June 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Laptop speakers nonfunctional, jacks functional
by Takashi Iwai 30 Jun '09

30 Jun '09

At Mon, 29 Jun 2009 20:06:47 +0200, Ludwig Maes wrote: > > Tried the snapshot, > still the same, alsa-info.txt attached Hm, just through a quick look at the output, there is no obvious problem. It's only the problem from speaker, not about the headphone, right? If so, the problem is likely the missing amplifier setup, which is usually powered up via COEF verbs or GPIO verbs (ALC888/1200 have no EAPD verbs). Try to install hda-verb program and check the command below. # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 GET_PROC_COEF 0 Which value do you get? What happens if you set # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3060 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x3070 or # hda-verb /dev/snd/hwC0D0 0x20 SET_COEF_INDEX 7 # hda-verb /dev/snd/hwC0D0 0x20 SET_PROC_COEF 0x0830 or so? Takashi

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[alsa-devel] Understanding _snd_pcm_channel_area Struct (PCM Interface)!
by Guilherme 30 Jun '09

30 Jun '09

Hi all. I am a bit confused on what I get with this function. I finished my first project but I did not understand what exactly the _snd_pcm_channel_area at http://www.alsa-project.org/alsa-doc/alsa-lib/struct__snd__pcm__channel__ar… does. There are 3 fields in the documentation. addr, first and step. It seems that addr is the memory address of the channel samples... so far so good. The "step" is the distance between 2 sample (but I am not sure) and the least, "first", I could not figure out what does it means. Could I get some help from here? Thanks in advanced! Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

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[alsa-devel] Merged patch_alc882() and patch_alc883()
by Takashi Iwai 30 Jun '09

30 Jun '09

Hi, I hacked around patch_realtek.c to clean it up a bit. As a result, now patch_alc883() gets merged to patch_alc882(). This cuts off lots of lines. I think this action is mostly OK as long as I tested with hda-emu. But, if you find any problems with this, please let me know. At best, get alsa-info.sh output before updating. It's always recommended to keep alsa-info.sh output once on a working system. This will make easier to track down the problem. thanks, Takashi

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[alsa-devel] External PCM hook SO is dlclosed after calling install, making it hard to have hooks in it
by Paul Fertser 30 Jun '09

30 Jun '09

Hi, I've tried to implement an external shared object to be able to execute arbitrary functions on PCM device opening and closing. After quite some source code reading i had an almost working but segfaulting example. This segfault is because snd_dlclose is called right after calling a hook install function and therefore i can't snd_pcm_hook_add functions from the same shared object. To me it looks like a bug but probably i just don't see how this functionality is intended to be used. Takashi, git-blame showed that it's you who stuffed this dlclose call that bothers me ;) TIA Background information: We're working on audio subsystem for OpenMoko Freerunner. It's an idea of Joerg Reisenweber to have virtual alsa devices for every reasonable use-case (like dedicated device for stereout, dedicated device for recording from gsm etc), for that we need a hook on every device opening and closing mostly to set up routing inside WM8753 codec. ctl_elems is not enough because we need a more complex locking scheme (with priorities, some kind of callbacks to the apps using particular devices etc) and other flexibility. -- Be free, use free (http://www.gnu.org/philosophy/free-sw.html) software! mailto:fercerpav@gmail.com

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[alsa-devel] [snd-hdsp] module mapping missing
by u2＠whoplays.asia 30 Jun '09

30 Jun '09

Module-hdsp maintainers Hi, Thanks for your great sound system, but I found a missing or invalid map of the soundcard and module for "RME HDSPe AIO". [RME HDSPe AIO] http://www.rme-audio.de/en_products_hdspe_aio.php However the soundcard shoud be recognized as the product "HDSPe ExpressCard" as in soundcard matrix I think, but was recognized as the product "Hammerfall DSP Madi". [Matrix:Vendor-RME] http://www.alsa-project.org/main/index.php/Matrix:Vendor-RME [RME HDSP MADI] http://www.rme-audio.de/en_products_hdsp_madi.php Because product-id looks categorized as "HDSP MADI", the module snd-hdspm for "HDSP MADI" was installed for "HDSPe AIO". $ uname -srvmo Linux 2.6.26-2-amd64 #1 SMP Sun Jun 21 04:47:08 UTC 2009 x86_64 GNU/Linux $ lspci -nn | grep RME 02:00.0 Multimedia audio controller [0401]: Xilinx Corporation RME Hammerfall DSP MADI [10ee:3fc6] (rev d4) $ cat /proc/asound/modules 0 snd_hdspm $ pcimodules | grep hdsp snd-hdspm $ cat /lib/modules/2.6.26-2-amd64/modules.pcimap | grep hdsp snd-hdsp 0x000010ee 0x00003fc5 0xffffffff 0xffffffff 0x00000000 0x00000000 0x0 snd-hdspm 0x000010ee 0x00003fc6 0xffffffff 0xffffffff 0x00000000 0x00000000 0x0 $ setpci -d 10ee:3fc6 vendor_id device_id subsystem_vendor_id subsystem_id 10ee 3fc6 00000000 0000 How can I help for it ? ---- Toshiya Miyagi

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[PATCH] pcm_hooks: cosmetic removal of unused variables
by Paul Fertser 30 Jun '09

30 Jun '09

Signed-off-by: Paul Fertser <fercerpav(a)gmail.com> --- src/pcm/pcm_hooks.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 2fbee73..3a99d55 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -99,7 +99,6 @@ static int snd_pcm_hooks_close(snd_pcm_t *pcm) } } while (!list_empty(&h->dllist)) { - struct snd_pcm_hook_dllist *dl; pos = h->dllist.next; hook_remove_dlobj(list_entry(pos, struct snd_pcm_hook_dllist, list)); } @@ -345,7 +344,6 @@ static int snd_pcm_hook_add_conf(snd_pcm_t *pcm, snd_config_t *root, snd_config_ snd_config_iterator_t i, next; int (*install_func)(snd_pcm_t *pcm, snd_config_t *args) = NULL; void *h = NULL; - struct snd_pcm_hook_dllist *dl; if (snd_config_get_type(conf) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid hook definition"); -- 1.6.1.3 --IMjqdzrDRly81ofr--

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[alsa-devel] [PATCH] [RFC] ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
by Janusz Krzysztofik 30 Jun '09

30 Jun '09

This patch tries to work around the problem of broken OMAP1510 PCM playback pointer calculation by replacing DMA function call that incorrectly tries to read the value form DMA hardware with a value computed locally from an already maintained variable omap_runtime_data.period_index. Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC driver. Based on linux-2.6-asoc.git v2.6.31-rc1. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- It seems that on OMAP1510, DMA Channel Progress Counter registers (DMA_CPC_CH[0-8]) always contain values derived from DMA channels destination port address, even if constant, and there are no DMA registers available that would follow DMA channels source port address. Because of this limitation, current implementation of omap_get_dma_src_pos() for OMAP1510 is broken and doesn't seem to be easy correctable. --- linux-2.6.31-rc1/sound/soc/omap/omap-pcm.c.orig 2009-06-27 20:20:16.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/omap-pcm.c 2009-06-27 23:21:42.000000000 +0200 @@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointe dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ptr = omap_get_dma_src_pos(prtd->dma_ch); - else + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else if (!(cpu_is_omap1510())) { + ptr = omap_get_dma_src_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else + offset = prtd->period_index * runtime->period_size; - offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); if (offset >= runtime->buffer_size) offset = 0;

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[alsa-devel] Can a phone hook switch follow alsa jack model?
by Janusz Krzysztofik 29 Jun '09

29 Jun '09

Hi, I am wondering if it is a good idea to create support for a phone hook switch, or a handset pick up switch, like that found on Amstrad E3 (Delta) videophone, using alsa jack framework. After my initial attempt to add support for the switch using gpio-keys driver, I am no longer sure if it is a good idea to follow the keyboard model, that the driver has been designed after, for driving a switch that has nothing to do with keyboards and may required a different approach. OTOH, the switch is closely related to a handset, and handsets can be seen as sound devices, can't they? So maybe alsa jack would fit better than keyboard? However, I am not sure if the switch in question matches the alsa jack model closely enough. I see the switch usage not as simple as turning handset microphone/speaker on or off. It can be used for other purposes as well, like accepting a phone call or actually dialing a number that has been just typed in. Furthermore, it can be used to turn off a speakerphone function, while, in turn, the related handset microphone/speaker pair can be turned off not only with this switch, but with a handsfree button as well, for example. All that extra functionality looks like belonging to userspace rather then kernel, not like other alsa jack implementations that seem to do all the job of switching media paths inside the kernel. That is why I am not sure if the jack model is suitable for the purpose. Thanks, Janusz -- 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] IMX31 Audio Driver issue
by Harish Kumar 29 Jun '09

29 Jun '09

Hi, I am using a imx31 based board from Garz & fricke. I need to configure the audio driver from the linux 2.6.23 BSP available. I configured the audio driver by selecting the SSI, ALSA & PMIC Audio options in the kernel configuration. The PMIC Audio card is listing /proc/asound/cards and all the necessary devices are created using the udev available on the rootfs. I used the check_audio.out application available from IMX PDk BSP test programs. We are getting the following error. I am using the /unit_tests/check_audio.out 0 /unit_tests/audio8k16S.wav Hi... playing on /dev/sound/adsp (SSI1) Configuration : 16 8000 2 audio driver write error: Input/output error audio driver write error: Input/output error a The program is not exiting. also tried with SSI2 as below /unit_tests/check_audio.out 1 /unit_tests/audio8k16S.wav Hi... playing on /dev/sound/dsp (SSI2) Configuration : 16 8000 2 audio driver write error: Input/output error audio driver write error: Input/output error But the same hardware is working with WINCE without any issue. I feel there might be issues with drivers that i am using or may some configuration mismatch. Request your help in resolving the same. Request you to provide the troubleshooting steps so that issue can be resolved. The applications used and the capture of the command execution is as attached. Thanks & Regards Harish Kumar.B

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[alsa-devel] warning "hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj."
by Simos Xenitellis 29 Jun '09

29 Jun '09

Hi All, I am getting the above warning when trying to play audio files. Contrary to other similar cases with this warning, in my case the playback of the media files fails. For every single media file, the playback fails after a random amount of time (a few seconds to a few minutes). I am running Ubuntu 9.04 (kernel 2.6.28-13-generic, x86_64, upgraded to Alsa 1.0.20 using AlsaUpgrade, http://ubuntuforums.org/showthread.php?p=6589810). I increased bdl_pos_adj to 32 and there is no effect (I added "options snd-hda-intel bdl_pos_adj=32" to /etc/modprobe.d/alsa-base.conf, then "/sbin/alsa force-reload"). lspci -vvv says: 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Dell Device 02bb Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 2293 Region 0: Memory at f6500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Address: 00000000fee0300c Data: 41d1 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel lshw says: *-multimedia description: Audio device product: 82801I (ICH9 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 03 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=HDA Intel latency=0 module=snd_hda_intel Simos

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[alsa-devel] Bug #4560: USB webcam Philips SPC 530NC micro not working
by Nicolas George 29 Jun '09

29 Jun '09

Hi. Some time ago, I submitted a bug about the integrated micro of the USB webcam Philips SPC 530NC: record: pcm_read:1617: read error: Input/output error This is bug #4560: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4560 I would really appreciate if someone familiar with the USB audio system could give it some attention. It is probably very similar to bug #3927. Raymond seems to have been able to help somewhat, but the command that is said to work for the reporter does not work for me. Regards, -- Nicolas George

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[alsa-devel] [PATCH 1/2] alsa/soc: Fix typo in MPC5200 PSC AC97 driver Kconfig
by Grant Likely 29 Jun '09

29 Jun '09

From: Grant Likely <grant.likely(a)secretlab.ca> ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS. Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly registered on the AC97 bus, which prevents thinks like the WM9712 touchscreen driver from getting probed. Tested against 2.6.31-rc1. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5dbebf8..5661876 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S config SND_SOC_MPC5200_AC97 tristate "Freescale MPC5200 PSC in AC97 mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select AC97_BUS + select SND_SOC_AC97_BUS select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help

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Re: [alsa-devel] [PATCH 62/62] sound: Remove unnecessary semicolons
by Mark Brown 29 Jun '09

29 Jun '09

On Sun, Jun 28, 2009 at 09:27:07AM -0700, Joe Perches wrote: > Signed-off-by: Joe Perches <joe(a)perches.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

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[alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Andreas Nuesslein 29 Jun '09

29 Jun '09

I just got a macbookpro (mid2009 version) with a new nvidia chipset. there is no sound at all. neither via the built-in speakers nor via out-jack. i tried aplay -D myfile.wav with everything from aplay -L as well. i also tried a recent snapshot (alsa-driver-20090626.tar.bz2) as well as gentoo ebuilds alsa-headers-9999 and alsa-driver-9999 (which uses git to pull recent versions). so here's a little information i can gather: $ lspci -s 00:08.0 -v 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: nVidia Corporation Device cb79 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 23 Memory at d3480000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel alsa-info: http://www.alsa-project.org/db/?f=80e89b82fec184b68a88b9ed4128544a276775fc modinfo snd-hda-intel: http://www.nopaste.com/p/a3B7miINP i also found a thread about that problem (w/o solution obviously): http://www.uluga.ubuntuforums.org/showthread.php?t=1188849&page=2 hope that's all information necessary. if i can i'd love to help in some way. thanks

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[alsa-devel] Laptop speakers nonfunctional, jacks functional
by Ludwig Maes 29 Jun '09

29 Jun '09

---------- Forwarded message ---------- From: Ludwig Maes <ludwig.maes(a)gmail.com> Date: 2009/6/26 Subject: Laptop speakers nonfunctional, jacks functional To: alsa-user(a)lists.sourceforge.net Realtek ALC888S (according to preinstalled Realtek HD audio manager on windows) visible jacks working, builtin laptop speakers and subwoofer silent. Running Ubuntu 9.04 compiled and installed alsa 1.0.20. http://www.alsa-project.org/db/?f=fb64ba9a8ab28d8c16532ece21e3adba7219e4b9 Notice it thinks it is ALC1200. I searched this and found it in patch_realtek.c, I changed the equality test so we would end up in ALC888 section... result was I got an extra volume control on mixer for BEEP. previously did not hear beeps when hitting down in terminal. BEEP is in headphones or line out. still not on speakers. aplay -l listed ALC888 too but codec#0 still showed ALC1200... I dont know the flow of information changes in ALSA to change this codec... I undid the equality test change and now its back to original realtek patch, recompiled reinstalled... added model=generic, to asound.state, this gave me more controls IIRC.. downloaded codecgraph utility, result: http://img20.imageshack.us/img20/8692/codec0.png I dont know how these codecgraphs work, but 4 jacks are right! the internal speakers appear twice?? how can one rewire these, or enable assuming dotted is disabled, does every amplifier triangle correspond to a volume control? is there a direct way to make changes? dmesg |grep hda: [ 16.172695] hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... [ 18.037866] ALSA /usr/src/alsa/alsa-driver-1.0.20/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:636: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x00bb0000 Realtek datasheet for ALC888S, reference design, if helpful: http://www.realtek.com.tw/products/productsView.aspx?Langid=1&PFid=28&Level… I am very willing to test proposals... friendly greetings, Ludwig

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[alsa-devel] Samsung P50 Heaphone Jack-Sense not working with latest ALSA
by Martin Egge 29 Jun '09

29 Jun '09

Hi, last year Takashi did help me to get the digital output and the internal/external mic up and running on my Samsung P50 notebook with an integrated Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller. Some time ago I switched from Ubuntu 7.10 to 8.10 and a new problem with the soundcard occurred. The Headphone/Line-Out Jack-Sense is not working. When I plug in my Headphones the internal speakers of the notebook aren't switched off. I've ignored that problem for a long time but now I'm willing to change this. I've already upgraded my system to ALSA 1.0.20 with a script provided in the Ubuntu Forums [2]. I've used the build-in support for the latest snapshots provided at [3] of the script. But the problem still exists. Can anyone (perhaps Takashi again) help me? You can find all ALSA related information about my system at [4]. Regards, Martin [1] http://tinyurl.com/mkw8mz [2] http://ubuntuforums.org/showthread.php?p=6589810#post6589810 [3] http://tinyurl.com/5ux2nf [4] http://pastebin.ca/1471032

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[alsa-devel] cmi8330: revert comments about AD1848 back
by Ondrej Zary 29 Jun '09

29 Jun '09

In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848. The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments were correct and are misleading now. Revert them back. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.30-orig/sound/isa/cmi8330.c 2009-06-10 05:05:27.000000000 +0200 +++ linux-test/sound/isa/cmi8330.c 2009-06-27 16:20:51.000000000 +0200 @@ -35,7 +35,7 @@ * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the CMI8330 codec, + * The driver "magically" routes the capturing to the AD1848 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -345,7 +345,7 @@ err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -527,11 +527,11 @@ wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); + snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); + snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } -- Ondrej Zary

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[alsa-devel] cmi8330: fix MPU-401 PnP init copy&paste bug
by Ondrej Zary 29 Jun '09

29 Jun '09

Fix copy&paste bug in PnP MPU-401 initialization. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.30-orig/sound/isa/cmi8330.c 2009-06-10 05:05:27.000000000 +0200 +++ linux-test/sound/isa/cmi8330.c 2009-06-27 13:37:56.000000000 +0200 @@ -338,7 +338,7 @@ return -EBUSY; acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); - if (acard->play == NULL) + if (acard->mpu == NULL) return -EBUSY; pdev = acard->cap; -- Ondrej Zary

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Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sun, 07 Jun 2009 14:00:26 +0200, MonsieurWeller wrote: > > Thanks for reply. Hope Creative engineers read this :D. > Two other little things :D: > - The sound seems really really to loud compared to my audigy2 zs ( for > the same level: pcm/master/etc... 74 in alsamixer for the audigy vs 26 > for the xfi) Yes, it's a known issue. 100% doesn't correspond to 0dB but overdriving. 0dB is about 70% or so, and the volume scale doesn't look logarithmic. > - There's no cursor for bass and treble in alsamixer: not supported? Not supported. Some DSP code would be required, which isn't open yet. thanks, Takashi

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Re: [alsa-devel] ice1712 error
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sun, 28 Jun 2009 10:43:04 +0200, Udo van den Heuvel wrote: > > On 2009-06-28 10:39, Takashi Iwai wrote: > > At Sat, 27 Jun 2009 13:33:42 +0200, > > Udo van den Heuvel wrote: > >> Hello Takashi, > >> > >> WHat to do when I have 3 cards total, two cards of these 3 using the > >> same driver: hda-intel. > >> Hwo can I fix this one specific card to a snd slot? The computer somehow > >> puts the card in the wrong order. > >> (i.e.: C-B-A instead of A-B-C where A and C use hda-intel) > >> Can I specify PCI-id's similar to the USB vid/pid that can be passed? > > > > You can specify index option, e.g. index=2,1,0. > > Yes, OK, but hen index 0 and 2 are both hda-intel, how do I tie a > *specific* hda-intel card to each slot? There is no generic way. The assignment order is usually static and unchanged for such PCI devices. (Or, you can change the slot later via sysfs, if any.) In general, you shouldn't stick with the assignment order but rather use the id string to each device. Takashi

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Re: [alsa-devel] ice1712 error
by Takashi Iwai 28 Jun '09

28 Jun '09

At Sat, 27 Jun 2009 13:33:42 +0200, Udo van den Heuvel wrote: > > Hello Takashi, > > WHat to do when I have 3 cards total, two cards of these 3 using the > same driver: hda-intel. > Hwo can I fix this one specific card to a snd slot? The computer somehow > puts the card in the wrong order. > (i.e.: C-B-A instead of A-B-C where A and C use hda-intel) > Can I specify PCI-id's similar to the USB vid/pid that can be passed? You can specify index option, e.g. index=2,1,0. Takashi

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Re: [alsa-devel] snd_pcm_lib_read1 timeout
by Harsha, Priya 27 Jun '09

27 Jun '09

I found another clue to this but can anyone please explain what could be going wrong? The execution enters into the following function >static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream >*substream) >{ > .... > new_hw_ptr = hw_base + pos; this gets updated correctly > ... > hdelta = new_hw_ptr - old_hw_ptr; > jdelta = jiffies - runtime->hw_ptr_jiffies; as far as I don't get into the following loop, am good...the moment I get into this loop my new_hw_ptr gets modified and it does not progress in capture. Any clues? > if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { > delta = jdelta / > (((runtime->period_size * HZ) / runtime->rate) > + HZ/100); > hw_ptr_error(substream, > "hw_ptr skipping! [Q] " > "(pos=%ld, delta=%ld, period=%ld, " > "jdelta=%lu/%lu/%lu)\n", > (long)pos, (long)hdelta, > (long)runtime->period_size, jdelta, > ((hdelta * HZ) / runtime->rate), delta); > hw_ptr_interrupt = runtime->hw_ptr_interrupt + > runtime->period_size * delta; > if (hw_ptr_interrupt >= runtime->boundary) > hw_ptr_interrupt -= runtime->boundary; > /* rebase to interrupt position */ > hw_base = new_hw_ptr = hw_ptr_interrupt; > /* align hw_base to buffer_size */ > hw_base -= hw_base % runtime->buffer_size; > delta = 0; > } > ... >} Thanks, Harsha ______________________________________ From: Harsha, Priya Sent: Saturday, June 27, 2009 9:18 AM To: alsa-devel(a)alsa-project.org Subject: snd_pcm_lib_read1 timeout Hi, I have a basic question; There is a schedule_timeout() in snd_pcm_lib_read1 -> wait_for_avail_min. When and where does this timeout get signaled? What would be the reason that this timeout happens? Thanks, Harsha

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Re: [alsa-devel] Should my Creative X-fi card be loading the snd-hda-intel module
by Susan Cragin 27 Jun '09

27 Jun '09

>> Shouldn't it be loading snd-hda-codec-ca0110 instead? > >snd-hda-codec-ca0110 is a slave module loaded by snd-hda-intel. > >Takashi The problem is, my Creative X-fi isn't being recognized any more, even though the modules are there and have been loaded. My onboard card is an hda-intel. And my utilities and programs don't find the Creative card. They only find the onboard card. I have added asoundconf to my alsa-utils, and asoundconf list shows only: Intel system (my usb card) winecfg shows the same. My goal is to get the card running again under wine. Susan

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[alsa-devel] snd_pcm_lib_read1 timeout
by Harsha, Priya 27 Jun '09

27 Jun '09

Hi, I have a basic question; There is a schedule_timeout() in snd_pcm_lib_read1 -> wait_for_avail_min. When and where does this timeout get signaled? What would be the reason that this timeout happens? Thanks, Harsha

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[alsa-devel] [PATCH] ASoC: TWL4030: Add EXT_MUTE gpio to reduce pop-noise effect
by Jorge Eduardo Candelaria 27 Jun '09

27 Jun '09

According to TRM, an external FET controlled by a 1.8V output signal can be used to reduce the pop-noise heard when the audio amplifier is switched on. The signal is controlled by a gpio pin and should be defined in the machine driver. However, the codec driver takes care of enabling and disabling this output during the headset pop attenuation sequence. If the board does not have an EXT_MUTE, then the machine driver should set the value of ext_mute_gpio to -EINVAL through ramp delay setup data, to bypass execution of the code that manages the gpio line. Also add a delay to let VMID settle in ramp up sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/codecs/twl4030.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2..e850b30 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -27,6 +27,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/i2c/twl4030.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -620,6 +621,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +633,11 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute, this dramatically reduces + * the pop-noise */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 1); + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +645,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +664,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && gpio_is_valid(setup->ext_mute_gpio)) + gpio_set_value(setup->ext_mute_gpio, 0); } static int headsetlpga_event(struct snd_soc_dapm_widget *w, @@ -2101,6 +2117,7 @@ static int twl4030_init(struct snd_soc_device *socdev) /* Configuration for headset ramp delay from setup data */ if (setup) { unsigned char hs_pop; + unsigned int ext_mute; if (setup->sysclk) twl4030->sysclk = setup->sysclk; @@ -2111,6 +2128,12 @@ static int twl4030_init(struct snd_soc_device *socdev) hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + + if (gpio_is_valid(setup->ext_mute_gpio)) { + ext_mute = setup->ext_mute_gpio; + BUG_ON(gpio_request(ext_mute, "ext_mute") < 0); + gpio_direction_output(ext_mute, 0); + } } else { twl4030->sysclk = 26000; } diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395..23a75e2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int ext_mute_gpio; }; #endif /* End of __TWL4030_AUDIO_H__ */ -- 1.6.0.4

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