April 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 22 Apr '10

22 Apr '10

The following changes since commit d4a8ca246146099e9068fc87cd2bc565896bc186: Takashi Iwai (1): ASoC: missing conversions to snd_soc_codec_*_drvdata() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (4): ASoC: Support FLL input clock selection on WM8994 ASoC: Set full range of WM8994 FLL Fratio values ASoC: Tone down debugging for WM8994 class W ASoC: Allow unspecified source when stopping WM8994 FLLs Phil Carmody (1): ASoC: da7210: Fencepost error in reg cache read sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/wm8994.c | 47 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.h | 5 ++++ 3 files changed, 43 insertions(+), 11 deletions(-)

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[alsa-devel] caps-bug (fwd)
by Tim Goetze 22 Apr '10

22 Apr '10

Hello alsa team, please find attached a bug report I received a short while ago. The finder isolated the problem on his own. It turned out that the "type equal" pcm slave ran the stereo plugin "Eq2x2" without connecting all of the plugin's input and output ports. This is in violation of ladspa.h, which mandates: """The plugin may assume that all its input and output ports have been connected to the relevant data location (see the `connect_port()' function below) before it is asked to run.""" -> http://www.ladspa.org/ladspa_sdk/ladspa.h.txt Cheers, Tim PS: not subscribed to this list. ---------- Forwarded message ---------- Subject: caps-bug From: x(a)y.z To: tim(a)quitte.de Date: Tue, 13 Apr 2010 11:35:41 +0200 Dear caps team, I'm using caps and the alsaequal alsa plugin together in my application. There is a failure when i want to close the pcm stream (snd_pcm_close). I'm working on redhat enterprise 5.4. Here is an extract of my alsa.conf: pcm.AudigySe { type hw card 0 device 0 } pcm.audigy_out1eq { type plug slave.pcm { type equal slave.pcm "plug:AudigySe" library "/usr/local/lib/ladspa/caps.so" module "Eq2x2" } } I've reproduced the bug using the aplay utility here: bash-3.2$ aplay -D pcm.audigy_out1eq abcdef.wav -d 1 Playing WAVE 'abcdef.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono *** glibc detected *** aplay: double free or corruption (!prev): 0x09776520 *** ======= Backtrace: ========= /lib/libc.so.6[0xb99595] /lib/libc.so.6(cfree+0x59)[0xb999d9] /usr/lib/libstdc++.so.6(_ZdlPv+0x21)[0x41b581] /usr/lib/libstdc++.so.6(_ZdaPv+0x1d)[0x41b5dd] /usr/local/lib/ladspa/caps.so(_fini+0x164)[0xebaf04] /lib/ld-linux.so.2[0xb24cc2] /lib/ld-linux.so.2[0xb25467] /lib/libdl.so.2[0xca1ce4] /lib/ld-linux.so.2[0xb1fe26] /lib/libdl.so.2[0xca22cc] /lib/libdl.so.2(dlclose+0x2a)[0xca1d1a] /usr/lib/alsa-lib/libasound_module_pcm_equal.so(LADSPAunload+0x1d)[0x3190ed] /usr/lib/alsa-lib/libasound_module_pcm_equal.so[0x3182c7] /lib/libasound.so.2[0x584f5c3] /lib/libasound.so.2(snd_pcm_close+0x76)[0x580e906] /lib/libasound.so.2[0x582a84b] /lib/libasound.so.2(snd_pcm_close+0x76)[0x580e906] aplay[0x805136a] /lib/libc.so.6(__libc_start_main+0xdc)[0xb45e9c] aplay(snd_pcm_mmap_readi+0x41)[0x804a3f1] ======= Memory map: ======== 00110000-0011b000 r-xp 00000000 08:02 2068216 /lib/libgcc_s-4.1.2-20080825.so.1 0011b000-0011c000 rwxp 0000a000 08:02 2068216 /lib/libgcc_s-4.1.2-20080825.so.1 001a6000-001ad000 r-xp 00000000 08:02 2068209 /lib/librt-2.5.so 001ad000-001ae000 r-xp 00006000 08:02 2068209 /lib/librt-2.5.so 001ae000-001af000 rwxp 00007000 08:02 2068209 /lib/librt-2.5.so 00317000-0031a000 r-xp 00000000 08:02 1775899 /usr/lib/alsa-lib/libasound_module_pcm_equal.so 0031a000-0031b000 rwxp 00002000 08:02 1775899 /usr/lib/alsa-lib/libasound_module_pcm_equal.so 00368000-00448000 r-xp 00000000 08:02 1154975 /usr/lib/libstdc++.so.6.0.8 00448000-0044c000 r-xp 000df000 08:02 1154975 /usr/lib/libstdc++.so.6.0.8 0044c000-0044d000 rwxp 000e3000 08:02 1154975 /usr/lib/libstdc++.so.6.0.8 0044d000-00453000 rwxp 0044d000 00:00 0 009f9000-009fa000 r-xp 009f9000 00:00 0 [vdso] 00b12000-00b2c000 r-xp 00000000 08:02 2068205 /lib/ld-2.5.so 00b2c000-00b2d000 r-xp 00019000 08:02 2068205 /lib/ld-2.5.so 00b2d000-00b2e000 rwxp 0001a000 08:02 2068205 /lib/ld-2.5.so 00b30000-00c6f000 r-xp 00000000 08:02 2068206 /lib/libc-2.5.so 00c6f000-00c70000 --xp 0013f000 08:02 2068206 /lib/libc-2.5.so 00c70000-00c72000 r-xp 0013f000 08:02 2068206 /lib/libc-2.5.so 00c72000-00c73000 rwxp 00141000 08:02 2068206 /lib/libc-2.5.so 00c73000-00c76000 rwxp 00c73000 00:00 0 00c78000-00c9d000 r-xp 00000000 08:02 2068213 /lib/libm-2.5.so 00c9d000-00c9e000 r-xp 00024000 08:02 2068213 /lib/libm-2.5.so 00c9e000-00c9f000 rwxp 00025000 08:02 2068213 /lib/libm-2.5.so 00ca1000-00ca3000 r-xp 00000000 08:02 2068207 /lib/libdl-2.5.so 00ca3000-00ca4000 r-xp 00001000 08:02 2068207 /lib/libdl-2.5.so 00ca4000-00ca5000 rwxp 00002000 08:02 2068207 /lib/libdl-2.5.so 00ca7000-00cba000 r-xp 00000000 08:02 2068208 /lib/libpthread-2.5.so 00cba000-00cbb000 r-xp 00013000 08:02 2068208 /lib/libpthread-2.5.so 00cbb000-00cbc000 rwxp 00014000 08:02 2068208 /lib/libpthread-2.5.so 00cbc000-00cbe000 rwxp 00cbc000 00:00 0 00e94000-00ee2000 r-xp 00000000 08:02 1937496 /usr/local/lib/ladspa/caps.so 00ee2000-00f83000 rwxp 0004d000 08:02 1937496 /usr/local/lib/ladspa/caps.so 057c3000-0589d000 r-xp 00000000 08:02 2068232 /lib/libasound.so.2.0.0 0589d000-058a2000 rwxp 000d9000 08:02 2068232 /lib/libasound.so.2.0.0 08048000-08054000 r-xp 00000000 08:02 1155406 /usr/bin/aplay 08054000-08056000 rw-p 0000b000 08:02 1155406 /usr/bin/aplay 0976a000-0978b000 rw-p 0976a000 00:00 0 [heap] b7d8d000-b7d8e000 r--p 00ca8000 08:02 1151208 /usr/lib/locale/locale-archive b7d8e000-b7f8e000 r--p 00000000 08:02 1151208 /usr/lib/locale/locale-archive b7f8e000-b7f91000 rw-p b7f8e000 00:00 0 bfe19000-bfe2e000 rw-p bffea000 00:00 0 [stack] Aborted by signal Abandon... Erreur de segmentation (core dumped) Is there a problem like a null descriptor here ?: void _fini() { for (ulong i = 0; i < N; ++i) delete descriptors[i]; }

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Re: [alsa-devel] The perennial asihpi question
by Eliot Blennerhassett 22 Apr '10

22 Apr '10

Bringing the outcome of a recent off-list push to get asihpi driver (for AudioScience soundcards) into kernel sources back on list: Takashi Iwai wrote: > Now merged, and finally moved to sound git tree. > > Your driver will appear in tomorrow's linux-next tree if no merge > problem occurs. > > Thanks for your efforts! Thank you very much for your help and patience, and work in merging many patches. It has been quite a long road. Looking back at my local history, the first commit of asihpi.c creating our local patches to alsa was on 2003-05-28, and Takashi's first commit to alsa-driver git on 2005-08-26. best regards Eliot Blennerhassett AudioScience Inc.

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[alsa-devel] [PATCH 1/4] pxa: add namespace on ssp
by Haojian Zhuang 22 Apr '10

22 Apr '10

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Re: [alsa-devel] T410s microphone is noisy
by Jens Taprogge 21 Apr '10

21 Apr '10

Hello Tobias, On Wed, Apr 21, 2010 at 10:35:58PM +0200, Tobias Grosser wrote: > I just tested the new v1.0.23 release including the Thinkpad T410s > support. I works really great, however the microphone input has > always > a white background noise. I am hearing some noise as well. I will try to look into the issue when I find the time. > I tried an external microphone as well as the internal ones. Either the > mixer is turned high enough that my voice can be heard good. In this > case there is always some background noise. If I turn the mixer (or > mic boost) down, the noise vanishes but I cannot hear myself. > > Can you give me a hint how to debug this problem? > > I already put my information here: > http://www.alsa-project.org/db/?f=6b218b5da3fcc298e6d677a3923ec66d98f996af > > Is this some input (maybe docking station) that is not disabled? Or > can I change some of the input amplifiers? I do not have access to datasheets and I am not an ALSA expert either. So I am not sure what the reason for the noise could be. Perhaps you could try adding mixers? I have read somewhere that the T410s has some sort of noise cancelation capabilities. I do not know if they are hard- or software based. Perhaps you could play with the GPIOs and see if that changes anything (e.g. enables noise cancelation). Best Regards Jens Taprogge

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[alsa-devel] [PATCH alsa-lib 1/2] Check for thread-specific locale support
by Rémi Denis-Courmont 21 Apr '10

21 Apr '10

--- configure.in | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/configure.in b/configure.in index f6df502..abc4687 100644 --- a/configure.in +++ b/configure.in @@ -64,6 +64,7 @@ dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL AC_CHECK_FUNC([hsearch_r], [HAVE_HSEARCH_R=yes]) AM_CONDITIONAL(ALSA_HSEARCH_R, [test "x$HAVE_HSEARCH_R" != xyes]) +AC_CHECK_FUNCS([uselocale]) SAVE_LIBRARY_VERSION AC_SUBST(LIBTOOL_VERSION_INFO) -- 1.7.0.4

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[alsa-devel] [PATCH] snd-maestro3: Make hardware volume buttons an input device
by Hans de Goede 21 Apr '10

21 Apr '10

While working on the sound suspend / resume problems with my laptop I noticed that the hardware volume handling code in essence just detects key presses, and then does some hardcoded modification of the master volume based on which key is pressed. This made me think that clearly the right thing to do here is just report these keypresses to userspace as keypresses using an input device and let userspace decide what to with them. This patch does this, getting rid of the ugly direct ac97 writes from the tasklet, the ac97lock and the need for using a tasklet in general. As an added bonus the keys now work identical to volume keys on a (usb) keyboard with multimedia keys, providing visual feedback of the volume level change, and a better range of the volume control (with a properly configured desktop environment). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/maestro3.c | 134 +++++++++++++++++++++++++++----------------------- 1 files changed, 73 insertions(+), 61 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0ed634a..135240d 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -41,6 +41,7 @@ #include <linux/vmalloc.h> #include <linux/moduleparam.h> #include <linux/firmware.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/info.h> #include <sound/control.h> @@ -806,6 +807,10 @@ struct m3_dma { struct snd_m3 { struct snd_card *card; +#ifdef CONFIG_INPUT + struct input_dev *input_dev; + char phys[64]; /* physical device path */ +#endif unsigned long iobase; @@ -844,11 +849,7 @@ struct snd_m3 { struct m3_dma *substreams; spinlock_t reg_lock; - spinlock_t ac97_lock; - struct snd_kcontrol *master_switch; - struct snd_kcontrol *master_volume; - struct tasklet_struct hwvol_tq; unsigned int in_suspend; #ifdef CONFIG_PM @@ -1602,11 +1603,9 @@ static void snd_m3_update_ptr(struct snd_m3 *chip, struct m3_dma *s) (without wrap around) in response to volume button presses and then generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7 of a byte wide register. The meaning of bits 0 and 4 is unknown. */ -static void snd_m3_update_hw_volume(unsigned long private_data) +static void snd_m3_update_hw_volume(struct snd_m3 *chip) { - struct snd_m3 *chip = (struct snd_m3 *) private_data; - int x, val; - unsigned long flags; + int x, key = 0; /* Figure out which volume control button was pushed, based on differences from the default register @@ -1632,51 +1631,34 @@ static void snd_m3_update_hw_volume(unsigned long private_data) if (chip->in_suspend) return; - if (!chip->master_switch || !chip->master_volume) +#ifdef CONFIG_INPUT + if (!chip->input_dev) return; - /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */ - spin_lock_irqsave(&chip->ac97_lock, flags); - - val = chip->ac97->regs[AC97_MASTER_VOL]; switch (x) { case 0x88: /* The counters have not changed, yet we've received a HV interrupt. According to tests run by various people this happens when pressing the mute button. */ - val ^= 0x8000; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_switch->id); + key = KEY_MUTE; break; case 0xaa: /* counters increased by 1 -> volume up */ - if ((val & 0x7f) > 0) - val--; - if ((val & 0x7f00) > 0) - val -= 0x0100; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_volume->id); + key = KEY_VOLUMEUP; break; case 0x66: /* counters decreased by 1 -> volume down */ - if ((val & 0x7f) < 0x1f) - val++; - if ((val & 0x7f00) < 0x1f00) - val += 0x0100; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_volume->id); + key = KEY_VOLUMEDOWN; break; } - spin_unlock_irqrestore(&chip->ac97_lock, flags); + + if (key) { + input_report_key(chip->input_dev, key, 1); + input_sync(chip->input_dev); + input_report_key(chip->input_dev, key, 0); + input_sync(chip->input_dev); + } +#endif } static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) @@ -1691,7 +1673,7 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) return IRQ_NONE; if (status & HV_INT_PENDING) - tasklet_schedule(&chip->hwvol_tq); + snd_m3_update_hw_volume(chip); /* * ack an assp int if its running @@ -1957,18 +1939,14 @@ static unsigned short snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct snd_m3 *chip = ac97->private_data; - unsigned long flags; unsigned short data = 0xffff; if (snd_m3_ac97_wait(chip)) goto fail; - spin_lock_irqsave(&chip->ac97_lock, flags); snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND); if (snd_m3_ac97_wait(chip)) - goto fail_unlock; + goto fail; data = snd_m3_inw(chip, CODEC_DATA); -fail_unlock: - spin_unlock_irqrestore(&chip->ac97_lock, flags); fail: return data; } @@ -1977,14 +1955,11 @@ static void snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct snd_m3 *chip = ac97->private_data; - unsigned long flags; if (snd_m3_ac97_wait(chip)) return; - spin_lock_irqsave(&chip->ac97_lock, flags); snd_m3_outw(chip, val, CODEC_DATA); snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); - spin_unlock_irqrestore(&chip->ac97_lock, flags); } @@ -2091,7 +2066,6 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - struct snd_ctl_elem_id elem_id; int err; static struct snd_ac97_bus_ops ops = { .write = snd_m3_ac97_write, @@ -2111,15 +2085,6 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) schedule_timeout_uninterruptible(msecs_to_jiffies(100)); snd_ac97_write(chip->ac97, AC97_PCM, 0); - memset(&elem_id, 0, sizeof(elem_id)); - elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(elem_id.name, "Master Playback Switch"); - chip->master_switch = snd_ctl_find_id(chip->card, &elem_id); - memset(&elem_id, 0, sizeof(elem_id)); - elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(elem_id.name, "Master Playback Volume"); - chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); - return 0; } @@ -2398,6 +2363,11 @@ static int snd_m3_free(struct snd_m3 *chip) struct m3_dma *s; int i; +#ifdef CONFIG_INPUT + if (chip->input_dev) + input_unregister_device(chip->input_dev); +#endif + if (chip->substreams) { spin_lock_irq(&chip->reg_lock); for (i = 0; i < chip->num_substreams; i++) { @@ -2524,6 +2494,42 @@ static int m3_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ +#ifdef CONFIG_INPUT +static int __devinit snd_m3_input_register(struct snd_m3 *chip) +{ + struct input_dev *input_dev; + int err; + + input_dev = input_allocate_device(); + if (!input_dev) + return -ENOMEM; + + snprintf(chip->phys, sizeof(chip->phys), "pci-%s/input0", + pci_name(chip->pci)); + + input_dev->name = chip->card->driver; + input_dev->phys = chip->phys; + input_dev->id.bustype = BUS_PCI; + input_dev->id.vendor = chip->pci->vendor; + input_dev->id.product = chip->pci->device; + input_dev->dev.parent = &chip->pci->dev; + + input_dev->evbit[0] = BIT_MASK(EV_KEY); + input_dev->keybit[BIT_WORD(KEY_MUTE)] |= BIT_MASK(KEY_MUTE); + input_dev->keybit[BIT_WORD(KEY_VOLUMEDOWN)] |= BIT_MASK(KEY_VOLUMEDOWN); + input_dev->keybit[BIT_WORD(KEY_VOLUMEUP)] |= BIT_MASK(KEY_VOLUMEUP); + + err = input_register_device(input_dev); + if (err) { + input_dev->dev.parent = NULL; + input_free_device(input_dev); + } else { + chip->input_dev = input_dev; + } + + return err; +} +#endif /* CONFIG_INPUT */ /* */ @@ -2567,7 +2573,6 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); - spin_lock_init(&chip->ac97_lock); switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: @@ -2650,8 +2655,6 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_hv_init(chip); - tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); - if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, card->driver, chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); @@ -2682,7 +2685,16 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if ((err = snd_m3_pcm(chip, 0)) < 0) return err; - + +#ifdef CONFIG_INPUT + if (chip->hv_config & HV_CTRL_ENABLE) { + err = snd_m3_input_register(chip); + if (err) + snd_printk(KERN_WARNING "Input device registration " + "failed with error %i", err); + } +#endif + snd_m3_enable_ints(chip); snd_m3_assp_continue(chip); -- 1.7.0.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] alsa-python: add register_poll to alsaseq
by Hector Martin 21 Apr '10

21 Apr '10

Attached patch adds register_poll to the sequencer bindings to support nonblocking sequencer access with the select module. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

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[alsa-devel] [PATCH 1/3] snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
by Hans de Goede 21 Apr '10

21 Apr '10

Without this quirk sound stops working after suspend resume. With this quirk, one still needs to manually unmute the master volume control after a suspend / / resume cycle. That is fixed in another patch in this set. Note that this patch was submitted to the alsa bug tracker a long time ago: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/maestro3.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fb..491a2fc 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -884,6 +884,7 @@ static struct pci_device_id snd_m3_ids[] = { MODULE_DEVICE_TABLE(pci, snd_m3_ids); static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = { + SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c), SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d), SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d), SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03), -- 1.7.0.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [REGRESSION] SB X-Fi Notebook stopped working in 1.0.23
by The Source 21 Apr '10

21 Apr '10

This device works fine with 1.0.22.1 driver, but doesn't with 1.0.23. snd-usb-audio & Ko are loaded but to no good. alsamixer doesn't list the device and no device files in /dev/snd. Reverting back to 1.0.22.1 solves the problem. System: ASUS N51Vf laptop, OS: Fedora 12 2.6.31.12-174.2.3.fc12.x86_64

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