigmatel.c in > the latest master branch or topic/fix/hda branch of sound git tree: > git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git Every codec can mute beep. > Another concern is that I don't want to add the digital beep control > unconditionally. On many devices, the beep is implemented in the old > good analog way. We'd need to add the check of analog/digital beep in > a new spec field and in the preset table not to duplicate beep > controls on these. I don't think such a check is needed. I presume that analog beep in is disabled while digital beep generetor is being used. Furthermore, keyboard bell rings only one device even if both HDA Digital PCBeep and PC Speaker have been registered.

2 9

Re: [alsa-devel] [PATCH] ASoC: Replace snd_soc_machine by snd_soc_card structure in SDP3430 machine driver
by Lopez Cruz, Misael 10 Feb '09

10 Feb '09

> On Tue, 10 Feb 2009 00:57:00 +0100 > "ext Lopez Cruz, Misael" <x0052729(a)ti.com> wrote: > > > This patch replaces "snd_soc_machine" structure by "snd_soc_card" in > > SP3430 driver. This change is needed in SDP3430 driver to reflect > > changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" > > patch (875065491fba8eb13219f16c36e79a6fb4e15c68). > > > > Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> > > --- > > sound/soc/omap/sdp3430.c | 4 ++-- > > 1 files changed, 2 insertions(+), 2 deletions(-) > > > Looks like you haven't subscribed to alsa-devel since I don't > see your patch there? You can subscribe but have mail > delivery disabled if you don't want to get mailing list > traffic to your account. I'm sorry I used 'alsa-devel-request' instead of 'alsa-devel'. I'll resend the patch right away. > Patch itself looks like for 2.6.29-rc since it fixes a > compilation error. Ack from me, compile tested on linux-omap. > > Acked-by: Jarkko Nikula <jarkko.nikula(a)nokia.com>

1 0

[alsa-devel] usr/include/linux/soundcard.h warnings for 'make headers_check'
by Jaswinder Singh Rajput 10 Feb '09

10 Feb '09

usr/include/linux/soundcard.h is giving following warnings for 'make headers_check': usr/include/linux/soundcard.h:1047: extern's make no sense in userspace usr/include/linux/soundcard.h:1048: extern's make no sense in userspace usr/include/linux/soundcard.h:1049: extern's make no sense in userspace usr/include/linux/soundcard.h:1050: extern's make no sense in userspace usr/include/linux/soundcard.h:1051: extern's make no sense in userspace usr/include/linux/soundcard.h:1053: extern's make no sense in userspace usr/include/linux/soundcard.h:1055: extern's make no sense in userspace usr/include/linux/soundcard.h:1056: extern's make no sense in userspace usr/include/linux/soundcard.h:1061: extern's make no sense in userspace usr/include/linux/soundcard.h:1062: extern's make no sense in userspace usr/include/linux/soundcard.h:1078: extern's make no sense in userspace usr/include/linux/soundcard.h:1079: extern's make no sense in userspace Do we need to fix these warnings. If yes, then how can we fix these warnings. Thanks -- JSR

5 19

[alsa-devel] [PATCH alsa-lib] Fix build error with -Werror=format-security
by Herton Ronaldo Krzesinski 09 Feb '09

09 Feb '09

src/alisp/alisp.c (F_princ): add missing format argument in call to snd_output_printf Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- src/alisp/alisp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/alisp/alisp.c b/src/alisp/alisp.c index 4ad1d09..de3defb 100644 --- a/src/alisp/alisp.c +++ b/src/alisp/alisp.c @@ -1708,7 +1708,7 @@ static struct alisp_object * F_princ(struct alisp_instance *instance, struct ali delete_tree(instance, p1); p1 = eval(instance, car(p)); if (alisp_compare_type(p1, ALISP_OBJ_STRING)) - snd_output_printf(instance->out, p1->value.s); + snd_output_printf(instance->out, "%s", p1->value.s); else princ_object(instance->out, p1); n = cdr(p); -- 1.6.1.3

1 0

[alsa-devel] [topic-asoc 2.6.30][PATCH 1/2] ASoC: TLV320AIC3X: Fix volume ranges
by Jarkko Nikula 09 Feb '09

09 Feb '09

This is a minor fix but helps to define dB ranges for volume controls. Only DAC digital volume has full register value range from 0 to 127 but ADC PGA gain and output stage volume controls don't. For ADC PGA, maximum value is 119 and then it saturates to the same gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB and is muted for values 118 and above. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 38 +++++++++++++++++++------------------- 1 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ac73e69..cdb6dec 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -255,51 +255,51 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_RLOPM_VOL, 0, 0x7f, 1), + DACR1_2_RLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + DACR1_2_LLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, - LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_LLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, - LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_RLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, - DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), + DACR1_2_MONOLOPM_VOL, 0, 118, 1), SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, - PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), + PGAR_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, - LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, - DACR1_2_HPROUT_VOL, 0, 0x7f, 1), + DACR1_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + PGAR_2_HPROUT_VOL, 0, 118, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, - LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), + LINE2R_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, - DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), + DACR1_2_HPRCOM_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, - LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), + LINE2R_2_HPRCOM_VOL, 0, 118, 1), /* * Note: enable Automatic input Gain Controller with care. It can @@ -308,7 +308,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), /* Input */ - SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), + SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), -- 1.5.6.5

2 2

[alsa-devel] capture hp dv3500
by Rodrigo Fresneda 09 Feb '09

09 Feb '09

The capture device for hp dv3500 (codec 92HD71B7X) does not work with alsa (1.0.19) There is no bug report for this, but maybe it has already been solved in newer version of alsa. Can anybody confirm? thanks, rodrigo.

2 5

[alsa-devel] No headphone with an Intel 82801I sound card
by Aurélien Croc 09 Feb '09

09 Feb '09

Hello, With the 2.6.27.10-0 (Mandriva) kernel driver, my sound card works quite well excepted the front digital microphone which didn't worked. Here is the ALSA script output: http://www.alsa-project.org/db/?f=31b5974fde70e1c5273f4248023481bb6766c46f I upgraded my kernel to the 2.6.27.10-1 and now the digital microphone works very well but the headphone output doesn't work. In other work, when i plug a speaker on it, the front integrated (to my laptop) speakers are muted but there is no sound in my external speaker. Here is the output of the ALSA script: http://www.alsa-project.org/db/?f=6def7ad107a215d22cef35a74d701216c5a496bd Thank you for your help. Regards, Aurélien

3 15

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 09 Feb '09

09 Feb '09

The following changes since commit 0664678a84c653bde844c7d91646259a25c6188b: Philipp Zabel (1): ASoC: pxa-ssp: fix SSP port request are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Eero Nurkkala (1): ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback Jarkko Nikula (1): ASoC: WM8990: Fix kcontrol's private value use in put callback Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 Mike Frysinger (2): ASoC: Blackfin: cleanup sport handling in ASoC Blackfin AC97 code ASoC: Blackfin: drop unnecessary dma casts Philipp Zabel (3): pxa/h5000: Setup I2S pins for pxa2xx-i2s pxa/spitz: Setup I2S pins for pxa2xx-i2s ASoC: pxa2xx-i2s: remove I2S pin setup Robert Jarzmik (1): ASoC: Add initial support of Mitac mioa701 device SoC. Roel Kluin (1): ASoC: count reaches 10001, not 10000. Timur Tabi (1): ASoC: optimize init sequence of Freescale MPC8610 sound drivers arch/arm/mach-pxa/h5000.c | 7 + arch/arm/mach-pxa/spitz.c | 6 + sound/soc/blackfin/bf5xx-ac97.c | 92 +++++--------- sound/soc/blackfin/bf5xx-ad73311.c | 4 +- sound/soc/blackfin/bf5xx-sport.c | 104 ++++++--------- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/fsl/fsl_dma.c | 178 ++++++++++++------------- sound/soc/fsl/fsl_ssi.c | 19 ++-- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_wm9713.c | 250 ++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-i2s.c | 36 ----- 13 files changed, 456 insertions(+), 266 deletions(-) create mode 100644 sound/soc/pxa/mioa701_wm9713.c

2 1

[alsa-devel] ASoC updates for 2.6.29
by Mark Brown 09 Feb '09

09 Feb '09

The following changes since commit 64ca0404eed57f6c92290d55e949a7f46cbe4bf4: Liam Girdwood (1): ALSA: ASoC: email - update email addresses. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Eero Nurkkala (1): ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback Jarkko Nikula (1): ASoC: WM8990: Fix kcontrol's private value use in put callback sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- sound/soc/codecs/wm8990.c | 4 +++- 2 files changed, 10 insertions(+), 5 deletions(-)

1 0

Re: [alsa-devel] HD-audio modularization
by Takashi Iwai 09 Feb '09

09 Feb '09

At Mon, 9 Feb 2009 09:39:34 +0100, Florian Mickler wrote: > > Hi! > > > For example, the recent SUSE init script has the following > > > > mod=$(grep -m1 -E '^(snd[^ ]*|ac97_bus) [0-9]+ 0' /proc/modules) > > while [ -n "$mod" ]; do > > mod=${mod%% *} > > /sbin/modprobe -r $mod > > mod=$(grep -m1 -E '^(snd[^ ]*|ac97_bus) [0-9]+ 0' /proc/modules) > > done > > > > But, basically it's stupid to unload modules at shtudown/reboot, just > > a waste of time. If possible, you should check the next runlevel and > > skip unloading if unnecessary. > > > > > > Takashi > > Yeah, it _is_ stupid to unload modules at shutdown. (and I kicked it > out of my init-script... but one wonders how it's supposed to work) > > modprobe -r snd_hda_codec_analog > gives me an error (''in use'') The codec driver is indeed used by the controller driver, snd-hda-intel. Takashi > # lsmod | grep snd_; modprobe -r snd_hda_codec_analog > snd_pcm_oss 42112 0 > snd_mixer_oss 16224 1 snd_pcm_oss > snd_hda_codec_analog 78144 1 > snd_hda_intel 28264 0 > snd_hda_codec 68992 2 snd_hda_codec_analog,snd_hda_intel > snd_pcm 86440 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec > snd_timer 23472 1 snd_pcm > snd_page_alloc 10288 2 snd_hda_intel,snd_pcm > FATAL: Module snd_hda_codec_analog is in use. > > ...so, if i accidentially don't unload snd_hda_intel before > snd_hda_codec_analog it doesn't work? or are my tools somehow broken?

1 0