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Re: [alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by krzysztof.h1＠poczta.fm 18 Aug '08

18 Aug '08

> At Mon, 18 Aug 2008 10:50:15 +0200, > Rene Herman wrote: > > > > On 17-08-08 22:38, Krzysztof Helt wrote: > > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > > > Mute sound by setting mute bit without > > > setting volume to 0. It makes both source code > > > and binary shorter. > > > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > Grmmbbled-over-by: Rene Herman <rene.herman(a)gmail.com> > > > > This should be safe and the current function indeed generates crap code > > > but in the sense of always expecting the worst broken hardware variants > > > to be out there I must say I'd rather have seen that GCC were beter. But > > > yes, sure. > > Yeah, the slightest concern is the intention of using 0x80 there. > If this patch was already tested on some real hardwares (and I guess > it should work), I'm willing to apply it. Krzystof, was this tested > on your machines? > I have tested it only on cs4236 (inside laptop) and opti931 I have in the old PC at the moment. If you want I can retest on more hardware. The second check I noticed that for some registers the volume is defined as gain (so 0 is the least audible) and for some (CS4231_MONO_CTRL, CS4231_LEFT/RIGHT_OUTPUT) is defined as attentuation (the 0 is the most audible). In two cases, the old code set volume to the loudest and mute it with mute bit, probably the opposite what was intended. It is not a crucial patch. You may drop it. Regards, Krzysztof ---------------------------------------------------------------------- Wygraj laptopa lub iPoda! Wez udzial w igrzyskach! Kliknij >>> http://link.interia.pl/f1eda

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[alsa-devel] [PATCH] wss_lib: snd_wss_calibrate_mute improvement
by Krzysztof Helt 18 Aug '08

18 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Mute sound by setting mute bit without setting volume to 0. It makes both source code and binary shorter. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp alsa-ref/sound/isa/wss/wss_lib.c alsa-driver/sound/isa/wss/wss_lib.c --- alsa-ref/sound/isa/wss/wss_lib.c 2008-08-14 00:05:30.000000000 +0200 +++ alsa-driver/sound/isa/wss/wss_lib.c 2008-08-17 22:33:24.620085406 +0200 @@ -574,7 +574,7 @@ static void snd_wss_calibrate_mute(struc { unsigned long flags; - mute = mute ? 1 : 0; + mute = mute ? 0x80 : 0; spin_lock_irqsave(&chip->reg_lock, flags); if (chip->calibrate_mute == mute) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -589,34 +589,34 @@ static void snd_wss_calibrate_mute(struc chip->image[CS4231_LOOPBACK]); } snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX1_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + mute | chip->image[CS4231_AUX2_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + mute | chip->image[CS4231_AUX2_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + mute | chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + mute | chip->image[CS4231_RIGHT_OUTPUT]); if (!(chip->hardware & WSS_HW_AD1848_MASK)) { snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + mute | chip->image[CS4231_LEFT_LINE_IN]); snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + mute | chip->image[CS4231_RIGHT_LINE_IN]); snd_wss_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + mute | chip->image[CS4231_LEFT_MIC_INPUT]); snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + mute | chip->image[CS4231_RIGHT_MIC_INPUT]); snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + mute | chip->image[CS4231_LINE_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + mute | chip->image[CS4231_LINE_RIGHT_OUTPUT]); } chip->calibrate_mute = mute; spin_unlock_irqrestore(&chip->reg_lock, flags); ---------------------------------------------------------------------- Igrzyska na Dzikim Zachodzie! Sprawdz >> http://link.interia.pl/f1edc

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Re: [alsa-devel] developing audio with ALSA
by Henry Ho 18 Aug '08

18 Aug '08

Hi After reading the cross-compilation instruction in INSTALL file, I already successfully configured the alsa-lib-2.0.17. I also successfully made the library. Adding --prefix=xxx during configuration can set the output directory to xxx. I also modify my Makefile for application, including the search path of libasound.so.2.***. Now I can successfully make it. However, when I run my application on the target, I got the following error message: ****************************************************************************************** ./pcm: error while loading shared libraries: libasound.so.2: cannot open shared object file: No such file or directory ****************************************************************************************** I have put libasound.so.2.0.0 in /tmp of the target. I also modified /etc/ld.so.conf, adding /tmp in ld.so.conf file. However, I cannot find ldconfig on the target. Is anyone can tell me how to solve it? Should I rebuild a busybox with this command? Henry ===================================================================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of RichWave is strictly prohibited; and any information in this email irrelevant to the official business of RichWave shall be deemed as neither given nor endorsed by RichWave.

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[alsa-devel] maestro3 & es1988: mic range too low
by ikke 18 Aug '08

18 Aug '08

Hi, I'm having mic problems with snd_maestro3 and es1988 allegro hw. The laptop is HP OmniBook Xe3. The mic is way too silent (unusable) even with 20db boost option on. I found this seems to have been really common problem, but only one came with a fix [1] to the problem, enabling one more bit to volume control. Could someone please help me apply that fix to current alsa? I tried compiling 2.6.26.2 kernel and made the first change of the patch from mail [1] attachement [2]. No change. The second part of patch I did not find anymore from the code. There are similar parts in ac97_patch.c. Any hint what should be added there? The original patched code is somewhat outdated compared to modern one... But the reasoning for the original patch sounds reasonable. The mic volume is tunable, only at the very low range. I recall the laptop came with Win98, and there was no problem then with the sound -> not a HW problem. I have never got this laptop mic working with linux, so this is not anything that broke recently. I see there is check_volume_resolution function which tries to guess some of the register ranges, but maybe it get's it wrong with this HW. I tried with external headset, the behavior is just the same. The headset is working fine with IBM T41, so not due the mic itself. Any help would be preciated, thanks, ikke [1] http://thread.gmane.org/gmane.linux.alsa.devel/2025 [2] http://cache.gmane.org//gmane/linux/alsa/devel/2025-001.bin Other posts related, with no solutions: http://article.gmane.org/gmane.linux.alsa.user/9802 http://article.gmane.org/gmane.linux.alsa.user/15260 http://article.gmane.org/gmane.linux.alsa.user/9663 http://article.gmane.org/gmane.linux.alsa.devel/27195 HW is HP Omnibook Xe3, 00:08.0 Multimedia audio controller: ESS Technology ES1988 Allegro-1 (rev 12) Ubuntu up to date hardy, with manually compiled linux-2.6.26.2. the patch: ----------------------------------- --- ac97_codec.c Fri Jun 21 16:09:15 2002 +++ /usr/src/linux/sound/pci/ac97/ac97_codec.c Sun Jun 16 21:56:00 2002 @@ -579,7 +579,7 @@ static const snd_kcontrol_new_t snd_ac97_controls_mic[3] = { AC97_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), -AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 15, 1), +AC97_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0) }; @@ -1137,7 +1137,7 @@ for (idx = 0; idx < 3; idx++) if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic[idx], ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_MIC, 0x801f); + snd_ac97_write_cache(ac97, AC97_MIC, 0x805f); /* build Line controls */ for (idx = 0; idx < 2; idx++) ----------------------------------- amixer settings: ----------------------------------- $ amixer Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 15 [100%] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 31 [100%] [12.00dB] [on] Capture [off] Front Right: Playback 31 [100%] [12.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 31 [100%] [12.00dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 15 [100%] [0.00dB] [on] Simple mixer control 'Aux',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 15 [100%] [22.50dB] [on] Front Right: Capture 15 [100%] [22.50dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] ----------------------------------- ---> looking at the mic range it seems to be 0-31 already, capture is only 0-15, I wonder if that's right...

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[alsa-devel] Configure and use dmix with a new ASoC driver
by Simon Tremblay 18 Aug '08

18 Aug '08

Hi, I don't know if there is something special I should do to be able to use dmix with my new ASoC driver I made (TLV320AIC23b codec running on a AT91RM9200 board). The problem is when I try to use dmix with aplay, it doesn't work and I got this error message: # aplay -v -D plug:dmix /media/usb0/tada.wav aplay: main:545: audio open error: Operation not permitted # Listing of my /dev/snd: # ls -al /dev/snd drwxr-xr-x 2 root root 0 Jan 1 00:01 . drwxr-xr-x 7 root root 0 Jan 1 00:01 .. crw-rw---- 1 root audio 116, 0 Jan 1 00:01 controlC0 crw-rw---- 1 root audio 116, 16 Jan 1 00:01 pcmC0D0p Notice I have a default /usr/share/alsa/alsa.conf file too. I'm running ALSA 1.0.15. Of course, it works well when I play a sound without the -D plug:dmix parameter... # aplay /media/usb0/tada.wav at91-ssc:ssc_startup: SSC_SR=0x000004c3 armus_tlv320aic23b: pck1 started asoc: aic23b <-> at91-ssc1 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Playing WAVE '/media/usb0/tada.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo # I tried to find some useful documentation about dmix but mainly it says that when we have a recent version of ALSA, we don't need to configure it because it's already done?

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[alsa-devel] [PATCH] opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
by Krzysztof Helt 17 Aug '08

17 Aug '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Remove card pointer from the snd_opl3sa2 structure and break circular reference snd_card->snd_opl3sa2->snd_card. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A very small opl3sa2 clean up. diff -urp linux-ref/sound/isa/opl3sa2.c linux-2.6/sound/isa/opl3sa2.c --- linux-ref/sound/isa/opl3sa2.c 2008-08-14 00:05:30.000000000 +0200 +++ linux-2.6/sound/isa/opl3sa2.c 2008-08-15 12:30:58.000000000 +0200 @@ -124,7 +124,6 @@ static int pnpc_registered; #define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX) struct snd_opl3sa2 { - struct snd_card *card; int version; /* 2 or 3 */ unsigned long port; /* control port */ struct resource *res_port; /* control port resource */ @@ -222,14 +221,13 @@ static void snd_opl3sa2_write(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static int __devinit snd_opl3sa2_detect(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_detect(struct snd_card *card) { - struct snd_card *card; + struct snd_opl3sa2 *chip = card->private_data; unsigned long port; unsigned char tmp, tmp1; char str[2]; - card = chip->card; port = chip->port; if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) { snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); @@ -298,12 +296,14 @@ static int __devinit snd_opl3sa2_detect( static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id) { unsigned short status; - struct snd_opl3sa2 *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_opl3sa2 *chip; int handled = 0; - if (chip == NULL || chip->card == NULL) + if (card == NULL || card->private_data == NULL) return IRQ_NONE; + chip = card->private_data; status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS); if (status & 0x20) { @@ -327,8 +327,10 @@ static irqreturn_t snd_opl3sa2_interrupt snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT); snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT); if (chip->master_switch && chip->master_volume) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } } return IRQ_RETVAL(handled); @@ -512,9 +514,9 @@ static void snd_opl3sa2_master_free(stru chip->master_volume = NULL; } -static int __devinit snd_opl3sa2_mixer(struct snd_opl3sa2 *chip) +static int __devinit snd_opl3sa2_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; + struct snd_opl3sa2 *chip = card->private_data; struct snd_ctl_elem_id id1, id2; struct snd_kcontrol *kctl; unsigned int idx; @@ -650,7 +652,6 @@ static struct snd_card *snd_opl3sa2_card chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; - chip->card = card; card->private_free = snd_opl3sa2_free; return card; } @@ -672,9 +673,12 @@ static int __devinit snd_opl3sa2_probe(s xdma2 = dma2[dev]; if (xdma2 < 0) chip->single_dma = 1; - if ((err = snd_opl3sa2_detect(chip)) < 0) + err = snd_opl3sa2_detect(card); + if (err < 0) return err; - if (request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, "OPL3-SA2", chip)) { + err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + "OPL3-SA2", card); + if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); return -ENODEV; } @@ -694,7 +698,8 @@ static int __devinit snd_opl3sa2_probe(s err = snd_wss_mixer(wss); if (err < 0) return err; - if ((err = snd_opl3sa2_mixer(chip)) < 0) + err = snd_opl3sa2_mixer(card); + if (err < 0) return err; err = snd_wss_timer(wss, 0, NULL); if (err < 0) ---------------------------------------------------------------------- Prosty kredyt na wszystkie potrzeby. Od 1000 zl do 120 000 zl. Bez poreczycieli i zabezpieczen >>> http://link.interia.pl/f1edf

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[alsa-devel] Problem with latest alsa-info.sh patch
by Travis Place 17 Aug '08

17 Aug '08

The latest patch to allow the uploading to alsa-project.org seems to work fine, for the most part, unless you want to use the --pastebin option for the legacy upload to pastebin.ca . It seems that the uploaded data gets truncated. Not sure if this is the script, or the pastebin having problems. Ill look into it a bit more when i have time (its 5:30am here, and i need sleep). Just thought id point this out. I realise the 'default' behaviour is to use alsa-project.org for uploading, but i feel the other pastebin.ca mode should still function(I personally get people to upload there first, and then 'submit' card details to alsa-project.org/db/ afterwards. Thanks, Travis Place (wishie)

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[alsa-devel] [SOUND/SOC] removed unused #include <version.h>
by Huang Weiyi 17 Aug '08

17 Aug '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/wm8753.c sound/soc/codecs/wm9712.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809..dc7b18f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a..2f1c91b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h>

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Re: [alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 16 Aug '08

16 Aug '08

Hello Stan, This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" 2. aplay does not produce sound, period! 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: sudo cat /var/log/messages | grep ALSA Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped Let me know if I should send something else. // Salvador // Northwestern University ----- Original Message ---- From: stan <ghjeold_i_mwee(a)cox.net> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Friday, August 15, 2008 11:54:12 AM Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues S. Aguinaga wrote: > Hello, > > I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. > > // Salvador > Fedora uses pulseaudio by default. How are you determining that there is no output? i.e. what program are you using, and how Try aplay -vv some.wav Try aplay -vv -Dhw:0,0 some.wav Try aplay -vv -Dplughw:0,0 some.wav Are there error messages or does it look like it is playing without sound? Are there messages in /var/log/messages regarding sound at boot or while trying to play. I don't use pulseaudio and don't know it very well, but there is more information here. http://www.pulseaudio.org Could you run a script which provides more information about your sound setup and post the link back here? The script is found at http://www.alsa-project.org/alsa-info.sh You can run the script so the results occur locally, by specifying the ---no-upload option. alsa-info.sh --no-upload You would then paste the output into your email instead of providing a link. This information will enable me to determine if your setup is OK, and look for other problems.

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[alsa-devel] [PATCH 1/2] wm8990: Fix routing of left DAC to speaker mixer
by Mark Brown 15 Aug '08

15 Aug '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) This ought to go into 2.6.27 - it's a fairly straightforward bug fix. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce51..9505a18 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- 1.5.6.3

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