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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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