May 2007 - Alsa-devel - mailman.alsa-project.org

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] 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 30 May '07

30 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 30 May '07

30 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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