Re: [Alsa-user] Fedora core 11. No sound
Dave Pawson wrote: 2009/7/24 Clemens Ladisch cladi...@googlemail.com: Does the onboard HDA controller show up in the output of lspci? # lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller 01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 4000 Series] Both should have been picked up by the driver. What are the contents of /sys/module/snd_hda_intel/parameters/enable? Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Fedora core 11. No sound
Dave Pawson wrote: 2009/7/24 Clemens Ladisch cladi...@googlemail.com: Dave Pawson wrote: # lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller 01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 4000 Series] Both should have been picked up by the driver. What are the contents of /sys/module/snd_hda_intel/parameters/enable? Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y Is there any error message in the system log when you do this: rmmod snd-hda-intel modprobe snd-hda-intel dmesg | tail Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Fedora core 11. No sound
Dave Pawson wrote: 2009/7/24 Clemens Ladisch cladi...@googlemail.com: Is there any error message in the system log when you do this: rmmod snd-hda-intel $ rmmod snd-hda-intel ERROR: Module snd_hda_intel is in use Please kill all applications that use it, then try again. (I guess you could try to temporatily disable PulseAudio.) Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Strange knocking while playing sound
Dennis Borgmann wrote: I am working with snd_pcm_mmap_begin and snd_pcm_mmap_commit - just the way it is done in the pcm.c under the test-directory of the alsa-lib. While working with snd_pcm_writei(), those problems did not occur, but I want to test this using the mmap technology. This sounds like a problem with your mmap code. As long as you are copying samples from some buffer into the sound card's buffer, using mmap does not give you any benefit because this is exactly the same as snd_pcm_writei() does. Best regards, Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Terratec Phase28
Christoph Köditz wrote: is the driver in work or do you need some more informations about this soundcard? According to the driver's source, this card has been supported since 2005. HTH Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] mmap_readi or mmap_begin
Marek Michalak wrote:
Is there any difference if I write in my code:
while (size 0) {
frames = size;
err = snd_pcm_mmap_begin(handle, my_areas, offset, frames);
record_buffor(my_areas, offset, frames, data-phase);
snd_pcm_mmap_commit(handle, offset, frames);
size -= frames;
}
OR
while (rest 0) {
size_t c = (rest = (off64_t)chunk_bytes) ? (size_t)rest :
chunk_bytes;
size_t f = c * 8 / bits_per_frame;
snd_pcm_mmap_readi(audiobuf, f);
write(fd, audiobuf, c);
rest -= c;
}
This depends on what the function record_buffor() does.
snd_pcm_readi() (or the equivalent snd_pcm_mmap_readi() from your second
example) copies data from the sound card's buffer into the user's buffer.
If record_buffor() copies the data elsewhere, too, then there would be
_no_ difference in what the code does.
Using a memory-mapped buffer makes sense only if you actually use the
data in the buffer instead of copying it, i.e., if you make your
computations directly on the data as it is in the buffer.
BTW: Using snd_pcm_mmap_readi() doesn't make much sense if don't also
use snd_pcm_mmap_begin() in your program, because snd_pcm_readi() does
the same but does not require a device that supports mmap.
I want to have low latency capture and compute fft - is it good way to do
this?
If your FFT algorithm
* can read the data directly from the sound card buffer,
* does not write to the sound card buffer, and
* can handle the period/buffer layout (which is hardware dependent; not
all sound cards may support your desired period/buffers sizes),
then you can use snd_pcm_mmap_*.
Otherwise, you must use snd_pcm_readi().
Assuming that the FFT needs floating-point numbers, the int-to-float
conversion would be an algorithm that could be easily made to work with
mmap. Or you could write the first step of the FFT so that it reads
integer data from the mmap buffer, converts it to floats, does the
computation, and writes the results to its own processing buffer.
BTW: When capturing, latency depends only on the period size, not on the
buffer size, so you should use as big a buffer as possible to avoid
overruns.
Is true that snd_async_handler_t *ahandler is better for realtime system
than nonblocking mode?
No. Asynchronous handlers notify your program with a signal, but you
are not allowed to do anything useful inside the signal handler, so
you'd have to notify your own thread somehow. Using blocking reads or
poll() is more efficient.
Best regards,
Clemens
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Re: [Alsa-user] Fwd: KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!
Matt Snow wrote: On Fri, Jul 10, 2009 at 2:20 AM, Clemens Ladischcladi...@googlemail.com wrote: So the headphones or speakers in the front jack work, and the to-RCA cable in the *same* jack does not? This is obviously a broken cable (or TV). You might think that, but its fine. I take the exact same hdmi+dvi adaptor, and the rca-to-stero cable, plug in to either my macbook pro, or macbook with mini displayport adaptor to hdmi and I get audio and video. What happens when you connect the MacBook to the TV with the HDMI+DVI adapter and, at the same time, the Xonar DX to the TV with the RCA-to- stereo cable? Best regards, Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Fwd: KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!
Matt Snow wrote: On Thu, Jul 9, 2009 at 12:05 AM, Clemens Ladischcladi...@googlemail.com wrote: Matt Snow wrote: After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out plays audio perfectly through head phones or powered speakers, but the 3.5mm stereo-to-RCA cable going to RCA input on the TV doesn't produce any sound at all. Isn't that the same jack on the card? Or where have you connected the headphones, the speakers and the TV? there are several analog jacks on the card for different channels. i'm using the front speaker jack, but have tried all jacks with the TV and ear phones. So the headphones or speakers in the front jack work, and the to-RCA cable in the *same* jack does not? This is obviously a broken cable (or TV). Best regards, Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!
Matt Snow wrote: I have a Asus Xonar DX sound card that was working perfectly in KnoppMyth/LinHES R5.5(Knoppix/debian based) after manually compiling alsa-driver 1.0.18b and 1.0.18 libs/utils/etc. After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out plays audio perfectly through head phones or powered speakers, but the 3.5mm stereo-to-RCA cable going to RCA input on the TV doesn't produce any sound at all. Isn't that the same jack on the card? Or where have you connected the headphones, the speakers and the TV? What version of ALSA are you using? (see /proc/asound/version) Best regards, Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Mic not working with arecord on RedHat el5
Saleem Hasan wrote: I found that although aplay was working on my computer, arecord was not. I played back the wav files provided with the linux distribution for various system beeps and sounds. I am able to navigate through the alsamixer screen using tab, space, etc in order to select the mic, capture, mic boost, master and other settings but arecord still does not record. I have used the alsamixer -vv and the vu meter either stays at 0% or varies between 0-5% and the file when played back is only static. Try disabling Headphone Jack Sense. HTH Clemens -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Edirol UA-101: Does record work?
Phil wrote: I'm trying to find a way to get audio in over USB for a project, and I think that the UA-101 might be about the only Linux device that supports 6 or more channels of audio over USB. Indeed. There are other multichannel USB devices, but there are none that have even rudimentary Linux support. As I understand from the kernel comments, there are glitches on playback but recording works, does this mean all 10 channels work on record? From what I've heard , recording works perfectly. If record does work, then that would probably be enough to buy the thing, and I'll sort out playback separately. Somebody is currently trying to write a driver that would make playback work, too. Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't independently adjust Mic levels
scar wrote: Clemens Ladisch @ 06/26/2009 03:30 AM: The hardware does not have any mechanism to change the line input's level. You have to adjust the volume after recording. shown in section 6.3.2 of the xonar dx manual is a picture of the levels for recording, indicating there is an adjustment for line-in. The Windows driver does this in software. also throughout section 6, i am noticing a bunch of other options for sound/speaker settings that i am not sure how to take advantage of in linux or with ALSA. can someone explain? All those effect and encoders are implemented in software, too. Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't independently adjust Mic levels
scar wrote: i have been experimenting, and what appears to be happening is this: while recording using the line, if the microphone levels are adjusted, the driver/card automatically jumps to recording from the mic. Neither the driver nor the card do this; I'd guess this is a feature of your mixer application. still remaining is the issue of not being able to adjust the left and right line volumes. it would be nice to have that extra functionality. The hardware does not have any mechanism to change the line input's level. You have to adjust the volume after recording. Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't independently adjust Mic levels
scar wrote: Clemens Ladisch @ 06/24/2009 12:53 AM: The microphone input is mono. There are no left/right levels; the driver pretending that they exist is a driver bug. I'll fix that. but that port is also the line-in, so i need stereo When used as line in, that port is stereo. (It then also expects line level and does not allow to change the volume.) Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] can't independently adjust Mic levels
scar wrote: so, i got my asus xonar dx ... the problem i am running into is i cannot independently adjust the left and right levels of the Mic capture. well, i can, kind of, but it doesn't have any effect. The microphone input is mono. There are no left/right levels; the driver pretending that they exist is a driver bug. I'll fix that. HTH Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Known C-Media 8738 issues?
Malte Gell wrote: Clemens Ladisch cladi...@googlemail.com wrote Malte Gell wrote: Do you think I could get rid of this Skype issue when I upgrade to a CMedia 8768 chip? That's exactly the the same hardware interface (and driver), so I'd guess you'd get the same error. Ok, same driver, but different chip, isn't it? The internals of the chip might be somewhat different, but the interface to the driver is exactly the same (except for a bit to enable 8 channels), and since the problem is apparently not with the hardware but with the way how Skype accesses the device, the same thing is likely to happen with the 8768. Best regards, Clemens -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Known C-Media 8738 issues?
Malte Gell wrote: when I use Skype 2.0.0.72 on Linux I get an error: RtApiAlsa: callback thread error (RtApiAlsa: audio write error for device (C- Media CMI8738 (hw:CMI8738,0)): unknown error 405.) ... closing stream. Unknown error isn't quite helpful. I did not have problems before when I use my good old Soundblaster Live! and so I wonder, are there any known issues with the CMedia 8738 chip? Nothing comes to mind. Do you think I could get rid of this Skype issue when I upgrade to a CMedia 8768 chip? That's exactly the the same hardware interface (and driver), so I'd guess you'd get the same error. Best regards, Clemens -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dmix oddities
Grant wrote: Are you sure the hardware actually supports S24_3LE? Most 24 bit soundcards require 24 bit audio to be packed into 32 bit words. See if you can output S24_3LE directly to the hw device. I don't think a USB DAC could do 24-in-32 because of the USB 24-bit limitation. I've tried specifying S24_3LE, S24LE, S24_BE, S24_3BE, FLOAT_LE, FLOAT_BE, and S16_LE in asound.conf. None of them produce sound except for S24_3LE and S16_LE. How can I output S24_3LE directly to the device for testing? 24-bit USB devices use S24_3LE. 24-bit PCI cards usually use S32_LE (where only 24 bits are used). HTH Clemens -- Crystal Reports - New Free Runtime and 30 Day Trial Check out the new simplified licensing option that enables unlimited royalty-free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] VIA8237 and no sound
Mike Knichel wrote: I have recently set up a computer for my daughter from old parts. It uses the VIA 8237 chip. I can get no sound from speakers. I even ran the alsa script and the results are here... Your mixer settings seem to be correct. However, some mainboards use another than the front output for their front jack. Try plugging the speakers into another output, and/or changing the settings so that it play to more outputs (Exchange Front/ Surround, or Channel Mode 6ch, Surround Jack Mode Independent, Spread Front to Surround and Center/LFE). HTH Clemens -- Crystal Reports - New Free Runtime and 30 Day Trial Check out the new simplified licensing option that enables unlimited royalty-free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] VIA8237 and no sound
Mike Knichel wrote: I do not have front mounted speaker jacks. Only the ones in the back. I meant front as opposed to surround or center/LFE. If you have only one jack intended for speakers, then your description looks as if that jack is connected to the surround output of the codec. Please try setting the Surround Jack Mode control to Independent and enabling the Exchange Front/Surround control. HTH Clemens -- Crystal Reports - New Free Runtime and 30 Day Trial Check out the new simplified licensing option that enables unlimited royalty-free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem
Raena wrote: Clemens Ladisch wrote: Raena wrote: checking for ncursesw5-config... yes This indicates that ncursesw is installed ... checking for new_panel in -lpanelw... no configure: error: panelw library not found ... but actually it isn't. What is the output of the following commands? ... locate libpanel.so locate libpanelw.so ra...@kubuntu:~$ locate libpanel.so /usr/lib/libpanel.so /usr/lib/libpanel.so.5 /usr/lib/libpanel.so.5.7 This is OK ... ra...@kubuntu:~$ locate libpanelw.so /usr/lib/libpanelw.so.5 /usr/lib/libpanelw.so.5.7 ... but this isn't: the libpanelw.so link is missing. Please file a bug report with your distribution. You could create the missing files yourself with these commands: ln -s libpanelw.so.5 /usr/lib/libpanelw.so ln -s libformw.so.5 /usr/lib/libformw.so ln -s libmenuw.so.5 /usr/lib/libmenuw.so ln -s libncursesw.so.5 /lib/libncursesw.so (You don't actually need the new alsa-utils, except if you want to see the Japanese translation of alsamixer. :-) Best regards, Clemens -- Crystal Reports - New Free Runtime and 30 Day Trial Check out the new simplified licensing option that enables unlimited royalty-free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] record vinyl at 24-bit 96khz
scar wrote: i was basically just looking for verification that the m-audio 2496 card would suit my needs. Yes, it certainly would. gary wrote: I often wonder has VIA came out of nowhere with a DSP audio chip (Envy24). Oversampled ADCs are not exactly rocket science, but to make one with a product history of previous chips is suspicious. VIA got the ICE1712 chip design when it took over IC Ensemble. It then develop a bunch of derived Envy24xxx chips from that. Best regards, Clemens -- Crystal Reports - New Free Runtime and 30 Day Trial Check out the new simplified licensing option that enables unlimited royalty-free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem
Raena Lea-Shannon wrote: Is there another ncurses-devel I have missed? libncursesw5-dev? HTH Clemens -- OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ymfpci drive capability - help needed with setup
robert crowther wrote: Can ymfpci access the WF-192XG wavetable? No. While documentation for basic DS-XG wavetable functionality exists, nobody has as yet added support for this to the driver. Best regards, Clemens -- OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MOTU micro lite (USB, 5in, 5out)
Aidan Dixon wrote: I have a MOTU micro lite (USB, 5x MIDI IN and 5x MIDI OUT, Vendor ID 0x07fd, product ID 0x0001). According to the kernel usb-audio device driver source code, this device should be supported as its USB vendor and product ID are listed in a table in linux/sound/usb/usbmidi.c. It seems MOTU didn't quite grasp the concept of product ID; they use the same ID 0x0001 for all their USB MIDI interfaces. The driver checks that the device is actually a FastLane before attaching; all other MOTU models use a completely different protocol and are not supported. What are the issues beyond simply lack of information from MOTU? Lack of developers who would have time to reverse engineer it and to write a driver. Best regards, Clemens -- OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Capture w/ M-Audio Fast Track USB?
Michael B Allen wrote: Does capture work with the M-Audio Fast Track USB? Maybe. Does some capture device show up in the output of arecord -l? Can someone give me some advice or a pointer to some documentation that explains how the whole ALSA tool chain works wrt external USB sound recording? You can tell any ALSA program to use a specific device by using a device name like plughw:x,y if you know the card and device numbers. If you're using PulseAudio, you have to tell it to use that device. HTH Clemens -- OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem
Raena wrote: I also installed the snapshot alsa-lib and that was fine but alsa-utils would not configure. configure: error: panelw library not found I have not been able to find out what package has this library. Here is my configure output; ... checking for ncursesw5-config... yes This indicates that ncursesw is installed ... checking for new_panel in -lpanelw... no configure: error: panelw library not found ... but actually it isn't. What is the output of the following commands? ncursesw5-config --libs ncursesw5-config --libdir locate libncurses.so locate libncursesw.so locate libpanel.so locate libpanelw.so Best regards, Clemens -- OpenSolaris 2009.06 is a cutting edge operating system for enterprises looking to deploy the next generation of Solaris that includes the latest innovations from Sun and the OpenSource community. Download a copy and enjoy capabilities such as Networking, Storage and Virtualization. Go to: http://p.sf.net/sfu/opensolaris-get ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] emu10k firmware
Ari Moisio wrote: Thanks:-) I downloaded the relevant firmware package from alsa-project site, configured, compled and installed. Everthing is working again:) I only wonder if the driver itself loaded the firmware earlier because there were no such issues with 2.4 kernels? In older driver versions, the firmware was included directly in the driver, but this was changed to allow the driver to be used with Debian. It is still possible to compile the firmware into the kernel, but it seems your distribution did not choose to do this. HTH Clemens -- Crystal Reports #45; New Free Runtime and 30 Day Trial Check out the new simplified licensign option that enables unlimited royalty#45;free distribution of the report engine for externally facing server and web deployment. http://p.sf.net/sfu/businessobjects ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] DELL SP2309W Monitor with Integrated microphone
Paul Hartman wrote: On Fri, Apr 17, 2009 at 2:27 AM, Clemens Ladisch cladi...@googlemail.com wrote: Phil Gorbett wrote: I am having difficulty getting the microphone(s) going with this monitor, and get the get the following results when I disable pulseaudio and run arecord: arecord -v -f cd -D plughw:0 file.wav Recording WAVE 'file.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable arecord: set_params:939: Broken configuration for this PCM: no configurations available Please show the output of lsusb -v for this device. I'm not the OP but I have the same monitor and same problem. Here's the lsusb output for the microphone: http://pastebin.com/f4c340014 These descriptors claim it is a simple standard-compliant device. When you run that arecord command, is there any error message in the system log (/var/log/messages, or the output of dmesg)? Best regards, Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] pcm_min.c trouble with modification
Michał Kowalczyk wrote: SND_PCM_FORMAT_U8, //changing this type for any other gives an error in the output: segmentation fault. U8 has one byte per sample; if you try to use a bigger sample format, you have to enlarge the buffer accordingly. 1, //argument responsible for channels; if I change it to 2(stereo) program gives an segmentation fault error. Using two channels would double the buffer size, too. 48000, //sampling rate; works good but if I modify this to lower value the time of generating noise will be longer, don't know why. If the sample rate is lower, you need more timer to play the same number of samples. frames = snd_pcm_writei(handle, buffer, sizeof(buffer)); The last parameter is measued in frames, not in bytes. When you use another sample format than U8/S8 or more then one channel, you have to divide the buffer size by the number of bytes per frame. HTH Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] pcm_min.c trouble with modification
MK wrote: I want to write a simple ALSA application. I have downloaded pcm_min.c from ALSA documentation and try to do some modifications in code like change channels to stereo or sampling rate to 44100Hz. But every time program fails. Can anybody explain me why this is happening? Please show your changes. Best regards, Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] AMT8
immanuel litzroth wrote: Apr 13 12:08:39 voodochile kernel: [ 169.622067] usb 2-10: new full speed USB device using ohci_hcd and address 5 Apr 13 12:08:39 voodochile kernel: [ 169.844566] usb 2-10: configuration #1 chosen from 1 choice Apr 13 12:08:39 voodochile kernel: [ 169.847594] snd-usb-audio: probe of 2-10:1.0 failed with error -5 Apr 13 12:08:39 voodochile kernel: [ 169.847616] snd-usb-audio: probe of 2-10:1.1 failed with error -5 Apr 13 12:08:44 voodochile kernel: [ 174.656857] usb 2-9: new full speed USB device using ohci_hcd and address 6 Apr 13 12:08:44 voodochile kernel: [ 174.879643] usb 2-9: configuration #1 chosen from 1 choice Apr 13 12:08:44 voodochile kernel: [ 174.882987] snd-usb-audio: probe of 2-9:1.0 failed with error -5 Apr 13 12:08:44 voodochile kernel: [ 174.883013] snd-usb-audio: probe of 2-9:1.1 failed with error -5 What's up with that? I'd guess the driver doesn't claim all the interfaces that the device has. This shouldn't matter. Then if I do and aplaymidi -l it lists the following ports: 40:0AMT8 AMT8 MIDI 1 40:1AMT8 AMT8 MIDI 2 40:2AMT8 AMT8 MIDI 3 40:3AMT8 AMT8 MIDI 4 40:4AMT8 AMT8 MIDI 5 40:5AMT8 AMT8 MIDI 6 40:6AMT8 AMT8 MIDI 7 40:7AMT8 AMT8 MIDI 8 40:8AMT8 AMT8 Broadcast 44:0AMT8 AMT8 MIDI 1 44:1AMT8 AMT8 MIDI 2 44:2AMT8 AMT8 MIDI 3 44:3AMT8 AMT8 MIDI 4 44:4AMT8 AMT8 MIDI 5 44:5AMT8 AMT8 MIDI 6 44:6AMT8 AMT8 MIDI 7 44:7AMT8 AMT8 MIDI 8 44:8AMT8 AMT8 Broadcast Is it possible to have other names generated for these ports through some magic? You could add some entries to the snd_usbmidi_port_info array in the driver source ... Preferably each port would be named after the synth connected to it. Changing port names at runtime isn't supported by the driver. Then when I play the devices I get messages in /var/log/messages saying Apr 11 23:03:56 voodochile kernel: [14700.085974] rtc: lost 28 interrupts Apr 11 23:04:26 voodochile kernel: [14730.054964] rtc: lost 28 interrupts Apr 11 23:04:56 voodochile kernel: [14760.023963] rtc: lost 28 interrupts What's up with that? Every thirty seconds, something on your computer eats some CPU and disables all interrupts for about 27 milliseconds. If you have a laptop, this it probably some fan control or power management stuff done by the BIOS. Best regards, Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] DELL SP2309W Monitor with Integrated microphone
Phil Gorbett wrote: I am having difficulty getting the microphone(s) going with this monitor, and get the get the following results when I disable pulseaudio and run arecord: arecord -v -f cd -D plughw:0 file.wav Recording WAVE 'file.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable arecord: set_params:939: Broken configuration for this PCM: no configurations available Please show the output of lsusb -v for this device. Best regards, Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] query alsa for supported sample rates and formats?
Matt Garman wrote:
Is there a way to query alsa to see what sample rates and formats
the sound hardware natively supports?
Try the attached program.
HTH
Clemens
/*
* hw_params.c - print hardware capabilities
*
* compile with: gcc -o hw_params hw_params.c -lasound
*/
#include stdio.h
#include alsa/asoundlib.h
#define ARRAY_SIZE(a) (sizeof(a) / sizeof *(a))
static const snd_pcm_access_t accesses[] = {
SND_PCM_ACCESS_MMAP_INTERLEAVED,
SND_PCM_ACCESS_MMAP_NONINTERLEAVED,
SND_PCM_ACCESS_MMAP_COMPLEX,
SND_PCM_ACCESS_RW_INTERLEAVED,
SND_PCM_ACCESS_RW_NONINTERLEAVED,
};
static const snd_pcm_format_t formats[] = {
SND_PCM_FORMAT_S8,
SND_PCM_FORMAT_U8,
SND_PCM_FORMAT_S16_LE,
SND_PCM_FORMAT_S16_BE,
SND_PCM_FORMAT_U16_LE,
SND_PCM_FORMAT_U16_BE,
SND_PCM_FORMAT_S24_LE,
SND_PCM_FORMAT_S24_BE,
SND_PCM_FORMAT_U24_LE,
SND_PCM_FORMAT_U24_BE,
SND_PCM_FORMAT_S32_LE,
SND_PCM_FORMAT_S32_BE,
SND_PCM_FORMAT_U32_LE,
SND_PCM_FORMAT_U32_BE,
SND_PCM_FORMAT_FLOAT_LE,
SND_PCM_FORMAT_FLOAT_BE,
SND_PCM_FORMAT_FLOAT64_LE,
SND_PCM_FORMAT_FLOAT64_BE,
SND_PCM_FORMAT_IEC958_SUBFRAME_LE,
SND_PCM_FORMAT_IEC958_SUBFRAME_BE,
SND_PCM_FORMAT_MU_LAW,
SND_PCM_FORMAT_A_LAW,
SND_PCM_FORMAT_IMA_ADPCM,
SND_PCM_FORMAT_MPEG,
SND_PCM_FORMAT_GSM,
SND_PCM_FORMAT_SPECIAL,
SND_PCM_FORMAT_S24_3LE,
SND_PCM_FORMAT_S24_3BE,
SND_PCM_FORMAT_U24_3LE,
SND_PCM_FORMAT_U24_3BE,
SND_PCM_FORMAT_S20_3LE,
SND_PCM_FORMAT_S20_3BE,
SND_PCM_FORMAT_U20_3LE,
SND_PCM_FORMAT_U20_3BE,
SND_PCM_FORMAT_S18_3LE,
SND_PCM_FORMAT_S18_3BE,
SND_PCM_FORMAT_U18_3LE,
SND_PCM_FORMAT_U18_3BE,
};
static const unsigned int rates[] = {
5512,
8000,
11025,
16000,
22050,
32000,
44100,
48000,
64000,
88200,
96000,
176400,
192000,
};
int main(int argc, char *argv[])
{
const char *device_name = hw;
snd_pcm_t *pcm;
snd_pcm_hw_params_t *hw_params;
unsigned int i;
unsigned int min, max;
int any_rate;
int err;
if (argc 1)
device_name = argv[1];
err = snd_pcm_open(pcm, device_name, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (err 0) {
fprintf(stderr, cannot open device '%s': %s\n, device_name,
snd_strerror(err));
return 1;
}
snd_pcm_hw_params_alloca(hw_params);
err = snd_pcm_hw_params_any(pcm, hw_params);
if (err 0) {
fprintf(stderr, cannot get hardware parameters: %s\n,
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf(Device: %s (type: %s)\n, device_name,
snd_pcm_type_name(snd_pcm_type(pcm)));
printf(Access types:);
for (i = 0; i ARRAY_SIZE(accesses); ++i) {
if (!snd_pcm_hw_params_test_access(pcm, hw_params, accesses[i]))
printf( %s, snd_pcm_access_name(accesses[i]));
}
putchar('\n');
printf(Formats:);
for (i = 0; i ARRAY_SIZE(formats); ++i) {
if (!snd_pcm_hw_params_test_format(pcm, hw_params, formats[i]))
printf( %s, snd_pcm_format_name(formats[i]));
}
putchar('\n');
err = snd_pcm_hw_params_get_channels_min(hw_params, min);
if (err 0) {
fprintf(stderr, cannot get minimum channels count: %s\n,
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_channels_max(hw_params, max);
if (err 0) {
fprintf(stderr, cannot get maximum channels count: %s\n,
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf(Channels:);
for (i = min; i = max; ++i) {
if (!snd_pcm_hw_params_test_channels(pcm, hw_params, i))
printf( %u, i);
}
putchar('\n');
err = snd_pcm_hw_params_get_rate_min(hw_params, min, NULL);
if (err 0) {
fprintf(stderr, cannot get minimum rate: %s\n,
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_rate_max(hw_params, max, NULL);
if (err 0) {
fprintf(stderr, cannot get maximum rate: %s\n,
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf(Sample rates:);
if (min == max)
printf( %u, min);
else if (!snd_pcm_hw_params_test_rate(pcm, hw_params, min + 1, 0))
printf( %u-%u, min, max);
Re: [Alsa-user] alsamixer does not save settings
Geoffrey Leach wrote: I'm havind a problem with alsamixer. It appears not to save the changes. The man page does not mention this, so perhaps its doing what it is designed to do. It is only designed to change the sound card's mixer settings. In that case, can anyone point me to a tool that makes permanent (mute, in this case) settings? alsactl store and alsactl restore do what you want. In theory, these calls should be made automatically by your distribution's shutdown/ startup scripts. HTH Clemens -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problems with ALC650
Jonathan Black wrote: I've been struggling as of lately to get ANY sound through the lime green output on the back of my motherboard. lsmod | grep snd shows: snd_mpu401 6252 0 snd_cs4232 11380 0 snd_via82xx20920 2 Please show the output of /proc/asound/cards. Is the AC'97 controller the default (first) sound card? If yes, please show the output of amixer contents. Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] building soc module drivers
Geiger Ho wrote: I would like to build the WM8510 SOC sound driver for my ARM platform. After downloading alsa-driver-1.0.19, ./configure --with-cross=arm-linux- --with-kernel=/home/source/linux-2.6.22.19 --with-build=/home/source/linux-2.6.22.19 --with-cards=all, and then make. I get no wm8510.c compiled. Is the kernel in /home/source/linux-2.6.22.19 configured for your target machine? Best regards, Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Can't open Asus Xonar D1 PCM output 6-channel in non-interleaved mode
Oliver Stephenson wrote:
Now though, I'm trying to open the PCM output in 6-channel mode
non-interleaved, and it all falls over here:
if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params,
SND_PCM_ACCESS_RW_NONINTERLEAVED)) 0)
{
fprintf(stderr, cannot set access type (%s)\n, snd_strerror (err));
return false;
}
What gives? Is this a feature of the hardware (C-Media chipset) or is
there a workaround?
This chip, like pratically every other sound chip, supports only
interleaved samples.
Try adding plug:x in front of the device name.
HTH
Clemens
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Re: [Alsa-user] Trying to use two sound cards at the same time
Poulet Fou wrote: I have two sound cards, one attached to pc speakers and one attached to a sound system in another room. Following instructions on the alsa project wiki, I am trying to use two sound cards (V8237 and CA0106). I configure my ~/.asoundc but I still can't manage to send a sound to both cards at the same time. Below is my ~/.asoundrc content. When I tested directly the ca0106 card (hw:0,0) with a play, it didn't work so I thought it might be some sort of rate or conversion problem. I added the pcm.via and pcm.audigy in my asoundrc file and then, when I test each card individually (via and audigy) with aplay it works well. When testing the multi pcm with aplay ( aplay -vD multi question.wav ), I get the following error aplay: set_params:959: Nombre de canaux non disponible (Channels count non available) Your multi device is a four-channel device. Try plug:multi. HTH Clemens -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to set USB-Headset volume with udev and amixer
Jens Rutschmann wrote:
I have a C-Media USB Headphone Set which is working well. One issue though is
that when plugging it in the mixer levels are initialized to bad values. The
Speaker level is set to 100% (*very* loud, like in ouch !!!) while the
Mic
Capture level is set to 0%.
Therefore I set the mixer levels by hand (using alsamixer) until now. So I
wrote
a script which is invoked by udev. The script tries to set the levels using
amixer.
[...]
So I wrote the following script and udev rule:
ATTR{idProduct}==000c, ATTR{idVendor}==0d8c, ACTION==add,
RUN+=/usr/local/bin/headset_volume.sh
...
The script is executed since I get the log file, but amixer obviously isn't
executed correctly,
I don't know why your script doesn't work, but the following udev rule
works for me, and it does not need to be adjusted for the specific sound
device but just requires that you've previously saved the the correct
settings with alsactl store:
SUBSYSTEM==sound, ACTION==add, KERNEL==controlC[0-9]*,
RUN+=/usr/sbin/alsactl restore $number
HTH
Clemens
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Re: [Alsa-user] Problem with SoundBlaster Live
Matias D'Ambrosio wrote: I have been using this system for months, no problem (and many distros before), but a few days ago sound stopped working, probably after an update. I have an SB Live, I'm running Debian testing AMD64 kernel 2.6.26-1-amd64 #1 SMP. lspci shows the card: 01:07.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev 0a) cat /proc/asound/modules gives no output ... dmesg output: http://pastebin.ca/1365751 [9.394958] gameport: EMU10K1 is pci:01:07.1/gameport0, io 0xb800, speed 1201kHz The card is there ... [ 11.280004] AC'97 0 does not respond - RESET [ 11.280010] AC'97 0 access is not valid [0x0], removing mixer. [ 11.280842] ACPI: PCI interrupt for device :01:07.0 disabled [ 11.280857] EMU10K1_Audigy: probe of :01:07.0 failed with error -5 ... but the AC'97 codec on your card doesn't seem to work. This is probably a driver bug (try a newer kernel, or some older/newer live CD to be sure), but a hardware defect is also possible. Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Analog line in output via IEC958?
Antony Gelberg wrote: I have an onboard NVidia sound card using snd_hda_intel. Full info about my setup is at http://www.alsa-project.org/db/?f=f407c7aceb5abe36b68731ff50fd1d21f272fb98 I can output everything to a HT amp via the IEC958 optical out by means of the .asoundrc that you see in the above link. I would like to plug something into the analogue line in and have it be output via the IEC958 as well. However, it is only output via the analogue line out, not the optical. How can I get this to work? The ALC883 cannot do this alone. You have to run a program to record from the line in and to play the data back to the SPDIF output. arecord -f dat | aplay -D spdif might work, but then you cannot use these devices with other programs. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Unable to configure ASUS A7V sound card with Alsa
(please don't top-post) sto...@safe-mail.net wrote: Any suggestions with this output? The AC'97 controller does not show up. Please make sure that SB emulation is disabled. Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] modprobe snd-virtuoso error
Ron Pitts wrote: Analogue Output is working fine now however no luck with the HDMI audio. But HDMI video does work? Does your HDMI display show if any audio is sent, or in which format? What sample format did you try to play? I've rebuilt the alsa lib with full debugging turned on: No sound for HDMI but receiving the following in dmesg: [ 33.668846] : 4e 43 4f 4b NCOK [ 36.523606] message from Xonar HDAV HDMI chip received: [ 36.523632] : 53 54 41 52 54 4f 4b STARTOK [ 36.581530] message from Xonar HDAV HDMI chip received: [ 36.581551] : 32 34 50 4f 46 46 4f 4b 24POFF Is the plug iec958, the HDMI sound component? No, that is the SPDIF output. The HDMI sound must be driven by the same DMA channel as the analog outputs (Multichannel) because it is the only one that supports more than two uncompressed PCM channels. I take it the above commands are from the method xonar_hdav_init in virtuoso.c, No, the commands sent by the driver are not shown. The logged messages are responses from the HDMI chip. You can show the sent commands by adding the following at the end of the hdmi_write_command function: printk(KERN_DEBUG sent: fb ef %02x %02x, command, count); for (i = 0; i count; ++i) printk( %02x, params[i]); printk( %02x\n, checksum); its been awhile since I hacked some C++ code!! Well, the driver doesn't use C++ ... Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Diamond Xtreme XS51 ALSA support? (was ALSA Support for old ISA Soundcard)
Landis McGauhey wrote: I've decided to replace the old ISA card with a PCI card (several PCI slots are available). Can anyone happen to tell me if you have any ALSA experience with the Diamond Xtreme XS51? According to the driver's .inf file, that card uses a CMI8738/8768 chipset, which is supported by ALSA. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] modprobe snd-virtuoso error
Ron Pitts wrote: [ 653.504508] snd_mpu401_uart: Unknown symbol snd_rawmidi_receive ... The sound modules cannot be loaded because other sound modules they depend on cannot be loaded. Please show the first few of these messages (for module snd). Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] modprobe snd-virtuoso error
Ron Pitts wrote: Its loading now without error however doesnt report its found my ASUS HDAV1.3 .. It doesn't report anything when there is no error. Does it appear in /proc/asound/cards? Anyway, I've fixed a bug after 1.0.19 was released; please try http://www.alsa-project.org/snapshot/files/alsa-driver-1.0.19.30.gf0b3d.325.gc46d0.tar.bz2 Clemens, your the man when it comes to this card, I understand its only 10% finished at the moment? You're the first to ever try this code; I'm not sure if anything works. I estimate that analog sound output and video data passthrough should work, and that HDMI sound output possibly works. Video effects are not supported at the moment because I don't know how to enable them. Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA support for old ISA card?
Landis McGauhey wrote: ALSACONF recognizes the soundcard as SBAWE Creative SB AWE64 PnP and completes its work without errors. ... Command ALSAMIXER results in the error message function snd_ctl_open failed for default: no such device. In theory, all those old ISA cards are still supported. However, some distributions don't enable the ISA drivers. Does the card appear in /proc/asound/cards? If not, does it after running modprobe snd-sbawe? If you need some output from DMESG, just let me know. This would be a good idea. Best regards, Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] modprobe snd-virtuoso error
Ron Pitts wrote: ... FATAL: Error inserting snd_virtuoso (/lib/modules/2.6.27-11-generic/kernel/sound/pci/oxygen/snd-virtuoso.ko): Unknown symbol in module, or unknown parameter (see dmesg) The most likely cause of this problem is that you are trying to load a module of a new driver version while some other sound module of the old driver version is still loaded. It should work after you unload all snd_* modules, or reboot. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] (no subject)
dennism...@gmx.net wrote: Can please a admin have a look at my mail? There is no moderator; all such mails go into a black hole. the message was over the max size Compress it, ot put it somewhere on the web. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Need help connecting spdif input
Mårten Gustafsson wrote: mor...@linux-hyfd:~ arecord -D spdif:0 -f dat -t raw | ac3dec -D hw:0 -6 Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo 2.0 Mode 48.0 KHz 32 kbps Complete Main Audio Service unsupported 1/1 channels 6 Apparently, the recorded Dolby data is encoded as 2.0, not 5.1, and ac3dec does not supported playing 2.0 data on six channels. Try it without the -6. Best regards, Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Unable to configure ASUS A7V sound card with Alsa
sto...@safe-mail.net wrote: The card is enabled in the BIOS, Is the entry Enabled or Auto? The latter means the BIOS automatically disables it when it finds another sound card, because, apparently, it doesn't expect anybody would want to use it when there is a _real_ sound card. This is common with BIOSes for VIA chipsets. The Linux kernel has code to forcibly enable the AC'97 controller on VT8237-based mainboards. In the case of older mainboards like yours, which sometimes didn't have audio, this cannot be done automatically. Also check that Sound Blaster emulation is disabled. but does not appear in the lspci output. Please show the lspci output. I was also surprised to find that Asus supplies drivers that support RedHat 6.2. This is no surprise if you consider that RH 6.2 comes with kernel 2.2.14, which didn't yet have ALSA. I strongly suggest you forget those drivers and use the snd-via82xx driver of your distribution, which is likely to be newer by about eight years or so. HTH Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Need help connecting spdif input
rek...@holisticode.se wrote: I have been searching the web for an example of how to get sourround sound from my toslink input [...] I realise that an a52 filter should be used, The a52 plugin encodes PCM data into Dolby data, and as a plugin, it is only used when an application uses the specified sound device. What you want to do is to record from the digital input device, convert the data from Dolby to PCM, then output this to the analog output device. It should be possible to do this by using arecord to capture raw data, then piping it into ac3dec. untested, assuming card 0: arecord -D spdif:0 -f dat -t raw | ac3dec -D hw:0 -6 HTH Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] xonar essence stx
jed wrote: Damn, that's a substantial amt of stuff that won't be available don't u reckon? Yes. Best regards, Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Split stereo input?
deadrabbit wrote: I'm new to using ALSA, or at least to the internals of it. I'm trying to create a Icecast stream encoding computer for two audio streams. The computer as a single stereo line in, so I'm planning on feeding each mono stream through the single stereo input. I've looked at the asoundrc guide, but I'm not sure I understand it entirely: there's no documentation for some things, like ttable. Will the config below work for splitting the stereo input in to two mono devices? In theory, yes. Also, how do I know what the input device name is? It's the string after the pcm.. In this case, dshare is the slave device used by the two other devices which are named leftx and rightx. HTH Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] anyone using the M-Audio Fast Track Ultra 8R (or any other 8in 8 out USB device)
sonof...@iinet.net.au wrote: I'm looking to move over to a device that I can use with both my linux desktop and a 2nd hand powerbook i bought recently. I have been more focussed on FFADO compatible as from what I understand I am more likely to get latency issues on a USB device... However, I have been using my M-Audio Delta 1010 for years now and have been very happy with the audio quality and latency performance. This looks like a USB version but with preamps (which I don't need) The USB latency depends on the USB protocol itself and on the driver; these are the same for _all_ USB devices. Furthermore, there is no supported 8+8 device at the moment. Best regards, Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Griffin Imic
gary wrote: gary wrote: Now for the problem. I can't talk to the iMic. Using krecord -d /dev/dsp4 . I get the response ioctl SNDCTL_DSP_SETMFT: Input/output error There should be an error message in the system log (probably /var/log/messages, or at the end of the output of dmesg). I loaded the same two devices on Suse 10. That got a bit further. The error message is the usb sound module can't be loaded. There should be an error message in the system log. Best regards, Clemens -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] xonar essence stx
jed wrote: Is support for this coming soon? Yes. How does using ALSA compare to using drivers developed by the vender specifically for Windows? i.e. Will I not be getting most out of this card (hardware-wise) if I'm not using the drivers Asus developed, or is the difference negligible? The driver supports all features of the hardware; the output quality is the same as with the Windows driver. Please note that the Linux driver does not have the driver features, which are implemented in software (Dolby, SVN, Xear, etc.). Best regards, Clemens -- Create and Deploy Rich Internet Apps outside the browser with Adobe(R)AIR(TM) software. With Adobe AIR, Ajax developers can use existing skills and code to build responsive, highly engaging applications that combine the power of local resources and data with the reach of the web. Download the Adobe AIR SDK and Ajax docs to start building applications today-http://p.sf.net/sfu/adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] xonar essence stx
jed wrote: Is support for this coming soon? Yes. Awesome, is it likely to be within the next 6-months? Actually, it looks like this week ... Please note that the Linux driver does not have the driver features, which are implemented in software (Dolby, SVN, Xear, etc.). Can you please be a little more specific about these driver features implemented in software? Or provide a link to where it's outlined? They're listed on this page: http://www.asus.com/products.aspx?modelmenu=2model=2711l1=25l2=150l3=0l4=0 Best regards, Clemens -- Create and Deploy Rich Internet Apps outside the browser with Adobe(R)AIR(TM) software. With Adobe AIR, Ajax developers can use existing skills and code to build responsive, highly engaging applications that combine the power of local resources and data with the reach of the web. Download the Adobe AIR SDK and Ajax docs to start building applications today-http://p.sf.net/sfu/adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Surround over spdif
Floris wrote: Is there a way to encode a file on the fly? http://wiki.alsa-project.org/main/index.php/A52_plugin HTH Clemens -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] broken digital sound again
James wrote: $ speaker-test -D iec958 -c 6 ... Channels count (6) not available for playbacks: Invalid argument S/PDIF does not support uncompressed surround sound; you have to play stereo data (use -c 2) or AC-3/DTS-compressed data. Best regards, Clemens -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Device exists, PCM does not
Evan Leibovitch wrote: # arecord -D hw:1,0 /dev/null Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo arecord: set_params:932: Broken configuration for this PCM: no configurations available Try arecord -D default:1. HTH Clemens -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] textual tool for pause/resume
davide.bonfa...@bticino.it wrote: I'm searching for a textual tool in order to play, pause and resume the streaming on a certain device. Try mplayer. HTH Clemens -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to order multiple usb devices
Ari Moisio wrote: # Usb-headset options snd-usb-audio index=2 vid=0x046d,0x0a02 # M-audio fasttrack options snd-usb-audio index=3 vid=0x0763,0x2012 This does not work because the second 'options' line overrides the first, and the product ID must be put into the 'pid' parameter. Use this: options snd-usb-audio index=2,3 vid=0x046d,0x0763 pid=0x0a02,0x2012 (Since the vendor IDs are unique, you don't actually need 'pid' in this case.) HTH Clemens -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Digital bit perfect ouptut with ALSA
Bill Unruh wrote: On Mon, 15 Dec 2008, Paulo Moura Guedes wrote: [...] The ASRC, as the name implies, is not syncronized to the clock of the incoming digital signal. Therefore, its performance is independant of the This makes no sense at all. If the incoming signal is a digital signal, it has no clock. An S/PDIF signal does have a clock (it uses biphase mark coding for exactly this reason), and there are many DACs that derive their master clock directly from the input signal. (It's different in the case of USB where the sample clock is independent of the bus clock.) quality of that clock. In other words, it doesn't matter if the signal came from a cheap transport with cheap cables, or from a $10,000 signal chain. The And what the hell do cables have to do with anything? Cables do NOT affect the timing. If the cable's impedance is high enough, the transitions between the signal levels can be delayed. This wouldn't matter much except that due to the biphase mark coding, the jitter is data dependent. (It's a different question whether the distortions caused by the jitter are actually audible.) Best regards, Clemens -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Digital bit perfect ouptut with ALSA
Paulo Moura Guedes wrote: On Thursday 11 December 2008 07:35:25 Clemens Ladisch wrote: The default device (named default) uses automatic resampling, but the spdif device does not. So, I have to manually set the sample-rate, depending on the files I will play? No, ALSA automatically switches the device's sample rate to that of the audio data (or resamples to the nearest supported rate). - is it possible to set the bit-depth? (in my case to 24 bit) Yes, if the hardware supports it. It's a Benchmark DAC1 which supports 24 bit. Where can I set it? You don't need to do this manually; ALSA automatically uses 24 bits. In your case, the USB device is handled like a sound card; any other sound card in the system doesn't affect it. Best regards, Clemens -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Audio on ECS A780GM-A motherboard
Laurent E wrote: - According to the command cat /proc/asound/card0/codec#0, the chip is a SigmaTel ID 7645. [...] 1/ Is it possible that the chip is incorrectly detected? Your driver doesn't know about your chip. Support for this chip was added in May: http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=7bd3c0f73c9c5b47fd1ca49757c436e73f4cd55b Update your kernel, or try to apply the patch manually. HTH Clemens -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Digital bit perfect ouptut with ALSA
Paulo Moura Guedes wrote: - does Linux/ALSA features dynamic sample rates? The default device (named default) uses automatic resampling, but the spdif device does not. - is it possible to set the bit-depth? (in my case to 24 bit) Yes, if the hardware supports it. - what other variables do i have to consider in order to get bit transparent ouput, i.e., no resampling at all? Just using the spdif device should work. (Nowadays, there is no device that would want to forgo the ability to output AC-3/DTS data.) - is it possible to check if ouput is bit-perfect? Record it, then compare the data to the original. (However, there are sound chips that always resample when recording from a digital input.) HTH Clemens -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ICH7/ALC662 card won't initialise
Lindsay Roberts wrote: I have a board Which one? This is an Intel 945GC chipset (ICH7) with a Realtek ALC662 on it, both of which seem to be supported in alsa. Leaving the dist to its own devices sees it load snd_intel8x0, and error out as follows: codec_ready: codec is not ready[0x870] Assuming what I know about the division between snd_hda_intel and snd_intel8x0 is correct this doesn't suprise me a great deal because the ALC662 is actually a Realtek hda chip, not an ac97 chip. Seems like the pci id registered for the ICH7 in snd_hda_intel is 8086:27d8 and the one in snd_intel8x0 is 8086:27de . The device on my board is of course pci id 8086:27de . Hacking in that pci id to snd_hda_intel was completely unenlightening: hda-intel: ioremap error The BIOS is responsible for switching the controller into either AC97 or HDA mode. It appears this was not done correctly in your case. As the chip is currently configured, you _do_ have an AC97 controller (without an AC97 codec). I have no explanation for this, except maybe a wrong BIOS (for a different mainboard), or a hardware error. Did it ever work? One final juicy tidbit is that it reports no subsystem vendor/device (zeroes all round). IIRC these IDs are to be initialized by the BIOS, too. Best regards, Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] setting ALSA_CARD from /etc/bashrc
Yan Seiner wrote: I've added a line to /etc/bash.bashrc that sets ALSA_CARD depending on the value of $DISPLAY. Now this works fine if I open a terminal window and run my apps. But apps started from the menu don't have ALSA_CARD set. You could set ALSA_CARD when X is started (in .xinitrc or something like that). HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] AC-3 Passthrough doesn't works
[EMAIL PROTECTED] wrote: But when I try to replay a DVD with AC-3 data xine only tells me that the audio device is not available. What happens when you try a command like this: aplay -D spdif something.wav Best regards, Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Definition of a frame
Bankim Bhavsar wrote: For a stereo DAC channel with 16-bit samples, number of bytes per frame = 4 bytes, correct? Yes. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] choice of a new soundcard
Vedran Miletić wrote: http://www.via.com.tw/en/products/audio/controllers/envy24/ Claims that it has 20-channel, 26-bit wide built in mixer. What does that mean? The chip plays 10 channels in parallel (8 analog, 2 digital) and records 12 channels in parallel (8 analog, 2 digital, 2 monitoring). The two monitoring channels result from mixing together the other 20 channels. This is not the usual hardware mixing scenario because we don't have several independent stereo streams, and the mixing result does not go to the speaker outputs. Regards, Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] choice of a new soundcard
Vedran Miletić wrote: Drivers that support hardware mixing: * snd_emu10k1 - ... * snd_cs46xx - ... * snd_au88x0 - ... * snd_ymfpci - for YMF7xx chips. Like most of the others, has been discontinued and must be bought used. Used on several cards, e.g., Hoontech SoundTrack Digital XG (most of those cards have XG in their name). If you want a supported wavetable synthesizer, get a SB Live! or Audigy. If you want to do bit-exact recording from the SPDIF input, get a YMF754B-based card. Otherwise, the chip doesn't matter. Not sure about: * snd_ice1712 Does _not_ support hardware mixing. Hardware does support it, and it did work on Windows. The Windows driver, like the drivers of most other cards, claims to support multiple streams and does software mixing in the driver itself instead of letting Windows do the mixing. Since Windows assumes that no driver would do such a silly thing, the driver shows up as supports hardware mixing. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA timer interface and accuracy
Julien Claassen wrote: If I'd like to use the PCM slave timer, how do I access that? Or were the instructions you gave me at the end of your mail already for that prupose? Yes. For the values of the CARE/DEV/SUBDEV parameters, see /proc/asound/timers. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA timer interface and accuracy
Julien Claassen wrote: Does the ALSA timter interface take its timing info from the soundcard (sample clock) or from the system's timer (RTC)? ALSA's timer interface can use 1) the system timer (which is not the RTC timer), 2) the RTC timer (if it is available), 3) a sound card timer, if the sound card has a programmable timer (very few actually have one), 4) a PCM slave timer, which generates timer ticks at period boundaries (PCM slave timers are available for every hardware sound device, but only while running, i.e., while sound is actually being played/recorded). If it is the system: Is there a way to get a clock source from the soundcard using the ALSA API? The device name of the timer device hw has several parameters: 1) hw:CLASS=1,DEV=0 2) hw:CLASS=1,DEV=1 3) hw:CLASS=2,CARD=xxx,DEV=yyy 4) hw:CLASS=3,CARD=xxx,DEV=yyy,SUBDEV=zzz HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to interrupt a blocking call to snd_pcm_writei
Florian Winter wrote: I can't derive from the documentation when exactly my poll() call will wake up. It can wake up whenever the device generates an interrupt, i.e., at period boundaries. Is there a way to tell ALSA something like: I want to write at least N frames to the device without blocking. Please don't wake up my poll() call until I can do this (assuming that N is a meaningful value, i.e. less than the total buffer size)? You can use snd_pcm_sw_params_set_avail_min(), but it won't have any effect if N is lower then the period size. Will the ALSA device always become writeable when N bytes can be written, or is the ALSA implementation (driver implementation) free to decide when and when not to block? All drivers behave the same in this regard because waking up is handled by the ALSA framework. Please note that you can be waken up when an error occurs before N frames have become available. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How to interrupt a blocking call to snd_pcm_writei
Florian Winter wrote: Suppose, an ALSA playback device is opened in blocking mode, and one thread calls snd_pcm_writei. If the snd_pcm_writei call blocks, because the internal buffer of the ALSA device is full, is there a way by which another thread can interrupt the call, so it returns immediately without writing any data to the device? No. As far as the user-space library is concerned, ALSA is not thread safe. You have to use non-blocking mode and poll() so that you can send a message through a file descriptor (e.g. pipe or eventfd). Regards, Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] em28xx_alsa: disagrees about version of symbol xxx after upgrading CentOS 5.2 ALSA .14 for ALSA .17
Robert Vincent Krakora wrote: em28xx_alsa: disagrees about version of symbol snd_pcm_new This means that the em28xx_alsa module that you're trying to load and the currently loaded snd_pcm module have been compiled for different kernels. It is possible that the old ALSA modules are still loaded; try unloading them, or rebooting. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] SPDIF not working with AD1989B (Asus P5Q-E motherboard)
Jason Tackaberry wrote: I am testing with mplayer, and ensuring the digital device (as shown by aplay -l) is being used. In my case it's device 1, and I am specifying -ao alsa:device=hw=0.1 on the mplayer command line. Try spdif instead of hw:0,1. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Which process is using an ALSA driver.
Eliot Blennerhassett wrote: can anyone tell me how to find out which processes are using an ALSA driver and preventing it from being unloaded. I know that lsof /dev/snd will tell me who has these files open, but often this shows nothing and still the driver can't be unloaded. Where else should I look? The device files for the OSS emulation are in not in /dev/snd but in /dev, try lsof /dev/dsp*. Is there an equivalent to lsof for memory mapped regions? Yes, it's called lsof. ;-) Memory-mapping requires an open file handle. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] SPDIF woes
Jason Gauthier wrote: I can't find many cards except for the Creative Labs X-Fi that seems to fit both.. and from what I read SPDIF passthrough doesn't work.I'm not 100% sure what that means. (Pretty new to this level of audio) SPDIF was designed to transport two channels of 16-bit uncompressed audio at 48 kHz (this is 192,000 bytes per second). If you want to transport more channels (4.0/5.1/7.1), the data has to be compressed with a codec like Dolby Digital (AC-3) or DTS, and a special control bit must be set in the SPDIF stream to indicated that the data is compressed. (Since nobody wants to pay for an encoder license, this is mostly interesting for DVDs, which the audio data already in compressed form.) A sound card that can do SPDIF passthrough (usually called DTS/AC3 passthrough) can two things: 1) it does not modify the audio data in any way (every bit is needed for decoding), and 2) it allows to set the non-audio control bit. You don't need this feature if all you want is stereo. Does anyone have any recommended cards that fit both PCI Express, USB, or firewire and SPDIF out? USB: There are many devices with SPDIF output, but there isn't any one that does support passthrough in Linux. PCI-E: The X-Fi has SPDIF, and it should not take more than another few years before Creative adds passthrough support to the driver. The Asus Xonar DX and D2X cards do support passthrough, but you need a recent enough kernel; see http://www.alsa-project.org/main/index.php/Matrix:Vendor-Asus. Firewire: don't know, there is probably nothing that would be cheaper than the Xonar DX. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA
V Gabriele De Palo wrote: # cat /proc/asound/cards ... 1 [Ultra ]: USB-Audio - Fast Track Ultra M-Audio Fast Track Ultra at usb-:00:1d.7-3, high speed It seems the device is more or less class compliant, so, in theory, it could work. # speaker-test -c 2 -D hw:1 ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'defaults.namehint.extended' There is some problem with your alsa-lib installation that is unrelated to the sound hardware. Try reinstalling alsa-lib and/or alsa-utils, if possible. Remove /etc/asound.conf or ~/.asoundrc, if you have them. HTH Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] CMI8788.
Mark A Jenks wrote:
I have a Bluegears b-Enspirer 7.1 Sound Card CMI8788 that I am looking
for some help/advice for.
I would like to seperate out the stereo outs to multiple outputs, so I
can use it as a backend for my house with zones.
The hardware cannot control the four analog outputs independently, but
it's possible to use plugins for ALSA-aware applications.
Put the following into your ~/.asoundrc or /etc/asound.conf:
pcm_slave eightchannels {
pcm hw:0,0# or hw:1,0 for the second card
channels 8
}
pcm.stereo1 {
type plug
slave.pcm {
type dshare
slave eightchannels
}
bindings [ 0 1 ]
}
pcm.stereo2 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 2 3 ]
}
pcm.stereo3 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 4 5 ]
}
pcm.stereo4 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 6 7 ]
}
then use device names stereo1 to stereo4 instead of default or
hw:x.
I also would like to leave the Digital out as an 7.1 to work with my
stereo for Watching movies.
The SPDIF output is independent; it mirrors the front analog output only
when the spdif device is not being used.
HTH
Clemens
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Re: [Alsa-user] Record audio with Philips SPC520NC webcam
Artem Makhutov wrote: I have a Philips SPC520NC webcam, which works fine with the linux UVC driver. The only problem I have is to record audio using the build in microphone. (Under Windows the microphone works well.) After arround 5 seconds I am getting an read error: I/O Error message, and I am getting an empty wav file (only the wav header is in there). This error usually indicates that the device didn't send any data. Is there any error message in the system log? Regards, Clemens - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] dmixer
Stevens, Peter wrote: For alsa-lib-1.0.17rc2 the dmix plug-in is enabled by default (for the default stereo device). Would anybody know which setting I must override in /etc/asound.conf file to enable dmix for the surround51 device (among others)? I wish to be able to open any of the : stereo, surround40, or surround50 devices at the same time and have their outputs mixed automatically. I was able make this happen by creating my own alsa devices, but I assume there is a cleaner way to do this (via overriding an existing setting)? At the moment, there is no existing dmix configuration for the surround devices; writing your own device definitions is the only way. Regards, Clemens CONFIDENTIALITY NOTICE: ... *** DISCLAIMER *** This e-mail contains public information intended for any subscriber of this mailing list and for anybody else who bothers to read it; it will be copied, disclosed and distributed to the public. If you think you are not the intended recipient, please commit suicide immediately. These terms apply also to any e-mails quoted in, referenced from, or answering this e-mail, and supersede any disclaimers in those e-mails. Additionally, disclaimers in those e-mails will incur legal processing fees of $42 per line; you have agreed to this by reading this disclaimer. *** END OF DISCLAIMER *** - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Audio hardware with pause support
Florian Winter wrote: - What is the dmix plugin and what are the benefits of using it? - Is it possible to disable the dmix plugin? - What consequences does disabling the dmix plugin have? What essential features of ALSA will be missing without it? The dmix plugin allows multiple applications to play to one sound device by mixing multiple streams together. The default device automatically uses dmix when the sound card does not have hardware mixing; to disable this, configure your applications to use device name plughw instead of default. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
stan wrote: Florian Faber wrote: You want hardware monitoring - there are sound cards that support hardware mixing. With good converters you have latencies down to 5 samples at 192kHz, that would be 0.026ms for each way, 0.052ms over all. I'm not the original poster, but I'm curious about this. Can you name a card that has this capability supported in alsa? I know that CMI8788-based cards can do this. There may be other cards, like the ESI Julia, but I don't know if monitoring is supported by the driver. Regards, Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Output latency
Alexander Carôt wrote: 3.) Rather than using a double buffer for the playout wouldn't it be possible to choose only one physical playout buffer and parse the captured data in right at the interrupt. It's unlikely that any code could be fast enough to write the entire buffer before the hardware starts reading from the buffer; in fact, the hardware is likely to read the first sample from the buffer at the same time it is triggering the interrupt for the previous period. To reduce output latency, just reduce the output buffer size. For example, you could use a buffer with 128 samples (with 2 64-sample periods). HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Audio hardware with pause support
Florian Winter wrote: Is there another way to determine whether a certain hardware supports snd_pcm_pause without having to test the hardware? $ grep -rl SNDRV_PCM_INFO_PAUSE sound sound/arm/pxa2xx-pcm.c sound/arm/sa11xx-uda1341.c sound/core/pcm_native.c sound/drivers/vx/vx_pcm.c sound/pci/als300.c sound/pci/atiixp.c sound/pci/au88x0/au88x0_pcm.c sound/pci/cmipci.c sound/pci/cs4281.c sound/pci/cs5535audio/cs5535audio_pcm.c sound/pci/echoaudio/darla20.c sound/pci/echoaudio/darla24.c sound/pci/echoaudio/echo3g.c sound/pci/echoaudio/gina20.c sound/pci/echoaudio/gina24.c sound/pci/echoaudio/indigo.c sound/pci/echoaudio/indigodj.c sound/pci/echoaudio/indigoio.c sound/pci/echoaudio/layla20.c sound/pci/echoaudio/layla24.c sound/pci/echoaudio/mia.c sound/pci/echoaudio/mona.c sound/pci/emu10k1/emupcm.c sound/pci/ens1370.c sound/pci/es1968.c sound/pci/fm801.c sound/pci/hda/hda_intel.c sound/pci/ice1712/ice1712.c sound/pci/ice1712/ice1724.c sound/pci/intel8x0.c sound/pci/intel8x0m.c sound/pci/maestro3.c sound/pci/mixart/mixart.c sound/pci/nm256/nm256.c sound/pci/oxygen/oxygen_pcm.c sound/pci/pcxhr/pcxhr.c sound/pci/riptide/riptide.c sound/pci/rme32.c sound/pci/rme96.c sound/pci/trident/trident_main.c sound/pci/via82xx.c sound/pci/via82xx_modem.c sound/pci/ymfpci/ymfpci_main.c sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c sound/soc/at91/at91-pcm.c sound/soc/davinci/davinci-pcm.c sound/soc/omap/omap-pcm.c sound/soc/pxa/pxa2xx-pcm.c sound/soc/s3c24xx/s3c24xx-pcm.c sound/usb/usbaudio.c Please note that the dmix plugin does not support pausing even if the hardware device does. HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Combined USB-Serial Serial-MIDI device
Rangel Reale wrote: modprobe snd-serialmidi sdev=/dev/ttyUSB0 Hmm the serialmidi driver seems to be broken with the kernel Ubuntu 8.04 uses (2.6.24-17-generic) (I had to enable CONFIG_BROKEN on configure to be allowed to compile it) /usr/src/alsa-driver-1.0.16/drivers/serialmidi.c:117: error: conflicting types for ‘tty_ioctl’ Well, there is a reason it's marked as broken ... It should be possible to fix this driver, or to write a user-space driver. I'll see what I can do. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Info on edirol UA-101
Germano Carella wrote: I have an edirol UA-101 usb sound card. The UA-101 is not fully supported; capturing may work, but the driver does not correctly synchronize to the device's sample frequency when playing, and there are no mixer controls. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Looking for firmware lists
Karl Schmidt wrote: The linux kernel is no longer accepting closed firmware blobs The kernel does contain sourceless firmware blobs. Some distributions (Debian and derivatives) remove such drivers from the kernel. A list of audio chip sets that require firmware that is not GPL friendly CS46xx A list of audio chip sets that don't require firmware because they are GPL friendly. All others, although I'm not quite sure what you mean by GPL friendly. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Alsa error with 2.6.24 kernel
Mr. Man wrote: Dmesg says: Maestro3: probe of :02:03.0 failed with error -2 -2 means no such file or directory. I'd guess that it did not find the firmware files. Either enable CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL when compiling the kernel, or install the alsa-firmware package. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] random hang when using latest snd-oxygen driver
Peter A. Friend wrote: I have a bGears b-Enspirer card supported by the snd-oxygen driver of the latest ALSA release. Things work fine for a short time then the machine hangs and I am forced to reboot. ... Usually the system hangs when adjusting the volume or skipping tracks in Rhythmbox or Amarok. Which volume, playback or recording? Does this ever happen during normal playback, i.e., when you don't change mixer controls or start/stop something? Are there ever any messages from the driver in the system log? System is a quad core Xeon with 8 gigs of ram. Could you try with a single-processor kernel? Regards, Clemens - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd_config_search_alias_hooks
Jerry Geis wrote: When compiling Mplayer rc1 I am getting a link error about undefined reference to snd_config_search_alias_hooks() That code was changed over a year ago. Use rc2. HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA driver compile error (hg)
Stephen Stocker wrote: With a fresh checkout of the hg repository (alsa-driver and alsa-kernel), I'm getting a compile error. ../alsa-kernel/core/vmaster.c:258: error: parse error before this_object_must_be_defined_as_export_objs_in_the_Makefile Now fixed: http://hg.alsa-project.org/alsa-driver/rev/bd497c013329 HTH Clemens - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Redirecting Mic Input in software (no arecord / aplay involved)
I wrote: However, an even better solution for your problem would be to enable the input monitoring function of your sound card. Okay, monitoring is now supported, with this patch: http://hg.alsa-project.org/alsa-kernel/rev/0fff88f7f05b To enable monitoring from a script, run amixer cset name=Analog Input Monitor Switch on or name=Analog Input Monitor Switch,index=1 for the second analog input. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Trying to get arecord/aplay to work with M-Audio
Helge Fredriksen wrote: I just bought a couple of MAudio cards to test out (Revolution 5.1 and Audiophile 2496). I'm trying to test the sound using arecord and aplay, but no matter which format I choose I get the message: aplay: set_params:900: Sample format non available Use aplay -v something.wav, then look into what format it gets converted. HTH Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] S32_LE format
John Sigler wrote: format : S32_LE msbits : 24 ... The first sample is stored as 00 00 a5 f5. Does this represent the value 0xf5a5? Yes. For the majority of samples, the first two bytes are 00 00. Is this because the input stream was, in fact, a 16-bit stream? Apparently. For the rest of the samples, the first two byte are 08 00. Why don't all the samples start with 00 00? Does this sequence 08 00 have a special meaning? Since only the upper 24 bits are used for sample values, the lowest eight bits are undefined. The sound card can set them to any value, although most chips do set them to zero. I don't know why the HDSP sets this bit in about half of the samples, it may be some form of dithering. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
