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

[alsa-devel] snd_hda_intel: Conexant CX20551 (Waikiki)
by Edwin Smulders 03 Jun '07

03 Jun '07

Hi, I believe there's still a problem with the HDA Intel chip Conexant CX20551 (Waikiki) not working. However, im not entirely sure the problem the problem is with alsa, because it starts working when i boot with acpi=off (a fix suggested on the ubuntu forums, but not quite useful on a laptop [plus, it breaks my ethernet]) The reason I'm mailing is, can you confirm this is something alsa can/should/will fix or are we all looking in the wrong direction and this is really an acpi problem? If you can fix this or know of something that might work, I would be happy to hear it. I've attached my lspci -nv and cat /proc/asound/card0/codec\#* just in case. I would be happy to give you any more info you might need. ~~ Edwin

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[alsa-devel] [PATCH] Add support for another X-Fi Extreme Audio (really SB Live 24bit)
by Daniel T. Chen 03 Jun '07

03 Jun '07

Hi, Attached please find an hg changeset against current alsa-kernel tip. Thanks, Daniel Chen

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[alsa-devel] Next steps
by chris＠idlelion.net 03 Jun '07

03 Jun '07

Hey gang, I'm trying to work on issue 305, suspend/resume for cs46xx. I'm running CentOS 5, so I'd like to work on what's current for that distribution first, and patch to latest if needed after that. I think I have the correct source package, but the only mention of cs46xx I could find in the source is a configuration file. Running make doesn't make the cs46xx module, AFAICT. Which source (package)(s) should I have? How do I track down the source file(s) I need to examine? Thanks, Chris

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Re: [alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 01 Jun '07

01 Jun '07

Liam Girdwood wrote: >> I guess I just don't understand why the codec driver is acting like the "master" driver of >> ASOC. IMHO, the codec driver should be doing two things: > > I guess this is due to an ordering issue we had during early > development. We had to make sure that the I2C probe of the codec > succeeded before registering the card, pcms, etc. Fwiw, I do intend to > address this in future versions (I should probably add a road map to the > wiki now). In that case, I think we can solve this problem as well as the PowerPC device tree problem in one shot, because they're basically the same problem. Anyway, let me see if I get this right: 1. The first function called is eti_b1_init(). 2. eti_b1_init() calls platform_device_add() to add an soc-audio class device and register the eti_b1_snd_devdata structure. 3. The registration causes a number of things to happen, one of which is a call to soc_probe(). 4. soc_probe() sees that the snd_soc_machine_eti_b1.probe is NULL, so that part is skipped. 3. soc_probe() sees that snd_soc_machine_eti_b1.dai_link->cpu_dai->probe is also NULL, so that part is skipped. (eti_b1_snd_devdata.machine == snd_soc_machine_eti_b1) 4. soc_probe() sees that soc_codec_dev_wm8731.probe is not NULL, so it calls wm8731_probe(). 5. wm8731_probe() registers an I2C driver. 6. The I2C class driver calls wm8731_i2c_driver.attach_adapter, which is wm8731_i2c_attach(). 7. wm8731_i2c_attach() calls i2c_probe(), which in turn calls wm8731_codec_probe() 8. wm8731_codec_probe() allocates a snd_soc_codec structure. 8. wm8731_codec_probe() calls wm8731_init(). This means that if there is no I2C support, wm8731_init() will never be called. 9. wm8731_init() initializes the snd_soc_codec structure. Question: why doesn't wm8731_probe() initialize the non-I2C parts of the snd_soc_codec structure? For example, snd_soc_codec.dai, snd_soc_codec.name, and snd_soc_codec.owner? That way, the snd_soc_codec structure will be properly initialized even when there's no I2C. 10. wm8731_init() calls snd_soc_new_pcms. 11. wm8731_init() exits, wm8731_codec_probe() exits, wm8731_i2c_attach() exits, and then wm8731_probe() exits. Control returns to soc_probe(). 12. soc_probe() notices that eti_b1_snd_devdata.platform.probe is NULL, so it skips that. Question: why don't you define a function for at91_soc_platform.probe, and in that function you can call snd_soc_new_pcms()? That way, you guarantee the codec driver's I2C interface is initialized before snd_soc_new_pcms() is called, and you avoid have PCM calls in the codec driver. >> ASOC and the machine driver should then work in tandem to decide which driver will do what >> and which capabilities are *actually* supported. *Something* needs to look at the entire >> system and say to each device, "Well, yes, I know about this little feature of yours, but >> we're just not going to support that today." >> > > The machine driver pretty much already does this. It can override valid > hardware configurations and return -EINVAl if required. Do you have an example of that? Would that be eti_b1_hw_params()? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [1/3] 2.6.22-rc3: known regressions v2
by Michal Piotrowski 01 Jun '07

01 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : build failed in function `vic_sys_interrupt' References : http://lkml.org/lkml/2007/5/31/227 Submitter : Toralf Förster <toralf.foerster(a)gmx.de> Status : Unknown Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown Networking Subject : b44: regression in 2.6.22 References : http://lkml.org/lkml/2007/5/27/108 Submitter : Maximilian Engelhardt <maxi(a)daemonizer.de> Handled-By : Michael Buesch <mb(a)bu3sch.de> Status : problem is being debugged Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Configure problems for alsa-utils-1.0.4
by David Topper 01 Jun '07

01 Jun '07

So I've managed to compile alsa-drivers-1.0.4 for ARM. I can even load the modules. I've also successfully compiled alsa-libs-1.0.4 ... but I had to do it on the board. Cross compiling simply didn't seem to work. I've been using --prefix=/ and --exec-prefix=/ for configures. But when I do ./configure --prefix=/ --exec-prefix=/ for alsa-utils-1.0.4 it fails with: checking for libasound headers version >= 0.9.0... not present. configure: error: Sufficiently new version of libasound not found. But they're there, for sure. Perhaps it doesn't like looking in /include/alsa? Thanks, DT

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[alsa-devel] snd_soc_dai_link.init()
by Timur Tabi 01 Jun '07

01 Jun '07

Liam, Why does the init() function pointer in the snd_soc_dai_link structure take a snd_soc_codec pointer? Wouldn't it make more sense for it to take a snd_soc_dai_link pointer? That way, I could use the init() function to initialize the structure. I'm trying to write a function that will initialize the codec_dai pointer. I thought I could use snd_soc_dai_link.init(), but I don't know how to get the pointer to the snd_soc_dai_link structure from a snd_soc_codec structure. /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_dai_link { char *name; /* Codec name */ char *stream_name; /* Stream name */ /* DAI */ struct snd_soc_codec_dai *codec_dai; struct snd_soc_cpu_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); }; -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Power management state flow in ALSA
by Nobin Mathew 01 Jun '07

01 Jun '07

I am having a doubt regarding ALSA power management. My understanding of APM suspend() is like this. Freeze the ALSA apps Call ALSA driver suspend () in the ALSA suspend() function it saves the current state of substream and changes the state of substream to SUSPENDED. My understanding of APM resume() is like this Call ALSA driver resume () Activate the ALSA apps In ALSA resume function it restores the saved state of substream. So my question is when ALSA app will get to know the SUSPENDED state of substream.??? I am using linux kernel 2.6.17 ALSA driver version 1.0.11rc4 ASoC patch version 0.12.4

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[alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 31 May '07

31 May '07

I'm working on some ASoC drivers for a new board, and we're using a CS4270. The CS4270 is interesting in that if it is connected in stand-alone mode, there is no way to configure it. The board wirings determine all the parameters. Therefore, my CS4270 codec driver will probably be very skimpy. My question is: does this mean that my cs4270.c file will *never* call these functions: snd_ctl_add snd_soc_cnew snd_soc_dapm_new_control snd_soc_dapm_connect_input snd_soc_dapm_new_widgets These are the functions used to add new controls and widgets. So how does ALSA know that it needs to call my I2S driver *instead* of my codec driver to do stuff like change volume? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Will the real alsa wiki please stand up?
by j t 31 May '07

31 May '07

There something at: http://bugtrack.alsa-project.org/main/index.php/Main_Page which thinks it's the alsa wiki, and there something at: http://alsa.opensrc.org/Main_Page which alsa thinks it's the alsa wiki. Well, devs, which is the imposter? And would it be a good idea to reduce the number of distinct alsa wikis to something smaller, perhaps "1"? Jaime :-) PS: apologies to "To Tell the Truth".

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[alsa-devel] Native Instruments - Audio Kontrol 1 - snd_usb_caiaq module
by Alex Pintilie 31 May '07

31 May '07

Hallo! I just registered to this mailing list because a developer kindly advised me to do so. I own the external sound card Audio Kontrol 1 from Native Instruments and I was happy to find that drivers for it are already in the making. I downloaded alsa-driver-1.0.14rc4 and compiled successfully with ./hgcompile, then I installed the module with: # install -m 644 snd-usb-caiaq.ko /lib/modules/`uname -r`/kernel/drivers/snd-usb-caiaq.ko # /sbin/depmod -a After the command: # modprobe snd-usb-caiaq I get the following error: FATAL: Error inserting snd_usb_caiaq (/lib/modules/2.6.20-15-lowlatency/kernel/drivers/snd-usb-caiaq.ko): Unknown symbol in module, or unknown parameter (see dmesg) dmesg shows the following: [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_new [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_new [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_limit_hw_rates [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_limit_hw_rates [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_hidden_kfree [10095.331000] snd_usb_caiaq: Unknown symbol snd_verbose_printk [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_set_ops [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_set_ops [10095.333000] snd_usb_caiaq: Unknown symbol snd_hidden_kmalloc [10095.333000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_period_elapsed [10095.333000] snd_usb_caiaq: Unknown symbol snd_pcm_period_elapsed This may not look like much but it's the effort from a couple of days searching, reading and testing. I hope you can help me. -- Alex

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[alsa-devel] Hardware-accelerated OpenAL support with ALSA?
by Chris Rankin 31 May '07

31 May '07

Hi, I have been looking at getting OpenAL working on my machine, since OpenAL is the only way that games under Windows Vista can now provide hardware-accelerated sound effects. This suggests that a lot of PC games could be adding OpenAL support soon, which means that it would be handy if Linux had an efficient native OpenAL implementation that Wine could wrap. Fedora Core does provide an OpenAL library from www.openal.org, but it looks as if this library's ALSA backend does not provide any hardware acceleration. (I have an "Audigy 2 ZS" sound card, which can at least support hardware mixing.) Various alternative ALSA-based solutions do seem to exist on the Web, e.g. http://dino.e4a.it/openal-alsa/ and http://www.lost.org.uk/openal.html However, it seems to me that OpenAL support would be better served as an actual ALSA package that distros would be able to ship. Are there any plans to provide official OpenAL support in ALSA, please? Thanks for any information, Chris P.S. My Audigy 2 also seems to have EAX support. Does the ALSA driver support that functionality too? I can't say that "World of Warcraft" seems to use it, although Wine's DirectSound3D support still has lots of gaps. _________________________________________________________ Inbox cluttering up with junk? Clean up with Yahoo! Mail. http://uk.docs.yahoo.com/mail/isp_targeting.html

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[alsa-devel] Userspace Device Driver
by Klaus Zimmermann 31 May '07

31 May '07

Greetings List, on various places on the alsa site I read about the possibility of writing a userspace device driver. I am interested thatin, since I have a nice Edirol FA-66 ieee1394 interface which works nicely under jack with the ffado (http://ffado.org , former freebob) driver. >From what I read it seems I could either write an external plugin to the alsa-lib, which would be truly userspace and fine except I it would always have to be configured from .asoundrc and would probably not be found by legacy programs like ekiga. Or I could get really messy and have a userspace daemon relay audio data between the ffado library, i.e. librawieee1394, and a yet to be written alsa-driver module. This would at least yield the advantage, that the driver was indistinguishable from a true hardware driver. So: Is it possible to write a userspace driver that is indistinguishable from a hardware driver? TIA Klaus Zimmermann P.S.: I am aware of the jackplug plugin. However even with very conservative buffer settings I get underruns and x-runs quickly leading to a failing jack :(

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Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards
by Jose Henrique Spahn Torres 30 May '07

30 May '07

-----Original Message----- From: Zoltan Devai [mailto:zdevai@gmail.com] Sent: sexta-feira, 25 de Maio de 2007 14:40 To: Jose Henrique Spahn Torres Subject: Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards 2007/5/25, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > > 2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > usbcore: registered new interface driver hci_usb Advanced Linux Sound > Architecture Driver Version 1.0.14rc4 (Wed May 16 > 09:45:46 2007 UTC). > ASoC version 0.13.1 > SMDK2440 ALSA SoC Audio driver > UDA1341 SoC Audio Codec > L3 write > L3 transfer > L3 sendbyte: 16 > L3 sendbyte: 40 > asoc: UDA1341 <-> s3c24xx-i2s mapping ok ALSA device list: > #0: SMDK2440 (UDA1341) > TCP cubic registered > ... Fine. > However, when triyng to play something I get this: > > asoc: machine hw_params failed > > This is in sound/soc/soc-core.c, line 494. > I couldn't go any futher then this. > Any hint? This combination of processor, crystal and codec only supports the following rates: 11.025 Khz, 22.050Khz and 44.100 Khz Please try to play a file with these sample rates. (If you have alsa-tools, you could try 'speaker-test -r 44.100'). BTW, could you tell me the freq of the crystal on the processor board ? (Is it 16.934 Mhz?) Zoltan Hi, I used speaker-test as indicated above and the result is: pendrive $ ./speaker-test -r 44100 speaker-test 1.0.11 Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory I have also tried: /pendrive $ ./aplay -f cd 01-jack_johnson-times_like_these-rns.wav ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:547: audio open error: No such file or directory /pendrive $ so I don't know what is happening here. Does anyone have any idea? Thanks, Henrique

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[alsa-devel] Fw: hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Begin forwarded message: Date: Tue, 29 May 2007 15:19:12 -0400 From: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> To: stanl(a)cox.net, alsa-devel(a)alsa-project.org Subject: Re: hardware not asking for more data using asyn call back Hi, Thanks Stan, for your response - Instead of copying pcm data from a wav file, i decided to copy all the data to a master buffer first and then see if the circular buffer implementation works - so now instead of copying from the wave file, i m copying data from the master buffer to the circular buffer.. however, as per your suggestion, if i copy the data in the while loop in main, then, it might not always be copied between 2 consecutive callbacks, but maybe more, as the callbacks are asynchronous. however, the hardware still does not ask for more data after executing the callback function about twice -- is there a way to flush the hardware buffer before beginning playback? I have pasted my code below Regards, Ashlesha. <snip> I've been doing some thinking about what you are trying to accomplish and have a suggestion. I think you would be better off letting jack handle the back end/sound portion. Run a daemon/service listening on an arbitrary tcp/ip port. On startup it initializes alsa and jack. When it receives a request, it forks a process which opens a new channel on jack and creates a buffer for it. It then plays the sound it receives on that link via jack, silence when the buffer is empty. You would have to write client software for the user on the network to contact your daemon/service. You might even find something like this already out there.

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[alsa-devel] How the dma interrupt handlers have been registered in ALSA?
by Pharaoh . 30 May '07

30 May '07

Hi I am referring omap-alsa-dma.c and the related codec files. We usually register the IRQ handler by calling request_irq() function but in this mentioned files the IRQ handler etc is defined but it is not registered anywhere i.e. request_irq is not used anywhere. Is there any other way of registering irq handlers? The irq handlers address is passed to omap_request_dma, but there also it is not registered. I am missing the entire thing?

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[alsa-devel] [1/4] 2.6.22-rc3: known regressions
by Michal Piotrowski 30 May '07

30 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : Oops in dentry_iput with 2.6.22-rc2 on AMD64 References : http://lkml.org/lkml/2007/5/22/4 Submitter : Florin Iucha <florin(a)iucha.net> Status : Unknown Kbuild Subject : make M=$PWD modules_install does nothing References : http://lkml.org/lkml/2007/5/27/190 Submitter : Andrey Borzenkov <arvidjaar(a)mail.ru> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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Re: [alsa-devel] hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Somehow I replied to the wrong message and it went to a personal email address. Putting it on the list so searches can find it (for what that is worth :-) ) Begin forwarded message: Date: Tue, 29 May 2007 18:18:22 -0700 From: stan <stanl(a)cox.net> To: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> Subject: Re: hardware not asking for more data using asyn call back On Tue, 29 May 2007 15:19:12 -0400 "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> wrote: <snip> I didn't understand what you were asking before. It sounds to me like you want to have a thread running that accepts input from arbitrary processes and puts it in the main buffer. Then when the alsa callback occurs you want to take any new sound samples and send them to alsa. If that is the case, it makes more sense for the main buffer to be the circular buffer and the alsa buffer to be just a regular buffer. That's just personal preference though. As far as the handle error, I don't know what it means beyond what it says. You would have to look in the sources for the alsa library to see what causes that error to be generated. I don't see anything wrong in the logic of the code below, but the output indicates that you are not keeping up with the play rate. This is astonishing. At each play you can see that the available room in the buffer grows larger until larger than 1000, the buffer you have allocated. I'm not sure what infinite loop means and how your program is recovering from it. I presume that is where the time is being eaten up that isn't allowing you to keep up with the requests for data. I don't think you need to check if there is room when you get a callback. You should be able to just write the data to alsa using the writei call. i.e. eliminate the snd_pcm_avail_update call. I'm not familiar enough with alsa to help you further. Perhaps someone else can give you a hint or point you to the right place to look. > > /* Reading wav file into a buffer and then trying to play it */ > > > > #include <stdio.h> > > #include </usr/include/alsa/asoundlib.h> > > #include </usr/include/alsa/pcm.h> > > #include <sys/types.h> > > #include <unistd.h> > > #include <fcntl.h> > > > > #define SIZE 128 > > //mpos is the position in the master buffer to write from > > short int *buffer, *mbuf, *mpos; > > int ind=10, fd1, length; > > static snd_pcm_t *handle; > > static char *device = "default"; /* playback > > device */ > > snd_output_t *output = NULL; > > static short int *wto, *rfrom, buflen; > > > > void MyCallback(snd_async_handler_t *pcm_callback); > > > > int main() > > { > > int i,j,n,count,a,err,k=0; > > char c; > > snd_pcm_sframes_t frames; > > snd_async_handler_t *pcm_callback; > > > > > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > > printf("error opening wav file\n"); > > > > count=0; > > while(count++<40) > > a=read(fd1,&c,sizeof(char)); > > > > a=read(fd1,&length,sizeof(int)); > > > > n=length/SIZE; > > > > printf("length = %d\n",length); > > mbuf = (short int *) malloc (length * sizeof(short int)); > > mpos = &mbuf[0]; > > > > buflen = 4*SIZE; > > buffer = (short int *) malloc (buflen*sizeof(short int)); > > > > wto = NULL; > > rfrom = &buffer[0]; > > > > /* reading to master buffer */ > > count=0; > > a=1; > > while(count<length && a>0) > > a=read(fd1,&mbuf[count++],sizeof(short int)); > > if(a<0){ > > printf("error in reading from wav file\n"); > > exit(1); > > } > > close (fd1); > > > > for(count =0; count < buflen; count++) > > { > > buffer[count]=mbuf[count]; > > } > > /******************************/ > > > > /*ALSA DEV INIT*/ > > if((err = snd_pcm_open(&handle, device, > > SND_PCM_STREAM_PLAYBACK, 0)) < 0) { > > printf("error opening device: %s\n",snd_strerror(err)); > > return -1; > > } > > > > if((err = snd_pcm_set_params(handle, > > SND_PCM_FORMAT_S16_LE, > > SND_PCM_ACCESS_RW_INTERLEAVED, > > 1, > > 16000, > > 1, > > 1000)) < 0) { /* 2 sec */ > > printf("Playback open error: %s\n", snd_strerror(err)); > > return -2; > > } > > /*****************************/ > > > > printf("n=%d\n",n); > > for (i=0; i<50; i++) > > { > > printf("i=%d\n",i); > > frames = snd_pcm_writei(handle, rfrom, SIZE); > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_recover(handle, frames,0); > > if(frames < 0){ > > printf("error in recovery\n"); > > return -3; > > } > > } > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote %d\n",SIZE,frames); > > > > printf("distance between ptrs before reinit is %d\n",rfrom > > - wto); if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > printf("rfm at the end, rfrom -buffer[buflen-1] > > =%d\n",(*rfrom - buffer[buflen-1])); > > //wto = rfrom; > > rfrom = &buffer[0]; > > } > > > > if((a = writetobuf())<0) > > printf("error in reading from wav file k = %d\n"); > > > > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > > *rfrom)); > > > > } > > > > /* Async Handler */ > > > > > > err = > > snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > > if(err<0) printf("add pcm handler error = > > %d\n%s\n",err,snd_strerror(err)); > > > > err = snd_pcm_start(handle); > > if(err<0) > > printf("error in starting snd pcm start err > > :%s\n",snd_strerror(err)); > > /******************************/ > > > > while (1) { > > > > if(wto == NULL) > > break; // redundancy -- already checking > > for this condition in MyCallback > > printf("infinite loop\n"); > > sleep(1); > > } > > > > err = snd_pcm_close(handle); > > if(err<0) > > printf("error in closing pcm device: > > %s\n",snd_strerror(err)); > > > > return 0; > > } > > > > void MyCallback(snd_async_handler_t *pcm_callback) > > { > > > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > > snd_pcm_sframes_t avail; > > int count,a; > > snd_pcm_uframes_t period_size = 64; > > snd_pcm_sframes_t frames; > > > > if((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) ^ From the alsa lib api docs. | | Using of this function is useless for the standard read/write operations. Use it only for mmap access. See to snd_pcm_delay. Obtain delay for a running PCM handle. Parameters: pcm PCM handle delayp Returned delay in frames Returns: 0 on success otherwise a negative error code Delay is distance between current application frame position and sound frame position. It's positive and less than buffer size in normal situation, negative on playback underrun and greater than buffer size on capture overrun. > > { > > printf("available frames = %d\n",avail); > > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); Using rfrom is going to be problematic if you get to a > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_prepare(pcm_handle); > > printf("error from snd_pcm_prepare is: > > %d\n",frames); if(frames < 0){ > > printf("error in recovery\n"); > > > > } > > } > > > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote > > %d\n",SIZE,frames); > > > > > > if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > // wto = rfrom; > > rfrom = &buffer[0]; > > } //location of where to start > > writing from next > > > > if(writetobuf()<0){ > > wto=NULL; > > exit(0); // returns 0 to the operating system -- so > > it ends the program! > > } > > ++ind; > > } > > > > printf("going out of callback, ind = %d\n",ind); > > } > > > > /*Function to copy data from master buffer (mbuf) to circular buffer > > (buffer)*/ > > int writetobuf() > > { > > int count=-1; > > > > while((++count < SIZE) && (mpos < &mbuf[length])) > > *(mpos++) = *(wto+count); > > > > if(mpos >= &mbuf[length-1]) > > return -1; > > else > > return 0; > > } > > i=49 error in starting snd pcm start err :File descriptor in bad state infinite loop inside callback, available frames = 364 going out of callback, ind = 1 infinite loop inside callback, available frames = 577 going out of callback, ind = 2 infinite loop inside callback, available frames = 790 underrun recovery error from snd_pcm_prepare is: 0 going out of callback, ind = 3 infinite loop infinite loop infinite loop

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Re: [alsa-devel] hardware not asking for more data using asyn call back
by Ashlesha Shintre 29 May '07

29 May '07

Hi, Thanks Stan, for your response - Instead of copying pcm data from a wav file, i decided to copy all the data to a master buffer first and then see if the circular buffer implementation works - so now instead of copying from the wave file, i m copying data from the master buffer to the circular buffer.. however, as per your suggestion, if i copy the data in the while loop in main, then, it might not always be copied between 2 consecutive callbacks, but maybe more, as the callbacks are asynchronous. however, the hardware still does not ask for more data after executing the callback function about twice -- is there a way to flush the hardware buffer before beginning playback? I have pasted my code below Regards, Ashlesha. /* Reading wav file into a buffer and then trying to play it */ > > #include <stdio.h> > #include </usr/include/alsa/asoundlib.h> > #include </usr/include/alsa/pcm.h> > #include <sys/types.h> > #include <unistd.h> > #include <fcntl.h> > > #define SIZE 128 > //mpos is the position in the master buffer to write from > short int *buffer, *mbuf, *mpos; > int ind=10, fd1, length; > static snd_pcm_t *handle; > static char *device = "default"; /* playback device > */ > snd_output_t *output = NULL; > static short int *wto, *rfrom, buflen; > > void MyCallback(snd_async_handler_t *pcm_callback); > > int main() > { > int i,j,n,count,a,err,k=0; > char c; > snd_pcm_sframes_t frames; > snd_async_handler_t *pcm_callback; > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > printf("error opening wav file\n"); > > count=0; > while(count++<40) > a=read(fd1,&c,sizeof(char)); > > a=read(fd1,&length,sizeof(int)); > > n=length/SIZE; > > printf("length = %d\n",length); > mbuf = (short int *) malloc (length * sizeof(short int)); > mpos = &mbuf[0]; > > buflen = 4*SIZE; > buffer = (short int *) malloc (buflen*sizeof(short int)); > > wto = NULL; > rfrom = &buffer[0]; > > /* reading to master buffer */ > count=0; > a=1; > while(count<length && a>0) > a=read(fd1,&mbuf[count++],sizeof(short int)); > if(a<0){ > printf("error in reading from wav file\n"); > exit(1); > } > close (fd1); > > for(count =0; count < buflen; count++) > { > buffer[count]=mbuf[count]; > } > /******************************/ > > /*ALSA DEV INIT*/ > if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < > 0) { > printf("error opening device: %s\n",snd_strerror(err)); > return -1; > } > > if((err = snd_pcm_set_params(handle, > SND_PCM_FORMAT_S16_LE, > SND_PCM_ACCESS_RW_INTERLEAVED, > 1, > 16000, > 1, > 1000)) < 0) { /* 2 sec */ > printf("Playback open error: %s\n", snd_strerror(err)); > return -2; > } > /*****************************/ > > printf("n=%d\n",n); > for (i=0; i<50; i++) > { > printf("i=%d\n",i); > frames = snd_pcm_writei(handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_recover(handle, frames,0); > if(frames < 0){ > printf("error in recovery\n"); > return -3; > } > } > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > > printf("distance between ptrs before reinit is %d\n",rfrom - wto); > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > if(rfrom >= &buffer[buflen-1]) > { > printf("rfm at the end, rfrom -buffer[buflen-1] > =%d\n",(*rfrom - buffer[buflen-1])); > //wto = rfrom; > rfrom = &buffer[0]; > } > > if((a = writetobuf())<0) > printf("error in reading from wav file k = %d\n"); > > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > *rfrom)); > > } > > /* Async Handler */ > > > err = snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > if(err<0) > printf("add pcm handler error = %d\n%s\n",err,snd_strerror(err)); > > err = snd_pcm_start(handle); > if(err<0) > printf("error in starting snd pcm start err > :%s\n",snd_strerror(err)); > /******************************/ > > while (1) { > > if(wto == NULL) > break; // redundancy -- already checking for this > condition in MyCallback > printf("infinite loop\n"); > sleep(1); > } > > err = snd_pcm_close(handle); > if(err<0) > printf("error in closing pcm device: %s\n",snd_strerror(err)); > > return 0; > } > > void MyCallback(snd_async_handler_t *pcm_callback) > { > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > snd_pcm_sframes_t avail; > int count,a; > snd_pcm_uframes_t period_size = 64; > snd_pcm_sframes_t frames; > > if((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) > { > printf("available frames = %d\n",avail); > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_prepare(pcm_handle); > printf("error from snd_pcm_prepare is: %d\n",frames); > if(frames < 0){ > printf("error in recovery\n"); > > } > } > > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > > > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > if(rfrom >= &buffer[buflen-1]) > { > // wto = rfrom; > rfrom = &buffer[0]; > } //location of where to start writing > from next > > if(writetobuf()<0){ > wto=NULL; > exit(0); // returns 0 to the operating system -- so it > ends the program! > } > ++ind; > } > > printf("going out of callback, ind = %d\n",ind); > } > > /*Function to copy data from master buffer (mbuf) to circular buffer > (buffer)*/ > int writetobuf() > { > int count=-1; > > while((++count < SIZE) && (mpos < &mbuf[length])) > *(mpos++) = *(wto+count); > > if(mpos >= &mbuf[length-1]) > return -1; > else > return 0; > } >

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[alsa-devel] CMI8788............
by Khalid Paden 29 May '07

29 May '07

Does anyone have a driver for the CMI8788 chipset? AKA The X-Meridian? -- Khalid Paden _______________________________________________________________________ So that's it?!?!?! Not quite. Right now he is taking the safety's off all his weapons. He won't make the same mistake twice. -Hunt for Red October

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[alsa-devel] Reasonable sound hardware parameters?
by Lorenz Kolb 29 May '07

29 May '07

> Message: 4 > Date: Mon, 28 May 2007 10:52:18 +0200 (CEST) > From: Jaroslav Kysela <perex(a)suse.cz> > Subject: Re: [alsa-devel] Reasonable sound hardware parameters? > To: Joachim F?rster <mls.JOFT(a)gmx.de> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <Pine.LNX.4.61.0705281045310.7894(a)tm8103.perex-int.cz> > Content-Type: text/plain; charset="iso8859-2" > > On Sat, 26 May 2007, Joachim F?rster wrote: > > > > > First question: To implement mmap on such kind of device IO memory, > > I have to use snd_pcm_lib_mmap_iomem() as mmap() callback - like it > > is used in the rme32 driver, right? > > Yes. rme32 is quite similar. > > > Second question: Do the following parameters make sense regarding > > the size of the HW ring buffer and period sizes: > > > > .buffer_bytes_max=16*1024; > > .period_bytes_max=8*1024; > > .period_bytes_min=?? resonable value ?? > > 32 or 64 > > > .periods_min=2; > > .periods_max=?? resonable value ?? > > 16 > > > Furthermore, the sound controller will fire an interrupt after one > > period is played. Are there any arguments against these values? Any > > comments? > > Yes, I think that the access through a PCI I/O window to a PCI device > is much slower than PCI DMA transfers with bursts. I would really > consider to design hardware with real DMA. Of course, if it's only > experimental stuff, then don't worry. > > Jaroslav > > ----- > Jaroslav Kysela <perex(a)suse.cz> > Linux Kernel Sound Maintainer > ALSA Project, SUSE Labs Hi Jaroslav, First of all: thanks for Your answer. Second: We do not have a PCI Bus. It is an embedded system with an OCP-Bus (aka OPB). The ML403 holds a SoC with PPC and FPGA. So the only difference between DMA with bursts and a dedicated memory (as far as we do see it) is the amount of time the bus gets occupied. With dedicated memory the system will (hopefully ;-)) do a burst write each period. So what happens in DMA case (except the more complex hardware as the DMA controller has to be written (in VHDL) as well and even more memory is located at an more complex bus (PLB - processor local bus): the bus will be occupied twice: (1) CPU writes data to memory. --> start DMA --> (2) Controller fetches data from memory. While (2) is in progress the poor CPU is "disconnected" from the bus. In an embedded system with a not too powerful CPU (PPC 405 @300MHz) this might be quite inefficient. Or did we miss something? Regards, Lorenz -- Lorenz Kolb <alsa-dev(a)lkmail.de> Embedded specialist ESIC-Solutions

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[alsa-devel] SigmaTel STAC9221 - Mux Amp-Out > 0x02 == microphone not working
by Ivan N. Zlatev 29 May '07

29 May '07

Hello, I am on an iMac ("*imac-20" model/pinconfig) with a Intel HD card with the SigmaTel STAC9221 codec. I have attached the inital codec info after modprobe. I am using Alsa HG. Setting the mux gain to a value > 20 (0x02 Amp-Out value) results a non-working microphone. Steps and codec diffs as follows: 1) initial - microphone working 2) Mux Gain = 10 - microphone working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x00 0x00] + Amp-Out vals: [0x01 0x01] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 3) Mux Gain = 20 - microphone working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x01 0x01] + Amp-Out vals: [0x02 0x02] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 4) Mux Gain = 30 - microphone not working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x02 0x02] + Amp-Out vals: [0x03 0x03] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a 4) Mux Gain = 40 - microphone not working Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 - Amp-Out vals: [0x03 0x03] + Amp-Out vals: [0x04 0x04] Connection: 7 0x0e 0x15 0x0f 0x0b* 0x0c 0x0d 0x0a Regards -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] Intel HD SigmaTel STAC9221 - Wrong input after output change
by Ivan N. Zlatev 29 May '07

29 May '07

Hello, I am on an iMac ("*imac-20" model/pinconfig) with a Intel HD card with the SigmaTel STAC9221 codec. I have attached the inital codec info after modprobe. When I plug in and then unplug my speakers the microphone stops working. Reproduction steps with codec dump diffs as follows 1) Initial codec info - attached - Microphone working 2) Plug in speakers - Microphone working Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x03 3) Unplug speakers - Microphone not working Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown - Pin-ctls: 0x44: OUT + Pin-ctls: 0x04: Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Connection: 1 0x03 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0181e020: [Jack] Line In at Ext Rear Conn = 1/8, Color = White - Pin-ctls: 0x40: OUT + Pin-ctls: 0x00: Connection: 1 0x05 4) Plug in speakers again - Microphone not working Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown - Pin-ctls: 0x04: + Pin-ctls: 0x44: OUT Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x9017e110: [Fixed] Speaker at Int N/A Conn = Analog, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x03 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x0181e020: [Jack] Line In at Ext Rear Conn = 1/8, Color = White - Pin-ctls: 0x00: + Pin-ctls: 0x40: OUT Connection: 1 0x05 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] Question on snd_soc_dai_link
by Timur Tabi 29 May '07

29 May '07

I have a question about these DAI structs for ASoC. Let's take eti_b1_wm8731.c as an example. It contains this structure: static struct snd_soc_dai_link eti_b1_dai = { .name = "WM8731", .stream_name = "WM8731", .cpu_dai = &at91_i2s_dai[1], .codec_dai = &wm8731_dai, .init = eti_b1_wm8731_init, .ops = &eti_b1_ops, }; Notice that the .cpu_dai field is hard-coded to use the 2nd I2S device on the AT91. The problem I'm having with this approach is that everything is hard-coded. On PowerPC, this approach doesn't work well, because we have a "device tree" that dictates what devices are present on the SOC. For a Freescale SOC that has two I2S devices (for example), both devices would be listed in the device tree, with the base address of the memory-mapped registers. This means that a Freescale I2S driver would have its 'probe' function called (much like PCI does) each time the kernel processes an I2S node in the device tree. This means that when the machine driver (which has the snd_soc_dai_link structure) loads, there's no guarantee that the I2S driver has also loaded and the probe functions have been called. It gets even more complicated if only one I2S device has a codec connected to it, because then I need to kmalloc the snd_soc_cpu_dai structures and somehow tell the snd_soc_dai_link structure which one to use. Has any thought been given to making these structures more dynamic and handling interdependencies among the drivers? As it stands now, it looks like I'm going to have to hard-code everything in the drivers and break the device tree paradigm. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] Sigmatel STAC9221 - Add Configs for all Apple Intel Macs
by Ivan N. Zlatev 29 May '07

29 May '07

This patch: * adds the pinconfigs for all 5 Apple boards and 14 Subsystem IDs (support for possibly all iMac, Mac, MacMini etc etc) * adds "intel-mac-v1" to v5 models which replace the current * reflects changes in Alsa-Configuration.txt Please review. Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

> ----- Original Message ----- > From: "Takashi Iwai" <tiwai(a)suse.de> > To: "Ash Willis" <ashwillis(a)programmer.net> > Subject: Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver > Date: Tue, 29 May 2007 15:08:30 +0200 > > > At Sat, 26 May 2007 13:26:35 +0000, > Ash Willis wrote: > > > > Disables debugging output in the ALS300 driver. > > Also contains a whitespace cleanup and a fix for a potential bug. > > > > Signed-off-by: Ash Willis <ashwillis(a)programmer.net> > > Thanks for the patch. It seems that your MUA breaks the embedded > patch, so I had to apply it manually. Could you check whether the > latest HG is OK for you? I'm away from home right now. I'll test it as soon as I have a chance but it looks fine. > > Also I changed "if ((err = ...))" to two individual lines, which is a > newer standard kernel coding style. > Yeah, I left that for now. I'll go through the tree and tidy up remaining occurences of this style when I have some time. It seems that when people add drivers to the tree, they generally copy already existing drivers. It doesn't make much sense to be copying code that doesn't fit CodingStyle. Ash = Sports Memorabilia License Plates Signs All license plates $6 all signs $10 all frames $8 - same day shipping - fun company and selection. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=a44f49e42c5a283d05f0c96…

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[alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

Disables debugging output in the ALS300 driver. Also contains a whitespace cleanup and a fix for a potential bug. Signed-off-by: Ash Willis <ashwillis(a)programmer.net> ---------------------------------------------------- diff -r 337962e19de6 -r 8ea4fa014563 pci/als300.c --- a/pci/als300.c Fri May 25 12:18:49 2007 +0200 +++ b/pci/als300.c Sat May 26 14:04:14 2007 +0000 @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -730,10 +730,9 @@ static int __devinit snd_als300_create(s } chip->irq = pci->irq; - snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + if ((err = snd_als300_ac97(chip)) < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err;

2 1

[alsa-devel] sound on smdk2410
by Jose Henrique Spahn Torres 29 May '07

29 May '07

Hi, I am trying recent Zoltan Devai's patchs for kernel 2.6.22-rc2 on a smdk2410 board. I reached the point where the kernel recognized the uda1341 sound chip, however, when I try to play something, I get the output below. Can anyone, please, show me the path to follow to get sound working? Thanks, Henrique /home $ l drwxr-xr-x 3 root root 0 Jan 1 00:00 . drwxr-xr-x 18 root root 1024 May 18 2007 .. -r--r--r-- 1 root root 3419671 Jan 1 1970 jj.mp3 /home $ mpg123 jj.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 anasoc: UDA1341 <-> s3c24xx-i2s info: asoc: rate mask 0xfe asoc: min ch 2 max ch 2 asoc: min rate 8000 max rate 48000 version 0.65; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes asoc: machine hw_params failed ... (lots of asoc: messages) ... asoc: machine hw_params failed pop wq checking: Playback status: inactive waiting: yes pop wq D3 UDA1341 Playback Playing MPEG stream 1 of 1: jj.mp3 ... Title: Times Like These Artist: Jack Johnson Comment: Album: On And On Year: 2003 Genre: Acoustic MPEG 1.0 layer III, 192 kbits/s, 44100 Hz stereo [audio.c:264] error: No supported rate found! /home $

3 3

[alsa-devel] HDA Codec: Add support for MSI K9N Ultra
by Tobin Davis 29 May '07

29 May '07

This patch adds the MSI K9N Ultra system to the realtek patch. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] ICE1724 alsa driver
by Rossco 29 May '07

29 May '07

Hi, I hope someone can help me. (tried the user list but no response) I have a hitpoint 7.1ch Envy24ht-s based card with the usual vt1616 multichannel codec and the wolfson WM8728. The card is recognised as a chaintech av710 with the only difference being that the card has spdif in as well as out. It appears to be physically similar to the maddog entertainer (would be good to know if this is also recognised as an AV710) By default 2 channel stereo in played through the "front" channel of the card through the vt1616 codec and I appear to be able to set spdif out through the mixer (but dont have another device to test at the moment). I have been able to change default output to go through the wolfson dac through an asound.state file posted on the web. I have to admit I do not quite understand what the file has done as it only appears to have removed some of the controls rather than change or add anything. Only two devices are detected for the card the vt1616 and IEC958, neither are shown as having any subdevices, therefore I am unsure how the asound.state file was able to address the wolfson chip which requires i2s input rather ac-link. This seems to indicate i2s and ac-link modes of the envy24ht-s chip are active. Most AV710 users seem happy enough to be able to route to toslink out and/or the higher quality wolfson chip, therefore there is little discussion of addressing the other channels of the card. I really want to be able to address each of the channels of the card so that I can use the convolution programme Brutefir with the card to create a multiway crossover for my loudspeakers. Is there anyway I can define each of the cards outputs so that I can select to send audio to any one of them? I am also keen to get the spdif in working and particularly the syncing to spdif in of the clock that the ice1724 chip supports. Can these features be added to the driver? Please advise of any required info, I have all the datasheets and can trace chip pins also. Thanks in advance. Ross

2 5

[alsa-devel] [PATCH] Add quirk for Supermicro PDSBA to alc883_cfg_tbl[]
by Daniel T. Chen 29 May '07

29 May '07

Attached is a diff against current alsa-kernel hg tip. Thanks, Daniel Chen

1 1

[alsa-devel] hardware not asking for more data using asyn call back
by Ashlesha Shintre 29 May '07

29 May '07

Hi, In my asynchronous playback program, the hardware stops asking for any data and the program gets stuck after one call to the callback function. I have pasted the program below -- please let me know if something is wrong - Thanks, Ashlesha. /* Reading wav file into a buffer and then trying to play it */ > > #include <stdio.h> > #include </usr/include/alsa/asoundlib.h> > #include </usr/include/alsa/pcm.h> > #include <sys/types.h> > #include <unistd.h> > #include <fcntl.h> > > #define SIZE 128 > > short int *buffer; > int ind=10, fd1; > static snd_pcm_t *handle; > static char *device = "default"; /* playback device > */ > snd_output_t *output = NULL; > static short int *wto, *rfrom, buflen; > > void MyCallback(snd_async_handler_t *pcm_callback); > > int main() > { > int i,j,n,length,count,a,err,k=0; > char c; > snd_pcm_sframes_t frames; > snd_async_handler_t *pcm_callback; > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > printf("error opening wav file\n"); > > count=0; > while(count++<40) > a=read(fd1,&c,sizeof(char)); > > a=read(fd1,&length,sizeof(int)); > > n=length/SIZE; > > printf("length = %d\n",length); > > buflen = 4*SIZE; > buffer = (short int *) malloc (buflen*sizeof(short int)); > > wto = NULL; > rfrom = &buffer[0]; > > count=0; > a=1; > while(count<buflen && a>0) > a=read(fd1,&buffer[count++],sizeof(short int)); > if(a<0) > printf("error in reading from wav file\n"); > > if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < > 0) { > printf("error opening device: %s\n",snd_strerror(err)); > return -1; > } > > if((err = snd_pcm_set_params(handle, > SND_PCM_FORMAT_S16_LE, > SND_PCM_ACCESS_RW_INTERLEAVED, > 1, > 16000, > 1, > 1000)) < 0) { /* 2 sec */ > printf("Playback open error: %s\n", snd_strerror(err)); > return -2; > } > printf("n=%d\n",n); > for (i=0; i<50; i++) > { > printf("i=%d\n",i); > frames = snd_pcm_writei(handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_recover(handle, frames,0); > if(frames < 0){ > printf("error in recovery\n"); > return -3; > } > } > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > printf("distance between ptrs before reinit is %d\n",rfrom - wto); > if(rfrom < &buffer[buflen-1]) > { > wto = rfrom; > rfrom = rfrom + SIZE; > } > else > { > printf("rfm at the end, rfrom -buffer[buflen-1] > =%d\n",(*rfrom - buffer[buflen-1])); > wto = rfrom; > rfrom = &buffer[0]; > } > for(k=0; k<SIZE; k++) > { > a=read(fd1,(wto+k),sizeof(short int)); > if(a<0) > printf("error in reading from wav file k = %d\n"); > } > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > *rfrom)); > > } > > /* Async Handler */ > > // err = > snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,buffer); > err = snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > if(err<0) > printf("add pcm handler error = %d\n%s\n",err,snd_strerror(err)); > > err = snd_pcm_start(handle); > if(err<0) > printf("error in starting snd pcm start err > :%s\n",snd_strerror(err)); > > while (1) { > if(wto == NULL) > break; > sleep(1); > } > > err = snd_pcm_close(handle); > if(err<0) > printf("error in closing pcm device: %s\n",snd_strerror(err)); > close(fd1); > return 0; > } > > void MyCallback(snd_async_handler_t *pcm_callback) > { > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > snd_pcm_sframes_t avail; > int count,a; > snd_pcm_uframes_t period_size = 64; > snd_pcm_sframes_t frames; > //short int *bufferin = > snd_async_handler_get_callback_private(pcm_callback); > > while((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) > { > printf("available frames = %d\n",avail); > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); > if(frames < 0) // underrun > { > printf("underrun recovery\n"); > frames = snd_pcm_prepare(pcm_handle); > printf("error from snd_pcm_prepare is: %d\n",frames); > if(frames < 0){ > printf("error in recovery\n"); > // break; > } > } > > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > count = 0; > a=1; > while(count<SIZE && a>0){ > a=read(fd1,(wto+count),sizeof(short int)); //write to > the buffer from file > ++count; > } > printf("count = %d\n",count); > if(a <0){ //EOF > wto = NULL; > printf("end of file reached\n"); > break; > } > else{ > if(rfrom < &buffer[buflen-1]) > { > wto = wto+count; > rfrom = rfrom + SIZE; > } > else > { > wto = wto + count; > rfrom = &buffer[0]; > } //location of where to start > writing from next > } > ++ind; > } > > printf("going out of callback, ind = %d\n",ind); > } > > >

2 1

[alsa-devel] async playback works despite error in snd_pcm_start.
by Ashlesha Shintre 28 May '07

28 May '07

Hi, I managed to write code to playback sound asynchronously -- however, this code works (i.e data is read from .wav file and output to the playback device) despite the snd_pcm_start giving a "file descriptor in bad state" error.. can someone explain this please? Thanks, Ashlesha.

1 0

Re: [alsa-devel] another ALSA question
by Jaroslav Kysela 28 May '07

28 May '07

I'm CCing alsa-devel because this discussion might be interesting for other developers, too. On Mon, 28 May 2007, Maarten Lankhorst wrote: > Jaroslav Kysela schreef: > > On Mon, 28 May 2007, Maarten Lankhorst wrote: > > > >> After playing around a little more with alsa I tried to do manipulation > >> of pointer using the instructions, I wrote some simple code either does > >> rewind or forward to get to the right point, taking in respect the > >> amount of delay and where to go. > >> > >> However it doesn't seem to work: Looking at the source code it seems > >> rewinding isn't implemented, using the #if 0'd code there gives problems > >> when doing snd_pcm_forward later: It then ends up at the pointer where > >> the sound is currently being played at. > >> > > > > Not all alsa-lib plugins are capable or have implemented this fine > > position control (check error value). > > > For a plugin like dmix, I would expect it to be implemented, especially > since most cards sold are integrated cards that don't have multiple > sound buffers. Go ahead and send us patches. We're doing open source :-) Anyway any rewind operation mens "undo" - it will eat extra CPU ticks for nothing. > >> As workaround I used the following in my own code: > >> typeof(snd_pcm_forward) snd_pcm_mmap_appl_forward; > >> #define snd_pcm_forward snd_pcm_mmap_appl_forward > >> > >> typeof(snd_pcm_rewind) snd_pcm_mmap_appl_backward; > >> #define snd_pcm_rewind snd_pcm_mmap_appl_backward > >> > >> This works, but I wonder if there is a cleaner way to do it. > >> > > > > It won't work with all plugins and mentioned functions are not part of > > public API. > > > As I work in wine on it's DirectSound implementation, I can tell it > relies heavily on the fact that it can lock a buffer at an arbitrary > position, and that it works with absolute pointers, rather then > relative. So if snd_pcm_rewind/snd_pcm_forward don't work as expected, > an alsa driver for it would become nearly useless, I don't expect it to > always work, but I would expect it to work for a common plugin like dmix > at least. > >> Also, I noticed some audio glitches when using this in combination with > >> alsa-lib's resampling, is that because of the way it is done or is > >> something else wrong? > >> > > > > You should not use mentioned functions. > > > > You can try to use the direct hardware access (hw:CARD_NAME) devices > > (you'll lose all stream conversions from the ALSA side) or use an > > intermediate buffer for fine position control and configure ALSA to use > > smallest ring buffer as possible (to minimalize latency). > > > > Jarosla > I wish that would work, directsound expects to have access to a fixed > size buffer, and if I go for direct hardware access, I lose the ability > to work properly with most cheap cards on the market, since people want > to be able to listen to music on background while gaming. For now it > seems I have to use the internal calls in case of dmix, or modify > already committed data, which is what winealsa's current driver does. I > don't know what is the least of those 2 evils, or if there is a third > option. I'm open for advice. I'm afraid, that you're trying to exactly copy DirectSound API. Playing with the buffer position in this way is always dangerous, because it's really hard to predict the final result in the multitasking environment. Think about these proposals: 1) what sample (ring buffer) rewritting is supposed to do - maybe you can change only not yet commited samples with satisfactory results 2) use a small ALSA ring buffer and use own "big" ring buffer for changes before you commit samples to ALSA's ring buffer - you'll increase a bit sample flow latency, but you can do anything you want (you just add extra buffering to avoid rewind / forward for ALSA API) 3) fix (or better implement) dmix plugin rewind function Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

1 0

[alsa-devel] [ALSA] ALSA Power Management, Drivers behaving unexpectedly after suspend/resume cycle
by Nobin Mathew 28 May '07

28 May '07

I am implementing power management in my ALSA sound drivers. ALSA drivers are not working properly after resume. All registers values are proper and I am able to read back the codec register contents. If I do a playback after resume then I am getting lots of "underrun". After a reboot playback is working fine. Is there any known issues with ALSA power management? Why Ubuntu and Redhat does ALSA modules removal before suspend and insertion after resume?

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[alsa-devel] Reasonable sound hardware parameters?
by Joachim Förster 28 May '07

28 May '07

Hi ALSA devs, as I already described in one of my last posts on the list, I'm going to write a ALSA driver for a sound controller which will also be "written" at the same time (VHDL for FPGA) - on top of an embedded system (Xilinx ML403 EvalBoard, PowerPC 405). We will do playback feature only. The point is, that we will have _no_ real DMA and we don't want to use the copy/silence or ack/workqueue (with intermediate buffer). So we decided to move the ring buffer, which usually exists in RAM into the sound controller and handle it as some kind of device IO memory, which should be mmap'able (kernel function io_remap_pfn_range()). So, the important goal is, to have mmap feature in the ALSA driver. First question: To implement mmap on such kind of device IO memory, I have to use snd_pcm_lib_mmap_iomem() as mmap() callback - like it is used in the rme32 driver, right? Second question: Do the following parameters make sense regarding the size of the HW ring buffer and period sizes: .buffer_bytes_max=16*1024; .period_bytes_max=8*1024; .period_bytes_min=?? resonable value ?? .periods_min=2; .periods_max=?? resonable value ?? Furthermore, the sound controller will fire an interrupt after one period is played. Are there any arguments against these values? Any comments? Thanks, Joachim

2 1

[alsa-devel] alsa playback channel interleaving question
by Matthias Nyffenegger 27 May '07

27 May '07

Hi, Earlier this year I wrote to this list: > I am trying to write an alsa driver for my audiowerk8 soundcard. > The aw8 uses a Philips saa7146 chipset. There are 2 capture and > 8 playback channels mapped on the saa7146's DMA as follows: > > DMA1: 2xcapture > DMA2: 4xplayback > DMA3: 4xplayback > > Sample format is 16bit. > > Frame format is as follows: > (A number corresponds to a channel, a stereo channel-pair is embraced > by [] and 1 digit corresponds to one sample byte) > > capture channels [0,1] on DMA1: 00--11-- > playback channels [2,3],[4,5] on DMA2: 22443355 > playback channels [6,7],[8,9] on DMA3: 66887799 I tried to find a way to handle this in the card, but this special interleaving for the playback channels is by hardware restriction and can't be worked around in the soundcard itself. AFAIK alsa provides the samples interleaved like 22334455 (66778899). To avoid channel switching I guess some special processing in the alsa driver is required. Where is this done best? Do I have to move around the samples in the interrupt handler or does alsa provide some (possibly better) means to achive the correct (for the soundcard) interleaving? Regards, Matthias

1 0

[alsa-devel] Suggestion for getting alsa to use Conexant
by Victor Schneider 25 May '07

25 May '07

How about being able to parameterize Alsa modules so that the intel 80xm module finds any hda modem that also acts as an Intel hda chip? I am referring to the Conexant combination modem and hda chip that gets identified as "Conexant Waikiki" by alsamixer, but that 80xm can't find as a modem. Giving the 80xm module the device:identifier value to override its detection of the Intel version of the Conexant hda modem might let slmodemd work.

3 5

Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support for SMDK2440boards
by Zoltan Devai 25 May '07

25 May '07

2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > I have a SMDK2410 and I am trying to run kernel 2.6.20 on it. The kernel > is running Ok, but the sound is dead. I gonna try your patch and I would > like to know which files from the RK's patch have you used? The attached two patches should enable the sound support fully. One of them adds the l3-bus, the other one the necessary Kconfig and Makefile changes, along with an ASoC hack. Add this patch also: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/001105.html Everything should apply cleanly to 2.6.22-rc2, but probably also to 2.6.21. Zoltan

2 3

[alsa-devel] Sound on MSI S420
by Baruch Even 25 May '07

25 May '07

Hello, I've succeeded by guessing to get my MSI S420 laptop sound to work. Attached is a one-liner patch to do that. The patch is against kernel 2.6.22-rc2. Since the patch is based on guesswork it might not be optimal but it does work for me. Baruch

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 25 May '07

25 May '07

> Thanks. Now committed to HG tree. > > One more possible fix is about the build of snd-sb16-dsp and dependent > modules. We don't need snd-sb16 itself. > > Could you try the patch below (both for alsa-driver and alsa-kernel)? > It introduces the intermediate configs for snd-sb16-dsp, snd-sb8-dsp > and snd-sb-common modules. > Yeah, that makes more sense...and it works fine. Thanks, Ash =

2 1

[alsa-devel] Testing ALSA and OSS drivers
by Pharaoh . 24 May '07

24 May '07

Hi I am planning to write an ALSA driver for an OMAP based board, I already have an OSS driver written for my board, but it is a 2.4 kernel. Question: I am trying to bring up Linux for my board (the board is supported), how can I test whether the oss drivers are working fine or not? I just have to check whether operations like playback and recording etc work as expected. Any pointers will be helpful. Regards, Pharaoh.

2 1

[alsa-devel] [RFC] [PATCH 2/2] ASoC sound support for SMDK2440 boards: board driver
by Zoltan Devai 24 May '07

24 May '07

--- /dev/null +++ /sound/soc/s3c24xx/smdk2440.c @@ -0,0 +1,162 @@ +/* + * smdk2440.c -- SMDK2440 ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/l3.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/regs-iis.h> + +#include "../codecs/uda1341.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +static int smdk2440_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + unsigned int div = 0; + int ret = 0; + + switch (params_rate(params)) { + case 11025: + div = 16; + clk = 4233600; + break; + case 22050: + div = 8; + clk = 8467200; + break; + case 44100: + div = 4; + clk = 16934400; + break; + default: + return -EINVAL; + } + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div,div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static const char* smdk2440_l3_link_name = "s3c24xx-l3"; + +static struct snd_soc_ops smdk2440_ops = { + .hw_params = smdk2440_hw_params, +}; + +static struct snd_soc_dai_link smdk2440_dai_link = { + .name = "UDA1341", + .stream_name = "UDA1341", + .codec_dai = &uda1341_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &smdk2440_ops, +}; + +static struct snd_soc_machine snd_soc_machine_smdk2440 = { + .name = "SMDK2440", + .dai_link = &smdk2440_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device smdk2440_snd_devdata = { + .machine = &snd_soc_machine_smdk2440, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda1341, + .codec_data = &smdk2440_l3_link_name, +}; + +static struct platform_device *smdk2440_snd_device; + +static int __init smdk2440_uda1341_init(void) +{ + int ret; + + printk(KERN_INFO "SMDK2440 ALSA SoC Audio driver\n"); + + if (!(machine_is_s3c2440())) + return -ENODEV; + + smdk2440_snd_devdata.codec_data = l3_get_adapter("s3c24xx-l3"); + if (smdk2440_snd_devdata.codec_data == NULL) { + printk(KERN_WARNING "SMDK2440 SoC Audio: \ + Unable to find L3 bus adapter\n"); + return -ENODEV; + } + + smdk2440_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk2440_snd_device) { + printk(KERN_ERR "SMDK2440 SoC Audio: Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(smdk2440_snd_device, &smdk2440_snd_devdata); + smdk2440_snd_devdata.dev = &smdk2440_snd_device->dev; + ret = platform_device_add(smdk2440_snd_device); + + if (ret) + platform_device_put(smdk2440_snd_device); + + return ret; +} + +static void __exit smdk2440_uda1341_exit(void) +{ + platform_device_unregister(smdk2440_snd_device); +} + +module_init(smdk2440_uda1341_init); +module_exit(smdk2440_uda1341_exit); + +MODULE_AUTHOR("Zoltan Devai"); +MODULE_DESCRIPTION("SMDK2440 ALSA SoC audio driver"); +MODULE_LICENSE("GPL");

2 1

[alsa-devel] Default Sample rate Conversion
by Prabhu Sivaraja 24 May '07

24 May '07

Does ALSA automatically convert to my devices sample rate? My alsa.conf file is below pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } When I play a 32KHz file with aplay does it get up sampled to my device's 48KHz sample rate? I am assuming it does as it plays just fine but I am not sure where the upsampling happens? Thanks.

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

> > Takashi, here are my modified patches. > > One small one for alsa-driver. > > I took your advice and merged the rest into alsa-kernel. > > Everything should be in order this time. > > > > Add support for Cyrix/NatSemi Geode SC5530 (VSA1) > > > > Signed-off-by Ash Willis <ashwillis(a)programmer.net> > > Thanks. One thing I overlooked at the previous review is the kconfig > dependency below: Ok, hopefully we're finished now :) Here are the two patches. Ash --- diff -r fb391138d29c -r 994136cf989e pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/cs5530.c" +EXPORT_NO_SYMBOLS; --- diff -r 45179b325c8e -r 9ca89d69569c Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 +++ b/Documentation/ALSA-Configuration.txt Thu May 24 23:16:59 2007 +0100 @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- diff -r 45179b325c8e -r 9ca89d69569c include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 23:16:59 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 23:16:59 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 23:16:59 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r 9ca89d69569c isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 23:16:59 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff -r 45179b325c8e -r 9ca89d69569c pci/Kconfig --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 +++ b/pci/Kconfig Thu May 24 23:16:59 2007 +0100 @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP Say Y here to use a new DSP image for SPDIF and dual codecs. This works better than the old code, so say Y. + +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" diff -r 45179b325c8e -r 9ca89d69569c pci/Makefile --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 +++ b/pci/Makefile Thu May 24 23:16:59 2007 +0100 @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff -r 45179b325c8e -r 9ca89d69569c pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 23:16:59 2007 +0100 @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + =

2 1

[alsa-devel] [RFC] [PATCH 0/2] ASoC sound support for SMDK2440 boards
by Zoltan Devai 24 May '07

24 May '07

Hi, The following patches are the first working version of drivers for the UDA1341 codec and SMDK2440 boards. Any comments for improvement are welcome. I also would like to ask for help in the following topics: - The codec requires an L3-bus driver. I'm using one made by RMK, but it's not mainline, and probably needs adoption to the driver model. (http://lkml.org/lkml/2005/10/28/194) It could also make more sense to integrate the driver into the i2c or spi subsystems, or just make it a S3C24xx specific gpio hack. Any volunteers ? - I only have access to a SMDK2440 board. Could someone verify if this would work on other SMDK's as well ? Regards, Zoltan

1 0

[alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

Takashi, here are my modified patches. One small one for alsa-driver. I took your advice and merged the rest into alsa-kernel. Everything should be in order this time. Add support for Cyrix/NatSemi Geode SC5530 (VSA1) Signed-off-by Ash Willis <ashwillis(a)programmer.net> --- diff -r fb391138d29c -r 994136cf989e pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/cs5530.c" +EXPORT_NO_SYMBOLS; --- diff -r 45179b325c8e -r 0b0c03c6c6ca Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 +++ b/Documentation/ALSA-Configuration.txt Thu May 24 20:58:27 2007 +0100 @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a above explicitly. The power-management is supported. - + + Module snd-cs5530 + _________________ + + Module for Cyrix/NatSemi Geode 5530 chip. + Module snd-cs5535audio ---------------------- diff -r 45179b325c8e -r 0b0c03c6c6ca include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 20:58:27 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 20:58:27 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 20:58:27 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 20:58:27 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Kconfig --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 +++ b/pci/Kconfig Thu May 24 20:58:27 2007 +0100 @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP Say Y here to use a new DSP image for SPDIF and dual codecs. This works better than the old code, so say Y. + +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Makefile --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 +++ b/pci/Makefile Thu May 24 20:58:27 2007 +0100 @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff -r 45179b325c8e -r 0b0c03c6c6ca pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 20:58:27 2007 +0100 @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + =

2 1

Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

I'm sorry. I forgot to mention that these patches replace the two previous patches :) Ash > ----- Original Message ----- > From: "Ash Willis" <ashwillis(a)programmer.net> > To: "Takashi Iwai" <tiwai(a)suse.de> > Subject: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1) > Date: Thu, 24 May 2007 09:09:29 -0500 > > > Takashi, here are my modified patches. > One small one for alsa-driver. > I took your advice and merged the rest into alsa-kernel. > Everything should be in order this time. > > Add support for Cyrix/NatSemi Geode SC5530 (VSA1) > > Signed-off-by Ash Willis <ashwillis(a)programmer.net> > --- > > diff -r fb391138d29c -r 994136cf989e pci/cs5530.c > --- /dev/null Thu Jan 01 00:00:00 1970 +0000 > +++ b/pci/cs5530.c Thu May 24 20:45:25 2007 +0100 > @@ -0,0 +1,2 @@ > +#include "../alsa-kernel/pci/cs5530.c" > +EXPORT_NO_SYMBOLS; > > > --- > > diff -r 45179b325c8e -r 0b0c03c6c6ca Documentation/ALSA-Configuration.txt > --- a/Documentation/ALSA-Configuration.txt Wed May 23 16:27:32 2007 +0200 > +++ b/Documentation/ALSA-Configuration.txt Thu May 24 20:58:27 2007 +0100 > @@ -467,7 +467,12 @@ Prior to version 0.9.0rc4 options had a > above explicitly. > > The power-management is supported. > - > + > + Module snd-cs5530 > + _________________ > + > + Module for Cyrix/NatSemi Geode 5530 chip. > + > Module snd-cs5535audio > ---------------------- > > diff -r 45179b325c8e -r 0b0c03c6c6ca include/sb.h > --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 > +++ b/include/sb.h Thu May 24 20:58:27 2007 +0100 > @@ -38,6 +38,7 @@ enum sb_hw_type { > SB_HW_ALS100, /* Avance Logic ALS100 chip */ > SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ > SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ > + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ > }; > > #define SB_OPEN_PCM 0x01 > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb16_main.c > --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb16_main.c Thu May 24 20:58:27 2007 +0100 > @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct > __open_ok: > if (chip->hardware == SB_HW_ALS100) > runtime->hw.rate_max = 48000; > + if (chip->hardware == SB_HW_CS5530) { > + runtime->hw.buffer_bytes_max = 32 * 1024; > + runtime->hw.periods_min = 2; > + runtime->hw.rate_min = 44100; > + } > if (chip->mode & SB_RATE_LOCK) > runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; > chip->playback_substream = substream; > @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct > __open_ok: > if (chip->hardware == SB_HW_ALS100) > runtime->hw.rate_max = 48000; > + if (chip->hardware == SB_HW_CS5530) { > + runtime->hw.buffer_bytes_max = 32 * 1024; > + runtime->hw.periods_min = 2; > + runtime->hw.rate_min = 44100; > + } > if (chip->mode & SB_RATE_LOCK) > runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; > chip->capture_substream = substream; > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_common.c > --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb_common.c Thu May 24 20:58:27 2007 +0100 > @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb > minor = version & 0xff; > snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", > chip->port, major, minor); > - > + > switch (chip->hardware) { > case SB_HW_AUTO: > switch (major) { > @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb > break; > case SB_HW_DT019X: > str = "(DT019X/ALS007)"; > + break; > + case SB_HW_CS5530: > + str = "16 (CS5530)"; > break; > default: > return -ENODEV; > diff -r 45179b325c8e -r 0b0c03c6c6ca isa/sb/sb_mixer.c > --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 > +++ b/isa/sb/sb_mixer.c Thu May 24 20:58:27 2007 +0100 > @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > if ((err = snd_sbmixer_init(chip, > snd_sb16_controls, > ARRAY_SIZE(snd_sb16_controls), > @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); > break; > case SB_HW_ALS4000: > @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c > break; > case SB_HW_16: > case SB_HW_ALS100: > + case SB_HW_CS5530: > restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); > break; > case SB_HW_ALS4000: > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Kconfig > --- a/pci/Kconfig Wed May 23 16:27:32 2007 +0200 > +++ b/pci/Kconfig Thu May 24 20:58:27 2007 +0100 > @@ -214,6 +214,16 @@ config SND_CS46XX_NEW_DSP > Say Y here to use a new DSP image for SPDIF and dual codecs. > > This works better than the old code, so say Y. > + > +config SND_CS5530 > + tristate "CS5530 Audio" > + depends on SND > + select SND_SB16 > + help > + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. > + > + To compile this driver as a module, choose M here: the module > + will be called snd-cs5530. > > config SND_CS5535AUDIO > tristate "CS5535/CS5536 Audio" > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/Makefile > --- a/pci/Makefile Wed May 23 16:27:32 2007 +0200 > +++ b/pci/Makefile Thu May 24 20:58:27 2007 +0100 > @@ -12,6 +12,7 @@ snd-bt87x-objs := bt87x.o > snd-bt87x-objs := bt87x.o > snd-cmipci-objs := cmipci.o > snd-cs4281-objs := cs4281.o > +snd-cs5530-objs := cs5530.o > snd-ens1370-objs := ens1370.o > snd-ens1371-objs := ens1371.o > snd-es1938-objs := es1938.o > @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_BT87X) += snd-bt87x.o > obj-$(CONFIG_SND_BT87X) += snd-bt87x.o > obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o > obj-$(CONFIG_SND_CS4281) += snd-cs4281.o > +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o > obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o > obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o > obj-$(CONFIG_SND_ES1938) += snd-es1938.o > diff -r 45179b325c8e -r 0b0c03c6c6ca pci/cs5530.c > --- /dev/null Thu Jan 01 00:00:00 1970 +0000 > +++ b/pci/cs5530.c Thu May 24 20:58:27 2007 +0100 > @@ -0,0 +1,306 @@ > +/* > + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio > + * > + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> > + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> > + * > + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did > + * mess with it a bit. The chip seems to have to have trouble with > full duplex > + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to > + * simultaneously play back audio at 16bit 44100kHz, the device > actually plays > + * back in the same format in which it is capturing. By forcing the chip to > + * always play/capture in 16/44100, we can let alsa-lib convert > the samples and > + * that way we can hack up some full duplex audio. > + * > + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. > + * The older version (VSA1) provides fairly good soundblaster emulation > + * although there are a couple of bugs: large DMA buffers break record, > + * and the MPU event handling seems suspect. VSA2 allows the native driver > + * to control the AC97 audio engine directly and requires a different driver. > + * > + * Thanks to National Semiconductor for providing the needed information > + * on the XpressAudio(tm) internals. > + * > + * This program is free software; you can redistribute it and/or modify it > + * under the terms of the GNU General Public License as published by the > + * Free Software Foundation; either version 2, or (at your option) any > + * later version. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + * TO DO: > + * Investigate whether we can portably support Cognac (5520) in the > + * same manner. > + */ > + > +#include <sound/driver.h> > +#include <linux/delay.h> > +#include <linux/moduleparam.h> > +#include <linux/pci.h> > +#include <sound/core.h> > +#include <sound/sb.h> > +#include <sound/initval.h> > + > +MODULE_AUTHOR("Ash Willis"); > +MODULE_DESCRIPTION("CS5530 Audio"); > +MODULE_LICENSE("GPL"); > + > +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > + > +struct snd_cs5530 { > + struct snd_card *card; > + struct pci_dev *pci; > + struct snd_sb *sb; > + unsigned long pci_base; > +}; > + > +static struct pci_device_id snd_cs5530_ids[] = { > + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, > + PCI_ANY_ID, 0, 0}, > + {0,} > +}; > + > +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); > + > +static int snd_cs5530_free(struct snd_cs5530 *chip) > +{ > + pci_release_regions(chip->pci); > + pci_disable_device(chip->pci); > + kfree(chip); > + return 0; > +} > + > +static int snd_cs5530_dev_free(struct snd_device *device) > +{ > + struct snd_cs5530 *chip = device->device_data; > + return snd_cs5530_free(chip); > +} > + > +static void __devexit snd_cs5530_remove(struct pci_dev *pci) > +{ > + snd_card_free(pci_get_drvdata(pci)); > + pci_set_drvdata(pci, NULL); > +} > + > +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) > +{ > + outb(reg, io + 4); > + udelay(20); > + reg = inb(io + 5); > + udelay(20); > + return reg; > +} > + > +static int __devinit snd_cs5530_create(struct snd_card *card, > + struct pci_dev *pci, > + struct snd_cs5530 **rchip) > +{ > + struct snd_cs5530 *chip; > + unsigned long sb_base; > + u8 irq, dma8, dma16 = 0; > + u16 map; > + void __iomem *mem; > + int err; > + > + static struct snd_device_ops ops = { > + .dev_free = snd_cs5530_dev_free, > + }; > + *rchip = NULL; > + > + err = pci_enable_device(pci); > + if (err < 0) > + return err; > + > + chip = kzalloc(sizeof(*chip), GFP_KERNEL); > + if (chip == NULL) { > + pci_disable_device(pci); > + return -ENOMEM; > + } > + > + chip->card = card; > + chip->pci = pci; > + > + err = pci_request_regions(pci, "CS5530"); > + if (err < 0) { > + kfree(chip); > + pci_disable_device(pci); > + return err; > + } > + chip->pci_base = pci_resource_start(pci, 0); > + > + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); > + if (mem == NULL) { > + kfree(chip); > + pci_disable_device(pci); > + return -EBUSY; > + } > + > + map = readw(mem + 0x18); > + iounmap(mem); > + > + /* Map bits > + 0:1 * 0x20 + 0x200 = sb base > + 2 sb enable > + 3 adlib enable > + 5 MPU enable 0x330 > + 6 MPU enable 0x300 > + > + The other bits may be used internally so must be masked */ > + > + sb_base = 0x220 + 0x20 * (map & 3); > + > + if (map & (1<<2)) > + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); > + else { > + printk(KERN_ERR "Could not find XpressAudio!\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (map & (1<<5)) > + printk(KERN_INFO "CS5530: MPU at 0x300\n"); > + else if (map & (1<<6)) > + printk(KERN_INFO "CS5530: MPU at 0x330\n"); > + > + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; > + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); > + > + if (dma8 & 0x20) > + dma16 = 5; > + else if (dma8 & 0x40) > + dma16 = 6; > + else if (dma8 & 0x80) > + dma16 = 7; > + else { > + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (dma8 & 0x01) > + dma8 = 0; > + else if (dma8 & 02) > + dma8 = 1; > + else if (dma8 & 0x08) > + dma8 = 3; > + else { > + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + if (irq & 1) > + irq = 9; > + else if (irq & 2) > + irq = 5; > + else if (irq & 4) > + irq = 7; > + else if (irq & 8) > + irq = 10; > + else { > + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); > + snd_cs5530_free(chip); > + return -ENODEV; > + } > + > + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, > + dma16); > + > + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, > + dma16, SB_HW_CS5530, &chip->sb); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create PCM\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_sbmixer_new(chip->sb); > + if (err < 0) { > + printk(KERN_ERR "CS5530: Could not create Mixer\n"); > + snd_cs5530_free(chip); > + return err; > + } > + > + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); > + if (err < 0) { > + snd_cs5530_free(chip); > + return err; > + } > + > + snd_card_set_dev(card, &pci->dev); > + *rchip = chip; > + return 0; > +} > + > +static int __devinit snd_cs5530_probe(struct pci_dev *pci, > + const struct pci_device_id *pci_id) > +{ > + static int dev; > + struct snd_card *card; > + struct snd_cs5530 *chip = NULL; > + int err; > + > + if (dev >= SNDRV_CARDS) > + return -ENODEV; > + if (!enable[dev]) { > + dev++; > + return -ENOENT; > + } > + > + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); > + > + if (card == NULL) > + return -ENOMEM; > + > + err = snd_cs5530_create(card, pci, &chip); > + if (err < 0) { > + snd_card_free(card); > + return err; > + } > + > + strcpy(card->driver, "CS5530"); > + strcpy(card->shortname, "CS5530 Audio"); > + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); > + > + err = snd_card_register(card); > + if (err < 0) { > + snd_card_free(card); > + return err; > + } > + pci_set_drvdata(pci, card); > + dev++; > + return 0; > +} > + > +static struct pci_driver driver = { > + .name = "CS5530_Audio", > + .id_table = snd_cs5530_ids, > + .probe = snd_cs5530_probe, > + .remove = __devexit_p(snd_cs5530_remove), > +}; > + > +static int __init alsa_card_cs5530_init(void) > +{ > + return pci_register_driver(&driver); > +} > + > +static void __exit alsa_card_cs5530_exit(void) > +{ > + pci_unregister_driver(&driver); > +} > + > +module_init(alsa_card_cs5530_init) > +module_exit(alsa_card_cs5530_exit) > + > > > > = > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > =

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[alsa-devel] [1/3] 2.6.22-rc2: known regressions v2
by Michal Piotrowski 24 May '07

24 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc2. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : lag problems while browsing sites with heavy JS and or flash usage References : http://lkml.org/lkml/2007/5/19/189 Submitter : Anant Nitya <kernel(a)prachanda.info> Handled-By : Ingo Molnar <mingo(a)elte.hu> David Miller <davem(a)davemloft.net> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown Subject : BAT0 no longer seen, ACPI errors References : http://bugzilla.kernel.org/show_bug.cgi?id=8493 Submitter : Ben Collins <bcollins(a)ubuntu.com> Workaround : BIOS upgrade Status : Unknown Subject : nx6125 has lost fan control References : http://lkml.org/lkml/2007/5/16/249 Submitter : Ray Lee <ray-lk(a)madrabbit.org> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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Re: [alsa-devel] [PATCH 2/2] Add support for Cyrix/NatSemi Geode SC5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

> Avoid the style "if ((err = xx) < 0)" in new codes as much as > possible. Just write: > err = pci_enable(pci); > if (err < 0) > return err; > Now the kernel coding style recommends this. Sorry, I wasn't aware this had changed :) > > + if (dma8 & 0x20) > > + dma16 = 5; > > + else if (dma8 & 0x40) > > + dma16 = 6; > > + else if (dma8 & 0x80) > > + dma16 = 7; > > + else > > + { > > + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); > > + snd_cs5530_free(chip); > > + return -ENODEV; > > + } > > Ditto. > (BTW, is dma16 really necessary to be zero-initialized at beginning?) > It is, I think. It supresses a compiler warning, due the to fact that the final else path does not set dma16 and so if this path is taken the variable is uninitialied...not that we care, but we certainly don't need a compiler warning. > > Is the driver confirmed to work with the recent kernel? > If yes, we can put cs5530.c to alsa-kernel so that it'll be merged to > the next linux kernel. alsa-driver tree still needs cs5530.c just > including alsa-kernel code, though. > Also, don't forget to add a description to ALSA-Configuration.txt. > > Could you fix these issues and resend the patch? It was devleoped on 2.6.21.1. OK, I'll make all the necessary changes. Thanks, Ash =

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[alsa-devel] [PATCH 2/2] Add support for Cyrix/NatSemi Geode SC5530 (VSA1)
by Ash Willis 24 May '07

24 May '07

Add support for the Cyrix/NatSemi Geode SC5530 chips. This driver is ported from oss/kahlua.c and is made up of two patches. Patch 1/2 adds some stuff to the SB layer to allow for some hardware quirks and is to be applied to alsa-kernel. Patch 2/2 adds the actual driver code to be applied to alsa-driver. Patch 2 depends on patch 1. Signed off by: Ash Willis <ashwillis(a)programmer.net> --- summary: Add Cyrix/NatSemi Geode CS5530 driver diff -r 5c25d0e8819c -r c9be7b8611e7 pci/Kconfig --- a/pci/Kconfig Fri May 18 17:56:53 2007 +0200 +++ b/pci/Kconfig Thu May 24 02:24:57 2007 +0100 @@ -20,5 +20,15 @@ config SND_ASIHPI help Say 'Y' or 'M' to include support for AudioScience ASIxxxx soundcards. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND + select SND_SB16 + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + endmenu diff -r 5c25d0e8819c -r c9be7b8611e7 pci/Makefile --- a/pci/Makefile Fri May 18 17:56:53 2007 +0200 +++ b/pci/Makefile Thu May 24 02:24:57 2007 +0100 @@ -13,6 +13,9 @@ clean-files := ad1889.c bt87x.c intel8x0 obj-$(CONFIG_SND) += pdplus/ asihpi/ +snd-cs5530-objs := cs5530.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o + # # # diff -r 5c25d0e8819c -r c9be7b8611e7 pci/cs5530.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/cs5530.c Thu May 24 02:24:57 2007 +0100 @@ -0,0 +1,296 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis(a)programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan(a)redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + if ((err = pci_enable_device(pci)) < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + if ((err = pci_request_regions(pci, "CS5530")) < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else + { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else + { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else + { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else + { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + if ((err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, + dma8, dma16, SB_HW_CS5530, &chip->sb)) < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm)) < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_sbmixer_new(chip->sb)) < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + if ((err = snd_cs5530_create(card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + = Work From Home & Learn How To Make Money Learn How To Make Money Online With Help & Guidance From The Pro's. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=db055100ca96c7d9527d2fa…

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[alsa-devel] [patch] bugfix and clean driver/pci/asihpi/hpi6205.[ch]
by Eliot Blennerhassett 24 May '07

24 May '07

Attached patch Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Patch makes power of 2 buffering unconditional. and fixes incorrect buffer allocation when requested size was rounded up to a power of 2. -- Eliot

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[alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Nobin Mathew 24 May '07

24 May '07

Hi, Is there any standard ALSA appplication which does simultaneous capture and playback?? What are the complications in implementing simultaneous playback and capture in ALSA??? What are the plugin (.asoundrc) configurations needed for simultaneous playback and capture???? Thanks Nobin Mathew

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Re: [alsa-devel] support for Medion laptops
by albox＠free.fr 24 May '07

24 May '07

I haven't get any news since my last email 2 weeks ago. Can anybody help me to find where the problem comes from ? Quoting albox(a)free.fr: > I have just installed 1.0.14rc4. Here are the logs : > > [ 20.132000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > codec_mask = 0x3 > [ 20.308000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1820: > hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) > [ 21.032000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:245: > si3054: cannot initialize. EXT MID = 0000 > [ 21.040000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:257: > Link Frame Detect(FDT) is not ready (line status: 0000) > [ 71.352000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.352000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.360000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.368000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.380000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > [ 71.480000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 > [ 71.488000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 > [ 71.496000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 > [ 71.504000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 > [ 71.524000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.524000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.532000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.540000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.548000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > [ 71.560000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 > [ 71.568000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 > [ 71.576000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 > [ 71.584000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 > [ 71.752000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > [ 71.752000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 > [ 71.760000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 > [ 71.768000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 > [ 71.776000] ALSA > /home/alboot/Sources/alsa1.0.14rc4/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:630: > hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 > > If you want more information, let me know...

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[alsa-devel] [PATCH 1/2] Add SB_HW_CS5530 to the SB code
by Ash Willis 23 May '07

23 May '07

Add support for the Cyrix/NatSemi Geode SC5530 chips. This driver is ported from oss/kahlua.c and is made up of two patches. Patch 1/2 adds some stuff to the SB layer to allow for some hardware quirks and is to be applied to alsa-kernel. Patch 2/2 adds the actual driver code to be applied to alsa-driver. Patch 2 depends on patch 1. Signed off by: Ash Willis <ashwillis(a)programmer.net> --- summary: Add SB_HW_CS5530 to the SB code diff -r 45179b325c8e -r b8441e56e853 include/sb.h --- a/include/sb.h Wed May 23 16:27:32 2007 +0200 +++ b/include/sb.h Thu May 24 03:09:49 2007 +0100 @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_CS5530, /* Cyrix/NatSemi 5530 VSA1 */ }; #define SB_OPEN_PCM 0x01 diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb16_main.c --- a/isa/sb/sb16_main.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb16_main.c Thu May 24 03:09:49 2007 +0100 @@ -563,6 +563,11 @@ static int snd_sb16_playback_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->playback_substream = substream; @@ -633,6 +638,11 @@ static int snd_sb16_capture_open(struct __open_ok: if (chip->hardware == SB_HW_ALS100) runtime->hw.rate_max = 48000; + if (chip->hardware == SB_HW_CS5530) { + runtime->hw.buffer_bytes_max = 32 * 1024; + runtime->hw.periods_min = 2; + runtime->hw.rate_min = 44100; + } if (chip->mode & SB_RATE_LOCK) runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate; chip->capture_substream = substream; diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb_common.c --- a/isa/sb/sb_common.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_common.c Thu May 24 03:09:49 2007 +0100 @@ -128,7 +128,7 @@ static int snd_sbdsp_probe(struct snd_sb minor = version & 0xff; snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", chip->port, major, minor); - + switch (chip->hardware) { case SB_HW_AUTO: switch (major) { @@ -167,6 +167,9 @@ static int snd_sbdsp_probe(struct snd_sb break; case SB_HW_DT019X: str = "(DT019X/ALS007)"; + break; + case SB_HW_CS5530: + str = "16 (CS5530)"; break; default: return -ENODEV; diff -r 45179b325c8e -r b8441e56e853 isa/sb/sb_mixer.c --- a/isa/sb/sb_mixer.c Wed May 23 16:27:32 2007 +0200 +++ b/isa/sb/sb_mixer.c Thu May 24 03:09:49 2007 +0100 @@ -821,6 +821,7 @@ int snd_sbmixer_new(struct snd_sb *chip) break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: if ((err = snd_sbmixer_init(chip, snd_sb16_controls, ARRAY_SIZE(snd_sb16_controls), @@ -950,6 +951,7 @@ void snd_sbmixer_suspend(struct snd_sb * break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: @@ -975,6 +977,7 @@ void snd_sbmixer_resume(struct snd_sb *c break; case SB_HW_16: case SB_HW_ALS100: + case SB_HW_CS5530: restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs)); break; case SB_HW_ALS4000: =

1 0

Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 23 May '07

23 May '07

EPPROM ON: OK, I formated that dos partition and simply sys a: c:, then I installed the driver (copying them to the partition from the linux install), in theinstall I selected the defaults settings (220 irq5 dma1...) and rebooted, now in dos again I used diagnose to test the audio and hwset /mr /ms to reset and save the epprom. now I rebooted and loged to linux: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 FATAL: Error inserting snd_sb8 (/lib/modules/2.6.20-15-generic/kernel/sound/isa/sb/snd-sb8.ko): No such device dmesg last lines ( [ 319.827327] Sound Blaster soundcard not found or device busy -reboot- EPPROM OFF (SOFTWARE ON): in dos: diagnose works, reboot and back to lin: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 no error, no mention in dmesg, but xmms gives the usual error... Heres lsmod just for kicks: Module Size Used by snd_sb8 9216 0 snd_sb8_dsp 12288 1 snd_sb8 snd_sb_common 17792 2 snd_sb8,snd_sb8_dsp snd_opl3_lib 11520 1 snd_sb8 snd_hwdep 9988 1 snd_opl3_lib snd_pcm_oss 44544 0 snd_pcm 79876 3 snd_sb8,snd_sb8_dsp,snd_pcm_oss snd_mixer_oss 17408 1 snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 32896 0 snd_seq_midi 9600 0 snd_rawmidi 25472 2 snd_sb8_dsp,snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 52592 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 23684 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 9100 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54020 13 snd_sb8,snd_sb8_dsp,snd_sb_common,snd_opl3_lib,snd_hwdep,snd_pcm_oss,snd_pcm,snd_mixer_oss,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 10888 1 snd_pcm nfs 240876 0 nfsd 218992 17 exportfs 6912 1 nfsd lockd 64904 3 nfs,nfsd sunrpc 161340 12 nfs,nfsd,lockd apm 22752 2 ppdev 10116 0 cpufreq_stats 7360 0 cpufreq_ondemand 9228 0 cpufreq_powersave 2688 0 freq_table 5792 2 cpufreq_stats,cpufreq_ondemand cpufreq_conservative 8200 0 cpufreq_userspace 5408 0 nls_iso8859_1 5120 1 nls_cp437 6784 1 vfat 14208 1 fat 53916 1 vfat ext2 66824 1 lp 12452 0 parport_pc 36388 1 parport 36936 3 ppdev,lp,parport_pc pcspkr 4224 0 ipv6 268704 8 tsdev 8768 0 evdev 11008 1 ext3 133128 1 jbd 59816 1 ext3 mbcache 9604 2 ext2,ext3 ide_disk 17024 5 8139too 27648 0 ide_cd 32672 0 cdrom 37664 1 ide_cd floppy 59524 0 ehci_hcd 34188 0 8139cp 25088 0 mii 6528 2 8139too,8139cp uhci_hcd 25360 0 usbcore 134280 3 ehci_hcd,uhci_hcd piix 10756 0 [permanent] generic 5124 0 [permanent] ata_generic 9092 0 libata 125720 1 ata_generic scsi_mod 142348 1 libata fbcon 42656 0 tileblit 3584 1 fbcon font 9216 1 fbcon bitblit 6912 1 fbcon softcursor 3200 1 bitblit vesafb 9220 0 capability 5896 0 commoncap 8192 1 capability So now atleast we know its not the jumpers ;-) Is their the slight chance xmms is the problem ? gxine crushs too... Is their something like "beep" only for the sound card ? Maybe droping some data to /dev/### to produce a beep ? Ram Kromberg ramkromberg(a)mail.com

4 13

[alsa-devel] No state is present for card CMI8738
by Tomas Pospisek's Mailing Lists 23 May '07

23 May '07

Since alsa-user is rather ... errr ... silent, I dare to repost the same question here. I've seen that Alsa likes the Terratec cards, however since they're also based on the CMI chipset, and at least /my/ onboard CMI chipset here sucks badly I wonder what low cost card I can use? I hope somebody can give me a clue, *t PS: Mic input is barely loud enough with my onboard sound. Mic boost has no effect. -- ----------------------------------------------------------- Tomas Pospisek http://sourcepole.com - Linux & Open Source Solutions ----------------------------------------------------------- ---------- Forwarded message ---------- Date: Mon, 14 May 2007 14:39:36 +0200 (CEST) From: Tomas Pospisek's Mailing Lists <tpo2(a)sourcepole.ch> To: alsa-user(a)lists.sourceforge.net Subject: [Alsa-user] No state is present for card CMI8738 The following is my problem: $ alsactl -f .alsavoip restore No state is present for card CMI8738 I see this "sometimes". Sometimes the above works and sometimes it returns with that error message. It needs to be noted that I am suspending to ram. Nevertheless, the behaveour is not consistent in between "s2ram"s and I don't know whether that has anything to do with it either. It also needs to be noted that alsamixer *never* has any problems accessing the card. But even after changing the card settings with alsamixer, alsactl will keep on repeating that error message. * $ lspci 00:0d.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) * alsa userspace is at 1.0.13 * kernel is Debian's 2.6.18-4-686 Further, minor problems: * After waking up from suspend the card looses its microphone settings - that is - "Mic Capture" is allway reset to zero. * Further on the choice to hide the microphone setting "Mic Capture" in alsamixer behind F5 irritates me a bit - there are *a lot* of very obscure settings (such as "IEC958 5V" - huh?), that are displayed by default. What's up with that error above? What does it mean? Is there any way I can fix or circumvent it? Anybody a hint? *t -- ----------------------------------------------------------- Tomas Pospisek http://sourcepole.com - Linux & Open Source Solutions ----------------------------------------------------------- ------------------------------------------------------------------------- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

2 4

[alsa-devel] A52 plugin segfault and patch
by Fabian van der Werf 23 May '07

23 May '07

Hi All, I bumped into segfault when trying to use the a52 plugin. I linked the plugin with the latest svn revision of ffmpeg (r9102). Starting program: /usr/bin/aplay -Dsurround test.wav Failed to read a valid object file image from memory. [Thread debugging using libthread_db enabled] [New Thread -1211078432 (LWP 9835)] Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1211078432 (LWP 9835)] av_crc (ctx=0x0, crc=0, buffer=0x8072c14 , length=1116) at crc.c:77 77 if(!ctx[256]) (gdb) bt #0 av_crc (ctx=0x0, crc=0, buffer=0x8072c14 , length=1116) at crc.c:77 #1 0xb78ce8a1 in output_frame_end (s=0x0) at ac3enc.c:1135 #2 0xb78d1010 in AC3_encode_frame (avctx=0x806d3c0, frame=0x8072c10 "\vw", buf_size=6136, data=0x806e400) at ac3enc.c:1261 #3 0xb7730f68 in avcodec_encode_audio (avctx=0x806d3c0, buf=0x8072c10 "\vw", buf_size=6136, samples=0x0) at utils.c:863 #4 0xb772f66c in convert_data (rec=0x805c868) at pcm_a52.c:57 #5 0xb7730101 in a52_transfer (io=0x805c868, areas=0x80551f0, offset=0, size=1536) at pcm_a52.c:197 #6 0xb7efcf11 in ioplug_priv_transfer_areas (pcm=0x806c138, areas=0x80551f0, offset=0, size=0) at pcm_ioplug.c:521 #7 0xb7efcfd6 in snd_pcm_ioplug_mmap_commit (pcm=0x806c138, offset=0, size=1536) at pcm_ioplug.c:590 #8 0xb7ebf7b7 in snd_pcm_mmap_commit (pcm=0x0, offset=0, frames=1536) at pcm.c:6305 #9 0xb7ecc7e4 in snd_pcm_plugin_write_areas (pcm=0x8054f48, areas=0xbfc8ea10, offset=0, size=1536) at pcm_plugin.c:282 #10 0xb7ebfc42 in snd_pcm_write_areas (pcm=0x8054f48, areas=0xbfc8ea10, offset=0, size=1536, func=0xb7ecc740 <snd_pcm_plugin_write_areas>) at pcm.c:6492 #11 0xb7ecbdff in snd_pcm_plugin_writei (pcm=0x8054f48, buffer=0x8074410, size=1536) at pcm_plugin.c:353 #12 0xb7eb7da5 in snd_pcm_writei (pcm=0x8068488, buffer=0x8074410, size=0) at pcm_local.h:401 #13 0x0804d05a in pcm_write (data=0x8074410 "", count=1536) at aplay.c:1260 #14 0x0804d66f in playback_go (fd=8, loaded=0, count=48457728, rtype=2, name=0xbfc8fa1d "test.wav") at aplay.c:1986 #15 0x0804e9b2 in playback (name=0xbfc8fa1d "test.wav") at aplay.c:2044 #16 0x08050ea9 in main (argc=3, argv=0xbfc8ed64) at aplay.c:601 The patch below fixes this segmentation fault by adding a call to avcodec_init() diff -Naur alsa-plugins-1.0.13/a52/pcm_a52.c alsa-plugins-1.0.13-patched/a52/pcm_a52.c --- alsa-plugins-1.0.13/a52/pcm_a52.c 2006-09-29 13:51:37.000000000 +0200 +++ alsa-plugins-1.0.13-patched/a52/pcm_a52.c 2007-05-22 18:49:56.000000000 +0200 @@ -676,6 +676,7 @@ rec->channels = channels; rec->format = format; + avcodec_init(); avcodec_register_all(); rec->codec = avcodec_find_encoder(CODEC_ID_AC3); if (! rec->codec) { Regards, Fabian

2 1

Re: [alsa-devel] bios update alters hda-intel and alsamixer
by Tobin Davis 23 May '07

23 May '07

On Wed, 2007-05-23 at 01:21 -0500, Chris Pemberton wrote: > On 5/23/07, Tobin Davis <tdavis(a)dsl-only.net> wrote: > > Could you post the output from this script: > > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > > > It was written by a user on the #alsa irc channel, and has been very > > helpful in debugging sound driver issues, at least for me. > > > > Tobin > > > > On Tue, 2007-05-22 at 23:55 -0500, Chris Pemberton wrote: > > > > > After updating the bios on a Lenovo notebook, my hda-intel sound "changed" > > > > > > Before the bios update I used kernel 2.6.20.3, external > > > alsa-driver-1.0.14rc3 and the attached patch_realtek.c (in-kernel alsa > > > wouldn't work) > > > > > > After the bios update I use just the alsa contained in kernel 2.6.20.3. > > > > > > Before bios update, my mixer contained many items: including master, > > > front, mic, pcm, etc. > > > > > > Now it only contains master and pcm. > > > > > > So, I have a few questions: > > > > > > How can I determine what items are "supposed" to show up in alsamixer? > > > > > > Where can I submit my laptop soundcard information to help finalize > > > support of this model/card? > > > > > > Thank you > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > -- > > Tobin Davis <tdavis(a)dsl-only.net> > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > It is located here: > > http://pastebin.ca/504675 Ok, according to what I can tell, you don't have the most recent version of the driver (your original post didn't have an attachement). I do know that the Lenovo support for the ALC861VD was added after 1.0.14rc4 snapshot was made. Try installing the most recent HG daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver . If that doesn't fix it, rerun the alsa-info.sh script and repost the new data. -- Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] bios update alters hda-intel and alsamixer
by Chris Pemberton 23 May '07

23 May '07

After updating the bios on a Lenovo notebook, my hda-intel sound "changed" Before the bios update I used kernel 2.6.20.3, external alsa-driver-1.0.14rc3 and the attached patch_realtek.c (in-kernel alsa wouldn't work) After the bios update I use just the alsa contained in kernel 2.6.20.3. Before bios update, my mixer contained many items: including master, front, mic, pcm, etc. Now it only contains master and pcm. So, I have a few questions: How can I determine what items are "supposed" to show up in alsamixer? Where can I submit my laptop soundcard information to help finalize support of this model/card? Thank you

2 1

Re: [alsa-devel] I don't understand snd_pcm_ops
by Ash Willis 22 May '07

22 May '07

> First off, I thought PCM stood for Pulse Code Modulation, but that > just describes the > format of the digital data. I don't understand what a PCM operation could be. Exactly, it described the format of the digital data used i audio processing. PCM does stand for pulse code modulation, but here it is a reference to a device that processes sample based digital audio i.e not midi etc. The ALSA middle layer (which includes PCM) takes care of all the common stuff that all drivers need to do. In sound drivers, we have multiple types of devices: PCM, midi, mixers etc. The PCM layer is responsible for doing all the digital audio work. I.e preparing the card for playback, initiaing transfer to and from the device etc. In short, if you want to playback or capture sound, you're going to need a PCM :) snd_pcm_ops is a structure of callbacks that relate to different events regarding the PCM interface. Use this to run your own code at specific events. I.e. it's most probable that you'll want to prepare the device in some manner before audio transfer starts, so you would put the code to do this into your prepare callback. > > I've seen documentation where PCM is considered an alternative to > I2S or AC97. But if you > look at eti_b1_wm8731.c, you'll see this: > > #include "at91-pcm.h" > #include "at91-i2s.h" > > So is the AT91 a PCM device or an I2S device??? None of these are alternatives to each other. They're all different things. The anser to your question would be that this is an i2c device that has a PCM. Have you read "Writing an ALSA driver"? It generally describes ALSA drivers as they relate to PCI, and how to write them, but it does a good job at explaining the general ALSA driver architecture. Ash =

2 1

[alsa-devel] [PATCH] Fix ASoC s3c24xx-pcm spinlock bug
by Zoltan Devai 22 May '07

22 May '07

Hi, This should fix a spinlock lockup bug on the s3c24xx arch. Regards, Zoltan --- --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -337,6 +337,8 @@ static int s3c24xx_pcm_open(struct snd_p if (prtd == NULL) return -ENOMEM; + spin_lock_init(&prtd->lock); + runtime->private_data = prtd; return 0; }

2 1

[alsa-devel] Broken JACK plugin in alsa-plugins and a way to fix it.
by Jasiu 22 May '07

22 May '07

Hi. I want to note that JACK plugin in alsa-plugins in broken. (At least on my two systems.) It just plays for a few seconds and then stops hanging the application that was sending data to it. When it hangs jackd spits out this: subgraph starting at alsa-jack.jackP.3635.0 timed out (subgraph_wait_fd=7, status = 0, state = Running) What I want to note is that there is a fix to that and I would like it to be included in the tree. I'm not sure if it doesn't break anything else, but it works for me ok. Open pcm_jack.c and on the line 146 you'll find: write(jack->fd, buf, 1); /* for polling */ Deleting this seems to get rid of the problem. So please if you can patch this out. (I didn't make a patch myself since this is so tiny fix that it isn't really worth.) Cheers.

2 3

[alsa-devel] [PATCH] remove email addresses from MODULE_AUTHOR tag
by Rene Herman 22 May '07

22 May '07

Hi Takashi. Following up the recent MODULE_MAINTAINER discussion on LKML: http://lkml.org/lkml/2007/4/4/170 that concluded with MODULE_MAINTAINER not being a good idea, here's a patch that just deletes the email addresses from the MODULE_AUTHOR tags for ALSA. The email address is the problem I was trying to fix; with multiple current and non-current authors and maintainers who might not even be authors the address(es) available from the tag confuse the issue of whom to contact. It's moreover also information that easily outdated. A bit more than half of the tags in the kernel don't include an email address already (the core contributors typically don't include any) and I'll submit patches removing more. (this was generated against current -linus). aoa/codecs/snd-aoa-codec-onyx.c | 2 +- aoa/codecs/snd-aoa-codec-tas.c | 2 +- aoa/codecs/snd-aoa-codec-toonie.c | 2 +- aoa/core/snd-aoa-core.c | 2 +- aoa/fabrics/snd-aoa-fabric-layout.c | 2 +- aoa/soundbus/core.c | 2 +- aoa/soundbus/i2sbus/i2sbus-core.c | 2 +- arm/sa11xx-uda1341.c | 2 +- core/hwdep.c | 2 +- core/memalloc.c | 2 +- core/oss/mixer_oss.c | 2 +- core/oss/pcm_oss.c | 2 +- core/pcm.c | 2 +- core/rawmidi.c | 2 +- core/seq/instr/ainstr_fm.c | 2 +- core/seq/instr/ainstr_gf1.c | 2 +- core/seq/instr/ainstr_iw.c | 2 +- core/seq/instr/ainstr_simple.c | 2 +- core/seq/oss/seq_oss.c | 2 +- core/seq/seq.c | 2 +- core/seq/seq_device.c | 2 +- core/seq/seq_dummy.c | 2 +- core/seq/seq_instr.c | 2 +- core/seq/seq_midi.c | 2 +- core/seq/seq_midi_emul.c | 2 +- core/seq/seq_midi_event.c | 2 +- core/seq/seq_virmidi.c | 2 +- core/sound.c | 2 +- core/timer.c | 2 +- drivers/dummy.c | 2 +- drivers/mpu401/mpu401.c | 2 +- drivers/mpu401/mpu401_uart.c | 2 +- drivers/mts64.c | 2 +- drivers/opl3/opl3_lib.c | 2 +- drivers/opl3/opl3_seq.c | 2 +- drivers/opl4/opl4_lib.c | 2 +- drivers/opl4/opl4_seq.c | 2 +- drivers/virmidi.c | 2 +- drivers/vx/vx_core.c | 2 +- i2c/cs8427.c | 2 +- i2c/i2c.c | 2 +- i2c/l3/uda1341.c | 2 +- i2c/other/ak4114.c | 2 +- i2c/other/ak4117.c | 2 +- i2c/other/ak4xxx-adda.c | 2 +- i2c/other/pt2258.c | 2 +- i2c/other/tea575x-tuner.c | 2 +- i2c/tea6330t.c | 2 +- isa/ad1816a/ad1816a.c | 2 +- isa/ad1848/ad1848.c | 2 +- isa/ad1848/ad1848_lib.c | 2 +- isa/als100.c | 2 +- isa/azt2320.c | 2 +- isa/cmi8330.c | 2 +- isa/cs423x/cs4231.c | 2 +- isa/cs423x/cs4231_lib.c | 2 +- isa/cs423x/cs4236.c | 2 +- isa/cs423x/cs4236_lib.c | 2 +- isa/dt019x.c | 2 +- isa/es1688/es1688.c | 2 +- isa/es1688/es1688_lib.c | 2 +- isa/es18xx.c | 2 +- isa/gus/gus_main.c | 2 +- isa/gus/gus_synth.c | 2 +- isa/gus/gusclassic.c | 2 +- isa/gus/gusextreme.c | 2 +- isa/gus/gusmax.c | 2 +- isa/gus/interwave.c | 2 +- isa/opl3sa2.c | 2 +- isa/opti9xx/miro.c | 2 +- isa/opti9xx/opti92x-ad1848.c | 2 +- isa/sb/es968.c | 2 +- isa/sb/sb16.c | 2 +- isa/sb/sb16_csp.c | 2 +- isa/sb/sb16_main.c | 2 +- isa/sb/sb8.c | 2 +- isa/sb/sb8_main.c | 2 +- isa/sb/sb_common.c | 2 +- isa/sgalaxy.c | 2 +- isa/wavefront/wavefront.c | 2 +- mips/au1x00.c | 2 +- parisc/harmony.c | 2 +- pci/ac97/ac97_codec.c | 2 +- pci/ac97/ak4531_codec.c | 2 +- pci/ad1889.c | 2 +- pci/ali5451/ali5451.c | 2 +- pci/als300.c | 2 +- pci/als4000.c | 2 +- pci/atiixp.c | 2 +- pci/atiixp_modem.c | 2 +- pci/azt3328.c | 2 +- pci/bt87x.c | 2 +- pci/ca0106/ca0106_main.c | 2 +- pci/cmipci.c | 2 +- pci/cs4281.c | 2 +- pci/cs46xx/cs46xx.c | 2 +- pci/echoaudio/echoaudio.c | 2 +- pci/emu10k1/emu10k1.c | 2 +- pci/emu10k1/emu10k1x.c | 2 +- pci/ens1370.c | 2 +- pci/es1938.c | 2 +- pci/fm801.c | 2 +- pci/hda/hda_codec.c | 2 +- pci/ice1712/ak4xxx.c | 2 +- pci/ice1712/ice1712.c | 2 +- pci/ice1712/ice1724.c | 2 +- pci/intel8x0.c | 2 +- pci/intel8x0m.c | 2 +- pci/korg1212/korg1212.c | 2 +- pci/maestro3.c | 2 +- pci/mixart/mixart.c | 2 +- pci/nm256/nm256.c | 2 +- pci/pcxhr/pcxhr.c | 2 +- pci/riptide/riptide.c | 2 +- pci/rme32.c | 2 +- pci/rme96.c | 2 +- pci/rme9652/hdsp.c | 2 +- pci/rme9652/rme9652.c | 2 +- pci/sonicvibes.c | 2 +- pci/trident/trident.c | 2 +- pci/trident/trident_synth.c | 2 +- pci/via82xx.c | 2 +- pci/via82xx_modem.c | 2 +- pci/vx222/vx222.c | 2 +- pci/ymfpci/ymfpci.c | 2 +- pcmcia/pdaudiocf/pdaudiocf.c | 2 +- pcmcia/vx/vxpocket.c | 2 +- soc/at91/at91-i2s.c | 2 +- soc/at91/at91-pcm.c | 2 +- soc/at91/eti_b1_wm8731.c | 2 +- soc/pxa/pxa2xx-i2s.c | 2 +- soc/soc-core.c | 2 +- soc/soc-dapm.c | 2 +- usb/usbaudio.c | 2 +- usb/usbmidi.c | 2 +- usb/usx2y/usbusx2y.c | 2 +- 136 files changed, 136 insertions(+), 136 deletions(-) Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. commit 7c179369bc673d2bac7f67fb94aee1d39b4c3f4d Author: Rene Herman <rene.herman(a)gmail.com> Date: Fri May 11 03:03:00 2007 +0200 module_author: alsa Signed-off-by: Rene Herman <rene.herman(a)gmail.com> diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c index e91f9f6..f9b8159 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c @@ -33,7 +33,7 @@ */ #include <linux/delay.h> #include <linux/module.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 041fe52..b1d2287 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -67,7 +67,7 @@ #include <linux/module.h> #include <linux/mutex.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("tas codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/snd-aoa-codec-toonie.c index 3c7d1d8..0ccdaf9 100644 --- a/sound/aoa/codecs/snd-aoa-codec-toonie.c +++ b/sound/aoa/codecs/snd-aoa-codec-toonie.c @@ -11,7 +11,7 @@ */ #include <linux/delay.h> #include <linux/module.h> -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/snd-aoa-core.c index 19fdae4..23372e3 100644 --- a/sound/aoa/core/snd-aoa-core.c +++ b/sound/aoa/core/snd-aoa-core.c @@ -13,7 +13,7 @@ #include "snd-aoa-alsa.h" MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); /* We allow only one fabric. This simplifies things, diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c index 9880628..f421ae7 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c @@ -17,7 +17,7 @@ #include "../aoa.h" #include "../soundbus/soundbus.h" -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa"); diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 8b2e9b9..d5f6851 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -9,7 +9,7 @@ #include <linux/module.h> #include "soundbus.h" -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Apple Soundbus"); diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index 0fccdbf..a99e26c 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -21,7 +21,7 @@ #include "i2sbus.h" MODULE_LICENSE("GPL"); -MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); +MODULE_AUTHOR("Johannes Berg"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); /* for auto-loading, declare that we handle this weird * string that macio puts into the relevant device */ diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index c7e1b26..0eaf1d0 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -106,7 +106,7 @@ /* {{{ Type definitions */ -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); +MODULE_AUTHOR("Tomas Kasparek"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 51ad95b..2198ecb 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -31,7 +31,7 @@ #include <sound/hwdep.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Hardware dependent layer"); MODULE_LICENSE("GPL"); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f057430..0f79a51 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -37,7 +37,7 @@ #endif -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Takashi Iwai, Jaroslav Kysela"); MODULE_DESCRIPTION("Memory allocator for ALSA system."); MODULE_LICENSE("GPL"); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index fccad8f..36d417a 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -33,7 +33,7 @@ #define OSS_ALSAEMULVER _SIOR ('M', 249, int) -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Mixer OSS emulation for ALSA."); MODULE_LICENSE("GPL"); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index fc11572..97dcf58 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -48,7 +48,7 @@ static int dsp_map[SNDRV_CARDS]; static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int nonblock_open = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela, Abramo Bagnara"); MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); MODULE_LICENSE("GPL"); module_param_array(dsp_map, int, NULL, 0444); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 2743414..8be04b2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -30,7 +30,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Jaroslav Kysela, Abramo Bagnara"); MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index e470c3c..1419c4a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -37,7 +37,7 @@ #include <sound/minors.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_fm.c b/sound/core/seq/instr/ainstr_fm.c index f80fab8..e9d680d 100644 --- a/sound/core/seq/instr/ainstr_fm.c +++ b/sound/core/seq/instr/ainstr_fm.c @@ -25,7 +25,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture FM Instrument support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c index c640e1c..3d4a411 100644 --- a/sound/core/seq/instr/ainstr_gf1.c +++ b/sound/core/seq/instr/ainstr_gf1.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture GF1 (GUS) Patch support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c index 5367bae..d2d42c9 100644 --- a/sound/core/seq/instr/ainstr_iw.c +++ b/sound/core/seq/instr/ainstr_iw.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture IWFFFF support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/instr/ainstr_simple.c b/sound/core/seq/instr/ainstr_simple.c index ac717be..12aeb66 100644 --- a/sound/core/seq/instr/ainstr_simple.c +++ b/sound/core/seq/instr/ainstr_simple.c @@ -26,7 +26,7 @@ #include <sound/initval.h> #include <asm/uaccess.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture Simple Instrument support."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index bc09923..fcb85f0 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -33,7 +33,7 @@ /* * module option */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("OSS-compatible sequencer module"); MODULE_LICENSE("GPL"); /* Takashi says this is really only for sound-service-0-, but this is OK. */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 2f0d877..c5f98e0 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -53,7 +53,7 @@ int seq_default_timer_device = int seq_default_timer_subdevice = 0; int seq_default_timer_resolution = 0; /* Hz */ -MODULE_AUTHOR("Frank van de Pol <fvdpol(a)coil.demon.nl>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Frank van de Pol, Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 37852cd..c05f067 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -47,7 +47,7 @@ #include <linux/slab.h> #include <linux/mutex.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index e55488d..bc84eb0 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -60,7 +60,7 @@ */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ALSA sequencer MIDI-through client"); MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-seq-client-" __stringify(SNDRV_SEQ_CLIENT_DUMMY)); diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c index f30d171..1332de0 100644 --- a/sound/core/seq/seq_instr.c +++ b/sound/core/seq/seq_instr.c @@ -26,7 +26,7 @@ #include <sound/seq_instr.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer instrument library."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 1daa5b0..07e8738 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -40,7 +40,7 @@ Possible options for midisynth module: #include <sound/seq_midi_event.h> #include <sound/initval.h> -MODULE_AUTHOR("Frank van de Pol <fvdpol(a)coil.demon.nl>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Frank van de Pol, Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI synth."); MODULE_LICENSE("GPL"); static int output_buffer_size = PAGE_SIZE; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index d7c4fb8..bf23c5c 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -39,7 +39,7 @@ #include <sound/initval.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Takashi Iwai / Steve Ratcliffe"); +MODULE_AUTHOR("Takashi Iwai, Steve Ratcliffe"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI emulation."); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 5ff80b7..e0e7557 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -28,7 +28,7 @@ #include <sound/seq_midi_event.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Takashi Iwai, Jaroslav Kysela"); MODULE_DESCRIPTION("MIDI byte <-> sequencer event coder"); MODULE_LICENSE("GPL"); diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 061a7c6..52ce7c9 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -48,7 +48,7 @@ #include <sound/seq_midi_event.h> #include <sound/seq_virmidi.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Virtual Raw MIDI client on Sequencer"); MODULE_LICENSE("GPL"); diff --git a/sound/core/sound.c b/sound/core/sound.c index 70600df..fac94ea 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -42,7 +42,7 @@ EXPORT_SYMBOL(snd_major); static int cards_limit = 1; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Advanced Linux Sound Architecture driver for soundcards."); MODULE_LICENSE("GPL"); module_param(major, int, 0444); diff --git a/sound/core/timer.c b/sound/core/timer.c index 67520b3..e7c0552 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -44,7 +44,7 @@ #endif static int timer_limit = DEFAULT_TIMER_LIMIT; -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela, Takashi Iwai"); MODULE_DESCRIPTION("ALSA timer interface"); MODULE_LICENSE("GPL"); module_param(timer_limit, int, 0444); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index a0f28f5..5a9e5cc 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -34,7 +34,7 @@ #include <sound/rawmidi.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Dummy soundcard (/dev/null)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 2de181a..6f7af4c 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -30,7 +30,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("MPU-401 UART"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 3daa9fa..9aa30a7 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -39,7 +39,7 @@ #include <sound/core.h> #include <sound/mpu401.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of MPU-401 in UART mode"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 6c9f4c9..cd4e604 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -47,7 +47,7 @@ MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); module_param_array(enable, bool, NULL, S_IRUGO); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -MODULE_AUTHOR("Matthias Koenig <mk(a)phasorlab.de>"); +MODULE_AUTHOR("Matthias Koenig"); MODULE_DESCRIPTION("ESI Miditerminal 4140"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESI,Miditerminal 4140}}"); diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 87fe376..f041d4e 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -31,7 +31,7 @@ #include <linux/ioport.h> #include <sound/minors.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Hannu Savolainen 1993-1996, Rob Hooft"); +MODULE_AUTHOR("Jaroslav Kysela, Hannu Savolainen, Rob Hooft"); MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index 96762c9..d5771d9 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -27,7 +27,7 @@ #include <linux/moduleparam.h> #include <sound/initval.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ALSA driver for OPL3 FM synth"); diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 01997f2..8611a8e 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -23,7 +23,7 @@ #include <linux/init.h> #include <asm/io.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("OPL4 driver"); MODULE_LICENSE("GPL"); diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index 43d8a2b..96f5b50 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -36,7 +36,7 @@ #include <linux/moduleparam.h> #include <sound/initval.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("OPL4 wavetable synth driver"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 46f3d34..fb324c2 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -55,7 +55,7 @@ /* hack: OSS defines midi_devs, so undefine it (versioned symbols) */ #undef midi_devs -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Dummy soundcard for virtual rawmidi devices"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}"); diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index ed19bc1..594f577 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -35,7 +35,7 @@ #include <sound/vx_core.h> #include "vx_cmd.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Common routines for Digigram VX drivers"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 64388cb..73586e7 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -32,7 +32,7 @@ static void snd_cs8427_reset(struct snd_i2c_device *cs8427); -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index b60fb18..1218b9b 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/i2c.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Generic i2c interface for ALSA"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index b074fdd..7d18dd6 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -921,7 +921,7 @@ static void __exit uda1341_exit(void) module_init(uda1341_init); module_exit(uda1341_exit); -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek(a)seznam.cz>"); +MODULE_AUTHOR("Tomas Kasparek"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index adbfd58..c32b717 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -29,7 +29,7 @@ #include <sound/ak4114.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index c022f29..0391f31 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -29,7 +29,7 @@ #include <sound/ak4117.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("AK4117 IEC958 (S/PDIF) receiver by Asahi Kasei"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 8805110..a38d2a2 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -31,7 +31,7 @@ #include <sound/tlv.h> #include <sound/ak4xxx-adda.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Jaroslav Kysela, Takashi Iwai"); MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index e91cc3b..c8499ac 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -26,7 +26,7 @@ #include <sound/i2c.h> #include <sound/pt2258.h> -MODULE_AUTHOR("Jochen Voss <voss(a)seehuhn.de>"); +MODULE_AUTHOR("Jochen Voss"); MODULE_DESCRIPTION("PT2258 volume controller (Princeton Technology Corp.)"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 4c2fd14..6ef2981 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -28,7 +28,7 @@ #include <sound/core.h> #include <sound/tea575x-tuner.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index ae5b1e3..91e9ab5 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -27,7 +27,7 @@ #include <sound/control.h> #include <sound/tea6330t.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of the TEA6330T circuit via i2c bus"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 5903450..74adb20 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -32,7 +32,7 @@ #define PFX "ad1816a: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("AD1816A, AD1815"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Highscreen,Sound-Boostar 16 3D}," diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index 74e501d..49f1ef3 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -32,7 +32,7 @@ #include <sound/ad1848.h> #include <sound/initval.h> -MODULE_AUTHOR("Tugrul Galatali <galatalt(a)stuy.edu>, Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Tugrul Galatali, Jaroslav Kysela"); MODULE_DESCRIPTION("AD1848/AD1847/CS4248"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848}," diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 8094282..edad9ce 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -35,7 +35,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of AD1848/AD1847/CS4248"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/als100.c b/sound/isa/als100.c index f2bcfb2..8939193 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -34,7 +34,7 @@ #define PFX "als100: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Avance Logic ALS1X0"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP}," diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index b615538..e974387 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -45,7 +45,7 @@ #define PFX "azt2320: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Aztech Systems AZT2320"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech Systems,PRO16V}," diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index c09a800..3d4b2f3 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -62,7 +62,7 @@ /* */ -MODULE_AUTHOR("George Talusan <gstalusan(a)uwaterloo.ca>"); +MODULE_AUTHOR("George Talusan"); MODULE_DESCRIPTION("C-Media CMI8330"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}"); diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index 696a5c8..4a0e35d 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -32,7 +32,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Generic CS4231"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}"); diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 75c7c5f..77b7af7 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -39,7 +39,7 @@ #include <asm/dma.h> #include <asm/irq.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 07ffd5c..9dbcd2f 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -32,7 +32,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifdef CS4232 MODULE_DESCRIPTION("Cirrus Logic CS4232"); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 7a5a6c7..2cd2055 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -89,7 +89,7 @@ #include <sound/cs4231.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index ce57d52..7955a1e 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -34,7 +34,7 @@ #define PFX "dt019x: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X}," diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 65f97ff..f496783 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -35,7 +35,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ESS ESx688 AudioDrive"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100}," diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index a2ab99f..2e407e8 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ESS ESx688 lowlevel module"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 725c115..e951f05 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1983,7 +1983,7 @@ static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) /* Card level */ -MODULE_AUTHOR("Christian Fischbach <fishbach(a)pool.informatik.rwth-aachen.de>, Abramo Bagnara <abramo(a)alsa-project.org>"); +MODULE_AUTHOR("Christian Fischbach, Abramo Bagnara"); MODULE_DESCRIPTION("ESS ES18xx AudioDrive"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES1868 PnP AudioDrive}," diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 8ced5e8..45ca31c 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -31,7 +31,7 @@ #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c index 3e4d4d6..4ebd902 100644 --- a/sound/isa/gus/gus_synth.c +++ b/sound/isa/gus/gus_synth.c @@ -26,7 +26,7 @@ #include <sound/gus.h> #include <sound/seq_device.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards - Synthesizer"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 0395e2e..ef6edeb 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -33,7 +33,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound Classic"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}"); diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 4f55fc3..76c04f5 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -37,7 +37,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound Extreme"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}"); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index d1ad90c..9ec9486 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -34,7 +34,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Gravis UltraSound MAX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}"); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 4ec2d79..08f1099 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -41,7 +41,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifndef SNDRV_STB MODULE_DESCRIPTION("AMD InterWave"); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index f3db686..34f84d0 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -37,7 +37,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Yamaha OPL3SA2+"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S}," diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 1dd9837..758c511 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -43,7 +43,7 @@ #include <sound/initval.h> #include "miro.h" -MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>"); +MODULE_AUTHOR("Martin Langer"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Miro miroSOUND PCM1 pro, PCM12, PCM20 Radio"); MODULE_SUPPORTED_DEVICE("{{Miro,miroSOUND PCM1 pro}, " diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index df22737..bf172e9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -53,7 +53,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_LICENSE("GPL"); #ifdef OPTi93X MODULE_DESCRIPTION("OPTi93X"); diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index d4b2187..2d8b942 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -31,7 +31,7 @@ #define PFX "es968: " -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); +MODULE_AUTHOR("Massimo Piccioni"); MODULE_DESCRIPTION("ESS AudioDrive ES968"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,AudioDrive ES968}}"); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index d64e67f..ac84169 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -44,7 +44,7 @@ #define PFX "sb16: " #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_LICENSE("GPL"); #ifndef SNDRV_SBAWE MODULE_DESCRIPTION("Sound Blaster 16"); diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 3d9d7e0..fd27bac 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -33,7 +33,7 @@ #include <sound/sb16_csp.h> #include <sound/initval.h> -MODULE_AUTHOR("Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Uros Bizjak"); MODULE_DESCRIPTION("ALSA driver for SB16 Creative Signal Processor"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 383911b..1c290a8 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -45,7 +45,7 @@ #include <sound/control.h> #include <sound/info.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index be1e83e..f11e672 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -31,7 +31,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}"); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5e..8bdca08 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -38,7 +38,7 @@ #include <sound/core.h> #include <sound/sb.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Uros Bizjak <uros(a)kss-loka.si>"); +MODULE_AUTHOR("Jaroslav Kysela, Uros Bizjak"); MODULE_DESCRIPTION("Routines for control of 8-bit SoundBlaster cards and clones"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f38..658b571 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -33,7 +33,7 @@ #include <asm/io.h> #include <asm/dma.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ALSA lowlevel driver for Sound Blaster cards"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c index 4fcd0f4..76d7cbf 100644 --- a/sound/isa/sgalaxy.c +++ b/sound/isa/sgalaxy.c @@ -38,7 +38,7 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -MODULE_AUTHOR("Christopher Butler <chrisb(a)sandy.force9.co.uk>"); +MODULE_AUTHOR("Christopher Butler"); MODULE_DESCRIPTION("Aztech Sound Galaxy"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech Systems,Sound Galaxy}}"); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index e2fdd5f..826d6d8 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -32,7 +32,7 @@ #include <sound/opl3.h> #include <sound/snd_wavefront.h> -MODULE_AUTHOR("Paul Barton-Davis <pbd(a)op.net>"); +MODULE_AUTHOR("Paul Barton-Davis"); MODULE_DESCRIPTION("Turtle Beach Wavefront"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,Maui/Tropez/Tropez+}}"); diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 8a61a11..3d4986a 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -48,7 +48,7 @@ #include <asm/mach-au1x00/au1000.h> #include <asm/mach-au1x00/au1000_dma.h> -MODULE_AUTHOR("Charles Eidsness <charles(a)cooper-street.com>"); +MODULE_AUTHOR("Charles Eidsness"); MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}"); diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index ff705c6..d549168 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -1036,7 +1036,7 @@ alsa_harmony_fini(void) } MODULE_LICENSE("GPL"); -MODULE_AUTHOR("Kyle McMartin <kyle(a)parisc-linux.org>"); +MODULE_AUTHOR("Kyle McMartin"); MODULE_DESCRIPTION("Harmony sound driver"); module_init(alsa_harmony_init); diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 3bfb210..fe48c1b 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -39,7 +39,7 @@ #include "ac97_id.h" #include "ac97_patch.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Universal interface for Audio Codec '97"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index dc26820..6b43fd4 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -29,7 +29,7 @@ #include <sound/ak4531_codec.h> #include <sound/tlv.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 98970d4..eb1ae50 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -53,7 +53,7 @@ #define AD1889_DRVVER "Version: 1.7" -MODULE_AUTHOR("Kyle McMartin <kyle(a)parisc-linux.org>, Thibaut Varene <t-bone(a)parisc-linux.org>"); +MODULE_AUTHOR("Kyle McMartin, Thibaut Varene"); MODULE_DESCRIPTION("Analog Devices AD1889 ALSA sound driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1889}}"); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index ba7fa22..14b5537 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -41,7 +41,7 @@ #include <sound/mpu401.h> #include <sound/initval.h> -MODULE_AUTHOR("Matt Wu <Matt_Wu(a)acersoftech.com.cn>"); +MODULE_AUTHOR("Matt Wu"); MODULE_DESCRIPTION("ALI M5451"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8afcb98..b0c922f 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -109,7 +109,7 @@ enum {DEVICE_ALS300, DEVICE_ALS300_PLUS}; -MODULE_AUTHOR("Ash Willis <ashwillis(a)programmer.net>"); +MODULE_AUTHOR("Ash Willis"); MODULE_DESCRIPTION("Avance Logic ALS300"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS300},{Avance Logic,ALS300+}}"); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 8fb55d3..af56e9e 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -79,7 +79,7 @@ #include <sound/sb.h> #include <sound/initval.h> -MODULE_AUTHOR("Bart Hartgers <bart(a)etpmod.phys.tue.nl>"); +MODULE_AUTHOR("Bart Hartgers"); MODULE_DESCRIPTION("Avance Logic ALS4000"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}"); diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7d8053b..3ce34a3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -35,7 +35,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ATI IXP AC97 controller"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250/300/400}}"); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 904023f..346912a 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -35,7 +35,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("ATI IXP MC97 controller"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250}}"); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 43edd28..8321f55 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -126,7 +126,7 @@ #include <sound/initval.h> #include "azt3328.h" -MODULE_AUTHOR("Andreas Mohr <andi AT lisas.de>"); +MODULE_AUTHOR("Andreas Mohr"); MODULE_DESCRIPTION("Aztech AZF3328 (PCI168)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index e9b029e..b8a2130 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -35,7 +35,7 @@ #include <sound/control.h> #include <sound/initval.h> -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("Brooktree Bt87x audio driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878}," diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index ea6712b..416567f 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -145,7 +145,7 @@ #include <sound/ac97_codec.h> #include <sound/info.h> -MODULE_AUTHOR("James Courtier-Dutton <James(a)superbug.demon.co.uk>"); +MODULE_AUTHOR("James Courtier-Dutton"); MODULE_DESCRIPTION("CA0106"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}"); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 7d3c5ee..ed43681 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -41,7 +41,7 @@ #include <sound/asoundef.h> #include <sound/initval.h> -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("C-Media CMI8x38 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738}," diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 44cf546..9bec625 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -38,7 +38,7 @@ #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Cirrus Logic CS4281"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}"); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 8b6cd14..6e0926a 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -34,7 +34,7 @@ #include <sound/cs46xx.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e413da0..2f7c3f6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -16,7 +16,7 @@ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ -MODULE_AUTHOR("Giuliano Pochini <pochini(a)shiny.it>"); +MODULE_AUTHOR("Giuliano Pochini"); MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver"); MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}"); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 55caf34..34b9c35 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -32,7 +32,7 @@ #include <sound/emu10k1.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("EMU10K1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index bb0fec7..f299cf3 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -43,7 +43,7 @@ #include <sound/info.h> #include <sound/rawmidi.h> -MODULE_AUTHOR("Francisco Moraes <fmoraes(a)nc.rr.com>"); +MODULE_AUTHOR("Francisco Moraes"); MODULE_DESCRIPTION("EMU10K1X"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Dell Creative Labs,SB Live!}"); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 425b167..48ff186 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -61,7 +61,7 @@ #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, Thomas Sailer <sailer(a)ife.ee.ethz.ch>"); +MODULE_AUTHOR("Jaroslav Kysela, Thomas Sailer"); MODULE_LICENSE("GPL"); #ifdef CHIP1370 MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370"); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fec29a1..0e084d3 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -66,7 +66,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaromir Koutek <miri(a)punknet.cz>"); +MODULE_AUTHOR("Jaromir Koutek"); MODULE_DESCRIPTION("ESS Solo-1"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,ES1938}," diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 6dc578b..02256f2 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -42,7 +42,7 @@ #define TEA575X_RADIO 1 #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ForteMedia FM801"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801}," diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8f34fb4..48d05e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -34,7 +34,7 @@ #include "hda_local.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Universal interface for High Definition Audio Codec"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index ab00cce..fab0813 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -30,7 +30,7 @@ #include <sound/initval.h> #include "ice1712.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ICEnsemble ICE17xx <-> AK4xxx AD/DA chip interface"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 830a1bb..2d62c5e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -73,7 +73,7 @@ #include "ews.h" #include "hoontech.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("ICEnsemble ICE1712 (Envy24)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{" diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1127ebd..aeff346 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -52,7 +52,7 @@ #include "phase.h" #include "wtm.h" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{" diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 7cf2dcb..ba95dc2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -43,7 +43,7 @@ #include <asm/pgtable.h> #include <asm/cacheflush.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index c155e1f..8736c02 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -37,7 +37,7 @@ #include <sound/info.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; " "SiS 7013; NVidia MCP/2/2S/3 modems"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 21d0899..9b98090 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -429,7 +429,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Korg 1212 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); -MODULE_AUTHOR("Haroldo Gamal <gamal(a)alternex.com.br>"); +MODULE_AUTHOR("Haroldo Gamal"); static struct pci_device_id snd_korg1212_ids[] = { { diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 4526904..0f9f218 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -51,7 +51,7 @@ #include <sound/initval.h> #include <asm/byteorder.h> -MODULE_AUTHOR("Zach Brown <zab(a)zabbo.net>, Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Zach Brown, Takashi Iwai"); MODULE_DESCRIPTION("ESS Maestro3 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ESS,Maestro3 PCI}," diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index ac007ce..c2ebdd9 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -42,7 +42,7 @@ #define CARD_NAME "miXart" -MODULE_AUTHOR("Digigram <alsa(a)digigram.com>"); +MODULE_AUTHOR("Digigram"); MODULE_DESCRIPTION("Digigram " CARD_NAME); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 03b3a47..a47a5e8 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -44,7 +44,7 @@ #define CARD_NAME "NeoMagic 256AV/ZX" #define DRIVER_NAME "NM256" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("NeoMagic NM256AV/ZX"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{NeoMagic,NM256AV}," diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index d974134..cd29197 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -44,7 +44,7 @@ #define DRIVER_NAME "pcxhr" -MODULE_AUTHOR("Markus Bollinger <bollinger(a)digigram.com>"); +MODULE_AUTHOR("Markus Bollinger"); MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}"); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 952625d..6854b62 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -113,7 +113,7 @@ #define SUPPORT_JOYSTICK 1 #endif -MODULE_AUTHOR("Peter Gruber <nokos(a)gmx.net>"); +MODULE_AUTHOR("Peter Gruber"); MODULE_DESCRIPTION("riptide"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Conexant,Riptide}}"); diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 6bb7ac6..19fb89a 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -101,7 +101,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable RME Digi32 soundcard."); module_param_array(fullduplex, bool, NULL, 0444); MODULE_PARM_DESC(fullduplex, "Support full-duplex mode."); -MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>, Pilo Chambert <pilo.c(a)wanadoo.fr>"); +MODULE_AUTHOR("Martin Langer, Pilo Chambert"); MODULE_DESCRIPTION("RME Digi32, Digi32/8, Digi32 PRO"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}"); diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index e3304b7..aed6e67 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -43,7 +43,7 @@ /* note, two last pcis should be equal, it is not a bug */ -MODULE_AUTHOR("Anders Torger <torger(a)ludd.luth.se>"); +MODULE_AUTHOR("Anders Torger"); MODULE_DESCRIPTION("RME Digi96, Digi96/8, Digi96/8 PRO, Digi96/8 PST, " "Digi96/8 PAD"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 89b3c7f..70ce246 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -54,7 +54,7 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for RME Hammerfall DSP interface."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable/disable specific Hammerfall DSP soundcards."); -MODULE_AUTHOR("Paul Davis <paul(a)linuxaudiosystems.com>, Marcus Andersson, Thomas Charbonnel <thomas(a)undata.org>"); +MODULE_AUTHOR("Paul Davis, Marcus Andersson, Thomas Charbonnel"); MODULE_DESCRIPTION("RME Hammerfall DSP"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index cc3bdec..8235870 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -51,7 +51,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable/disable specific RME96{52,36} soundcards."); module_param_array(precise_ptr, bool, NULL, 0444); MODULE_PARM_DESC(precise_ptr, "Enable precise pointer (doesn't work reliably)."); -MODULE_AUTHOR("Paul Davis <pbd(a)op.net>, Winfried Ritsch"); +MODULE_AUTHOR("Paul Davis, Winfried Ritsch"); MODULE_DESCRIPTION("RME Digi9652/Digi9636"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall}," diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 9f25d93..1fff81b 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -42,7 +42,7 @@ #include <asm/io.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("S3 SonicVibes PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}"); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 9145f7c..f6019a4 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -30,7 +30,7 @@ #include <sound/trident.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>, <audio(a)tridentmicro.com>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Trident 4D-WaveDX/NX & SiS SI7018"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Trident,4DWave DX}," diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c index 9b7dee8..e29553e 100644 --- a/sound/pci/trident/trident_synth.c +++ b/sound/pci/trident/trident_synth.c @@ -28,7 +28,7 @@ #include <sound/trident.h> #include <sound/seq_device.h> -MODULE_AUTHOR("Scott McNab <jedi(a)tartarus.uwa.edu.au>"); +MODULE_AUTHOR("Scott McNab"); MODULE_DESCRIPTION("Routines for Trident 4DWave NX/DX soundcards - Synthesizer"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index a289922..e905bd3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -68,7 +68,7 @@ #define POINTER_DEBUG #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA VT82xx audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C,pci},{VIA,VT8233A/C,8235}}"); diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index b338e15..17a2cb0 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -50,7 +50,7 @@ #define POINTER_DEBUG #endif -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("VIA VT82xx modem"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C modem,pci}}"); diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 474eac9..7aedb2f 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -31,7 +31,7 @@ #define CARD_NAME "VX222" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Digigram VX222 V2/Mic"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index fd9b7b8..73e204f 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -30,7 +30,7 @@ #include <sound/opl3.h> #include <sound/initval.h> -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Yamaha DS-1 PCI"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724}," diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 2d40cc7..3176554 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -33,7 +33,7 @@ #define CARD_NAME "PDAudio-CF" -MODULE_AUTHOR("Jaroslav Kysela <perex(a)suse.cz>"); +MODULE_AUTHOR("Jaroslav Kysela"); MODULE_DESCRIPTION("Sound Core " CARD_NAME); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Sound Core," CARD_NAME "}}"); diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 363bcb5..7d58d16 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -32,7 +32,7 @@ /* */ -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("Digigram VXPocket"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram,VXPocket},{Digigram,VXPocket440}}"); diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c index 9fc0c03..3b33274 100644 --- a/sound/soc/at91/at91-i2s.c +++ b/sound/soc/at91/at91-i2s.c @@ -716,6 +716,6 @@ struct snd_soc_cpu_dai at91_i2s_dai[NUM_SSC_DEVICES] = { EXPORT_SYMBOL_GPL(at91_i2s_dai); /* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino(a)endrelia.com, www.endrelia.com"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("AT91 I2S ASoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index b39b95a..f548391 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -427,6 +427,6 @@ struct snd_soc_platform at91_soc_platform = { EXPORT_SYMBOL_GPL(at91_soc_platform); -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("Atmel AT91 PCM module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 8179df3..58bcbb8 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -370,6 +370,6 @@ module_init(eti_b1_init); module_exit(eti_b1_exit); /* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino(a)endrelia.com>"); +MODULE_AUTHOR("Frank Mandarino"); MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 50c5c83..3da33b6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -313,6 +313,6 @@ struct snd_soc_cpu_dai pxa_i2s_dai = { EXPORT_SYMBOL_GPL(pxa_i2s_dai); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 36519ae..49be095 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1582,6 +1582,6 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7caf8c7..303846a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1318,6 +1318,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b6d8863..21ec11c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -57,7 +57,7 @@ #include "usbaudio.h" -MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); +MODULE_AUTHOR("Takashi Iwai"); MODULE_DESCRIPTION("USB Audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}"); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 24f5a26..b0d304e 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -64,7 +64,7 @@ #define ERROR_DELAY_JIFFIES (HZ / 10) -MODULE_AUTHOR("Clemens Ladisch <clemens(a)ladisch.de>"); +MODULE_AUTHOR("Clemens Ladisch"); MODULE_DESCRIPTION("USB Audio/MIDI helper module"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index e011fca..926d652 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -147,7 +147,7 @@ -MODULE_AUTHOR("Karsten Wiese <annabellesgarden(a)yahoo.de>"); +MODULE_AUTHOR("Karsten Wiese"); MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604), "NAME_ALLCAPS"(0x8001)(0x8005)(0x8007) }}");

3 6

Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Ciaccia 22 May '07

22 May '07

It looks like your application uses the following pattern: for (;;) { ReadFromAlsa WriteToAlsa } The ReadFormAlsa function waits for the capture channel to become available (i.e., there is some data to be read), while WriteToAlsa waits till the output buffer becomes available (i.e., it's possible to send data to the card without blocking). Ho do you know that this is the correct order? Could it be that you should before write data to the output buffer and then read data from the capture stream? I think your design is not completely correct... Look at my code, it works fine, but maybe there is a better approach... Andrea ----- Original Message ---- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Ciaccia <ciacciax(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Tuesday, May 22, 2007 10:53:54 AM Subject: Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ????? I have a single threaded ALSA application which does simultaneous capture and playback. But i am getting overflow error XRUN from capture side. I am attaching the code to this mail. Is there any problem with single threaded approach? Is there any global configuration for ALSA lib? When i run this app, ALSA state is getting corrupted, even aplay and arecord is not working properly after that. Any comments on this issue? Thanks in advance Nobin Mathew On 5/22/07, Ciaccia <ciacciax(a)yahoo.com> wrote: > > What are the complications in implementing simultaneous playback and > > capture in ALSA??? > > > I would like to add something to this post: what is the "best" way to > program a full duplex application? I would like to have an application > that does full-duplex, with the lowest possible latency and using less > CPU resource as possible. I tried different approaches, but I was not > completely satisfied with none of them.... > > > Here is what I tried: > -having two threads, one for playback and one for capture > -defining 2 callbacks with async_handles and do all the processing in the callback functions > (the last attempt) > -using poll() to wait for one of the streams to be ready (see attached file fullduplex.c) > > Could some of you look at the attached code (it is not generic and contains lots of asserts, but it's supposed to work only on one single machine) and tell me if the last approach using poll is well structured? > > The main loop looks as following: > > > while (1) { > > err = poll(ufds, 2, -1); > > > > for (i = 0; i < 2; i++) > > { > > if (snd_pcm_poll_descriptors_revents(handles[i], &ufds[i], 1, &revents) < 0) > > { > > printf("Error getting revents for %s\n", descriptors[i].name); > > exit(1); > > } > > > > if (revents & descriptors[i].poll_flag) > > { > > if (transfer_loop(handles[i], &first[i], descriptors[i].func) < 0) > > { > > printf("transfer_loop error for %s\n", descriptors[i].name); > > // TODO exit?!? > > } > > } > > } > > } > > and the transfer_loop is a modified version of the loop found in the ALSA examples (pcm.c and latency.c). > Is this strategy optimal in respect to low latency and low CPU usage? > > Every comment is greatly appreciated > Andrea > > > > > > ____________________________________________________________________________________Luggage? GPS? Comic books? > Check out fitting gifts for grads at Yahoo! Search > http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz > ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

1 0

Re: [alsa-devel] Any standard ALSA application for simultaneous capture and playabck ?????
by Ciaccia 22 May '07

22 May '07

> What are the complications in implementing simultaneous playback and > capture in ALSA??? I would like to add something to this post: what is the "best" way to program a full duplex application? I would like to have an application that does full-duplex, with the lowest possible latency and using less CPU resource as possible. I tried different approaches, but I was not completely satisfied with none of them.... Here is what I tried: -having two threads, one for playback and one for capture -defining 2 callbacks with async_handles and do all the processing in the callback functions (the last attempt) -using poll() to wait for one of the streams to be ready (see attached file fullduplex.c) Could some of you look at the attached code (it is not generic and contains lots of asserts, but it's supposed to work only on one single machine) and tell me if the last approach using poll is well structured? The main loop looks as following: > while (1) { > err = poll(ufds, 2, -1); > > for (i = 0; i < 2; i++) > { > if (snd_pcm_poll_descriptors_revents(handles[i], &ufds[i], 1, &revents) < 0) > { > printf("Error getting revents for %s\n", descriptors[i].name); > exit(1); > } > > if (revents & descriptors[i].poll_flag) > { > if (transfer_loop(handles[i], &first[i], descriptors[i].func) < 0) > { > printf("transfer_loop error for %s\n", descriptors[i].name); > // TODO exit?!? > } > } > } > } and the transfer_loop is a modified version of the loop found in the ALSA examples (pcm.c and latency.c). Is this strategy optimal in respect to low latency and low CPU usage? Every comment is greatly appreciated Andrea ____________________________________________________________________________________Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz

2 1

[alsa-devel] Firmware to add - Dreamcast AICA
by Adrian McMenamin 22 May '07

22 May '07

Please add this to the firmware repo

2 4

[alsa-devel] Question on buffer size setting
by Allyn, Mark A 21 May '07

21 May '07

Hello: I am attempting to do some debugging of a new audio device, and I need to change the number of buffer descriptor list entries from the default of 16 to something much smaller, like 3. I am a bit confused while looking at the code. There seems to be a value labeled SNDRV_PCM_HW_PARAM_BUFFER_SIZE, but I can't figure how it is set. Is this a parameter that is passed as part of the configure prior to doing a make for the driver? Is this a parameter passed to modprobe when loading the driver? Thanks for your help. Mark

2 1

[alsa-devel] [PATCH] hda-intel: fix ASUS M2V detection
by Daniel Drake 21 May '07

21 May '07

Commit f32610edab47f36946d23b883aeae91e15986121 added ALC660VD support, but this caused a 2.6.21 regression for some users. The ASUS M2V device is now detected as ALC660VD rather than ALC660/861 but the PCI quirk was not carried over. This patch allows affected users to use audio again. http://bugzilla.kernel.org/show_bug.cgi?id=8273 https://bugs.gentoo.org/show_bug.cgi?id=178243 Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/pci/hda/patch_realtek.c =================================================================== --- linux.orig/sound/pci/hda/patch_realtek.c +++ linux/sound/pci/hda/patch_realtek.c @@ -8384,7 +8384,6 @@ static struct snd_pci_quirk alc861_cfg_t SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), @@ -8990,6 +8989,7 @@ static const char *alc861vd_models[ALC86 static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),

2 1

Re: [alsa-devel] porter wanted: AMD Geode SCx200 driver OSS to ALSA
by Ash Willis 20 May '07

20 May '07

Sorry for this really late reply, I just wanted to add a note: > > (Would now also be a good time to ask for snd-sc5535audio to be > > renamed snd-sc5535 for consistence with other driver names?) > > I agree with you, I think it should be renamed. I've just ported the SND_KAHLUA driver from OSS (patch not released yet). It covers the cs5530 (VSA1 only). I've named it snd-cs5530. This audio suffix seems a bit wacky to me. Surely the snd prefix is adequate to distinguish it from an IDE driver, for example :) Ash = Nose Surgery Specialist in Manhattan, NY Rhinoplasty surgeon Geoffrey W. Tobias has pioneered the "un-nose job," the nose job that doesn't look like one. Chosen by "New York Magazine" as a nose expert. Video clips. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=1b57edb21974e0b43470c0c…

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[alsa-devel] Strange findings about the microphone in SigmaTel STAC9221 A1
by Ivan N. Zlatev 19 May '07

19 May '07

My hda-intel card is with the following codec: Codec: SigmaTel STAC9221 A1 Address: 0 Vendor Id: 0x83847680 Subsystem Id: 0x106b1700 Revision Id: 0x103401 I am using alsa 1.0.14rc4. I have a built-in microphone, which doesn't work, but today I found out something interesting. If I set the capture to minimum and place Mux at specific positions (Works with middle-ish and top, not bottom-ish) suddenly the mic started recording, but the quality is extremly bad. I have attached amixer and co. Does anyone have a clue what the issue is and how could possibly this little bit of random foundings to be used to possibly fix the mic? Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] ICE1712/1724 HW mixing?
by Rene Herman 19 May '07

19 May '07

Good day. I've been updating the TerraTec vendor page in the new wiki matrix: http://bugtrack.alsa-project.org/main/index.php/Matrix:Vendor-Terratec The problem with information that looks maintained though is that when something's _not_ mentioned, it might be interpreted to mean it's not actually working.... I have the older TerraTec cards (both ISA and PCI) but not any of the ICE1712/ICE1724 ones. Could someone verify/confirm that the ICE1712 (Envy24?) does not support 24-bit samples? And that ICE1724 supports only 32-bit in hardware? Does either chip/driver support hardware mixing? Filling out some of the Digital I/O tags for any Aureon card that you know would be nice as well. Rene.

3 5

[alsa-devel] I don't understand snd_pcm_ops
by Timur Tabi 19 May '07

19 May '07

I'm new to ALSA driver development, and I'm writing an SOC driver for new hardware. Can someone explain to me what the snd_pcm_ops structure is for? And it sounds like a stupid question, but after studying documentation and sample code, I'm still confused. First off, I thought PCM stood for Pulse Code Modulation, but that just describes the format of the digital data. I don't understand what a PCM operation could be. I've seen documentation where PCM is considered an alternative to I2S or AC97. But if you look at eti_b1_wm8731.c, you'll see this: #include "at91-pcm.h" #include "at91-i2s.h" So is the AT91 a PCM device or an I2S device??? Some background: I'm writing driver for a new SOC that has an I2S interface, and it's talking to a CS4270 codec, but the CS4270 codec is wired up in "stand-alone" mode, so there's no software configuration. -- Timur Tabi Linux Kernel Developer @ Freescale

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Re: [alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric De Vito 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear Tobin thank you for your so helpful recommendations ! In fact the key was probe_mask=1. But I did not see this option in the hda-intel module description :-( ... Now I added probe_mask)=1 in /etc/modprobe.d/alsa-base and it works either with 1.0.13 or 14rc4 :-). In fact when alsa is compiled with debug=detect this option is not necessary any more and sound works even without this option ... Does this option (probe_mask=1) limitates something in the sound system ? Is it better to not use it or is it painless for audio possibilities ? Thank you again for your help !!! Now to be more precise : 1) If I compile 14rc4 with debug=detect it can be compiled, modules can be inserted : - --> when inserted with # modprobe -v snd-hda-intel model=clevo I get in /var/log/messages : May 18 22:32:05 gondor kernel: [ 6012.711362] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 22:32:05 gondor kernel: [ 6012.774627] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 May 18 22:32:06 gondor kernel: [ 6013.404197] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1795: hda_codec: model 'clevo' is selected May 18 22:32:08 gondor kernel: [ 6015.083303] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:558: hda_intel: azx_get_response timeout, switching to polling mode... - -> when inserted with # modprobe -v snd-hda-intel model=clevo probe_mask=1 I get in /var/log/messages : May 18 22:32:50 gondor kernel: [ 6057.162063] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 22:32:50 gondor kernel: [ 6057.243574] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0x3 May 18 22:32:50 gondor kernel: [ 6057.873098] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/hda_codec.c:1795: hda_codec: model 'clevo' is selected 2) If I compile 14rc4 with debug=none it can be compiled, modules can be inserted : - --> when inserted with # modprobe -v snd-hda-intel model=clevo I get in /var/log/messages : May 18 19:39:46 gondor kernel: [ 2366.086099] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 May 18 19:39:49 gondor kernel: [ 2368.491630] ALSA /usr/src/modules/alsa-driver-1.0.14rc4/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:558: hda_intel: azx_get_response timeout, switching to polling mode... and aplay gives : eric@gondor:/usr/src/modules/alsa-driver-1.0.14rc4$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:/usr/src/modules/alsa-driver-1.0.14rc4$ - --> when inserted with # modprobe -v snd-hda-intel model=clevo probe_mask=1 May 18 19:41:45 gondor kernel: [ 2484.283312] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 17 (level, low) -> IRQ 20 And it works !!!! Eric Tobin Davis a écrit : > Ok, you have 14rc4 (according to your email). Try loading the driver > with "model=clevo", as it appears that is the system type you have. > Make sure the drivers are built with "--with-debug=detect" so that we > can get good dmesg output. You shouldn't need any other parameters. > The clevo model was added not too long ago. The other model to try is > "model=test". This will give you control over all nodes and functions. > > You may also need to add "probe_mask=1" (or something like that) to > limit the number of codecs being detected. I don't know why > cat /proc/asound/card0/codec#0 would hang. > > Tobin > > On Mon, 2007-05-14 at 14:56 +0200, Eric wrote: > >> I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio >> chipset >> >> eric@gondor:~$ lspci |grep Audio >> 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 >> Family) High Definition Audio Controller (rev 03) >> >> When modprobing the modules I get : >> eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout >> insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko >> insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout >> >> while /var/log/messages says : >> May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: >> azx_get_response timeout, switching to polling mode... >> >> >> Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the >> results of aplay : >> >> eric@gondor:~$ aplay >> ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) >> snd_pcm_hw_params_any failed >> ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave >> aplay: main:550: Erreur d'ouverture audio: Argument invalide >> eric@gondor:~$ >> >> Here are some other infos : >> >> eric@gondor:~$ tail -2 /proc/asound/oss/sndstat >> 0: Realtek ALC880 >> 1: SAA7134 Mixer >> >> amixer shows this : >> eric@gondor:~$ amixer >> Simple mixer control 'Headphone',0 >> Capabilities: pswitch >> Playback channels: Front Left - Front Right >> Mono: >> Front Left: Playback [on] >> Front Right: Playback [on] >> Simple mixer control 'PCM',0 >> Capabilities: pvolume >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 255 >> Mono: >> Front Left: Playback 228 [89%] >> Front Right: Playback 228 [89%] >> Simple mixer control 'Front',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 60 [47%] [-2144.00dB] [on] >> Front Right: Playback 60 [47%] [-2144.00dB] [on] >> Simple mixer control 'Front Mic',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 55 [43%] [-2304.00dB] [on] >> Front Right: Playback 55 [43%] [-2304.00dB] [on] >> Simple mixer control 'Surround',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 64 [50%] [-2016.00dB] [on] >> Front Right: Playback 64 [50%] [-2016.00dB] [on] >> Simple mixer control 'Center',0 >> Capabilities: pvolume pvolume-joined pswitch pswitch-joined >> Playback channels: Mono >> Limits: Playback 0 - 127 >> Mono: Playback 56 [44%] [-2272.00dB] [on] >> Simple mixer control 'LFE',0 >> Capabilities: pvolume pvolume-joined pswitch pswitch-joined >> Playback channels: Mono >> Limits: Playback 0 - 127 >> Mono: Playback 58 [46%] [-2208.00dB] [on] >> Simple mixer control 'Line',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 60 [47%] [-2144.00dB] [on] >> Front Right: Playback 60 [47%] [-2144.00dB] [on] >> Simple mixer control 'CD',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 58 [46%] [-2208.00dB] [on] >> Front Right: Playback 58 [46%] [-2208.00dB] [on] >> Simple mixer control 'Mic',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 57 [45%] [-2240.00dB] [on] >> Front Right: Playback 57 [45%] [-2240.00dB] [on] >> Simple mixer control 'IEC958',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> Simple mixer control 'PC Speaker',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 127 >> Mono: >> Front Left: Playback 53 [42%] [-2368.00dB] [on] >> Front Right: Playback 53 [42%] [-2368.00dB] [on] >> Simple mixer control 'Capture',0 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 31 [24%] [-3072.00dB] [on] >> Front Right: Capture 31 [24%] [-3072.00dB] [on] >> Simple mixer control 'Capture',1 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 27 [21%] [-3200.00dB] [on] >> Front Right: Capture 27 [21%] [-3200.00dB] [on] >> Simple mixer control 'Capture',2 >> Capabilities: cvolume cswitch >> Capture channels: Front Left - Front Right >> Limits: Capture 0 - 127 >> Front Left: Capture 24 [19%] [-3296.00dB] [on] >> Front Right: Capture 24 [19%] [-3296.00dB] [on] >> Simple mixer control 'Caller ID',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [on] >> Simple mixer control 'Channel Mode',0 >> Capabilities: enum >> Items: '2ch' '6ch' >> Item0: '6ch' >> Simple mixer control 'Input Source',0 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Input Source',1 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Input Source',2 >> Capabilities: enum >> Items: 'Mic' 'Front Mic' 'Line' 'CD' >> Item0: 'Mic' >> Simple mixer control 'Off-hook',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [on] >> >> aplay -l seems to work : >> >> eric@gondor:~$ aplay -l >> **** Liste des PLAYBACK périphériques **** >> carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 >> Analog] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 >> Digital] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 >> Modem] >> Sous-périphériques: 1/1 >> Sous-périphérique: #0: subdevice #0 >> >> >> eric@gondor:~$ cat /proc/asound/cards >> 0 [Intel ]: HDA-Intel - HDA Intel >> HDA Intel at 0xb0000000 irq 17 >> 1 [SAA7134 ]: SAA7134 - SAA7134 >> saa7133[0] at 0xb3006000 irq 22 >> >> if I try to do a cat /proc/asound/card0/codec#0 it hangs >> while /var/log/messages gives a lot of >> >> May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid >> dep_range_val 0:7fff >> May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid >> dep_range_val 0:7fff >> ........................ >> >> eric@gondor:~$ sudo lspci -nv >> 00:00.0 0600: 8086:2580 (rev 04) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0 >> Capabilities: [e0] Vendor Specific Information >> >> 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) >> Flags: bus master, fast devsel, latency 0 >> Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 >> Memory behind bridge: b1000000-b2ffffff >> Prefetchable memory behind bridge: c0000000-cfffffff >> Capabilities: [88] Subsystem: 8086:2582 >> Capabilities: [80] Power Management version 2 >> Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- >> Queue=0/0 Enable- >> Capabilities: [a0] Express Root Port (Slot+) IRQ 0 >> >> 00:1b.0 0403: 8086:2668 (rev 03) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0, IRQ 20 >> Memory at b0000000 (64-bit, non-prefetchable) [size=16K] >> Capabilities: [50] Power Management version 2 >> Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ >> Queue=0/0 Enable- >> Capabilities: [70] Express Unknown type IRQ 0 >> >> 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 18 >> I/O ports at 1800 [size=32] >> >> 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 19 >> I/O ports at 3000 [size=32] >> >> 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 17 >> I/O ports at 3020 [size=32] >> >> 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 20 >> I/O ports at 3040 [size=32] >> >> 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 18 >> Memory at b0004000 (32-bit, non-prefetchable) [size=1K] >> Capabilities: [50] Power Management version 2 >> Capabilities: [58] Debug port >> >> 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) >> Flags: bus master, fast devsel, latency 0 >> Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 >> I/O behind bridge: 00004000-00004fff >> Memory behind bridge: b3000000-b30fffff >> Prefetchable memory behind bridge: >> 0000000050000000-0000000055ffffff >> Capabilities: [50] Subsystem: 1558:0900 >> >> 00:1f.0 0601: 8086:2640 (rev 03) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0 >> >> 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 0, IRQ 17 >> I/O ports at 01f0 [size=8] >> I/O ports at 03f4 [size=1] >> I/O ports at 0170 [size=8] >> I/O ports at 0374 [size=1] >> I/O ports at 3080 [size=16] >> >> 00:1f.3 0c05: 8086:266a (rev 03) >> Subsystem: 1558:0900 >> Flags: medium devsel, IRQ 10 >> I/O ports at 3060 [size=32] >> >> 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) >> Subsystem: 1558:0900 >> Flags: bus master, fast devsel, latency 0, IRQ 11 >> Memory at b2000000 (32-bit, non-prefetchable) [size=16M] >> Memory at c0000000 (64-bit, prefetchable) [size=256M] >> Memory at b1000000 (64-bit, non-prefetchable) [size=16M] >> Capabilities: [60] Power Management version 2 >> Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ >> Queue=0/0 Enable- >> Capabilities: [78] Express Endpoint IRQ 0 >> >> 0a:00.0 0607: 104c:ac50 (rev 02) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 168, IRQ 17 >> Memory at b3007000 (32-bit, non-prefetchable) [size=4K] >> Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 >> Memory window 0: 50000000-53fff000 (prefetchable) >> Memory window 1: 58000000-5bfff000 >> I/O window 0: 00004800-000048ff >> I/O window 1: 00004c00-00004cff >> 16-bit legacy interface ports at 0001 >> >> 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) >> Subsystem: 1558:0900 >> Flags: bus master, medium devsel, latency 64, IRQ 21 >> Memory at b3004000 (32-bit, non-prefetchable) [size=2K] >> Memory at b3000000 (32-bit, non-prefetchable) [size=16K] >> Capabilities: [44] Power Management version 2 >> >> 0a:02.0 0104: 105a:3373 (rev 02) >> Subsystem: 1558:0900 >> Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 >> I/O ports at 4480 [size=64] >> I/O ports at 44c0 [size=16] >> I/O ports at 4400 [size=128] >> Memory at b3005000 (32-bit, non-prefetchable) [size=4K] >> Memory at b3020000 (32-bit, non-prefetchable) [size=128K] >> Capabilities: [60] Power Management version 2 >> >> 0a:03.0 0200: 10ec:8169 (rev 10) >> Subsystem: 1558:0900 >> Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 >> I/O ports at 4000 [size=256] >> Memory at b3004800 (32-bit, non-prefetchable) [size=256] >> [virtual] Expansion ROM at 54000000 [disabled] [size=128K] >> Capabilities: [dc] Power Management version 2 >> >> 0a:04.0 0480: 1131:7133 (rev f0) >> Subsystem: 1461:f31e >> Flags: bus master, medium devsel, latency 64, IRQ 23 >> Memory at b3006000 (32-bit, non-prefetchable) [size=2K] >> Capabilities: [40] Power Management version 2 >> >> 0a:05.0 0280: 1814:0201 (rev 01) >> Subsystem: 1462:6833 >> Flags: bus master, slow devsel, latency 64, IRQ 20 >> Memory at b3008000 (32-bit, non-prefetchable) [size=8K] >> Capabilities: [40] Power Management version 2 >> >> asoundconf output : >> eric@gondor:~$ asoundconf list >> Names of available sound cards: >> Intel >> SAA7134 >> >> I tried a lot of workarounds but none worked : boot options (irqpoll, >> pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel >> driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... >> >> I must add that I already had this problem with previous distributions >> of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, >> it did worked ONLY when I self-compiled the driver with debug option >> enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu >> kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it >> does not work. I cannot have one sound :-(. >> >> Please help me .... >> >> >> Eric >> >> >> >> >> >> >> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGThYExI+RZbxK37ARAl6lAKDOecNscFCyIk5LSG2J1Tu8thsxiACgvSTX zk0O2gqEl6t/bFmi76jztqw= =LN4P -----END PGP SIGNATURE-----

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[alsa-devel] Linux 2.6.21 regression with hda-intel (Conexant 5045)
by Gunter Ohrner 18 May '07

18 May '07

Hi! First of all: Sorry if this question is dumb or a fix should be obvious, but as a non-ALSA_expert I'm a bit a a loss at the moment... I searched the bugtracker and this list but could not find anything appropriate - most probably I missed a relevant post or didn't recognize it as being relevant... Any pointers are greatly appreciated! The situation is as follows: Using Linux 2.6.20.4, my laptop's soundcard works just fine. Using 2.6.21.1, it also works, but the headphone socket is dead. Plugging a headphone into it correctly mustes the internal speakers, but no sound is output via the headphones. Details: - Fujitsu Simens Amilo Si1520 - Codec: Conexant ID 5045 (as reported by Linux 2.6.20.4) Alsa device details between both Linux kernels differ completely, with Linux 2.6.20.4 only reporting a single a single analog PCM device and Linux 2.6.21.1 reporting a Conexant Analog and a Conexant Digital device. Should I still post the complete information from /proc/asound/* for both kernels? I tried applying conexant-test22.patch by Tobin Davis, but it does not seem to match the in-kernel ALSA code. It also does not seem to be related to the 5045 chip... Any ideas? Greetings, Gunter

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Re: [alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > > It's a process to merge to the upstream kernel. We need to give a > > kind of approval from the patch authors and contributors. > > It consists of a line ok .. something new for me :-) Here it goes: Signed-off-by: Christian Rothlaender <christian(a)rothlaender.net> ================================================================ hda-codec - patch_si3045.c This patch adds support for the ASUS A8J Series 56k Modem (Motorola SM56) Signed-off-by: Christian Rothlaender <christian(a)rothlaender.net> THE PATCH STARTS BELOW ================================================================ > Just a small issue is left. With the alsa driver buildin into 2.6.19 > slmodem is working, with any newer one (either > standalone or buildin driver) we have a "hda-intel: invalid > format_val, rate=9600, ch=1, format=2" error in dmesg. Seems > something has been broken on that step (probably 1.0.13 to > 1.0.14?). Sadly I'm not able to tell right now which alsa was > included into 2.6.19. I'm trying to compare the files to get an idea > what's wrong (#2799 and #2820 are reporting similar > problems). > > > > This should have been fixed in the recent version. > > The driver dropped the support of 9600HZ accidentally. That's really good to hear. Can you close the Bug entry #2732 then? (#2799 and #2820 can be closed too, if the 9600 Hz problem is solved) So thanks a lot ... Christian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTbwUYMBp7Qf+VpkRApyKAJ4xfBkrLiG1D7whZIV1CteUTRhFagCfd/LU 2jeMsy1MQ/WxX5A7P/iThjM= =A9s6 -----END PGP SIGNATURE-----

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Re: [alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there It's support for the SI3054 Chip build into the Asus A8J Notebook Series. In Windows it's called "Motorola SM56". What do you mean with "sign-off"? I'm pretty happy when it walks into upstream, so that the requests on my homepage about how to patch alsa to run that modem are ceasing down. Just a small issue is left. With the alsa driver buildin into 2.6.19 slmodem is working, with any newer one (either standalone or buildin driver) we have a "hda-intel: invalid format_val, rate=9600, ch=1, format=2" error in dmesg. Seems something has been broken on that step (probably 1.0.13 to 1.0.14?). Sadly I'm not able to tell right now which alsa was included into 2.6.19. I'm trying to compare the files to get an idea what's wrong (#2799 and #2820 are reporting similar problems). Christian Takashi Iwai wrote: > Thanks, the patch looks good to me. > Could you give a brief description of the patch and your sign-off to > merge to the upstream? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTbafYMBp7Qf+VpkRAgl1AJ4uWR1vVVE6APAMB2SQjcPmhByJpQCfdweP vJZuXa7/0njBJ0h8vFWdUH0= =Dbm9 -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH] SM56 Modem in Asus A8J Notebook Series
by Christian Rothländer 18 May '07

18 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there after I submitted the solution 5 month ago in the Bug system, I am trying it now directly. Attached to this posting is the necessary patch to get this modem working with slmodemd. The Patch is against alsa-driver-1.0.14rc4/alsa-kernel/pci/hda/patch_si3054.c if included it's Bug #0002732, assigned to tiwai. Greetings Christian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGTadQYMBp7Qf+VpkRAp8MAJ9lii0FA/lBZPRabFxWIiVVJnAh7wCdGEkk fH0mFw4iOP3S9BHcTor3qVo= =suhj -----END PGP SIGNATURE----- --- patch_si3054.c 2007-05-18 15:06:23.000000000 +0200 +++ patch_si3054.c 2007-05-18 15:06:56.000000000 +0200 @@ -305,6 +305,7 @@ { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, + { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, /*Asus A8J Modem (SM56)*/ {} };

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[alsa-devel] [PATCH] Add support for gain in softvol plugin
by Steve Longerbeam 18 May '07

18 May '07

Hi all, What do you think of this patch? I tested this using a stac9205 HDA codec with all sample formats supported by softvol (s32, s16, s24, big and little endian). The potential drawback I see is that softvol now has to use a intermediate long long multiply and add when computing amplified 32-bit samples. This wasn't an issue with the system I tested on, but possibly could be on slower machines. Also, are their any portability issues with using a long long datatype in alsa-lib? What is the list of embedded architectures that alsa-lib is targeted to build on? I see long long used in many other places in alsa-lib, so I'm hoping this isn't an issue. Steve

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[alsa-devel] submitting a patch against the ALSA repos
by Adrian McMenamin 18 May '07

18 May '07

I have submitted my patch against the 2.6.21 kernel tree but I dare say I need to actually submit it against the alsa repos. I have a few questions: 1. Which set of sources do I patch - I assume alsa-driver - is that right? 2. How do I patch the firmware repo? As it is a binary I assume it is by some means other than patch(1) 3. This is a new architecture - there are no other SuperH drivers in alsa (afaik) - how do I add a new architecture? My memory is that when I tried this a year ago it was a bit of a nightmare.

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Re: [alsa-devel] [PATCH] asihpi 3.07.04
by Takashi Iwai 18 May '07

18 May '07

At Fri, 18 May 2007 13:16:44 +1200, Eliot Blennerhassett wrote: > > Greetings, > > Attached patch for alsa-driver/pci/asihpi > > Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> > The accompanying patch also requires updated firmware > http://audioscience.com/internet/download/firmware/asihpi_firmware-3.07.04.… > > Based on > http://audioscience.com/internet/download/hpklinux-3.07.04.tar.bz2 > Release notes > http://audioscience.com/internet/download/drvnotes.txt > http://audioscience.com/internet/download/drvnotes_rc.txt > > Main points > The bulk of this patch is addition of doxygen documentation to hpi.h and > hpifunc.c. (hopefully coming to the end of this soon) > > Add several new adapter types ASI6618, ASI6518. ASI8900 > Bug fixes to ASI6416 DSP code > Filtered out some more conditionally not compiled for ALSA. > > asihpi.c: experimental MMAP mode (disabled) > hpidebug.h: simplify debug macros > hpios_linux_kernel.h: remove a layer of macros wrapping spinlocks > hpicmn.c: pull more common code from hpi backends > hpifunc.c: remove more obsolete functions > hpimsgx.c: remove dead code > hpios.h pOsReference now struct pci_dev* not void* > hpi6205.c: use power of 2 buffers (REQUIRES matching firmware) Thanks. Many chunks look like a good clean up. Now committed to ALSA HG tree (both driver and firmware). Takashi

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[alsa-devel] HDA: Fix headphone mute issue on non-eapd Conexant systems.
by Tobin Davis 18 May '07

18 May '07

This patch fixes an automute code issue for systems that do not rely on eapd support. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] [PATCH] asihpi 3.07.04
by Eliot Blennerhassett 18 May '07

18 May '07

Greetings, There is a new patch for alsa-driver/pci/asihpi here http://audioscience.com/internet/download/alsa_driver3282_asihpi30704.patch Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> The accompanying patch also requires updated firmware http://audioscience.com/internet/download/firmware/asihpi_firmware-3.07.04.… Based on http://audioscience.com/internet/download/hpklinux-3.07.04.tar.bz2 Release notes http://audioscience.com/internet/download/drvnotes.txt http://audioscience.com/internet/download/drvnotes_rc.txt Main points The bulk of this patch is addition of doxygen documentation to hpi.h and hpifunc.c. (hopefully coming to the end of this soon) Add several new adapter types ASI6618, ASI6518. ASI8900 Bug fixes to ASI6416 DSP code Filtered out some more conditionally not compiled for ALSA. asihpi.c: experimental MMAP mode (disabled) hpidebug.h: simplify debug macros hpios_linux_kernel.h: remove a layer of macros wrapping spinlocks hpicmn.c: pull more common code from hpi backends hpifunc.c: remove more obsolete functions hpimsgx.c: remove dead code hpios.h pOsReference now struct pci_dev* not void* hpi6205.c: use power of 2 buffers (REQUIRES matching firmware) -- Eliot

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[alsa-devel] Newbie questions regarding ALSA drivers
by Pharaoh . 17 May '07

17 May '07

Hi, I am very new to ALSA and just trying to get familier with the things, I have got few questions. 1. Is it true that all the OSS drivers are included only in directory linux/sound/oss and no where else? 2. There are arch specific folders like ARM, MIPS etc, so all the drivers in these folders are ALSA or OSS? 3. If I write a new driver for an OMAP based board, it should go in to linux/sound/arm/omap/ right? Or only ALSA drivers for this board should go there? 4. I have some old OSS drivers which were written for my board, do I need to make them ALSA drivers to be in line with the latest trend i.e. phasing out OSS drivers? -Pharaoh.

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[alsa-devel] Audio driver for OMAP 730 based board - hpw to migrate from OSS to ALSA
by Pharaoh . 17 May '07

17 May '07

Hi I am new here and have slight idea about the Linux audio drivers. I am working on OMAP 730 based p2-sample board. AFAIK, there is no audio driver support for this board. I have found 2.4 drivers for the same board which I have ported to 2.6, but it is an OSS driver. So, the questions: 1. Has anyone written audio drivers for the above mentioned board? Any plans to make them open source? Any pointers will be useful regarding the same. 2. How should I go about migrating the OSS driver to ALSA, if something like this is necessary? 3. Any resources which will be helpful for writing ALSA drivers? Thanks & regards, Pharaoh.

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 17 May '07

17 May '07

Yes, its like you said Rene, sb8 works although the sound is distorted and the module can't be removed... Later I can modprobe sgalaxy too, and the sound works without a hitch. Also since now I can compile some code - your "eeprom.c" compiles and works - I'm available for further testing... I also noticed that the binary you sent was 506kb while my compilation is 9kb, maybe you sent an early/later version with some additional includes for verbose ? I compiled it with "gcc -W -Wall -o eeprom eeprom.c" like you said (on my xubuntu 7.04 with build-essentials deb package installed). See attachment... Ram Kromberg ramkromberg(a)mail.com

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 17 May '07

17 May '07

You did it Rene !!! >sudo ./eeprom >sudo modprobe snd-sgalaxy sbport=0x220 wssport=0x530 It's working now !! I downloaded the binary (i didn't compile the code), chmod it, sudo it and the sgalaxy worked !!! the sb8 didn't work but I don't really care for that anyhow :-) For the record, I tested it on xubuntu 7.04 with xmms. Are you now planning to implement the code to the driver itself ? If you want me to test anything, let me know. i'm also considering installing slackware 11 since I need the preformance... THANKS !!! Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] [1/3] 2.6.22-rc1: known regressions v2
by Michal Piotrowski 16 May '07

16 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc1. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : AMD k6-III/450 won't boot w/2.6.22-rc1 References : http://lkml.org/lkml/2007/5/16/192 Submitter : Bob Tracy <rct(a)gherkin.frus.com> Handled-By : Dave Jones <davej(a)codemonkey.org.uk> Status : problem is being debugged ACPI Subject : nx6125 has lost fan control References : http://lkml.org/lkml/2007/5/16/249 Submitter : Ray Lee <ray-lk(a)madrabbit.org> Status : Unknown ALSA Subject : lost snd_4236 device References : http://lkml.org/lkml/2007/5/13/87 Submitter : Pete Clements <clem(a)clem.clem-digital.net> Handled-By : Rene Herman <rene.herman(a)gmail.com> Patch : http://lkml.org/lkml/2007/5/15/23 Status : patch available Block devices Subject : loop devices limited to one single device References : http://lkml.org/lkml/2007/5/16/229 Submitter : Uwe Bugla <uwe.bugla(a)gmx.de> Status : Unknown Cryptography Subject : cryptomgr oops References : http://lkml.org/lkml/2007/5/14/283 Submitter : Luca Tettamanti <kronos.it(a)gmail.com> Handled-By : Herbert Xu <herbert(a)gondor.apana.org.au> Status : problem is being debugged Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

1 0

[alsa-devel] HDA: Add more systems to Sigmatel codec.
by Tobin Davis 16 May '07

16 May '07

This patch adds more Dell systems and a Panasonic laptop with Stac9200 codecs. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] HDA: Add support for Gateway NX860
by Tobin Davis 16 May '07

16 May '07

This patch adds support for the Gateway NX860 system. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

1 0

[alsa-devel] Audio DMA buffer layout
by Markus Korber 16 May '07

16 May '07

Hi, I'm working with a custom sound chip with its own DMA hardware. The figure shows how the I2S DMA expects data to be prepared: IRQ1 +----------------+ | page_3 (right) |\ +----------------+ > double buffer 1 | page_2 (left) |/ IRQ0 +----------------+ | page_1 (right) |\ +----------------+ > double buffer 0 | page_0 (left) |/ +----------------+ The hardware has 2 double buffers, each containing left and right sample regions, i.e. it alternately transmits samples from page_0 and page_1, i.e. double buffer 0, then generates an interrupt and starts transmitting page_2 and page_3. After that another interrupt is generated and the process begins again with double buffer 0. So my question is, how to tell ALSA about this memory layout? Is it possible to operate in MMAP mode with SNDRV_PCM_INFO_NONINTERLEAVED? But then, how to tell the application that it should either fill double buffer 0 or 1? Or should I use RW mode and copy samples manually to another buffer? Or something else? SG-DMA? page()-callback? Best regards, Markus Korber

2 4

[alsa-devel] Steinberg Midex 8 driver development
by Norman Schleicher 16 May '07

16 May '07

I am currently on development for the Steinberg Midex8 USB driver. At the moment I can send notes to my Midi equipment via this device. Receiving is currently not implemented. My questions are: 1. Are there any tools to test the Alsa-Midi compatiblity ? 2. May I release some milestone results to the Alsa project ? Eg when the driver supports a small subset of send commands supported by midi. 3. What are the requirements to get released in an official alsa release package ? 4. Who is responsible for the version management ? Where can I read how the source code is organized in the repository and how to manage tags and branches for the development ?(I don't need an introduction how to use CVS, thats clear for me.) Regards Norman

2 1

[alsa-devel] [PATCH] Add ALSA support for the SEGA Dreamcast PCM device
by Adrian McMenamin 15 May '07

15 May '07

Incorporating further small changes to tidy the code, please commit if ok. ALSA support for the SEGA Dreamcast Yamaha AICA sound device (pcm) This patch adds ALSA sound support for pcm playback on two channels on the SEGA Dreamcast built-in sound device (the Yamaha AICA) Add driver for the AICA sound device built into the SEGA Dreamcast Hook it all up with the build system. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk>

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[alsa-devel] Question to integrate a new driver
by Jean-Christian Hassler 15 May '07

15 May '07

Hello ! I wrote with a friend an ALSA driver for the emagic AUDIOWERK 2 soundcard. Now it works for the analog inputs/outputs. The project page is here : http://home.gna.org/aw2-alsa/index.html We have 2 questions to continue our work : 1- what is the way to submit our code to be integrated in the ALSA tree ? 2- We develop on two different version of the kernel (2.6.17-14 & 2.6.20) and the request_irq function prototype has changed between these two versions(the "struct pt_regs *" does no longer exists), how is this managed in ALSA ? Thank you

2 1

[alsa-devel] [PATCH] ALSA sound driver for SEGA Dreamcast AICA (pcm)
by Adrian McMenamin 15 May '07

15 May '07

(This is a cleaned-up version following comments - please commit) ALSA support for the SEGA Dreamcast Yamaha AICA sound device (pcm) This patch adds ALSA sound support for pcm playback on two channels on the SEGA Dreamcast built-in sound device (the Yamaha AICA) Add driver for the AICA sound device built into the SEGA Dreamcast Hook it all up with the build system. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk>

2 4

[alsa-devel] Jackd dies with snd-usb-usx2y (DRIVER NT error)
by Michael Bourgeous 15 May '07

15 May '07

Hi, I recently upgraded my system from 32-bit Ubuntu Edgy, with a custom hard-realtime-enabled 2.6.19 kernel, to 64-bit Ubuntu Feisty, with a hard-realtime-enabled 2.6.20 kernel. I also switched from an Athlon XP 2600 to an Athlon 64 X2 4600. In every way my new system should be faster, but jackd+usx2y worked on the last system, and doesn't work now. Here's what I'm doing to test the audio setup: ---------- sudo chrt -f 90 `pidof IRQ-23` # usb irq chrt -v -f 70 jackd -d alsa -dhw:1,2 -p 64 -n 2 # also tried defaults, and values up to -p 256 -n 8 chrt -v -f 25 zynaddsubfx ---------- After a short time (< 1 or 2 minutes) of switching instruments and playing notes with zynaddsubfx, all audio stops and jackd dies with the following error: ---------- ALSA: poll time out, polled for 7998858 usecs DRIVER NT: could not run driver cycle jack main caught signal 12 no message buffer overruns ---------- There is a corresponding message in the kernel log: ---------- [12546.132286] ALSA [...]/usx2yhwdeppcm.c:94: activ frame status -70. Most propably some hardware problem. ---------- I am loading the snd-usb-usx2y module with the nrpacks=1 option. I've gone through the Ubuntu HOWTOs I used to get things working on my last system (to some of which I have even contributed), and I am still having this problem. I already tried on alsa-user, so I'm taking this to the developer list (since I plan to help fix this if anybody will point me in the right direction). Additional note: I tried modifying the source code of the driver, commenting out the return statement after the "active status" error message. This began producing a new error message: "shouldn't be here with count=xx". So, I commented out the "return -EPIPE" after that error message, and now jackd doesn't crash. There are noticeable audio clicks whenever one of these errors occurs, but the driver seems to have no problem recovering, as audio continues. This error happens with the stock Ubuntu kernels as well. Thanks in advance, Mike Bourgeous

2 1

[alsa-devel] [Alsa-devel] Crosstool for PDA iPAQ hx4700
by K 15 May '07

15 May '07

Hi, I want to port a audio program on my hx4700, So I will use a prebuilt toolchain. I have downloaded ftp://ftp.handhelds.org/projects/toolchain/arm-linux-gcc-3.4.1.tar.bz2 and I extract this file to /usr/local/arm I have downloaded alsa-lib from ftp://ftp.alsa-project.org/pub/lib/stable/alsa-lib-0.0.9.tar.gz I have compiled it : #./configure --prefix=/usr/local/arm-linux --host=arm-linux #make #make install after I compile my program : #arm-linux-gcc -Wa,-mfpu=fpa11 -mcpu=xscale record.c -lasound -I /usr/local/arm/3.4.1/include/ /usr/local/arm/3.4.1/lib/gcc/arm-linux/3.4.1/../../../../arm-linux/bin/ld: cannot find -lasound collect2: ld returned 1 exit status Do you have an idea ? Can you help me ? I want to port a Vumeter, Peakmeter and other audio tools on hx4700. Thank you -- K

1 0

Re: [alsa-devel] 2.6.22-rc1 lost snd_4236 device
by Rene Herman 15 May '07

15 May '07

On 05/13/2007 06:23 PM, Pete Clements wrote: > Fyi: > With 2.6.22-rc1 get following when loading sound device module. Introduced > with 2.6.21-git16. > > FATAL: Error inserting snd_cs4236 (/lib/modules/2.6.22-rc1/kernel/sound/isa/cs423x/snd-cs4236.ko): No such device > FATAL: Error running install command for snd_cs4236 Yup; the conversion of the mixed legacy/pnp ALSA drivers to the isa_bus framework is not quite right. Wants the attached. Takashi, this needs to go to Linus. Can you make sure it gets there? This is against -linus HEAD. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 214d65d..f471f8a 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -109,6 +109,7 @@ module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; #endif @@ -686,14 +687,18 @@ static int __init alsa_card_cmi8330_init(void) int err; err = isa_register_driver(&snd_cmi8330_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&cmi8330_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_cmi8330_exit(void) @@ -701,8 +706,10 @@ static void __exit alsa_card_cmi8330_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&cmi8330_pnpc_driver); + + if (isa_registered) #endif - isa_unregister_driver(&snd_cmi8330_driver); + isa_unregister_driver(&snd_cmi8330_driver); } module_init(alsa_card_cmi8330_init) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 87f1392..1a14f33 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -127,6 +127,7 @@ module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnpc_registered; #ifdef CS4232 static int pnp_registered; @@ -770,9 +771,9 @@ static int __init alsa_card_cs423x_init(void) int err; err = isa_register_driver(&cs423x_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; #ifdef CS4232 err = pnp_register_driver(&cs4232_pnp_driver); if (!err) @@ -781,8 +782,14 @@ static int __init alsa_card_cs423x_init(void) err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#endif /* CONFIG_PNP */ - return 0; +#ifdef CS4232 + if (pnp_registered) + err = 0; +#endif + if (isa_registered) + err = 0; +#endif + return err; } static void __exit alsa_card_cs423x_exit(void) @@ -794,8 +801,9 @@ static void __exit alsa_card_cs423x_exit(void) if (pnp_registered) pnp_unregister_driver(&cs4232_pnp_driver); #endif -#endif /* CONFIG_PNP */ - isa_unregister_driver(&cs423x_isa_driver); + if (isa_registered) +#endif + isa_unregister_driver(&cs423x_isa_driver); } module_init(alsa_card_cs423x_init) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index d2a9c7d..f7732bf 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2036,7 +2036,9 @@ module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA 2 # for ES18xx driver."); #ifdef CONFIG_PNP -static int pnp_registered, pnpc_registered; +static int isa_registered; +static int pnp_registered; +static int pnpc_registered; static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { { .id = "ESS1869" }, @@ -2466,18 +2468,22 @@ static int __init alsa_card_es18xx_init(void) int err; err = isa_register_driver(&snd_es18xx_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; - #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_driver(&es18xx_pnp_driver); if (!err) pnp_registered = 1; + err = pnp_register_card_driver(&es18xx_pnpc_driver); if (!err) pnpc_registered = 1; + + if (isa_registered || pnp_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_es18xx_exit(void) @@ -2487,8 +2493,9 @@ static void __exit alsa_card_es18xx_exit(void) pnp_unregister_card_driver(&es18xx_pnpc_driver); if (pnp_registered) pnp_unregister_driver(&es18xx_pnp_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_es18xx_isa_driver); + isa_unregister_driver(&snd_es18xx_isa_driver); } module_init(alsa_card_es18xx_init) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 3e46572..0220cdb 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -135,6 +135,7 @@ struct snd_interwave { #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static struct pnp_card_device_id snd_interwave_pnpids[] = { @@ -934,15 +935,18 @@ static int __init alsa_card_interwave_init(void) int err; err = isa_register_driver(&snd_interwave_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP - /* ISA PnP cards */ + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&interwave_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_interwave_exit(void) @@ -950,8 +954,9 @@ static void __exit alsa_card_interwave_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&interwave_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_interwave_driver); + isa_unregister_driver(&snd_interwave_driver); } module_init(alsa_card_interwave_init) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 48743eb..61a323c 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -92,6 +92,7 @@ module_param_array(opl3sa3_ymode, int, NULL, 0444); MODULE_PARM_DESC(opl3sa3_ymode, "Speaker size selection for 3D Enhancement mode: Desktop/Large Notebook/Small Notebook/HiFi."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static int pnpc_registered; #endif @@ -967,17 +968,22 @@ static int __init alsa_card_opl3sa2_init(void) int err; err = isa_register_driver(&snd_opl3sa2_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_driver(&opl3sa2_pnp_driver); if (!err) pnp_registered = 1; + err = pnp_register_card_driver(&opl3sa2_pnpc_driver); if (!err) pnpc_registered = 1; + + if (isa_registered || pnp_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_opl3sa2_exit(void) @@ -987,8 +993,9 @@ static void __exit alsa_card_opl3sa2_exit(void) pnp_unregister_card_driver(&opl3sa2_pnpc_driver); if (pnp_registered) pnp_unregister_driver(&opl3sa2_pnp_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_opl3sa2_isa_driver); + isa_unregister_driver(&snd_opl3sa2_isa_driver); } module_init(alsa_card_opl3sa2_init) diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 2a19b0a..c4ba24b 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -129,6 +129,7 @@ MODULE_PARM_DESC(seq_ports, "Number of sequencer ports for WaveTable synth."); #endif #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; #endif @@ -702,15 +703,18 @@ static int __init alsa_card_sb16_init(void) int err; err = isa_register_driver(&snd_sb16_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP - /* PnP cards at last */ + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&sb16_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_sb16_exit(void) @@ -718,8 +722,9 @@ static void __exit alsa_card_sb16_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&sb16_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_sb16_isa_driver); + isa_unregister_driver(&snd_sb16_isa_driver); } module_init(alsa_card_sb16_init) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 08c1497..9ea417b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -69,7 +69,9 @@ module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; + static struct pnp_card_device_id sscape_pnpids[] = { { .id = "ENS3081", .devs = { { "ENS0000" } } }, { .id = "" } /* end */ @@ -1405,22 +1407,21 @@ static struct pnp_card_driver sscape_pnpc_driver = { static int __init sscape_init(void) { - int ret; + int err; - /* - * First check whether we were passed any parameters. - * These MUST take precedence over ANY automatic way - * of allocating cards, because the operator is - * S-P-E-L-L-I-N-G it out for us... - */ - ret = isa_register_driver(&snd_sscape_driver, SNDRV_CARDS); - if (ret < 0) - return ret; + err = isa_register_driver(&snd_sscape_driver, SNDRV_CARDS); #ifdef CONFIG_PNP - if (pnp_register_card_driver(&sscape_pnpc_driver) == 0) + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&sscape_pnpc_driver); + if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit sscape_exit(void) @@ -1428,8 +1429,9 @@ static void __exit sscape_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&sscape_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_sscape_driver); + isa_unregister_driver(&snd_sscape_driver); } module_init(sscape_init); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 75673f7..83c2fc4 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -86,6 +86,7 @@ module_param_array(use_cs4232_midi, bool, NULL, 0444); MODULE_PARM_DESC(use_cs4232_midi, "Use CS4232 MPU-401 interface (inaccessibly located inside your computer)"); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static struct pnp_card_device_id snd_wavefront_pnpids[] = { @@ -706,14 +707,18 @@ static int __init alsa_card_wavefront_init(void) int err; err = isa_register_driver(&snd_wavefront_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&wavefront_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_wavefront_exit(void) @@ -721,8 +726,9 @@ static void __exit alsa_card_wavefront_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&wavefront_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_wavefront_driver); + isa_unregister_driver(&snd_wavefront_driver); } module_init(alsa_card_wavefront_init)

3 2

[alsa-devel] Multichannel capture support in E-MU 1212
by dustin＠seznam.cz 15 May '07

15 May '07

Hello, I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c), there is still a number of issues to solve. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Pavel.

1 0

[alsa-devel] intel8x0 / ALC655 no-capture solved
by Vegard Nossum 15 May '07

15 May '07

Hello, I've successfully debugged a problem I've had with being unable to record anything from microphones or line-in. Detailed information about the bug along with a temporary work-around can be found here: http://bugtrack.alsa-project.org/alsa-bug/view.php?id=3066 Now I'd just like to inform about the presence of the bug (at least I'm pretty sure it's a bug), and that it's solved by the means of a work-around. I'd be very happy to see a fix in the *code* for this, however. I'd even like to try it myself (heck, it's just a register write), but I'm not certain WHERE/WHEN to make the write, under which conditions, and for which range of cards (ie. is it specific to the ALC655 or ALC*, or perhaps Realtek chips, or should the fix be made for the Intel driver, or all AC'97 cards?). Guidance would be appreciated :) Maybe it's even as easy as a card config option, but I wouldn't know. Thanks. Kinds regards, Vegard Nossum PS: I'm not on the list, so please CC any replies.

2 1

Re: [alsa-devel] [PATCH] Coping with short replies in usbmixer
by Thomas Reitmayr 15 May '07

15 May '07

Hi, here is it again and signed off. This patch makes sure that short USB replies are treated as an error when requesting the value of a certain mixer control. Signed-off-by: Thomas Reitmayr <thomas(a)devbase.at> --- sound/usb/usbmixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -rup a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c --- a/sound/usb/usbmixer.c 2007-05-11 21:58:33.000000000 +0200 +++ b/sound/usb/usbmixer.c 2007-05-11 21:59:14.000000000 +0200 @@ -360,7 +360,7 @@ static int get_ctl_value(struct usb_mixe request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, cval->mixer->ctrlif | (cval->id << 8), - buf, val_len, 100) >= 0) { + buf, val_len, 100) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); return 0; }

2 1

[alsa-devel] [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty
by crimsun＠fungus.sh.nu 15 May '07

15 May '07

# HG changeset patch # User crimsun(a)Box.ncat.edu # Date 1179219408 14400 # Node ID f38d824e5df6fb7d9677cce13f91ad868e617fe4 # Parent 38fe50050f000c3a9660134a86f713dc23f7ec79 [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty Included is a patch against current alsa-kernel hg tip adding quirks that Ubuntu Dapper/Edgy/Feisty contains. Signed-off-by: Daniel T Chen <crimsun(a)ubuntu.com> diff -r 38fe50050f00 -r f38d824e5df6 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue May 15 04:56:48 2007 -0400 @@ -1782,6 +1782,11 @@ static unsigned int ad1981_jacks_blackli 0x10140534, /* Thinkpad X31 */ 0x10140537, /* Thinkpad T41p */ 0x10140554, /* Thinkpad T42p/R50p */ + 0x10140567, /* Thinkpad T43p 2668-G7U */ + 0x10140581, /* Thinkpad X41-2527 */ + 0x104380b0, /* Asus A7V8X-MX */ + 0x11790241, /* Toshiba Satellite A-15 S127 */ + 0x144dc01a, /* Samsung NP-X20C004/SEG */ 0 /* end */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Tue May 15 04:56:48 2007 -0400 @@ -467,6 +467,8 @@ static struct snd_pci_quirk stac9200_cfg "Dell XPS M1710", STAC_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cf, "Dell Precision M90", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d6, + "unknown Dell", STAC_REF), {} /* terminator */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/intel8x0.c --- a/pci/intel8x0.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/intel8x0.c Tue May 15 04:56:48 2007 -0400 @@ -1799,6 +1799,18 @@ static struct ac97_quirk ac97_quirks[] _ }, { .subvendor = 0x1028, + .subdevice = 0x0186, + .name = "Dell Latitude D810", /* cf. Malone #41015 */ + .type = AC97_TUNE_HP_MUTE_LED + }, + { + .subvendor = 0x1028, + .subdevice = 0x0188, + .name = "Dell Inspiron 6000", + .type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */ + }, + { + .subvendor = 0x1028, .subdevice = 0x0191, .name = "Dell Inspiron 8600", .type = AC97_TUNE_HP_ONLY @@ -1819,7 +1831,7 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x103c, .subdevice = 0x088c, .name = "HP nc8000", - .type = AC97_TUNE_MUTE_LED + .type = AC97_TUNE_HP_MUTE_LED }, { .subvendor = 0x103c, @@ -1909,6 +1921,12 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x10cf, .subdevice = 0x1253, .name = "Fujitsu S6210", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x10cf, + .subdevice = 0x127e, + .name = "Fujitsu Lifebook C1211D", .type = AC97_TUNE_HP_ONLY }, { -- Daniel T. Chen crimsun(a)ubuntu.com GPG key: 0xC88ABDA3

2 1

[alsa-devel] soft vol
by Michel Benoit 15 May '07

15 May '07

Hello Now that I have my pcm playback working with aplay with our at91 aSoc sound driver I would like to control the volume with softvol. I followed the instructions from http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume and I have the following asound.con file in /etc pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "plughw:0,0" } control { name "PCM volume" card 0 } } When I play a wav file directly to the device it works fine: # aplay -Dplughw:0,0 Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono but when I try to use the default softvol device I get the following output (and no sound): # aplay Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ... Does anyone have any ideas as to why this is happening? Michel

2 6

[alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric 15 May '07

15 May '07

I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio chipset eric@gondor:~$ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) When modprobing the modules I get : eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout while /var/log/messages says : May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: azx_get_response timeout, switching to polling mode... Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the results of aplay : eric@gondor:~$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:~$ Here are some other infos : eric@gondor:~$ tail -2 /proc/asound/oss/sndstat 0: Realtek ALC880 1: SAA7134 Mixer amixer shows this : eric@gondor:~$ amixer Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 228 [89%] Front Right: Playback 228 [89%] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 55 [43%] [-2304.00dB] [on] Front Right: Playback 55 [43%] [-2304.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 64 [50%] [-2016.00dB] [on] Front Right: Playback 64 [50%] [-2016.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 56 [44%] [-2272.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 58 [46%] [-2208.00dB] [on] Front Right: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 57 [45%] [-2240.00dB] [on] Front Right: Playback 57 [45%] [-2240.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 53 [42%] [-2368.00dB] [on] Front Right: Playback 53 [42%] [-2368.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 31 [24%] [-3072.00dB] [on] Front Right: Capture 31 [24%] [-3072.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 27 [21%] [-3200.00dB] [on] Front Right: Capture 27 [21%] [-3200.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 24 [19%] [-3296.00dB] [on] Front Right: Capture 24 [19%] [-3296.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '6ch' Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] aplay -l seems to work : eric@gondor:~$ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 Analog] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 eric@gondor:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xb3006000 irq 22 if I try to do a cat /proc/asound/card0/codec#0 it hangs while /var/log/messages gives a lot of May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid dep_range_val 0:7fff May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid dep_range_val 0:7fff ........................ eric@gondor:~$ sudo lspci -nv 00:00.0 0600: 8086:2580 (rev 04) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: b1000000-b2ffffff Prefetchable memory behind bridge: c0000000-cfffffff Capabilities: [88] Subsystem: 8086:2582 Capabilities: [80] Power Management version 2 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [a0] Express Root Port (Slot+) IRQ 0 00:1b.0 0403: 8086:2668 (rev 03) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 20 Memory at b0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1800 [size=32] 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3000 [size=32] 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3020 [size=32] 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3040 [size=32] 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at b0004000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 I/O behind bridge: 00004000-00004fff Memory behind bridge: b3000000-b30fffff Prefetchable memory behind bridge: 0000000050000000-0000000055ffffff Capabilities: [50] Subsystem: 1558:0900 00:1f.0 0601: 8086:2640 (rev 03) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 3080 [size=16] 00:1f.3 0c05: 8086:266a (rev 03) Subsystem: 1558:0900 Flags: medium devsel, IRQ 10 I/O ports at 3060 [size=32] 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at b2000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at b1000000 (64-bit, non-prefetchable) [size=16M] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 0a:00.0 0607: 104c:ac50 (rev 02) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b3007000 (32-bit, non-prefetchable) [size=4K] Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00004800-000048ff I/O window 1: 00004c00-00004cff 16-bit legacy interface ports at 0001 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at b3004000 (32-bit, non-prefetchable) [size=2K] Memory at b3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 0a:02.0 0104: 105a:3373 (rev 02) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 I/O ports at 4480 [size=64] I/O ports at 44c0 [size=16] I/O ports at 4400 [size=128] Memory at b3005000 (32-bit, non-prefetchable) [size=4K] Memory at b3020000 (32-bit, non-prefetchable) [size=128K] Capabilities: [60] Power Management version 2 0a:03.0 0200: 10ec:8169 (rev 10) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 I/O ports at 4000 [size=256] Memory at b3004800 (32-bit, non-prefetchable) [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [dc] Power Management version 2 0a:04.0 0480: 1131:7133 (rev f0) Subsystem: 1461:f31e Flags: bus master, medium devsel, latency 64, IRQ 23 Memory at b3006000 (32-bit, non-prefetchable) [size=2K] Capabilities: [40] Power Management version 2 0a:05.0 0280: 1814:0201 (rev 01) Subsystem: 1462:6833 Flags: bus master, slow devsel, latency 64, IRQ 20 Memory at b3008000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 asoundconf output : eric@gondor:~$ asoundconf list Names of available sound cards: Intel SAA7134 I tried a lot of workarounds but none worked : boot options (irqpoll, pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... I must add that I already had this problem with previous distributions of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, it did worked ONLY when I self-compiled the driver with debug option enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it does not work. I cannot have one sound :-(. Please help me .... Eric

3 2

[alsa-devel] default PCM volume
by Michel Benoit 15 May '07

15 May '07

Hello again, I am running alsa on a custom AT91SAM9260 arm based SOC board together with a a simple external PCM1725 codec. When using alsa-lib and the at91 soc driver (pcm+i2s) I can playback PCM samples using both aplay and test/pcm. Using softvol I can also reduce the volume of the playback. The problem I have is that the playback volume (with softvol=100%) is not very loud. The volume is the same even without the softvol plugin. The PCM codec has no hw volume control. According to our hw team the sound level should be much higher. Is there a default PCM attenuation (volume reduction) built into alsa lib? If so, how can it be adjusted? Thanks, Michel

1 0

[alsa-devel] [PATCH] ALSA sound driver for SEGA Dreamcast AICA (pcm)
by Adrian McMenamin 14 May '07

14 May '07

ALSA support for the SEGA Dreamcast Yamaha AICA sound device (pcm) This patch adds ALSA sound support for pcm playback on two channels on the SEGA Dreamcast built-in sound device (the Yamaha AICA) Add driver for the AICA sound device built into the SEGA Dreamcast Hook it all up with the build system. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk>

2 1

[alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 14 May '07

14 May '07

Hello, here goes attempt #3 for the SH7760 ASoC patch. Changes v2 -> v3 - added a generic demo board for AC97 Changes v1 -> v2 - incorporate Paul Mundt's suggestions - work around an oops when aplay is used in mmap mode ('aplay -M') by changing the allocation type to SNDRV_DMA_TYPE_CONTINUOUS As soon as I get Linux working on the EDOSK7780 board, I'll complete SH7780 support as well. Thanks, Manuel Lauss --- ALSA ASoC support for SH7760 - SH7760 DMA engine (dmabrg) - AC97 driver for HAC unit(s) found on SH7760/SH7780 - I2S driver for SSI unit(s) found on SH7760/SH7780 - generic SH7760-AC97 "demoboard" sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/sh/Kconfig | 39 +++++ sound/soc/sh/Makefile | 14 ++ sound/soc/sh/dma-sh7760.c | 354 +++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 322 +++++++++++++++++++++++++++++++++++ sound/soc/sh/sh7760-ac97.c | 92 ++++++++++ sound/soc/sh/ssi.c | 400 ++++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 1224 insertions(+), 1 deletions(-) Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..a332e51 --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,39 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +config SND_SH7760_AC97 + tristate "SH7760 AC97 sound support" + depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760 + select SND_SOC_SH4_HAC + select SND_SOC_AC97_CODEC + help + This option enables generic sound support for the first + AC97 unit of the SH7760. + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..a8e8ab8 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,14 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards +snd-soc-sh7760-ac97-objs := sh7760-ac97.o + +obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..cdee374 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,354 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> + + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of a + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + pr_debug("PCM data: addr 0x%08ulx len %d\n", + (u32)runtime->dma_addr, runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * position at the time the IRQ was triggered, which results in + * fast-playback sound in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-pcm", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..8e3f039 --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,322 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "hac: reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + pr_debug("hac: invalid depth %d bit\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c new file mode 100644 index 0000000..5563f14 --- /dev/null +++ b/sound/soc/sh/sh7760-ac97.c @@ -0,0 +1,92 @@ +/* + * Generic AC97 sound support for SH7760 + * + * (c) 2007 Manuel Lauss + * + * Licensed under the GPLv2. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/io.h> + +#include "../codecs/ac97.h" + +#define IPSEL 0xFE400034 + +/* platform specific structs can be declared here */ +extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_platform sh7760_soc_platform; + +static int machine_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync_endpoints(codec); + return 0; +} + +static struct snd_soc_dai_link sh7760_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &sh4_hac_dai[0], /* HAC0 */ + .codec_dai = &ac97_dai, + .init = machine_init, + .ops = NULL, +}; + +static struct snd_soc_machine sh7760_ac97_soc_machine = { + .name = "SH7760 AC97", + .dai_link = &sh7760_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device sh7760_ac97_snd_devdata = { + .machine = &sh7760_ac97_soc_machine, + .platform = &sh7760_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *sh7760_ac97_snd_device; + +static int __init sh7760_ac97_init(void) +{ + int ret; + unsigned short ipsel; + + /* enable both AC97 controllers in pinmux reg */ + ipsel = ctrl_inw(IPSEL); + ctrl_outw(ipsel | (3 << 10), IPSEL); + + ret = -ENOMEM; + sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1); + if (!sh7760_ac97_snd_device) + goto out; + + platform_set_drvdata(sh7760_ac97_snd_device, + &sh7760_ac97_snd_devdata); + sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev; + ret = platform_device_add(sh7760_ac97_snd_device); + + if (ret) + platform_device_put(sh7760_ac97_snd_device); + +out: + return ret; +} + +static void __exit sh7760_ac97_exit(void) +{ + platform_device_unregister(sh7760_ac97_snd_device); +} + +module_init(sh7760_ac97_init); +module_exit(sh7760_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..b72bc31 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,400 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT) +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK (7 << CR_DWL_SHIFT) +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK (7 << CR_SWL_SHIFT) +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_SHIFT 4 +#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */ +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + pr_debug("ssi: already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); + pr_debug("bits: %d channels: %d\n", bits, channels); + + ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | + CR_SWL_MASK); + + /* direction (send/receive) */ + if (!recv) + ssicr |= CR_TRMD; /* transmit */ + + /* channels */ + if ((channels < 2) || (channels > 8) || (channels & 1)) { + pr_debug("ssi: invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK; + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + pr_debug("ssi: invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP | + CR_SWS_MASTER | CR_SCK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + pr_debug("ssi: unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + pr_debug("ssi: invalid inversion\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + pr_debug("ssi: invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 14 May '07

14 May '07

Since the only successful modprobe is done with the jumper on SOFTWARE instead of EPPROM, I'll only tested the following: Jumper: Software: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 without an error, dmesg - gives no hint, verbose, mention... cat /proc/asound/cards: 0 [Pro ]: SB Pro - Sound Blaster Pro Sound Blaster Pro at 0x220, irq 5, dma 1 So I did the following: cd /proc/asound/ cat * cat: Pro: Is a directory cat: card0: Is a directory 0 [Pro ]: SB Pro - Sound Blaster Pro Sound Blaster Pro at 0x220, irq 5, dma 1 2: : timer 3: : sequencer 4: [ 0- 0]: raw midi 5: [ 0- 0]: hardware dependent 6: [ 0- 0]: digital audio playback 7: [ 0- 0]: digital audio capture 8: [ 0] : control 00-00: OPL3 FM 0 snd_sb8 cat: oss: Is a directory 00-00: SB8 DSP : DSP v3.1 : playback 1 : capture 1 cat: seq: Is a directory G0: system timer : 4000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 09:56:17 2007 UTC). Thanks for your help, Ram Kromberg, ramkromberg(a)mail.com

2 1

[alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric 14 May '07

14 May '07

I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio chipset eric@gondor:~$ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) When modprobing the modules I get : eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout while /var/log/messages says : May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: azx_get_response timeout, switching to polling mode... Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the results of aplay : eric@gondor:~$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:~$ Here are some other infos : eric@gondor:~$ tail -2 /proc/asound/oss/sndstat 0: Realtek ALC880 1: SAA7134 Mixer amixer shows this : eric@gondor:~$ amixer Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 228 [89%] Front Right: Playback 228 [89%] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 55 [43%] [-2304.00dB] [on] Front Right: Playback 55 [43%] [-2304.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 64 [50%] [-2016.00dB] [on] Front Right: Playback 64 [50%] [-2016.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 56 [44%] [-2272.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 58 [46%] [-2208.00dB] [on] Front Right: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 57 [45%] [-2240.00dB] [on] Front Right: Playback 57 [45%] [-2240.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 53 [42%] [-2368.00dB] [on] Front Right: Playback 53 [42%] [-2368.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 31 [24%] [-3072.00dB] [on] Front Right: Capture 31 [24%] [-3072.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 27 [21%] [-3200.00dB] [on] Front Right: Capture 27 [21%] [-3200.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 24 [19%] [-3296.00dB] [on] Front Right: Capture 24 [19%] [-3296.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '6ch' Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] aplay -l seems to work : eric@gondor:~$ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 Analog] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 eric@gondor:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xb3006000 irq 22 if I try to do a cat /proc/asound/card0/codec#0 it hangs while /var/log/messages gives a lot of May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid dep_range_val 0:7fff May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid dep_range_val 0:7fff ........................ eric@gondor:~$ sudo lspci -nv 00:00.0 0600: 8086:2580 (rev 04) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: b1000000-b2ffffff Prefetchable memory behind bridge: c0000000-cfffffff Capabilities: [88] Subsystem: 8086:2582 Capabilities: [80] Power Management version 2 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [a0] Express Root Port (Slot+) IRQ 0 00:1b.0 0403: 8086:2668 (rev 03) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 20 Memory at b0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1800 [size=32] 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3000 [size=32] 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3020 [size=32] 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3040 [size=32] 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at b0004000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 I/O behind bridge: 00004000-00004fff Memory behind bridge: b3000000-b30fffff Prefetchable memory behind bridge: 0000000050000000-0000000055ffffff Capabilities: [50] Subsystem: 1558:0900 00:1f.0 0601: 8086:2640 (rev 03) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 3080 [size=16] 00:1f.3 0c05: 8086:266a (rev 03) Subsystem: 1558:0900 Flags: medium devsel, IRQ 10 I/O ports at 3060 [size=32] 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at b2000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at b1000000 (64-bit, non-prefetchable) [size=16M] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 0a:00.0 0607: 104c:ac50 (rev 02) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b3007000 (32-bit, non-prefetchable) [size=4K] Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00004800-000048ff I/O window 1: 00004c00-00004cff 16-bit legacy interface ports at 0001 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at b3004000 (32-bit, non-prefetchable) [size=2K] Memory at b3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 0a:02.0 0104: 105a:3373 (rev 02) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 I/O ports at 4480 [size=64] I/O ports at 44c0 [size=16] I/O ports at 4400 [size=128] Memory at b3005000 (32-bit, non-prefetchable) [size=4K] Memory at b3020000 (32-bit, non-prefetchable) [size=128K] Capabilities: [60] Power Management version 2 0a:03.0 0200: 10ec:8169 (rev 10) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 I/O ports at 4000 [size=256] Memory at b3004800 (32-bit, non-prefetchable) [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [dc] Power Management version 2 0a:04.0 0480: 1131:7133 (rev f0) Subsystem: 1461:f31e Flags: bus master, medium devsel, latency 64, IRQ 23 Memory at b3006000 (32-bit, non-prefetchable) [size=2K] Capabilities: [40] Power Management version 2 0a:05.0 0280: 1814:0201 (rev 01) Subsystem: 1462:6833 Flags: bus master, slow devsel, latency 64, IRQ 20 Memory at b3008000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 asoundconf output : eric@gondor:~$ asoundconf list Names of available sound cards: Intel SAA7134 I tried a lot of workarounds but none worked : boot options (irqpoll, pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... I must add that I already had this problem with previous distributions of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, it did worked ONLY when I self-compiled the driver with debug option enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it does not work. I cannot have one sound :-(. Please help me .... Eric

1 0

[alsa-devel] alsa-lib functions reporting conflicting information with two sound cards
by nick smethurst 14 May '07

14 May '07

Hey guys I am rather confused that I am getting crossed information when calling snd_names_list and snd_pcm_info in order to gather information about pcm devices. I have two sound cards in my development pc: RME Digi 96/8 PCI and an onboard VIA 8237. I'm trying to develop a piece of code that generates a list of objects containing pcm device information, including information on virtual devices. My current idea is to use snd_names_list() to get a full list of devices, and then get information about them by opening each device and querying. The bizzare thing is that snd_names_list() is returning comments that do not agree with the associated device. For example, given the following code: #include <alsa/asoundlib.h> int main() { snd_devname_t * list; snd_config_update(); snd_names_list("pcm", &list); while (list) { if (!strcmp(list->name, "hw:0,0")) { printf("hw:0,0 comment: %s\n", list->comment); break; } list = list->next; } snd_names_list_free(list); snd_pcm_t * pcm; snd_pcm_info_t * pcm_info; snd_pcm_info_malloc(&pcm_info); snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_info(pcm, pcm_info); printf("id: %s\n", snd_pcm_info_get_id(pcm_info)); snd_pcm_close(pcm); snd_pcm_info_free(pcm_info); return 0; } The result on my pc is: hw:0,0 comment: Physical Device - VIA 8237 (Duplex) id: Digi96 IEC958 So I'm apparently doing something very wrong here. I would like to ask if there is an easier and more reliable way of getting a list of pcm devices and querying them, and if not, what I am doing wrong here? Regards Nicholas

6 16

[alsa-devel] [PATCH1/5] ASoC S3C24xx machine drivers - s3c2443-AC97
by Liam Girdwood 14 May '07

14 May '07

Sorry for the lateness in the current merge window of this patch series. From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds AC97 support to the Samsung S3C2443 CPU. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

3 4

[alsa-devel] Undefined Symbols: snd-aoa-codec-tas.c:944 warning: implicit declaration of function 'of_get_property'
by Elimar Riesebieter 14 May '07

14 May '07

Hi all, the aoa drivers in 1.0.14rc4 are not usable. Compiling the modules there are much warnings "implicit declaration of function 'of_get_property'" Loading the module dmesg tells: snd_aoa: Unknown symbol of_get_property. rc3 out of 2.6.21 works fine. Any hints? Thanks Elimar -- You cannot propel yourself forward by patting yourself on the back.

3 4

[alsa-devel] Dreamcast AICA
by Adrian McMenamin 14 May '07

14 May '07

I haven't touched this for a year, but after some small changes (to run queues) I can compile this and it runs well. Anyone care to comment on the prospects of getting it committed? I'll obviously be happy to be named as the maintainer in that case... diff -Naur ./linux-2.6.21-orig/sound/Kconfig ./linux-2.6.21/sound/Kconfig --- ./linux-2.6.21-orig/sound/Kconfig 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Kconfig 2007-05-07 21:20:54.000000000 +0100 @@ -64,6 +64,8 @@ source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -Naur ./linux-2.6.21-orig/sound/Makefile ./linux-2.6.21/sound/Makefile --- ./linux-2.6.21-orig/sound/Makefile 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Makefile 2007-05-07 21:37:05.000000000 +0100 @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -Naur ./linux-2.6.21-orig/sound/sh/aica.c ./linux-2.6.21/sound/sh/aica.c --- ./linux-2.6.21-orig/sound/sh/aica.c 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.c 2007-05-12 15:27:54.000000000 +0100 @@ -0,0 +1,677 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006 +* <adrian(a)mcmen.demon.co.uk> +* See also http://newgolddream.dyndns.info/cgi-bin/cvsweb +* +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License as published by +* the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/timer.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/dreamcast/sysasic.h> +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static inline void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + break; + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(uint32_t toi, uint32_t what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + spu_write_wait(); + for (i = 0; i < length; i++) { + writel(what, toi + SPU_MEMORY_BASE); + toi++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(uint32_t toi, void __iomem * from, int length) +{ + uint32_t __iomem *froml = from; + uint32_t __iomem *to = (uint32_t __iomem *) (SPU_MEMORY_BASE + toi); + int i, val; + if (length % 4) + length = (length / 4) + 1; + else + length = length / 4; + spu_write_wait(); + for (i = 0; i < length; i++) { + val = *froml; + writel(val, to); + froml++; + to++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + uint32_t regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + uint32_t regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (uint32_t *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (uint32_t *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +static void run_spu_dma(struct work_struct *work) +{ + int buffer_size; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_card_aica *dreamcastcard; + struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); + substream = holder-> sspointer; + dreamcastcard = substream->pcm->private_data; + runtime = substream->runtime; + if (unlikely(dreamcastcard->dma_check == 0)) { + buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels > 1) + dreamcastcard->channel->flags |= 0x01; + aica_dma_transfer(runtime->channels, buffer_size, substream); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer fuction - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + dreamcastcard->timer.expires = jiffies + 1; + add_timer(&(dreamcastcard->timer)); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Use queue to do the heavy lifting */ + spu_working.sspointer = substream; + INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); + queue_work(aica_queue, &(spu_working.spu_dma_work)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream + *substream, int cmd) +{ + struct snd_card_aica *dreamcastcard; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + spu_begin_dma(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + dreamcastcard = substream->pcm->private_data; + if (dreamcastcard->timer.data) + del_timer(&dreamcastcard->timer); + aica_chn_halt(); + break; + default: + return -EINVAL; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + err = 0; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue("aica"); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + err = load_aica_firmware(); + if (unlikely(err < 0)) + return err; + + return 0; +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff -Naur ./linux-2.6.21-orig/sound/sh/aica.h ./linux-2.6.21/sound/sh/aica.h --- ./linux-2.6.21-orig/sound/sh/aica.h 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.h 2007-05-07 21:47:31.000000000 +0100 @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * <adrian(a)mcmen.demon.co.uk> + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as published by + * the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; diff -Naur ./linux-2.6.21-orig/sound/sh/Kconfig ./linux-2.6.21/sound/sh/Kconfig --- ./linux-2.6.21-orig/sound/sh/Kconfig 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Kconfig 2007-05-07 21:28:30.000000000 +0100 @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff -Naur ./linux-2.6.21-orig/sound/sh/Makefile ./linux-2.6.21/sound/sh/Makefile --- ./linux-2.6.21-orig/sound/sh/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Makefile 2007-05-07 21:34:10.000000000 +0100 @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o Signed-off by: Adrian McMenamin

2 1

[alsa-devel] [PATCH] Coping with short replies in usbmixer
by thomas＠devbase.at 14 May '07

14 May '07

Hi, while trying to find out why a USB handset (Yealink P1K) frequently gets muted when changing volume, I found that short USB responses cause the problem. The function get_ctl_value(..) in usbmixer.c requests the state of a certain control and retries up to 10 times if an error occurs. However a short USB response is not treated as an error as neither the transfer flag URB_SHORT_NOT_OK is set nor is the return value of snd_usb_ctl_msg(..) compared with the requested amount of data. As a result the function get_ctl_value(..) might not retry the USB request but returns an unpredictable result without reporting an error. The attached patch forces the number of returned bytes to be at least val_len. It was generated from kernel version 2.6.21. If there are any issues with the patch please let me know. Otherwise I would be happy if it could be applied to the ALSA sources. Regards, -Thomas

2 1

[alsa-devel] Meaning of this Plug PCM Transformation table?
by Nobin Mathew 14 May '07

14 May '07

Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 This is capture Handle logs Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0.5 + 1*0.5 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 I am having an application which does capture and playback simultaneously and is single threaded? What is the meaning of this Transformation table?

1 0

[alsa-devel] [2/2] 2.6.22-rc1: known regressions
by Michal Piotrowski 13 May '07

13 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions (http://bugzilla.kernel.org/ is not available, information are not up-to-date) Networking Subject : panic with e1000 driver on HP Integrity servers References : http://bugzilla.kernel.org/show_bug.cgi?id=8455 Submitter : Doug Chapman <doug.chapman(a)hp.com> Caused-By : Auke Kok <auke-jan.h.kok(a)intel.com> commit e0aac5a289b1dacbc94bd9ae8c449bcdf9ab508c Status : problem is being debugged Timers/NOHZ Subject : 2.6.21-git4 BUG: soft lockup detected on CPU#1! References : http://lkml.org/lkml/2007/5/2/511 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Thomas Gleixner <tglx(a)linutronix.de> Status : problem is being debugged ALSA Subject : lost snd_4236 device References : http://lkml.org/lkml/2007/5/13/87 Submitter : Pete Clements <clem(a)clem.clem-digital.net> Status : Unknown Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

1 0

[alsa-devel] e-mu 1212m and 1820(m) patch
by Ctirad Fertr 13 May '07

13 May '07

Hello, here is a little patch which enables ADAT and S/PDIF functionality. It adds possibility to sync with either S/PDIF or ADAT clock as well. 44.1/48kHz only, though it works pretty well for me. Regards, Ctirad

2 6

[alsa-devel] embedded sound architecture question
by Joachim Förster 13 May '07

13 May '07

Hi ALSA devs, I'm going to write an ALSA driver for a not yet existing AC97 controller, which is going to be "written" (VHDL), too (at the same time). Platform/Board is a Xilinx ML403 with Virtex-4 FPGA, PowerPC 405 architecture, OPB/OCP bus, AC97 Codec LM4550. Before presenting my question, I have to say, that I'm a beginner with ALSA/Linux driver development. My question is: Does the architecture described below make sense/is reasonable with ALSA and Linux? The problem is, that there is no DMA controller and implementing an OPB master device which would be able to do DMA itself is not an option at this time. So our thoughts were: Integrate the "DMA ring buffer" (which is usually somewhere in main memory/RAM) into the AC97 controller. Make ALSA access this HW buffer as if it was in main memory. This way, the device (AC97 controller) has "direct access" to its buffer "memory" - this could be called "fake DMA". The AC97 controller would have tell us where it is while playing and firing interrupts after one period, so that we don't write to values which are in the current period, instead update the area where of the past played periods etc. ... The buffer should be a "ring buffer", right? Mapping this "IO memory" into kernel space should be possible with io_remap_page_range(), right? I would have to implement the mmap() callback in my driver, to setup the given VMA (with the above function), right? So, ALSA library/applications will be able to use MMAP mode, which is what we want to achieve? [We don't want copy()/silence(). An intermediate buffer with ack()/tasklet/workqueue + FIFO in HW would be an alternative.] I read, that many applications don't work, if MMAP mode is not supported and classic read/write (copy()/silence()) is used, only. Is there a black list of apps, which don't work? Thanks for reading & your time, Joachim

4 4

[alsa-devel] apologies for this email
by Adrian McMenamin 13 May '07

13 May '07

My last two made it in to the archive, but didn't get delivered to me. Did anyone else see them?

3 2

[alsa-devel] Sound driver for Sega Dreamcast
by Adrian McMenamin 12 May '07

12 May '07

My last email is in the archive, but seem to make it back to me - in any case this code is better... but in case my last email didn't get delivered anywhere, I am asking for views on what needs to be done to get this committed. diff -Naur ./linux-2.6.21-orig/sound/Kconfig ./linux-2.6.21/sound/Kconfig --- ./linux-2.6.21-orig/sound/Kconfig 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Kconfig 2007-05-07 21:20:54.000000000 +0100 @@ -64,6 +64,8 @@ source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -Naur ./linux-2.6.21-orig/sound/Makefile ./linux-2.6.21/sound/Makefile --- ./linux-2.6.21-orig/sound/Makefile 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Makefile 2007-05-07 21:37:05.000000000 +0100 @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -Naur ./linux-2.6.21-orig/sound/sh/aica.c ./linux-2.6.21/sound/sh/aica.c --- ./linux-2.6.21-orig/sound/sh/aica.c 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.c 2007-05-12 19:10:32.000000000 +0100 @@ -0,0 +1,676 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006 +* <adrian(a)mcmen.demon.co.uk> +* See also http://newgolddream.dyndns.info/cgi-bin/cvsweb +* +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License as published by +* the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/timer.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/dreamcast/sysasic.h> +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static inline void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + break; + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(uint32_t toi, uint32_t what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + spu_write_wait(); + for (i = 0; i < length; i++) { + writel(what, toi + SPU_MEMORY_BASE); + toi++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(uint32_t toi, void __iomem * from, int length) +{ + uint32_t __iomem *froml = from; + uint32_t __iomem *to = (uint32_t __iomem *) (SPU_MEMORY_BASE + toi); + int i, val; + if (length % 4) + length = (length / 4) + 1; + else + length = length / 4; + spu_write_wait(); + for (i = 0; i < length; i++) { + val = *froml; + writel(val, to); + froml++; + to++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + uint32_t regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + uint32_t regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (uint32_t *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (uint32_t *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +static void run_spu_dma(struct work_struct *work) +{ + int buffer_size; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct snd_card_aica *dreamcastcard; + struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); + substream = holder-> sspointer; + dreamcastcard = substream->pcm->private_data; + runtime = substream->runtime; + if (unlikely(dreamcastcard->dma_check == 0)) { + buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels > 1) + dreamcastcard->channel->flags |= 0x01; + aica_dma_transfer(runtime->channels, buffer_size, substream); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer fuction - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + mod_timer(&(dreamcastcard->timer), jiffies + 1); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Use queue to do the heavy lifting */ + spu_working.sspointer = substream; + INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); + queue_work(aica_queue, &(spu_working.spu_dma_work)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream + *substream, int cmd) +{ + struct snd_card_aica *dreamcastcard; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + spu_begin_dma(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + dreamcastcard = substream->pcm->private_data; + if (dreamcastcard->timer.data) + del_timer(&dreamcastcard->timer); + aica_chn_halt(); + break; + default: + return -EINVAL; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + err = 0; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue("aica"); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + err = load_aica_firmware(); + if (unlikely(err < 0)) + return err; + + return 0; +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff -Naur ./linux-2.6.21-orig/sound/sh/aica.h ./linux-2.6.21/sound/sh/aica.h --- ./linux-2.6.21-orig/sound/sh/aica.h 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.h 2007-05-07 21:47:31.000000000 +0100 @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * <adrian(a)mcmen.demon.co.uk> + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as published by + * the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; diff -Naur ./linux-2.6.21-orig/sound/sh/Kconfig ./linux-2.6.21/sound/sh/Kconfig --- ./linux-2.6.21-orig/sound/sh/Kconfig 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Kconfig 2007-05-07 21:28:30.000000000 +0100 @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff -Naur ./linux-2.6.21-orig/sound/sh/Makefile ./linux-2.6.21/sound/sh/Makefile --- ./linux-2.6.21-orig/sound/sh/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Makefile 2007-05-07 21:34:10.000000000 +0100 @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 11 May '07

11 May '07

o, I thought the epprom settings are default until the driver kicks in... So I set the jumpers accordingly and still no luck: IDESEL: 1-2 (Disabled - Its the CD IDE ON\OFF) JMPCFG: 1-2 (EPPROM ON) JMPB0: 1-2 (22XH) IDEIRQ: 1 (IRQ 15 - shouldn't make a different) JX1: 2-3 (Carbon MIC - again just letting you know though it shouldn't make a different) This should make sense with the picture I sent earlier... I also removed another card I had connected - ISA Modem, Just making sure their are no conflicts... Here's the grim results: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 FATAL: Error inserting snd_sb8 (/lib/modules/2.6.20-15-generic/kernel/sound/isa/sb/snd-sb8.ko): No such device P.S: Also an interesting note is that I now installed Win98 on top of that Dos and it worked too - It did also mention to use IRQ 11 so I also reserved in the bios, but again to no avail. Maybe I'm missing something ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] Mackie XD-2 Spike USB Drivers for Linux
by Sava Tatic 11 May '07

11 May '07

Hello, I have been trying to get the Mackie XD-2 Spike (http://www.mackie.com/products/spike/) to work on my Kubuntu Feisty machine, and so far I have only gotten it to play out (I had to use plughw instead of hw). I am using the generic USB drivers that came with the distro. I am having trouble recording anything on the card. Alsamixer (or Kmix) do not see the card. When I do alsamixer -c 1 (0 is my onboard card), it returns "no mixer elems found". I am aware that this card is not in the ALSA Matrix of supported cards, but I am wondering what would it take for it to become supported? All the best, Sava

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[alsa-devel] Capturing MIDI
by Steve Merrony 11 May '07

11 May '07

I hope this isn't too off-topic. I run an open source project (GENPO) which can be thought of as a kind of MIDI switchboard and filter. It uses ALSA throughout. I'd like to add a new feature to the software - a facility to simply record to a standard MIDI file anything that the program produces. I've searched though the API and I can't find anything documented which will help much, so here's my question... Is there a recommended library for C++ ALSA programs to use for handling (writing) standard MIDI files? Regards, Steve

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[alsa-devel] [PATCH2/5] ASoC S3C24xx machine drivers - Openmoko Neo1973
by Liam Girdwood 11 May '07

11 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds ALSA support for the Openmoko Neo1973 phone. Features:- * HiFi Playback and capture. * Phone calls supported. * Support for BT PCM in WM8753 voice interface. * Support for LM4857 audio amp. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH] ASoC AC97 device reg bugfix
by Liam Girdwood 11 May '07

11 May '07

This patch fixes a bug whereby AC97 bus device data was being clobbered when AC97 codecs using the generic ac97_codec.c driver were being registered. Codecs that didn't use the generic driver were unaffected (e.g. WM9712, WM9713). Changes:- o Add new AC97 codec class for custom (or need bus dev registration) AC97 codecs. o Only register/deregister this custom codec device with the AC97 bus. The generic AC97 driver already does this for generic codec devices. This may be related to bug #3038 :- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038 Manuel, can you try this on SH & Au1x00. Thanks Liam

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

Well, it didn't take long so here's the results : "sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1" gave no errors, and "lsmod" later gave: Module Size Used by snd_sb8 9216 0 snd_sb8_dsp 12288 1 snd_sb8 snd_sb_common 17792 2 snd_sb8,snd_sb8_dsp snd_opl3_lib 11520 1 snd_sb8 snd_hwdep 9988 1 snd_opl3_lib snd_pcm_oss 44544 0 snd_pcm 79876 3 snd_sb8,snd_sb8_dsp,snd_pcm_oss snd_mixer_oss 17408 1 snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 32896 0 snd_seq_midi 9600 0 snd_rawmidi 25472 2 snd_sb8_dsp,snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 52592 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 23684 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 9100 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54020 13 snd_sb8,snd_sb8_dsp,snd_sb_common,snd_opl3_lib,snd_hwdep,snd_pcm_oss,snd_pcm,snd_mixer_oss,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 10888 1 snd_pcm nfs 240876 0 nfsd 218992 17 exportfs 6912 1 nfsd lockd 64904 3 nfs,nfsd sunrpc 161340 12 nfs,nfsd,lockd apm 22752 1 ppdev 10116 0 cpufreq_stats 7360 0 cpufreq_ondemand 9228 0 cpufreq_powersave 2688 0 freq_table 5792 2 cpufreq_stats,cpufreq_ondemand cpufreq_conservative 8200 0 cpufreq_userspace 5408 0 nls_iso8859_1 5120 1 nls_cp437 6784 1 vfat 14208 1 fat 53916 1 vfat ext2 66824 1 lp 12452 0 parport_pc 36388 1 parport 36936 3 ppdev,lp,parport_pc pcspkr 4224 0 ipv6 268704 8 tsdev 8768 0 evdev 11008 1 ext3 133128 1 jbd 59816 1 ext3 mbcache 9604 2 ext2,ext3 8139too 27648 0 ide_cd 32672 0 cdrom 37664 1 ide_cd ide_disk 17024 5 floppy 59524 0 ehci_hcd 34188 0 8139cp 25088 0 mii 6528 2 8139too,8139cp uhci_hcd 25360 0 usbcore 134280 3 ehci_hcd,uhci_hcd piix 10756 0 [permanent] generic 5124 0 [permanent] ata_generic 9092 0 libata 125720 1 ata_generic scsi_mod 142348 1 libata fbcon 42656 0 tileblit 3584 1 fbcon font 9216 1 fbcon bitblit 6912 1 fbcon softcursor 3200 1 bitblit vesafb 9220 0 capability 5896 0 commoncap 8192 1 capability But when I try to play an ogg gxine crushes an xmms ignores, when trying an mp3 gxine again crushes and xmms gives the soundcard no configured error... Just a recap: I verifyed using dos the settings (hwset and volset - default and save), and used the bios to set "IRQ 5" as "ISA Legacy". I didn't use the jumper as I see no need since the epprom already does the job. So I'm stuck, What do you think I should do now ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] detect extern/internel mic
by xinyun liu 10 May '07

10 May '07

It's interesting that when I plug in a headphone or an external microphone, the build in headphone and microphone will not work. Is it a driver feature or the hardware do it? Can anyone give some clue? I'd like to see the code but don't know how to start. Please give me help. Thx!

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Re: [alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 10 May '07

10 May '07

I have test all modes for alc883. The result is here: 0. 3stack-dig. both extern/internal can play, but can't record sound 1. 3stack-6ch-dig same with 3stack-dig 2. 3stack-6ch can play, but can't record sound. Can hear noise from the Mic. 3. 6stack-dig both can play sound, can't record sound 4. tarta-dig With alsamixer, can set to 2ch or 6ch, both can play sound, can't 5. tarta-2ch-dig both can play sound, can't record sound. 6. 6stack-dig-demo both can play sound, can't record sound. 7. aser both can play sound, can't record sound 8. medion can't play sound 9. laptop-eapd can play sound, but can't record sound 10. lenovo-101e both can play sound, can't record sound 11. auto both can play sound, and can record sound. But the sound quality is too bad to use from the internel Mic. After test it a while on switching the extern/internal Mic, the quality both droped. So I use lenovo-101e model, ( rebuild the driver, rmmod/insmod, use alsamixer to adjust the limits). Then I use auto mode, and the sound record work again from external Mic. After I switch to internel Mic, the quality droped again. Do you have any idea about it? Thx! 2007/4/30, Ustun ERGENOGLU <ustun.ergenoglu(a)gmail.com>: > I have a Fujitsu Siemens Amilo pa1510, with the same sound hardware, > and the same problem. I have posted some messages here, made a bug > report but no answer came :( i'm going hopeless. > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > I know it's the system bios does not set the pin configuration right. > > But I don't know how to try out a correct model. > > > > Can anyone help to give some hints to fix it? > > > > > > Regards, > > Xinyun > > > > > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > > hi, > > > I have an Acer Aspire 5100. I can play sound but with the build in > > > mic, I can't record sound. The driver is the latest HG code. > > > http://pastebin.ca/464406 > > > > > > [root@localhost ~]# lspci -nnv -d1002:437b > > > 00:14.2 0403: 1002:437b (rev 01) > > > Subsystem: 1025:009f > > > Flags: bus master, slow devsel, latency 64, IRQ 19 > > > Memory at c0000000 (64-bit, non-prefetchable) [size=16K] > > > Capabilities: [50] Power Management version 2 > > > Capabilities: [60] Message Signalled Interrupts: 64bit+ > > > Queue=0/0 Enable- > > > > > > > > > [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 > > > [root@localhost ~]# vi /tmp/1 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x1, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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[alsa-devel] alsa-driver-1.0.14rc3 won't compile with kernel 2.6.20.3 when Processor family is Core2
by Chris Pemberton 10 May '07

10 May '07

Laptop has Pentium Dual Core and Intel HD Audio Kernel is 2.6.20.3 compiled for Processor family Core2. In-kernel alsa drivers are 1.0.14rc3 (cat /proc/asound/version) and compile ok. In-kernel drivers do not work for this sound card. Downloaded 1.0.14rc3 and applied patch_realtek.c ./configure --prefix=/usr --with-cards=hda-intel "make" gives the following error: make[1]: Entering directory `/usr/src/linux-2.6.20.3' > /usr/src/linux-2.6.20.3/Makefile:491: > /usr/src/linux-2.6.20.3/arch//Makefile: No such file or directory If the kernel is compiled with Processor family = Pentium M, the external alsa drivers compile clean and the soundcard works. So, Processor family = Core2 cannot be selected when using external alsa-driver-1.0.14rc3. Chris Pemberton

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[alsa-devel] Problem with ALSA Capture and Playback : start_delay
by Nobin Mathew 10 May '07

10 May '07

Why start_delay=1 for capture and start_delay=0 for playback what this means I have one app which does both capture and playback, but when i tried to read ALSA buffer(capture) periodically it gives ALSa buffer overflow error. what my app does is read from ALSA (capture mic) and write to ALSA(playback). In this case for both Capture and Playback setting i am using start_delay = 0. But if i am giving some delay between Read and Write no overflow error happens Any idea why this happens Thanks

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

I already installed dos + the drivers and ubuntu, So I'll try working with that. As far as I can tell from the win95 driver name, the "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" is full-duplex : In : http://www.aztech.com/support_download_soundcard01.html : 2316FD95.ZIP 90,526 Windows 95 Full Duplex Audio Driver Version 4.8.0 Build 1128 I will try working with the software first, then probably either the epprom or the jumper (since I have dos + drivers working I'm tempted to try the epprom first ;-) ) I can't at the moment compile in that machine (My poor little xubuntu is barely working as it is, I don't even dare installing samba let alone a dev-env of any sorts..., maybe when I'll get everything working I'll install slackware, I decided gentoo is not a good idea ;-) ) so I won't be testing your drivers any time soon :-( , I do however hope you'll submit your drivers cause a non PnP driver can't hurt and waiting for confirms on such old hw will be point-less... When I'll have results I'll let you know... Also, I'm CCing the mailing list now as pair request... Thanks Rene, Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] [PATCH] snd-aoa-i2sbus: use MODULE_DEVICE_TABLE instead of plain MODULE_ALIAS
by Johannes Berg 10 May '07

10 May '07

This patch changes snd-aoa-i2sbus to use MODULE_DEVICE_TABLE instead of a hardcoded MODULE_ALIAS. Thanks to Sylvain Munaut for pointing this out. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- Please apply to the alsa tree. sound/aoa/soundbus/i2sbus/i2sbus-core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) --- wireless-dev.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:08:25.903457795 +0200 +++ wireless-dev/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:09:21.253457795 +0200 @@ -23,9 +23,6 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); -/* for auto-loading, declare that we handle this weird - * string that macio puts into the relevant device */ -MODULE_ALIAS("of:Ni2sTi2sC"); static int force; module_param(force, int, 0444); @@ -37,6 +34,8 @@ static struct of_device_id i2sbus_match[ { } }; +MODULE_DEVICE_TABLE(of, i2sbus_match); + static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r, int numcmds)

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[alsa-devel] [PATCH] ASoC AC97 static GPL symbol fix
by Liam Girdwood 10 May '07

10 May '07

This patch fixes a bug whereby the GPL'ed symbol ac97_dai was being declared as static. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

Hello alsa devs, I have an old "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" which I know to work on windows 2000 natively and on other windows using drivers from http://www.aztech.com/support_download_soundcard01.html, however, I was unsuccessful in using it in linux ( I tried many dists, but my most recent failure was with xubuntu 7.04 which have the most recent alsa). I believe the card is unsupported in alsa and I was hoping someone might be free to help me with this legacy card ? Since the card is unsupported I figured this should go the devs. * Their is this piece of code which might be relevant: http://www.gelato.unsw.edu.au/lxr/source/sound/isa/azt2320.c 21 This driver should provide support for most Aztech AZT2320 based cards. 22 Several AZT2316 chips are also supported/tested, but autoprobe doesn't 23 work: all module option have to be set. * I'm unsure of what it means to configure manually or how to do so... I do however have the card working currently with windows 2000 so if its a question of IRQ and DMA, I can find those with relative ease. * I obviously tried modprobeing the snd-sgalaxy driver/module, with no success - possibly related ? Thanks Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] Sound on SMDK2410
by Jose Henrique Spahn Torres 10 May '07

10 May '07

Hi all, I am trying to get sound working on a SMDK-2410 evaluation board. I compiled kernel 2.6.20 with alsa-driver-1.0.14rc4. However, I could not find the specific uda-1341TS driver for this board. I would like to know the status of this driver, if there will be one some time soon. Thanks, Henrique

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Re: [alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 10 May '07

10 May '07

On Thu, May 10, 2007 at 09:58:17PM +0900, Paul Mundt wrote: > On Thu, May 10, 2007 at 02:08:33PM +0200, Manuel Lauss wrote: > > +## > > +## Audio unit modules > > +## > > + > > +config SND_SOC_SH4_HAC > > + select AC97_BUS > > + select SND_SOC_AC97_BUS > > + select SND_AC97_CODEC > > + tristate > > + > > +config SND_SOC_SH4_SSI > > + tristate > > + > This might seem like a silly question, but how are these two supposed to > be enabled? At the very least you should have some dependencies on the > CPU subtypes that support these blocks. > > > + > > + > > +## > > +## Boards > > +## > > + > Or are you expecting that the board-specific code will simply select > these? That was the idea; the other ASoC boards do similar things. > Are you planning on submitting a stub that enables these? Or is this just > to be a placeholder until some board wants them? I suppose I can submit my AC97 board code, it does not depend on anything that is not already submitted. I'll add the ac97 board code to the patch and re-submit it. Thanks, Manuel Lauss

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[alsa-devel] [PATCH5/5] ASoC S3C24xx machine drivers - Kconfig
by Liam Girdwood 10 May '07

10 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds Kconfig and build support for the Neo1973, SMDK2443 and S3C2443 AC97 ALSA audio drivers. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH4/5] ASoC S3C24xx machine drivers - I2C ID for LM4857
by Liam Girdwood 10 May '07

10 May '07

This patch adds I2C ID for the LM4857 audio amp and corrects the spacing of the WM8731, WM8750 and WM8753 ID's. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH3/5] ASoC S3C24xx machine drivers - SMDK 2443
by Liam Girdwood 10 May '07

10 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds ALSA support for the SMDK2443 reference board. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 10 May '07

10 May '07

Hello, here goes round 2 of the SH7760 ASoC patch. Changes since last submission: - incorporate Paul Mundt's suggestions - work around an oops when aplay is used in mmap mode ('aplay -M') by changing the allocation type to SNDRV_DMA_TYPE_CONTINUOUS Thanks, Manuel Lauss --- ALSA ASoC support for SH7760 - AC97 driver for HAC unit(s) found on SH7760/SH7780 - I2S driver for SSI unit(s) found on SH7760/SH7780 - SH7760 DMA engine (dmabrg) sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/sh/Kconfig | 30 ++++ sound/soc/sh/Makefile | 11 ++ sound/soc/sh/dma-sh7760.c | 350 +++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 323 ++++++++++++++++++++++++++++++++++++ sound/soc/sh/ssi.c | 400 +++++++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1117 insertions(+), 1 deletions(-) Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..6c90c2c --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,30 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..32bf713 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,11 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..d495cf5 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,350 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> + + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in e.g. MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + pr_debug("PCM data: addr 0x%08lx len %d\n", + (u32)runtime->dma_addr, runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = runtime->dma_area; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = runtime->dma_area; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * position at the time the IRQ was triggered, which results in + * fast-playback sound in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, GFP_KERNEL, + DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-pcm", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..696ecb8 --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,323 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* + * BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "hac: reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + pr_debug("hac: invalid depth %d bit\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..d69bf63 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,400 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT) +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK (7 << CR_DWL_SHIFT) +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK (7 << CR_SWL_SHIFT) +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_SHIFT 4 +#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */ +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + pr_debug("ssi: already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); + pr_debug("bits: %d channels: %d\n", bits, channels); + + ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | + CR_SWL_MASK); + + /* direction (send/receive) */ + if (!recv) + ssicr |= CR_TRMD; /* transmit */ + + /* channels */ + if ((channels < 2) || (channels > 8) || (channels & 1)) { + pr_debug("ssi: invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK; + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + pr_debug("ssi: invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP | + CR_SWS_MASTER | CR_SCK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + pr_debug("ssi: unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + pr_debug("ssi: invalid inversion\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + pr_debug("ssi: invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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[alsa-devel] Mic and Line In Capture AC97 slots issue
by Nobin Mathew 10 May '07

10 May '07

http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000835.html http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000837.html the overrun issue was because my application was not fast enough. I resolved that problem by disabling capture of AC-Link slot4 in my ac97 controller. So capture happens only in slot3. According to ac97 spec Mic comes in Slot6 of AC-Link. I am using wolfson wm9713 codec. In this LineR and LineL are routed to LeftADC and RightADC. And Mic1 (Mic A) is also coonected to LeftADC and RightADC. So i am assuming that Mic 1 will come in Slot3 and Slot4. But my application is taking only mono data, so XRUN will happen. What is the correct configuration for this? What should be the configuration if i want to capture both Line IN and Mic? means AC-Link settings Thanks

2 1

[alsa-devel] [PATCH] snd-usb-audio: Another Logitech QuickCam ID
by Daniel Drake 10 May '07

10 May '07

This patch adds the ID for another quickcam microphone, reported by freqmod on ALSA ticket #0003040 I'm going to submit a USB patch separately to provide a macro to simplify these entries, as suggested by Alan Stern. We could switch to using that in future. Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -49,6 +49,15 @@ USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, + .idProduct = 0x0850, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL

2 1

[alsa-devel] Recording stops after a short while on ATI SB450
by Kjeld Flarup 10 May '07

10 May '07

Hello This list first got to my attention after I filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3064 The short story is that then using arecord the input stops after a short while. The only bug report I get is this when starting to capture. hda-intel: Invalid position buffer, using LPIB read method instead. I'm actually on the edge of buying a USB headset for this computer. I unfortunately only have it for 2½ weeks, then my mother in law is going to have it. Meanwhile I can test almost anything if anybody has an idea. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Forssavej 49, 7600 Struer, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

[alsa-devel] VIA 8237 - random switch to 8kHz sample rate?
by dustin＠seznam.cz 09 May '07

09 May '07

Hello, Occasionally my integrated VIA 8237 + AD1888 stops transmitting SPDIF AC3 (non-audio) streams. When troubleshoting, aplay -v -Dplughw:0 44.1kHz.wav shows in the actual card info section sample rate of 8kHz. The sound via DAC reflects this - no trebles. Logically, my AVR cannot detect 8kHz SPDIF a simply mutes. Since there is no sample rate mixer control, the only remedy is to reload corresponding alsa modules. The source via82xx.c uses 48kHz only. Please would you point me at some direction to watch, check, debug, etc.? It happens very rarely, I do not know how to reproduce it yet. Thanks a lot, Pavel.

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[alsa-devel] Fwd: ALSA problems with simultaneous Capture and playback
by Nobin Mathew 09 May '07

09 May '07

---------- Forwarded message ---------- From: Nobin Mathew <nobin.mathew(a)gmail.com> Date: May 9, 2007 11:57 PM Subject: Re: [alsa-devel] ALSA problems with simultaneous Capture and playback To: Lee Revell <rlrevell(a)joe-job.com> when I do capture and Playback independently for any channel configurations and any sampling rate configuration it works perfectly. Following configuration of simutaneous playback and capture work. loop "line out" and "line in" externaly using stereo jack. start arecord and then start aplay. Then it records properly. The application which is giving erorr is custom application which sets params properly. if i run "aplay" and "arecord" first (after boot) and then i run my application then it works properly. reverse is not working (i.e. running my app first destroys whole ALSA, then even arecord and aplay is not working). On 5/9/07, Lee Revell <rlrevell(a)joe-job.com> wrote: > On 5/9/07, Nobin Mathew <nobin.mathew(a)gmail.com> wrote: > > One read and write handle 256 frames. > > > > Write seems to be OK, but read fails > > > > State 4 is SND_PCM_STATE_XRUN > > > > why this XRUN is happening? > > If your hardware buffer is small you may need to run your application > with realtime priority (SCHED_FIFO) to avoid xruns. > > I guess this is a custom driver? Please post a link to the source code. > > Lee >

1 0

[alsa-devel] ALSA problems with simultaneous Capture and playback
by Nobin Mathew 09 May '07

09 May '07

I am facing a problem with simultaneous playback and Capture. I am doing Mic Capture and Playing it back. this is format i am using SND_PCM_FORMAT_S16_LE. I am recording only one channel "Mono". this is my driver pcm_hardware configuration static struct snd_pcm_hardware pcm_hw_info = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME, /* * ALSA doesn't support 18-bit or 20-bit packed into 32-bit * words. It also doesn't support 12-bit at all. */ .formats = SNDRV_PCM_FMTBIT_S16_LE, /* should this be continuous or knot? */ .buffer_bytes_max = 64 * 1024, .period_bytes_min = 1024, .period_bytes_max = 1024, .periods_min = 4, .periods_max = 64, }; I am getting following errors when i do simultaneous capture and playback. Start the while looop MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=2048 ReadFromAlsa():State:2 Overflow state MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=1280 ReadFromAlsa():State:2 Overflow state MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=1280 ReadFromAlsa():State:2 Overflow state One read and write handle 256 frames. Write seems to be OK, but read fails State 4 is SND_PCM_STATE_XRUN why this XRUN is happening? any idea why this is coming? where i am going wrong? Thanks

2 1

[alsa-devel] FIXME for Rear Channels of CMI8738, how do i write a softvol to fix this defect?
by John L. Utz III 09 May '07

09 May '07

Hello; the CMI8738 allows for 4 channel output by providing a second DAC, but sadly the hardware has no volume control for said second DAC. A review of /usr/share/alsa/cards/CMI8738-MC6.conf shows that the implementor of said file was aware of the need to fix the problem: # 2nd DAC # FIXME: we need a volume attenuator for rear channel. CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 1 } I'd really like to fix this! Is it fixable by creating a softvol plugin? http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html#pcm_plugins if so, can i have a hint? :-) so, if i have to invoke speaker test with -Dhw:0,1 to get 4 channel out, what does that tell me about how i need to configure a softvol for this item? Or am i missing the point and it's actually the *driver* that needs to be doctored up to provide a softvol implementation. If so, how? tnx! johnu

2 3

[alsa-devel] Realtek: ALC265/268 Support?
by Henry Walpool 09 May '07

09 May '07

Thought I checked my specs but think I missed the mark. I have a new Toshiba Laptop A205-S4577 Laptop and the laptop has a realtek high definition card. The hda-intel driver detects the card but has no sound and only two output controls in alsamixer nothing else. I have tried the latest release candidate ALSA 1.0.14rc4 without any luck. Is this card supported. Details below? Distribution: Kubantu Feisty 7.04, Kernel 2.6.20-15-generic, and used a custom 2.6.20.1 kernel and tried the lastest release drivers. Realtek ID 268 $ lspci -v -v 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 0 2) Subsystem: Toshiba America Info Systems Unknown device ff10 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PER R- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 21 Region 0: Memory at f0b40000 (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: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 $ lspci -v -n 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1179:ff10 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at f0b40000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 $ cat /proc/asound/card0/codec#0 Codec: Realtek ID 268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff10 Revision Id: 0x100003 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x01 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0xc0: OUT HP Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30940: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015812d: [N/A] Speaker at Ext N/A Conn = Optical, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x01 0x01] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x14 0x14] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0b40000 irq 21 Thanks!

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[alsa-devel] Fwd: Fwd: [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

Forwarded Conversation Subject: [ASoC] Mic record is not working with wm9713 ASoC driver ------------------------ From: Nobin Mathew <nobin.mathew(a)gmail.com> To: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 2:48 PM I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew -------- From: Andrew Morton <akpm(a)linux-foundation.org> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org, Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de> Date: Wed, May 9, 2007 at 3:51 PM [Quoted text hidden](maintainers cc'ed) -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 3:55 PM On Wed, 2007-05-09 at 14:48 +0530, Nobin Mathew wrote: > I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". I've replied to this duplicate post on alsa-dev. Liam -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 4:14 PM Liam Sorry for disturbing you again, I could not find that reply on alsa devel mailing list Can u send that to me Thanks [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Date: Wed, May 9, 2007 at 4:22 PM The mail server is probably a little behind..... give it about 30 mins... Liam [Quoted text hidden]> - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 4:25 PM Attachments: identity_wm9713.c Liam i got the mail. I am not setting any audio paths in my machine driver [Quoted text hidden] -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 4:28 PM Attachments: identity_wm9713.c [Quoted text hidden] -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 5:00 PM Liam i have few more doubts i have only Mic1, i am not connecting to Mic2A and Mic 2B So can i remove audio path to Mic2A and Mic2B means my audio path will be like static const char* audio_map[][3] = { /* mic is connected to mic1 - with bias */ {"MIC1", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 1"}, #if 0 /* mic is connected to mic2A - with bias */ {"MIC2A", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 2"}, /* mic is connected to mic2B - with bias */ {"MIC2B", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 3"}, #endif {NULL, NULL, NULL}, }; instead of static const char* audio_map[][3] = { /* mic is connected to mic1 - with bias */ {"MIC1", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 1"}, /* mic is connected to mic2A - with bias */ {"MIC2A", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 2"}, /* mic is connected to mic2B - with bias */ {"MIC2B", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 3"}, {NULL, NULL, NULL}, }; what is the meaning of these lines, what is RXP and RXN /* set up mainstone codec pins */ // snd_soc_dapm_set_endpoint(codec, "RXP", 0); // snd_soc_dapm_set_endpoint(codec, "RXN", 0); snd_soc_dapm_set_endpoint(codec, "MIC1", 0); [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Date: Wed, May 9, 2007 at 5:30 PM On Wed, 2007-05-09 at 17:00 +0530, Nobin Mathew wrote: > Liam i have few more doubts > > i have only Mic1, i am not connecting to Mic2A and Mic 2B > So can i remove audio path to Mic2A and Mic2B > means > my audio path will be like > static const char* audio_map[][3] = { > > /* mic is connected to mic1 - with bias */ > {"MIC1", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 1"}, > #if 0 > /* mic is connected to mic2A - with bias */ > {"MIC2A", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 2"}, > /* mic is connected to mic2B - with bias */ > {"MIC2B", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 3"}, > #endif > > {NULL, NULL, NULL}, > }; > This is fine, it's safe to remove unused Mic's. > instead of > > static const char* audio_map[][3] = { > > /* mic is connected to mic1 - with bias */ > {"MIC1", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 1"}, > /* mic is connected to mic2A - with bias */ > {"MIC2A", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 2"}, > /* mic is connected to mic2B - with bias */ > {"MIC2B", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 3"}, > > {NULL, NULL, NULL}, > }; > > > what is the meaning of these lines, what is RXP and RXN > > /* set up mainstone codec pins */ > // snd_soc_dapm_set_endpoint(codec, "RXP", 0); > // snd_soc_dapm_set_endpoint(codec, "RXN", 0); > snd_soc_dapm_set_endpoint(codec, "MIC1", 0); RXP and RXN are from a different codec and should be removed. Also remove the MIC1 endpoint setting to 0 as this will disconnect it. Comments from driver :- static int identity_wm9713_ac97_init(struct snd_soc_codec *codec) { struct platform_controller *controller = codec->ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; snd_ac97_pcm_assign(codec->ac97->bus, 3, ac97_defs); aaci->ac97_bus = codec->ac97->bus; /* * * Disable AC97 PC Beep input on audio codecs. * */ if (ac97_is_audio(codec->ac97)) snd_ac97_write_cache(codec->ac97, AC97_PC_BEEP, 0x801e); return 0; } AC97 beep should be disabled in userspace. e.g. alsa configuration file or by changing mixers settings (alsactl can save and restore settings). unsigned int identity_wm9713_config_sysclk(struct snd_soc_pcm_runtime *rtd, struct snd_soc_clock_info *info) { return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12187500); } This function has been removed in current ASoC versions. It's best to upgrade from kernel.org or from here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Liam -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 5:30 PM Liam Thanks of lot i got it working Cheers [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 5:38 PM It's probably best to keep this thread on alsa-dev otherwise things start to get confusing when cross posting occurs. Liam --------

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[alsa-devel] [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew

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[alsa-devel] [RFC PATCH] ASoC AC97/I2S for SH7760
by Manuel Lauss 09 May '07

09 May '07

ALSA ASoC support for SH7760 - SH7760 DMA (requires DMABRG patch sent to linux-sh list) - AC97 support for HAC unit found on SH7760/SH7780 - I2S support for SSI unit found on SH7760/SH7780 sound/soc/Kconfig | 1 sound/soc/Makefile | 3 sound/soc/sh/Kconfig | 30 +++ sound/soc/sh/Makefile | 11 + sound/soc/sh/dma-sh7760.c | 361 ++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 335 +++++++++++++++++++++++++++++++++++++ sound/soc/sh/ssi.c | 413 ++++++++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1153 insertions(+), 1 deletion(-) Comments welcome! Thanks, Manuel Lauss diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..6c90c2c --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,30 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..32bf713 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,11 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..a2aa1a8 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,361 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> +#include <asm/io.h> + +/* #define PCM_DEBUG */ + +#define MSG(x...) printk(KERN_INFO "sh7760-pcm: " x) + +#ifdef PCM_DEBUG +#define DBG(x...) MSG(x) +#else +#define DBG(x...) do {} while (0) +#endif + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in e.g. MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + MSG("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + MSG("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + DBG("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, + runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = runtime->dma_addr; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = runtime->dma_addr; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * time the IRQ was triggered, which gives fast-playback sound + * in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, DMABRG_PREALLOC_BUFFER, + DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-dmabrg", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..157268c --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,335 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* + * BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +/* #define AC97_DEBUG */ + +#ifdef AC97_DEBUG +#define MSG(x...) printk(KERN_INFO "sh4-hac: " x) +#else +#define MSG(x...) do {} while (0) +#endif + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RIER_STARYIE (1 << 22) +#define RIER_STDRYIE (1 << 21) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "HAC reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + MSG("unsupported depth %d bits\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..79642e0 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,413 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + * + * Don't forget to set IPSEL register to enable SSI output! + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +/* #define SSI_DEBUG */ + +#ifdef SSI_DEBUG +#define MSG(x...) printk(KERN_INFO "sh4-i2s: " x) +#else +#define MSG(x...) do {} while (0) +#endif + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK 3 +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK 7 +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK 7 +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_MASK 7 /* bitclock divider */ +#define CR_CKDIV_SHIFT 4 +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + MSG("SSI already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + MSG("hw_params() enter\nssicr was %08lx\n", ssicr); + MSG("bits: %d channels: %d\n", bits, channels); + + /* direction (send/receive) */ + if (recv) + ssicr &= ~CR_TRMD; + else + ssicr |= CR_TRMD; + + /* channels */ + ssicr &= ~(CR_CHNL_MASK << CR_CHNL_SHIFT); + if ((channels < 2) || (channels > 8) || (channels & 1)) { + MSG("invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + ssicr &= ~(CR_DWL_MASK << CR_DWL_SHIFT); + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + MSG("invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + ssicr &= ~((CR_SWL_MASK << CR_SWL_SHIFT) | CR_PDTA); + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + MSG("invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + MSG("hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~(CR_CKDIV_MASK << CR_CKDIV_SHIFT); + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + MSG("invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + MSG("set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA); /* I2S format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + MSG("unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + ssicr &= ~CR_BREN; + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + ssicr &= ~(CR_SWSP | CR_SCKP); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + MSG("invalid inversion\n"); + return -EINVAL; + } + + ssicr &= ~(CR_SWS_MASTER | CR_SCK_MASTER); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + MSG("invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + MSG("set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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[alsa-devel] [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew

1 0

[alsa-devel] snd-serial-u16550 overflow problems?
by r10 kindsofpeople 08 May '07

08 May '07

{reposting (with apologies) from alsa-user at another user's suggestion} I'm having what appears to be overflow problems within the snd-serial-u16550 driver. I'm using 1.0.14rc1. When calling "amidi -p hw:0,0,0 -s FILE.SYX" where hw:0,0,0 is the snd-serial-u16550 port, the data coming from the serial port appears to be a small fraction (900 bytes) of the total file size (13000 bytes). If I repeat the experiment, but send to a USB MIDI port, all is well. If I send the data to the serial port using calls to snd_seq_event_output_direct(), and I add a 1 second pause between every sysex message, all is well (aside from it taking too long). I'm logging data directly at the PC serial port using another PC, so I've eliminated the possibility that the serial-to-MIDI hardware is losing the data. My code is using blocking mode, and besides, amidi behaves the same way as my code. I'm hesitant to blame the driver code (especially given that last time I did this, it was an interrupt configuration problem), but I could use some help figuring out where to look next. Please? Are there calls I can use to ensure that the previous message has gone out before writing the next one? Given the amount of data I'm sending, should I be using snd_seq_event_output and scheduling the events on the queue? John _________________________________________________________________ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-us&ocid=TXT_TAGHM_migrati…

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[alsa-devel] [PATCH] ppc-beep - switch to using input_dev->dev.parent
by Dmitry Torokhov 08 May '07

08 May '07

In preparation for struct class_device -> struct device input core conversion, switch to using input_dev->dev.parent when specifying device position in sysfs tree. Also, do not access input_dev->private directly, use helpers. Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> --- sound/ppc/beep.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) Index: work/sound/ppc/beep.c =================================================================== --- work.orig/sound/ppc/beep.c +++ work/sound/ppc/beep.c @@ -118,7 +118,7 @@ static int snd_pmac_beep_event(struct in default: return -1; } - chip = dev->private; + chip = input_get_drvdata(dev); if (! chip || (beep = chip->beep) == NULL) return -1; @@ -239,8 +239,8 @@ int __init snd_pmac_attach_beep(struct s input_dev->evbit[0] = BIT(EV_SND); input_dev->sndbit[0] = BIT(SND_BELL) | BIT(SND_TONE); input_dev->event = snd_pmac_beep_event; - input_dev->private = chip; - input_dev->cdev.dev = &chip->pdev->dev; + input_dev->dev.parent = &chip->pdev->dev; + input_set_drvdata(input_dev, chip); beep->dev = input_dev; beep->buf = dmabuf; @@ -251,8 +251,8 @@ int __init snd_pmac_attach_beep(struct s err = snd_ctl_add(chip->card, beep_ctl); if (err < 0) goto fail1; - - chip->beep = beep; + + chip->beep = beep; err = input_register_device(beep->dev); if (err)

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[alsa-devel] Fwd: [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 07 May '07

07 May '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K PS : if I sent this message twice, excuse-me. I haven't confirm my subscription the first time

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[alsa-devel] [PATCH] Add sequence parsing for speaker-outs in snd_hda_parse_pin_def_config()
by Steve Longerbeam 07 May '07

07 May '07

2 2

Re: [alsa-devel] Getting Started on Alsa
by Ash Willis 07 May '07

07 May '07

Hi Chris, > I'd like to take action on bug #305, since it's my own itch. I've done very > low-level programming, but not on Linux device drivers. I'm very familiar > with command line development tools. > If you're not familiar with drivers or audio drivers in particular then reading Writing an ALSA Driver by Takashi Iwai might be a good place to start. You're referring to the suspend/resume bug in the cs46xx driver, right? Maybe try dumping all the registers while it's working and dumping them while it's broken and comparing the two sets of data? I guess the suspend function is doing something that it shouldn't be, or the resume function isn't doing something that it should be...good luck on finding out what that is. Ash =

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Re: [alsa-devel] Getting Started on Alsa
by Ash Willis 07 May '07

07 May '07

> Thanks for the tips, Ash. I've found the Iwai document. Now there's some > light reading. I take it that it's still relevant, even though it hasn't > been updated since 2005. > I haven't actually read it since 2005 I'm afraid :) but I don't think anything's changed that drastically. The fundamentals are all there. > Yes, the cs46xx driver. That chip is in a lot of machines (at least ten > ThinkPad models), and with this bug nobody running Alsa likes Linux very > much. > Yeah, it's a pretty poor show that this bug has existed for so long. Let's hope you can zap it! > Is there some (handy) way to dump the registers from DOS or Windows as a > comparison? > Well, that depends. I'm not familiar with the hardware. There's a Windows program called PCItree. It allows you to view the current status of the ioport of a chosen device. Now, that's pretty handy if the regs for the card are mapped directly onto the io port. It's usually the case though, I think, that the registers aren't just mapped on and you'll need to query the card. If they're not mapped onto the io port (and forgive me again - not familiar with this chip) then you could use a debugger such as Numega SoftIce to manually issue queries to find out the contents of the registers. Sounds like a happy fun time. Writing the code to do a reg dump might be fun too. You may also want to use a debugger/disassembler with Windows (or other) drivers in order to find out what those drivers do to the card on resume. > For development: Since the drivers go into the kernel, I should be able to > find a page on getting a development environment set up... right? I'm You don't need to mess with the kernel as ALSA is available as a package. All you really need is the latest copy of alsa-driver and a compiler and you're good to go. Ash =

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[alsa-devel] 1.0.14~rc4 prepared for upload.
by Elimar Riesebieter 06 May '07

06 May '07

As the compile errors in alsa-lib fixed I've preparded 1.0.14~rc4 (svn r1849) for upload: alsa-driver alsa-lib alsa-oss alsa-plugins alsa-utils alsa-tools stays still as 1.0.13~rc3. Jordi, could you please upload? Now I go for a beer ;) Elimar -- You cannot propel yourself forward by patting yourself on the back.

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[alsa-devel] Getting Started on Alsa
by Chris Schumann 06 May '07

06 May '07

Hello, I'd like to take action on bug #305, since it's my own itch. I've done very low-level programming, but not on Linux device drivers. I'm very familiar with command line development tools. Is there any documentation on how to get started doing bugfixes or development for Alsa, or do I just look for source and start hacking? Thanks, Chrs

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[alsa-devel] [PATCH] hda_codec: restore control element values during resume
by Daniel Drake 06 May '07

06 May '07

After resume from suspend-to-RAM, there is no sound output from my Intel HDA hardware: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Found in a Dell Inspiron 640m, SigmaTel STAC9200 chip. After resuming, sound immediately returns when you adjust the volume one notch in either direction. I reported this on the ALSA bug tracker (#0002989). Further investigation shows that the AC_VERB_SET_AMP_GAIN_MUTE verb is not being executed during resume, which seems strange. Does other hardware really store this value during a suspend/resume cycle? Or maybe the volume is generally controlled through other means on other setups? I'm not entirely sure what the difference is between kcontrol and kcontrol_new structures, but I note that there is already some code to restore the values for kcontrol_new elements. I solved the issue with the following patch. Is it correct? Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux-2.6/sound/pci/hda/hda_codec.c =================================================================== --- linux-2.6.orig/sound/pci/hda/hda_codec.c +++ linux-2.6/sound/pci/hda/hda_codec.c @@ -2248,15 +2248,39 @@ EXPORT_SYMBOL(snd_hda_suspend); int snd_hda_resume(struct hda_bus *bus) { struct list_head *p; + struct snd_ctl_elem_value *val = kmalloc(sizeof(*val), GFP_KERNEL); + + if (!val) + return -ENOMEM; list_for_each(p, &bus->codec_list) { struct hda_codec *codec = list_entry(p, struct hda_codec, list); + struct snd_kcontrol *kctl; + + codec->in_resume = 1; + hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + + + list_for_each_entry(kctl, &bus->card->controls, list) { + memset(val, 0, sizeof(*val)); + val->id = kctl->id; + /* Assume that get callback reads only from cache, + * not accessing the real hardware + */ + if (snd_ctl_elem_read(bus->card, val) < 0) + continue; + snd_ctl_elem_write(bus->card, NULL, val); + } + if (codec->patch_ops.resume) codec->patch_ops.resume(codec); + + codec->in_resume = 0; } + kfree(val); return 0; }

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[alsa-devel] New Soundcard Matrix
by Jaroslav Kysela 05 May '07

05 May '07

Hello all, I moved the contents of matrix from XML files to Wiki - see http://bugtrack.alsa-project.org/main/index.php/Matrix:Main Please, no more updates to old matrix now. I left the information structure (we can update it later): Vendor -> Device -> ALSA Module If any volunteers are watching this list, we have many "wanted" pages now: http://bugtrack.alsa-project.org/main/index.php/Special:Wantedpages Example page for Matrix:Tag-* : http://bugtrack.alsa-project.org/main/index.php?title=Matrix:Tag-PCI&action… Example page for Matrix:Module-* : http://bugtrack.alsa-project.org/main/index.php?title=Matrix:Module-ali5451… Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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Re: [alsa-devel] Wiki Site
by Jaroslav Kysela 05 May '07

05 May '07

On Fri, 4 May 2007, cameron porter wrote: > Greetings, > > I noticed the notice on your front page needing people to help transfer the > site contents over to your wiki that you set up. I did a little bit of work, > and managed to get a navigation, downloads page, application page, and some > other links. Your work is greatly appreciated here. > I would be glad to continue, if you think I am doing things right. The next > thing I had planned to make some changes to was the documentation section, > add a changes section, add a structure whereby you have an article for each > package, and then the folder for that package contains all the different > versions that have been or will be released. Please give me any thoughts on > any of these, and tell me if you think they are a good idea, and whether I > should continue or not. We need to move the contents at first, but I agree that a good page structure should be created during this process. Please, go ahead - the nice thing on wiki is that we can rearrange information later (if we find a problem with the structure). I just started to move probably most problematic part - soundcard matrix to wiki. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Call for Explanation of ALSA Config Behavior
by Jeff Rush 05 May '07

05 May '07

I'm trying to get ALSA to support all the little features of my laptop, which is based on the Intel HDA chipset with the Realtek codec. I'm also playing with the softvol plug to add a volume controls over certain virtual devices. A few questions I can't figure out: 1. Isn't it the standard convention that there be a control named "Master" that adjusts the overall volume of a device? So why does the Intel HDA ALSA driver define one named "Headphone" that does the same thing instead of following that convention? Is there some reason, or perhaps a difference in opinion within the ALSA community on the naming of controls? Shouldn't every ALSA driver define a "Master" control? 2. When I add a control using the softvol plugin, the mixer GUI shows it in both the playback and the capture set of controls. Why and how can I tag a control to belong to only one set or the other? Being a volume control it seems it should only be in the playback collection. 3. Once I add such a control, it -never- goes away. What I mean is that if I rename it, or remove it, and restart ALSA or even restart the entire computer, the old name for the control always shows up in the mixer GUI. I've grepped *all* of /etc, looking in the /etc/asound.conf, /etc/asound.state and ~/.asoundrc files but the old name isn't anywhere -- where is it coming from? Thanks very much for any help you can provide, For years I've just tried to "use" ALSA but now I'm trying to actually understand its subtle nuances. -Jeff

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[alsa-devel] DMix period_size should always be half of buffer size?
by Prabhu Sivaraja 04 May '07

04 May '07

This is a followup to my earlier question about dmix not being able to mix small files. I investigated it more and here is what I found. Again background info. I have a 64KB as kernel buffer and a 64KB circular buffer for my device. In a poll thread I copy the data obtained from ALSA lib from the kernel buffer to my device circular buffer by keeping track of appl_ptr and hw_ptr(dmix only). Everything was working fine except when I was trying to mix files smaller than 64KB(my buffer size), the small file was not getting mixed through the dmix plugin. Anything over 64KB was getting mixed just fine. So I set about reducing my period size from 32KB to 4KB while maintaining my buffer size at 64KB. When I tried to do this my hw_ptr never gets incremented from 0. Hence my driver is stuck in a loop waiting for bytes to copy. This is only observed when I am using dmix plugin and NOT when using hw or plughw. When my period size is half the size of my buffer everything works fine with dmix (I verified this by reducing my buffer size to 8KB and making my period size 4KB. hw_ptr gets incremented then but I get buffer problems). It seems like dmix is forcing me to keep my period size half the size of my buffer. hw and plughw devices do not seem to do that. Is this a limitation of dmix. Could anyone comment on the dmix limitation that I am experiencing. Thanks a lot of all your help. Prabhu.

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[alsa-devel] Aureon 7.1 Firewire supported by ALSA ?
by Michael Gerdau 04 May '07

04 May '07

Hi list, looking for an external (USB or Firewire) interface to record at least a single stereo 44.1kHz/24bit S/P-DIF signal I've come across the Aureon 7.1 Firewire but have not been able to find out whether it is already supported by ALSA. Another option would be the Aureon 5.1 USB MKII but according to the specs it is 16bit only. However when feeding S/P-DIF with a stereo 44.1kHz/24bit would the driver be recognize that and give me that signal unaltered or would it be trimmed to 16bit ? [Or is my understanding of S/P-DIF just plain wrong and such a signal would not go through the box at all ?] Any advice welcome, best, Michael -- Technosis GmbH, Geschäftsführer: Michael Gerdau, Tobias Dittmar Sitz Hamburg; HRB 89145 Amtsgericht Hamburg Vote against SPAM - see http://www.politik-digital.de/spam/ Michael Gerdau email: mgd(a)technosis.de GPG-keys available on request or at public keyserver

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[alsa-devel] [PATCH] multi-channel playback and capture fixes for stac92xx codecs
by Steve Longerbeam 04 May '07

04 May '07

Hi, I have picked up where Matt Porter left off on stac92xx HDA codec devel. I've been working with the 9200, 9221, 9202, 9250, and 9205 codecs, and I found some problems with muti-channel playback on 922x and 9205. Also a problem with input mux selection for codecs with more than one configured input pin. I've tested the attached patch on all the above listed codecs. Steve

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[alsa-devel] [PATCH] snd-usb-audio: Add another QuickCam ID
by Daniel Drake 04 May '07

04 May '07

This adds another Logitech QuickCam USB ID. These devices have vendor-specific device class values, but standards-compliant audio interfaces. Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -53,6 +53,16 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08f6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, + /* * Yamaha devices

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[alsa-devel] support for Medion laptops
by albox＠free.fr 04 May '07

04 May '07

Hi, I installed Ubuntu Feisty a few days ago and my sound card driver doesn't work anymore. It was working perfectly before updating to Feisty but I'm not sure the problem comes from the update since I was not using the last ALSA version but an older one patched with a patch released by Tobin Davis a few months ago (something else could have been changed in the ALSA source code since then) : --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c.orig 2006-10-08 10:58:28.000000000 -0700 +++ alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2006-10-09 10:50:03.000000000 -0700 @@ -112,6 +112,7 @@ ALC883_6ST_DIG, ALC888_DEMO_BOARD, ALC883_ACER, + ALC883_MEDION, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5080,6 +5081,8 @@ .config = ALC883_ACER }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x009f, .config = ALC883_ACER }, + { .pci_subvendor = 0x161f, .pci_subdevice = 0xd82b, + .modelname = "medion", .config = ALC883_MEDION } { .modelname = "auto", .config = ALC883_AUTO }, {} }; @@ -5167,6 +5170,20 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_MEDION] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, + alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + } + }; You can find some parts of the emails about this subject on this web page : http://www.spinics.net/lists/alsa-devel/msg04396.html The driver (Developement release 1.0.14rc3) can detect my sound card but I can't figure out what's wrong. Can anybody help me ? $ dmesg | grep alsa [ 20.468000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: codec_mask = 0x3 [ 20.648000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:1768: hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) [ 21.376000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:245: si3054: cannot initialize. EXT MID = 0000 [ 21.384000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:257: Link Frame Detect(FDT) is not ready (line status: 0000) [ 76.256000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.256000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.264000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.272000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.280000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 76.292000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 76.300000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 76.308000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 76.320000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 76.332000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.332000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.344000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.356000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.364000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 76.376000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 76.384000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 76.392000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 76.408000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 76.592000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.592000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.600000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.608000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.616000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 153.564000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 153.572000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 153.580000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 153.588000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 216.896000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 216.896000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 216.904000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 216.912000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 216.920000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 216.932000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 216.940000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 216.948000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 216.956000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 216.972000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 216.972000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 216.980000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 216.988000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 216.996000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 218.824000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 218.832000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 218.840000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 218.848000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 Thanks

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[alsa-devel] dmix does not mix small files??
by Prabhu Sivaraja 03 May '07

03 May '07

Hi, I've been stuck on this dmix problem for the past week and have tried everything under the sun. Please any help or comment is much appreciated. Thanks a lot. Background info. ----------------------- In my ALSA driver I have thread that keeps track of hw_ptr(dmix) appl_ptr(for all other cases) and copies the incoming data from kernal buffer into device circular buffer(which is monitored by the device). I allocate 64KB kernal buffer in my module_init function . (See allocation below) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), 64*1024 , 64*1024); So I monitor this kernal buffer and copy data from this to my device circular buffer which is also 64KB. My period size is defined as 32KB. (See below) substream->runtime->hw.period_bytes_min = 1024*32; substream->runtime->hw.period_bytes_max = 1024*32; substream->runtime->hw.periods_min = 2; substream->runtime->hw.periods_max = 2; My asound.conf file pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } Everything was working fine until I noticed something weird. When I am playing a long wav file by invoking aplay and then try and play another small wav file simultaneously, the second file never gets mixed if the file is less than 64KB(my buffer size). If the second file is larger than 64KB then the second file gets mixed with the first one as it should since I am using dmix plugin. So in order to debug this I tried to decrease my period_size = 4KB but when I do this my hw_ptr never moves. My question. 1. Why do you think a file smaller than 64KB does not get mixed. (Obviously it has something to do with my buffer size but I dont get why?) 2. When I decrease my period_size to 4KB while keeping my buffer size at 64KB, why doesnt my hw_ptr move? 3. How else would I debug this? Thanks in advance. I would really appreciate any help. Regards Prabhu.

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[alsa-devel] alsa test function?
by Ashlesha Shintre 03 May '07

03 May '07

Hi, Is there a test program for alsa, that gives a report on whether alsa is working correct with the underlying hardware or just does elementary testing of alsa on a system? Thanks, Ashlesha.

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[alsa-devel] 1.0.14rc4 release
by Jaroslav Kysela 03 May '07

03 May '07

Hello all, 1.0.14rc4 packages are available for download on http://www.alsa-project.org . The full changelog can be obtained here: http://www.alsa-project.org/changes/v1-0-14rc3--v1-0-14rc4.txt . Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Add support for new HP DV series laptops
by Tobin Davis 03 May '07

03 May '07

This patch adds support for 3 new HP laptops to the Conexant "Venice" driver. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] ALSA TImers
by Nobin Mathew 03 May '07

03 May '07

Hi, I am new to ALSA. Can any body explain me the use of timers in ALSA, timers like the one in pcm_timer.c Thanks in advance Nobin Mathew

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Re: [alsa-devel] Support for SAM9407?
by Ash Willis 03 May '07

03 May '07

Hi Pieter, > Are you interested in joining the FFADO project? (short into: FFADO = > new FreeBoB = drivers for FireWire Audio Interfaces / www.ffado.org). > I have absolutely zero experience in firewire audio and firewire in general to be honest, but the project does sound interesting. > To be frank/direct here, I think that ALSA is better 'completed' with > FFADO support than with driver support for obsolete cards. But of course > I'm a little biased here :). Be as Frank as you want, I totally agree with you :) The thing is, I have limited time and experience (I have written one ALSA driver). Although supporting older hardware doesn't bring a whole lot to the project, it does provide me with a training ground so I can be of greater use in the future. I can work at my own pace on an area that nobody else will be working on. I can improve my skills without stepping on other people's toes. Your project sounds interesting but I don't feel I have the knowledge to keep up on a project like this. This is why I'm messing around with this stuff, so that I can do some 'real work' some day soon ;) regards, Ash =

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[alsa-devel] alsa driver api
by Michel Benoit 03 May '07

03 May '07

Hi Does anyone have some code examples that show how to send pcm samples directly to an alsa driver. I'm (still) having trouble building/using alsa-lib and I want to make sure that my driver works correctly. If I understand correctly it is possible to send sound samples using the alsa driver api only (no alsa-lib). All I want to do for now is send a fixed sine wave to the driver. I saw some code snippets that seemed to do just that in a recent email on this list but it would help a lot if I could have a look at the complete application. Thanks, Michel

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[alsa-devel] Newb wants to fix snd_cs46xx driver
by chris＠idlelion.net 03 May '07

03 May '07

There's been a bug on this chip's driver for at least four years. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=305 There are two ThinkPad 600X's in my home that have the problem. The chip is also in the ThinkPad A20m, A20p, A21m, A21p, A22m, A22p, T20, T21 and T22... and probably other brands as well. In the 600X, it happens to be paired with a CS4297A... hey, should I just try to get Linux to use that? (I use Fedora BTW.) How would I try to do that? Or... better still... how would I get either the '97A driver installed automatically by soundcard detection, or failing that, actually fix the driver? (I am a software developer who has written machine code for fun.) Thanks, Chris

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[alsa-devel] Support for SAM9407?
by Ash Willis 02 May '07

02 May '07

Hi, I've just been digging around for information on the old Dream SAM9407 chip. What's the current support status in ALSA for this chip? I know some work was done on producing drivers for cards based on this chip and I know there were plans to port the efforts to ALSA. Did that ever happen? It doesn't seem so. Was there a specific reason why it wasn't ported in the end? In the interest of making ALSA that bit more complete as well as trying to improve my skills, I've been looking around for some older hardware that I can write drivers for. This chip seems like a good candidate considering the cards haven't quite disappeared into oblivion :) I was looking into the possiblity of porting the remaining kernel OSS drivers to ALSA...but I can't even find any test hardware because they're so rare. Ash Willis =

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Re: [alsa-devel] alsa driver api
by Ciaccia 02 May '07

02 May '07

Hi Michel, Maybe you are referring to my post. I always use alsa-lib to send data to the driver, I think it is the easiest way and preferred way. Months ago I wanted to do the same (i.e. send data to the alsa-driver without alsa-lib) but then I understood this was not the right way. I'm not saying you cannot or should not try that, but from my experience (I spent a lot of time) it's not going to work. The problem I had was that my system did not support shared objects, nor -ldl, so I could not user the "standard" alsa-lib (which usually require pthread, dl, an *.so). Then Takashi gave me a patch for that, and now I'm an happy "static" alsa-lib user. Maybe one of the gurus (Takashi?) will give you some tips, but I think you really need a good reason for that. Good luck, sorry for not helping you Andrea ----- Original Message ---- From: Michel Benoit <murpme(a)gmail.com> To: ciacciax(a)yahoo.com; alsa-devel(a)alsa-project.org Sent: Wednesday, May 2, 2007 7:40:29 PM Subject: alsa driver api Hi Does anyone have some code examples that show how to send pcm samples directly to an alsa driver. I'm (still) having trouble building/using alsa-lib and I want to make sure that my driver works correctly. If I understand correctly it is possible to send sound samples using the alsa driver api only (no alsa-lib). All I want to do for now is send a fixed sine wave to the driver. I saw some code snippets that seemed to do just that in a recent email on this list but it would help a lot if I could have a look at the complete application. Thanks, Michel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] how does a buffer underrun / broken pipe error take place?
by Ashlesha Shintre 02 May '07

02 May '07

Hi, Following up from my previous email, is there a website that explains in greater detail exactly how a buffer underrun occurs? I think that might help greatly in debugging my program, attached with the previous email. Thanks, Ashlesha.

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[alsa-devel] error in recovery from buffer under-run.
by Ashlesha Shintre 02 May '07

02 May '07

Hi, Thanks for the link you emailed earlier. I tried writing my own code, based on the how to in the alsa wiki. I have attached that code with this email. The problem it faces is a buffer underrun every alternate time, i try writing to the device. In my code, to take care of underruns, I do the following: period_size=64; buffer is a short int array which is 16000 in length. every alternate call to snd_pcm_writei is successful but only write in the vicinity of 1350 frames, instead of size which is initialised to length of buffer. Now, in the code below, the snd_pcm_recover function prints a message indicating a buffer under-run has occured, however, it returns a 0, indicating success in recovery from the error! I am not sure how to deal with this -- is there a problem with my ALSA libraries? Sorry about the length of this email and thanks again, Ashlesha. while(size>(2*period_size)) > { > frames = snd_pcm_writei (pcm_handle, buffer, size); > printf("frames returned is %d\n",frames); > if (frames>0) > temp=frames; > if (frames == -32){ > frames = snd_pcm_recover(pcm_handle, frames, 0); > printf("recover returned %d\n",frames); > } > if (frames < 0){ > printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); > return -1; > } > printf("%d frames written\n",frames); > if(temp>0) > size=size-temp; > } >

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Re: [alsa-devel] [ALSA][hda-codec] fix-connexant-headphone-mute
by Takashi Iwai 02 May '07

02 May '07

[keep cc to alsa-devel please] At Wed, 02 May 2007 13:47:53 +0300, Andrei Popa wrote: > > On Wed, 2007-05-02 at 12:16 +0200, Takashi Iwai wrote: > > At Sun, 29 Apr 2007 05:19:50 +0300, > > Andrei Popa wrote: > > > > > > On Sun, 2007-04-29 at 04:43 +0300, Andrei Popa wrote: > > > > Hi, > > > > > > > > The sound works ok with the built-in speakers on my Fujitsu laptop(amilo > > > > pro v3205). But if I plug in headphones in the audio out jack there is > > > > no sound in the headphones. With kernel 2.6.20 it worked ok. With 2.6.21 > > > > it doesn't. > > > > This BUG is reported in ALSA bugtracking system with ID 0003039. > > > > > > > > Patch that fixes this bug: > > > > > > > > diff --git a/sound/pci/hda/patch_conexant.c > > > > b/sound/pci/hda/patch_conexant.c > > > > index 46e93c6..5edd330 100644 > > > > --- a/sound/pci/hda/patch_conexant.c > > > > +++ b/sound/pci/hda/patch_conexant.c > > > > @@ -636,7 +636,7 @@ static void cxt5045_hp_automute(struct hda_codec > > > > *codec) > > > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > > snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > } > > > > > > > > /* unsolicited event for HP jack sensing */ > > > > @@ -988,10 +988,10 @@ static void cxt5047_hp_automute(struct hda_codec > > > > *codec) > > > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > > snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > /* Mute/Unmute PCM 2 for good measure - some systems need this > > > > */ > > > > snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > } > > > > > > > > /* toggle input of built-in and mic jack appropriately */ > > > > > > > > > > first patch was broken. it only worked for the right channel. > > > this fixes this: > > > > > > diff --git a/sound/pci/hda/patch_conexant.c > > > b/sound/pci/hda/patch_conexant.c > > > index 46e93c6..0822df7 100644 > > > --- a/sound/pci/hda/patch_conexant.c > > > +++ b/sound/pci/hda/patch_conexant.c > > > @@ -635,8 +635,8 @@ static void cxt5045_hp_automute(struct hda_codec > > > *codec) > > > AC_VERB_GET_PIN_SENSE, 0) & > > > 0x80000000; > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > - snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > } > > > > > > /* unsolicited event for HP jack sensing */ > > > @@ -987,11 +987,11 @@ static void cxt5047_hp_automute(struct hda_codec > > > *codec) > > > AC_VERB_GET_PIN_SENSE, 0) & > > > 0x80000000; > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > - snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > /* Mute/Unmute PCM 2 for good measure - some systems need this > > > */ > > > - snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > } > > > > > > /* toggle input of built-in and mic jack appropriately */ > > > > But this means to _unmute_ the speaker (0x1d) when you plug HP. > > So, something else should be wrong. Is hp_present detected properly > > per plugging or unplugging? > > > > when I debugged the problem I think hp_present was "0" when the HP were > plugged in and "-28953535"(a big negative number) when the HP were > unplugged. Weird, then this returns exactly the inversed bit. Could you double-check? Anyway, having both this auto-mute and manual switches is confusing. If they exist, we'd need to add notifier to the control at unsol event handler. Takashi

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[alsa-devel] Intel ICH6 with AD1986: Mic not working
by Joerg Platte 02 May '07

02 May '07

Hi! On my computer both microphones are not working but playback and all other audio stuff seems to work. I'm not sure if the microphones ever worked with alsa because I never tried it before. On windows they are working as expected, hence the hardware seems to be OK. Is there anything I can do to further debug this? Is it a known problem? Below you can find more information about the hardware: root@ds19:~> uname -a Linux ds19 2.6.20-1-686 #1 SMP Tue Apr 24 21:52:11 UTC 2007 i686 GNU/Linux (Debian kernel) Doesn't work with 2.6.18 as well. root@ds19:~> cat /proc/asound/card0/codec97#0/ac97#0-0 0-0/0: Analog Devices AD1986 PCI Subsys Vendor: 0x1734 PCI Subsys Device: 0x105a Revision : 0x02 Compat. Class : 0x00 Subsys. Vendor ID: 0x0000 Subsys. ID : 0x0000 Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 16-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : +20dB [+20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Double rate slots: 10/11 Extended ID : codec=0 rev=2 AMAP LDAC SDAC CDAC DSA=0 SPDIF DRA VRA Extended status : SPCV LDAC SDAC CDAC SPDIF=10/11 VRA PCM front DAC : 48000Hz PCM Surr DAC : 48000Hz PCM LFE DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Copyright Category=0x22 Generation=1 Rate=48kHz Gain Inverted Buffer delay Location Master Out : 0.0 dBV - 16/fs Rear I/O Panel AUX Out : 0.0 dBV - 16/fs Rear I/O Panel Center/LFE Out : 0.0 dBV - 16/fs Rear I/O Panel Mic 1 : 0.0 dBV - 16/fs Rear I/O Panel Mic 2 : 0.0 dBV - 16/fs Rear I/O Panel Line In : 0.0 dBV - 16/fs Rear I/O Panel AD18XX configuration Unchained : 0x1000,0x0000,0x0000 Chained : 0x0000,0x0000,0x0000 root@ds19:~> cat /proc/asound/card0/codec97#0/ac97#0-0+regs 0:00 = 0090 0:02 = 0404 0:04 = 0606 0:06 = 0006 0:08 = 0000 0:0a = 0000 0:0c = 801f 0:0e = 0040 0:10 = 9f9f 0:12 = 0606 0:14 = 0000 0:16 = 9f9f 0:18 = 0606 0:1a = 0000 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0bc7 0:2a = 05f1 0:2c = bb80 0:2e = bb80 0:30 = bb80 0:32 = bb80 0:34 = 0000 0:36 = 0300 0:38 = 0202 0:3a = 2a20 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 1000 0:5e = 0000 0:60 = 8080 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 8080 0:6e = 8080 0:70 = 0000 0:72 = 0000 0:74 = 1001 0:76 = 7c7c 0:78 = 0000 0:7a = 0000 0:7c = 4144 0:7e = 5378 root@ds19:~> cat /proc/asound/cards 0 [ICH6 ]: ICH4 - Intel ICH6 Intel ICH6 with AD1986 at 0xf0140800, irq 16 root@ds19:~> lspci 00:00.0 Host bridge: Intel Corporation 82915G/P/GV/GL/PL/910GL Memory Controller Hub (rev 04) 00:02.0 VGA compatible controller: Intel Corporation 82915G/GV/910GL Integrated Graphics Controller (rev 04) 00:02.1 Display controller: Intel Corporation 82915G Integrated Graphics Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 1 (rev 03) 00:1c.1 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 2 (rev 03) 00:1c.2 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 3 (rev 03) 00:1c.3 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 4 (rev 03) 00:1d.0 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #1 (rev 03) 00:1d.1 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #2 (rev 03) 00:1d.2 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #3 (rev 03) 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) 00:1d.7 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB2 EHCI Controller (rev 03) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev d3) 00:1e.2 Multimedia audio controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) AC'97 Audio Controller (rev 03) 00:1f.0 ISA bridge: Intel Corporation 82801FB/FR (ICH6/ICH6R) LPC Interface Bridge (rev 03) 00:1f.1 IDE interface: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) IDE Controller (rev 03) 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) 00:1f.3 SMBus: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) SMBus Controller (rev 03) 07:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) root@ds19:~> lsmod | grep snd snd_intel8x0 31420 1 snd_ac97_codec 89412 1 snd_intel8x0 ac97_bus 2432 1 snd_ac97_codec snd_pcm_oss 39232 0 snd_mixer_oss 15488 1 snd_pcm_oss snd_pcm 72036 3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss snd_timer 20964 1 snd_pcm snd 48196 8 snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 7616 1 snd snd_page_alloc 9736 2 snd_intel8x0,snd_pcm regards, Jörg Grüße, Jörg -- PGP Key: send mail with subject 'SEND PGP-KEY' PGP Key-ID: FD 4E 21 1D PGP Fingerprint: 388A872AFC5649D3 BCEC65778BE0C605

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[alsa-devel] 2.6.21: known regressions with patches (for -stable team)
by Michal Piotrowski 02 May '07

02 May '07

Hi all, Here is a list of known regressions with patches available. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Subject : 2.6.21: longhaul freq-scaling broken References : http://lkml.org/lkml/2007/5/1/260 Submitter : Jan Kiszka <jan.kiszka(a)web.de> Patch : http://git.kernel.org/?p=linux/kernel/git/davej/cpufreq.git;a=commitdiff_pl… Status : patch available Subject : Kernel hang at "kernel: switched to high resolution mode on cpu 0" when either/both of CONFIG_NO_HZ, CONFIG_HIGH_RES_TIMERS is used References : http://lkml.org/lkml/2007/4/30/464 Submitter : Mark Lord <lkml(a)rtr.ca> Handled-By : Daniel Walker <dwalker(a)mvista.com> Thomas Gleixner <tglx(a)linutronix.de> Patch : http://lkml.org/lkml/2007/4/13/190 Status : patch was suggested Subject : ACPI related kernel panic References : http://lkml.org/lkml/2007/4/30/193 http://bugzilla.kernel.org/show_bug.cgi?id=8385 Submitter : Sebastian Kemper <sebastian_ml(a)gmx.net> Patches : http://bugzilla.kernel.org/attachment.cgi?id=11331&action=view http://bugzilla.kernel.org/attachment.cgi?id=11332&action=view Status : patches was suggested Subject : sis900 oops in 2.6.21.1 after bringing up the network References : http://lkml.org/lkml/2007/4/29/138 http://lkml.org/lkml/2007/4/29/151 Patch : http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Submitter : Brett Ryland <nappers(a)gmail.com> Status : patch available Subject : reiserfs -- circular locking References : http://lkml.org/lkml/2007/4/26/648 Submitter : David Brownell < david-b(a)pacbell.net > Handled-By : Jeff Mahoney < jeffm(a)suse.com > Patch : http://lkml.org/lkml/2007/4/27/332 Status : patch available Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger < markus.rechberger(a)amd.com > Status : patches available Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski < michal.k.k.piotrowski(a)gmail.com > Handled-By : Takashi Iwai < tiwai(a)suse.de > Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available Subject : qla2xxx driver no longer works References : http://lkml.org/lkml/2007/4/16/49 http://lkml.org/lkml/2007/4/20/224 Submitter : David Miller < davem(a)davemloft.net > Robert Peterson < rpeterso(a)redhat.com > Handled-By : David Miller < davem(a)davemloft.net > Patch : http://lkml.org/lkml/2007/4/16/234 Status : patch was suggested Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] [PATCH] snd-usb-audio: explicitly match Logitech QuickCam
by Daniel Drake 02 May '07

02 May '07

Commit 93c8bf45e083b89dffe3a708363c15c1b220c723 modified the USB device matching behaviour to ignore interface class matches if the device class is vendor-specific. This patch adds an explicit ID match for the Logitech QuickCam Messenger, which has a vendor specific device class (but standards-compliant audio interfaces). This fixes a 2.6.20 regression where the audio component of this device was no longer usable. http://bugs.gentoo.org/show_bug.cgi?id=175715 https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/93822 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3040 Based on a patch from sergiom Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -40,6 +40,21 @@ .bInterfaceClass = USB_CLASS_VENDOR_SPEC /* + * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface + * class matches do not take effect without an explicit ID match. + */ + +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x08f0, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, + +/* * Yamaha devices */

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[alsa-devel] ALC260 patch for Packard Bell Easy Note v7900
by Andrea Gozzi 02 May '07

02 May '07

Please see if you can accept this patch that fixes sound/headphones/mic on Packard Bell EN v7900 notebook (ALC260 chipset). The code is not mine, I diff'ed using pshou's (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2189) but since it works perfectly it would be nice to see it in the next alsa release. Andrea

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[alsa-devel] [ALSA][hda-codec] fix-connexant-headphone-mute
by Andrei Popa 02 May '07

02 May '07

Hi, The sound works ok with the built-in speakers on my Fujitsu laptop(amilo pro v3205). But if I plug in headphones in the audio out jack there is no sound in the headphones. With kernel 2.6.20 it worked ok. With 2.6.21 it doesn't. This BUG is reported in ALSA bugtracking system with ID 0003039. Patch that fixes this bug: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 46e93c6..5edd330 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -636,7 +636,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, bits); } /* unsolicited event for HP jack sensing */ @@ -988,10 +988,10 @@ static void cxt5047_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, bits); /* Mute/Unmute PCM 2 for good measure - some systems need this */ snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, bits); } /* toggle input of built-in and mic jack appropriately */

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Giridhar Pemmasani 02 May '07

02 May '07

--- Adrian Bunk <bunk(a)stusta.de> wrote: > On Sun, Apr 29, 2007 at 09:55:18AM -0700, Giridhar Pemmasani wrote: > > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails > to > > boot - it seems to stop at various points depending on what other modules > are > > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with > everything > > else being same, using maestro3.c from 2.6.20 kernel and using it in > 2.6.21 > > kernel also works - i.e., system boots fine. I will try to isolate the > issue > > and follow up. > > There were only the two attached patches between 2.6.20 and 2.6.21 to > this driver: > - maestro3: add request_firmware() > - maestro3 - Use quirk list helper function > > 2.6.21 + 2.6.20 maestro3.c works for you. > > Does 2.6.21 + 2.6.20 maestro3.c + patch-maestro3-1 work? 2.6.21 with 2.6.20's maestro3.c with patch-maestro3-1 exhibited the same issue. However, I identified the cause: maestro3 driver in 2.6.21 (even just patch-maestro3-1) tries to load firmware file from user space. But each time request_firmware is called it takes about 1 minute to timeout (because I don't have the firmware installed?). If I let the system take its time, even with maestro3.c from 2.6.21, it boots after timing out twice. So after about 2 minutes of no activity, system does boot. Following are trace messages I added to identify the issue: [ 17.004000] input: TPPS/2 IBM TrackPoint as /class/input/input5 [ 70.904000] sound/pci/maestro3.c (snd_m3_create:2763): -2 [ 70.920000] sound/pci/maestro3.c (snd_m3_create:2774): -2 [ 130.940000] sound/pci/maestro3.c (snd_m3_create:2777): -2 [ 130.956000] sound/pci/maestro3.c (snd_m3_create:2788): -2 [ 130.972000] sound/pci/maestro3.c (snd_m3_create:2794): 0 [ 130.992000] sound/pci/maestro3.c (snd_m3_create:2797): 0 [ 131.012000] sound/pci/maestro3.c (snd_m3_create:2802): 0 [ 131.044000] sound/pci/maestro3.c (snd_m3_create:2804): 0 [ 131.580000] sound/pci/maestro3.c (snd_m3_create:2807): 0 [ 131.596000] sound/pci/maestro3.c (snd_m3_create:2810): 0 [ 131.612000] sound/pci/maestro3.c (snd_m3_create:2814): 0 [ 131.628000] sound/pci/maestro3.c (snd_m3_create:2821): 0 [ 132.364000] device-mapper: ioctl: 4.11.0-ioctl (2006-10-12) initialised: dm-d evel(a)redhat.com Giri __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Audio DMA buffer layout and setup
by Markus Korber 02 May '07

02 May '07

Hi, I'm working with a custom sound chip with its own DMA hardware. The figure shows how the I2S DMA expects data to be prepared: IRQ1 +----------------+ | page_3 (right) |\ +----------------+ > double buffer 1 | page_2 (left) |/ IRQ0 +----------------+ | page_1 (right) |\ +----------------+ > double buffer 0 | page_0 (left) |/ +----------------+ The hardware has 2 double buffers, each containing left and right sample regions, i.e. it alternately transmits samples from page_0 and page_1, i.e. double buffer 0, then generates an interrupt and starts transmitting page_2 and page_3. After that another interrupt is generated and the process begins again with double buffer 0. So my question is, how to tell ALSA about this memory layout? Is it possible to operate in MMAP mode with SNDRV_PCM_INFO_NONINTERLEAVED? But then, how to tell the application that it should either fill double buffer 0 or 1? Or should I use RW mode? Or something else? Furthermore, since the hardware doesn't provide a 'current' DMA pointer containing the DMA address being transmitted at the moment (I only have interrupts generated after each double buffer has been transmitted), what should the pcm_pointer() callback return? (I guess it should return 0 when called before IRQ0 and sizeof(double buffer 0) between IRQ0 and IRQ1?) Best regards, Markus Korber

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[alsa-devel] driver mixer control questions
by Tim Harvey 02 May '07

02 May '07

I'm troubleshooting an issue with an ALSA driver I wrote for a very unique hardware platform. In this case I have an audio codec which only has a usable 'line-in' capture source. Currently the only mixer control I've created is a 'Master Playback Volume'. Do I need to create some kind of control for 'line-in'? I'm unclear how to 'register' capture inputs so that they will be returned by the OSS mixer emulation's ioctl SOUND_MIXER_READ_RECMASK. Currently that ioctl returns 0 for my mixer device. Also, is a control 'switch' such as 'Master Playback Switch' simply a mute control or is it intended to be any type of control which could have enumerated values? Thanks, Tim

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Re: [alsa-devel] Support for SAM9407?
by Ash Willis 02 May '07

02 May '07

Thanks for your detailed response. That chip definitely does seem like a beast that I really don't want to tame... > > How about a Media Vision Pro Audio Spectrum 16? Those can still be found > relatively easily and it's one the cards for which an OSS/Free driver is > still kept in the tree. I have a pair of those as well so I'll happily test > if you want to write a driver for it... > Yeah, we have a winner. I just ordered one. I'm busy right now but I will get this started in 4 weeks. Thanks for the offer of testing. I'll hunt you down when I have something to show you :) Ash =

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[alsa-devel] porter wanted: AMD Geode SCx200 driver OSS to ALSA
by Martin-Éric Racine 02 May '07

02 May '07

Greetings, I work for a company that develops hardware using the AMD Geode chipset. While our current product line is based on the Geode LX model (which is very well supported in ALSA), our legacy products are based on the earlier Geode SC1200 and SC2200, for which no ALSA support exists. However, AMD has OSS driver source code available at: http://www.amd.com/us-en/ConnectivitySolutions/TechnicalResources/0,,50_233… We basically need this ported to an ALSA snd-scx200 module to replace it. (Would now also be a good time to ask for snd-sc5535audio to be renamed snd-sc5535 for consistence with other driver names?) We could provide a sample SC2200-based unit to whoever volunteers for porting AMD's old OSS code to ALSA, Please contact me for details. The resulting code will be directly submitted to the ALSA project for inclusion in the mainstream kernel, to make sure that everyone can benefit from it. Best Regards, -- Martin-Éric Racine http://q-funk.iki.fi

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Re: [alsa-devel] Endiannes and alsa (in an embedded device)
by Ciaccia 01 May '07

01 May '07

Forget about that, it took 4 hours but I understood why it was not working... Since I'm working with 24bit resolution (and not 16 as I usually do), the samples were to little... Now I replaced the sin line with: sample = (int)(4000000.0f * sin((float)i * 2.0f * M_PI / (512.0f / 50.0f))); and it's working fine. I really thought it was too strange to be true... Sorry for the spam :-| Andrea ----- Original Message ---- From: Ciaccia <ciacciax(a)yahoo.com> To: alsa-devel(a)alsa-project.org Sent: Tuesday, May 1, 2007 8:49:18 PM Subject: [alsa-devel] Endiannes and alsa (in an embedded device) Dear all, I have a question that really puzzles me, I wonder if someone can explain to me what I'm doing wrong... :-| I have programmed an ALSA driver for an embedded device, everything works fine and all the application use my driver without any glitches. The snd_pcm_hardware for this device is defined as following: static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info= SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .formats = SNDRV_PCM_FMTBIT_S24_LE, .buffer_bytes_max = 131072, .period_bytes_min = 512, .period_bytes_max = 32768, .periods_min = 1, .periods_max = MAX_PERIODS, .fifo_size = 32, }; Then I tried to program a test application that uses the hardware device directly (without using alsa-lib to convert the format and so on). I open the device as following: if ((err = snd_pcm_open (&playback_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0)) < 0) { Everything works fine until I try to write data to the stream. I tried both SND_PCM_ACCESS_RW_INTERLEAVED and SND_PCM_ACCESS_MMAP_INTERLEAVED, but I always have the following problem. When I want to write data to the stream, I have to "reverse" the byte order, otherwise I hear noise on the speaker. The system I am working on is little-endian (it has cirrus ep93xx arm core), so I assume I could write data directly to the buffer. Here is what I have to do: // fill the buffer for (i = 0; i < 512; i++) { int sample = (int)(20000.0f * sin((float)i * 2.0f * M_PI / 512.0f)); int reversed; char *src = (char *)(&sample); char *dst = (char *)(&reversed); dst[0] = src[3]; dst[1] = src[2]; dst[2] = src[1]; dst[3] = src[0]; *(buffer++) = reversed; *(buffer++) = reversed; } It this normal? I thought that since the system is little-endian and the format of the driver is S24_LE, I could copy the data directly to the stream (either using snd_pcm_writei or snd_pcm_mmap_begin). I think the driver is ok, since all the applications run with no problem at all. Can someone explain to me how/why the applications such as aplay/arecord work ok? Every help is greatly appreciated Thanks in advance Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Endiannes and alsa (in an embedded device)
by Ciaccia 01 May '07

01 May '07

Dear all, I have a question that really puzzles me, I wonder if someone can explain to me what I'm doing wrong... :-| I have programmed an ALSA driver for an embedded device, everything works fine and all the application use my driver without any glitches. The snd_pcm_hardware for this device is defined as following: static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info= SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .formats = SNDRV_PCM_FMTBIT_S24_LE, .buffer_bytes_max = 131072, .period_bytes_min = 512, .period_bytes_max = 32768, .periods_min = 1, .periods_max = MAX_PERIODS, .fifo_size = 32, }; Then I tried to program a test application that uses the hardware device directly (without using alsa-lib to convert the format and so on). I open the device as following: if ((err = snd_pcm_open (&playback_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0)) < 0) { Everything works fine until I try to write data to the stream. I tried both SND_PCM_ACCESS_RW_INTERLEAVED and SND_PCM_ACCESS_MMAP_INTERLEAVED, but I always have the following problem. When I want to write data to the stream, I have to "reverse" the byte order, otherwise I hear noise on the speaker. The system I am working on is little-endian (it has cirrus ep93xx arm core), so I assume I could write data directly to the buffer. Here is what I have to do: // fill the buffer for (i = 0; i < 512; i++) { int sample = (int)(20000.0f * sin((float)i * 2.0f * M_PI / 512.0f)); int reversed; char *src = (char *)(&sample); char *dst = (char *)(&reversed); dst[0] = src[3]; dst[1] = src[2]; dst[2] = src[1]; dst[3] = src[0]; *(buffer++) = reversed; *(buffer++) = reversed; } It this normal? I thought that since the system is little-endian and the format of the driver is S24_LE, I could copy the data directly to the stream (either using snd_pcm_writei or snd_pcm_mmap_begin). I think the driver is ok, since all the applications run with no problem at all. Can someone explain to me how/why the applications such as aplay/arecord work ok? Every help is greatly appreciated Thanks in advance Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Implementing Priority Based Device ALSA device interface. ALSA drivers
by Nobin Mathew 01 May '07

01 May '07

Hi Friends, I am a new to ALSA, I am developing a ALSA ASoC driver. Now driver can support both audio playback and capture. Now the problem is handling of application with different priority. We have three applications say app1, app2 and app3; they are in increasing order of priority. One scenario of app1 and app2 is below If app2 is accessing the device while app1 playback is going on, then app1 should stop (clearly do pause) playback and allow app2 to playback, and app1 should continue playback when app2 exit. The requirement is that there should be no changes to apps, except the devices each app will access. There will be no daemons in user space which will handle priority. For that i implemented three interfaces like pcmC0p0, pcmC0p1 and pcmC0p2 there are in increaing order of priority. All these interfaces will access the same audio hardware. Only thing is each will have separate ALSA kernel buffer. In this implementation app1 will always access pcmC0p0, and app2 pcmC0p1 like that. What i am doing is, in driver i will pause the pcmC0p0 substream using snd_pcm_pause() when pcmC0p1 is accessed. Same for pcmc0p2. This call will change the state of pcmc0p0 substream to SNDRV_PCM_STATE_PAUSED, and this call should notify the user app also. When app2 playback is over i will change the state of pcmC0p0 substream to SNDRV_PCM_STATE_RUNNING by another call to snd_pcm_pause(). This implementation requires some minor modifications in the ALSA lib and ALSA app (handling of PAUSE from kernel). But what i found is kernel is not notifying the user using snd_timer_notify() (user call back) function because i am using the substream timer snd_pcm_timer which is there in pcm_timer.c. For this timer SNDRV_TIMER_HW_SLAVE flag is set so it will not notify the user. What is the use of these timers in ALSA? Am i going wrong? Is my design correct? Can u suggest some alternate design? Sorry for my poor English. Thanks Nobin Mathew

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[alsa-devel] Set fragment size for USB sound card
by Jason Heeris 01 May '07

01 May '07

System: NEC Versa M540 laptop Dist: Debian Etch/Lenny Kernel: 2.6.18 Alsa: using drivers in kernel, although no difference when compiled from alsa-source package. Card in question: C-Media USB sound card (more info below) Hi there, I realise that this may not be strictly a developer list question, but someone on the alsa-user list suggested I try here. Any help would be appreciated. The question is a continuation of: http://sourceforge.net/mailarchive/forum.php?thread_name=1176015858.4497.9.… and http://www.linuxquestions.org/questions/showthread.php?t=540205 Essentially, I have a C-Media USB audio adaptor with which to use my headphones, and it only works with a fragment size of 2048 or less (discovered using 'alsaplayer -f 2048'). I would like to be able to plug it in, have ALSA recognise it and use this fragment size for it. Without specifying this fragment size, I get: -- $ alsaplayer -i text -o alsa -d hw:2 /usr/share/sounds/generic.wav Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (23219 23220) PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: 8 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 1000 failed to configure output device...trying OSS ...done playing -- ...and the sound comes through the laptop speakers. I tried defining a slave PCM: -- pcm.usbtest { type dmix ipc_key 192837465 ipc_key_add_uid true slave { pcm "hw:2,0" # period_size 2048 buffer_size 2048 } } -- But trying 'alsaplayer -i text -o alsa -d usbtest /usr/share/sounds/generic.wav' gave exactly the same error. >From my previous post: -- ...I also tried forcing the index of the card by adding the following to /etc/modprobe.d/alsa-base, but even this only works half of the time: -- # Prevent abnormal drivers from grabbing index 0 options snd-bt87x index=-2 options snd-cx88_alsa index=-2 options snd-atiixp-modem index=-2 # options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-intel8x0 index=0 options snd-intel8x0m index=1 options snd-usb-audio index=2 vid=0d8c pid=000c options snd-usb-audio index=3 -- The contents of '/proc/asound/cards' are: -- 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with ALC655 at 0xf0000400, irq 5 1 [Modem ]: ICH-MODEM - Intel 82801DB-ICH4 Modem Intel 82801DB-ICH4 Modem at 0xe200, irq 5 2 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-0000:00:1d.1-1, full speed -- ...although sometimes the ICH-MODEM is missing and the C-Media card is at #1. Card information from 'lsusb': Bus 002 Device 003: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter Usually I boot with the card plugged in, but if I plug it in after booting, the new modules loaded are: snd_hwdep, snd_rawmidi, snd_usb_audio and snd_usb_lib. -- Anyway, as I said, any help would be appreciated. Cheers, Jason

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[alsa-devel] Possible bug in snd_mixer_selem_set_playback_switch_all
by Tim Burrell 30 Apr '07

30 Apr '07

Hi there, I've encountered a situation where if I very quickly, and repeatedly make calls to snd_mixer_selem_set_playback_switch_all I get some strange undefined behavior. The situation is I'm looping over all mixer elements (on all sound cards) to do a system wide mute / unmute. If I make numerous calls to this function it ends up that some mixers only toggle switches on specific channels (right or left), and sometimes the volume on certain (random) channels gets set to some also random (usually 0, but can be anywhere between 0 and max) level. I've been able to remedy the situation by inserting a 10ms delay in between calls to snd_mixer_selem_set_playback_switch_all. Clearly though this is undesirable. I don't have more than one sound card to test on so I can't be sure if it's a problem with all drivers or just mine (hda-intel). I can certainly provide more information if requested, just let me know what I can do. Or if I'm doing something wrong I'd love to be made aware of something I can do to fix this behavior. Thanks, Tim.

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[alsa-devel] List of all Apple Mac's hda-intel pin configurations
by Ivan N. Zlatev 30 Apr '07

30 Apr '07

I have attached all windows drivers' .ini files (taken from latest windows drivers cd of Boot Camp), which include all pin configuration as well as probably some more useful information (GPIO and stuff). I hope this will help you to make a nice update. [Subsystem ID - configuration] 106B0800 - Board1.ini 106B0600 - Board2.ini 106B0700 - Board2.ini 106B0E00 - Board3.ini 106B0F00 - Board3.ini 106B1600 - Board3.ini 106B1700 - Board3.ini 106B0200 - Board3.ini 106B1E00 - Board3.ini 106B0400 - Board4.ini 106B1A00 - Board4.ini 106B0A00 - Board5.ini 106B2200 - Board5.ini 00000100 - Board4.ini Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work? [Solved]
by Cody Jung 30 Apr '07

30 Apr '07

On 4/29/07, Rene Herman <rene.herman(a)gmail.com> wrote: > On 04/29/2007 09:31 AM, Rene Herman wrote: > > > I've attached (to try and avoid linewrap) the lines that you will need in > > /etc/modprobe.conf. > > No I haven't. > > Rene. > > alias snd-card-0 snd-cs4231 > alias sound-slot-0 snd-cs4231 > > options snd-cs4231 index=0 port=0x534 irq=5 dma1=1 dma2=0 > > install snd-cs4231 echo activate >/sys/bus/pnp/devices/01\:01.00/resources &&\ > sleep 1 &&\ > /sbin/modprobe --ignore-install snd-cs4231 > > remove snd-cs4231 /sbin/modprobe -r --ignore-remove snd-cs4231 &&\ > echo disable >/sys/bus/pnp/devices/01\:01.00/resources > > Wahoo! Thank you, thank you! I ended up not going the /etc/modules way, partially because it wasn't working and partially because I didn't know what I was doing. Instead I added the following 4 lines to a file and included it in my init scripts. echo activate > /sys/bus/pnp/devices/01\:01.00/resources sleep 1 modprobe snd modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 And it works great! Thanks again for putting up with my incompetence. ~Cody Jung

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Michal Piotrowski 29 Apr '07

29 Apr '07

Hi, On 29/04/07, Giridhar Pemmasani <pgiri(a)yahoo.com> wrote: > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails to > boot - it seems to stop at various points depending on what other modules are > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with everything > else being same, using maestro3.c from 2.6.20 kernel and using it in 2.6.21 > kernel also works - i.e., system boots fine. I will try to isolate the issue > and follow up. Please revert this patches http://www.stardust.webpages.pl/files/patches/alsa_revert/ and test kernel after each revert. Takashi, I'll add this to known regressions for -stable team. > > Giri Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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