On 18-12-12 14:13, Clemens Ladisch wrote:
> Rene Herman wrote:
>> I am setting up basic, "manual" network-transparency on my local network
>> using alsa. Getting sound data from "server" to "player" is easily
>> accomplished with the "file&
Good day.
I am setting up basic, "manual" network-transparency on my local network
using alsa. Getting sound data from "server" to "player" is easily
accomplished with the "file" plugin and netcat:
[rene@server ~]$ cat .asoundrc
pcm.player {
type file
slave.pcm "null"
On 23-08-08 00:00, James Shatto wrote:
>>> mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob
>> I'd have figured that out :-) Didn't work with the 96/24 audio...
>
> Well, you could probably do the arecord method.
>
> arecord -D copy -t wav -c 2 -f S24_BE -r 96000 audio_track.wav
> (unv
On 22-08-08 23:44, Grant wrote:
>>> http://dvd-audio.sourceforge.net/
>> Interesting. Thanks.
>
> Check this out:
>
> http://dvd-a.info/Index.asp?stitle=neil+young
>
> 2 pages of mainly Neil Young DVD-A releases all containing tracks up
> to 24bit/176khz. Lots of people are ripping their DVD-
On 22-08-08 23:44, James Shatto wrote:
>> Those are the two I have here. mplayer was giving me endless grief
>> actually ripping the tracks from the DVD so I haven't yet done that in
>> fact, but:
>
> mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob
I'd have figured that out :-) Didn'
On 22-08-08 23:09, Grant wrote:
> Can you just mount the DVD and cp the tracks over? Probably not or
> you would have done that.
Not as individual tracks no. It's a regular DVD-Video, so you can copy
over the VOBs ofcourse, but piecing things together requires processing
them from harddisk the
On 22-08-08 20:58, Grant wrote:
>>> Recording wise 24 bit gives you a greater dynamic sampling range. But
>>> for most people it's not needed for playback as your CDs and DVDs
>>> are already in a 16 bit format. And the benefit on playback is
>>> minimal. Aside from generally being a better soundca
On 22-08-08 17:37, Grant wrote:
>>> I'm going to donate a 24-bit capable sound card to the mpd project to
>>> add 24-bit support. Can anyone recommend an inexpensive one that is
>>> 24-bit capable and PCI, PCIe, or USB?
>> For inexpensive, I believe one of the CMI cards would be a good choice.
>>
On 22-08-08 15:07, James Shatto wrote:
> Recording wise 24 bit gives you a greater dynamic sampling range. But
> for most people it's not needed for playback as your CDs and DVDs
> are already in a 16 bit format. And the benefit on playback is
> minimal. Aside from generally being a better soundca
On 21-08-08 16:15, Grant wrote:
> I'm going to donate a 24-bit capable sound card to the mpd project to
> add 24-bit support. Can anyone recommend an inexpensive one that is
> 24-bit capable and PCI, PCIe, or USB?
For inexpensive, I believe one of the CMI cards would be a good choice.
Note, I h
On 19-08-08 03:32, Wesley Johnson wrote:
> I know, ALSA in the kernel in Linux 2.6 ! Havn't you noticed how
> many companies are keeping old "obsolete" versions of equipment
> running.
Definitely. And, like you and as said, they are for the most part on
their own for support as far as the gener
On 17-08-08 04:27, Phil Howard wrote:
> I'm building a monolithic Linux kernel, which means no modules. I'd
> like to find out how I can upgrade ALSA in a kernel source tree so I
> can build a complete kernel image with the latest version of ALSA
> (1.0.17) on the latest Linux kernel (2.6.26.2) th
On 17-08-08 01:03, Wesley Johnson wrote:
> I did not see any typo in my email address, and your email did get
> here, using the same address.
Yes, because I clicked on the link and completed the captcha I was then
presented with. I'll probably get another one of those for this message
but will
On 15-08-08 09:19, Rene Herman wrote:
> On 14-08-08 22:36, Wesley Johnson wrote:
>
>> How do you notify ALSA dev team about a bug that has been in 1.0.16
>> and 1.0.17. ???
>
> Here will do, although the alsa-devel mailing list would've been better
> still. Ad
On 14-08-08 22:36, Wesley Johnson wrote:
> How do you notify ALSA dev team about a bug that has been in 1.0.16
> and 1.0.17. ???
Here will do, although the alsa-devel mailing list would've been better
still. Added to CC.
> Bugtracker will not let me login, nor email an account confirmation.
>
On 31-07-08 00:39, Jason Hitt wrote:
I gave this one a shot and had the same results as before. No logging
beyond the ACPI line listed before, and no audio. Could there be
something more basic I'm missing here? It's almost as if the card never
leaves mute (though i do get a loud popping whe
On 29-07-08 20:56, Rene Herman wrote:
> Here's the next attempt. Again applies to a virgin (2.6.26) ens1370.c.
> If it's easier, the entire file as it should now look after this patch
> is at:
>
> http://members.home.nl/rene.herman/ens1371/ens1370-src-delay.c
Whil
On 30-07-08 16:29, Media Fan wrote:
> oops, kernel reported kernel bug, and the attempt to run
> 'system>preference>hardware>sound' hang the system. dmesg is attached.
Thanks. That's definitely not caused by my last patch though -- it just
added 4 udelay(1) calls in the codec communication path
On 22-07-08 09:56, Jason Hitt wrote:
Rene Herman keyaccess.nl> writes:
If you'd be able to test the new patch (I'll attach it again here for
convenience) that would be interesting.
Well, not as interesting as you may have hoped. I reverted the file and applied
the new pat
On 29-07-08 19:16, Roman Haefeli wrote:
> thanks for the hint. i will follow your advice and post there as well.
> however, the reason, why i posted here first, was, because jack works
> all well here. i suspect the problem to be on the alsa side, since the
> error i get is certainly an alsa speci
On 29-07-08 16:22, Roman Haefeli wrote:
> i would like to route the audio from firefox' flash plugin to jack in
> whatever way possible. on ubuntu dapper, i could start firefox with some
> environment variable, that forced the flash plugin to use OSS instead of
> alsa, so that i could route its ou
e...
Rene.
>From 86d9e8d586df4d5065464db57506b75c999a5647 Mon Sep 17 00:00:00 2001
From: Rene Herman <[EMAIL PROTECTED]>
Date: Mon, 21 Jul 2008 21:57:13 +0200
Subject: [PATCH] ALSA: ens1371: replace src_mutex by the reg_lock spinlock
Signed-off-by: Rene Herman <[EMAIL PROTECTED]>
--
It's again against kernel
2.6.26/2.6.25.x (and/or alsa-driver-1.0.17 which I saw you use) same as
the previous patch since you said you reverted that again.
No rush...
Rene.
>From 86d9e8d586df4d5065464db57506b75c999a5647 Mon Sep 17 00:00:00 2001
From: Rene Herman <[EMAIL PROTECTED]>
On 20-07-08 02:56, Media Fan wrote:
Just so so at them, but now I have to put this issue aside due to
limited bandwidth. Sorry about this.
Attached you'll find a patch that (with a wide brush) adds delays after
any and all AC97 related accesses. Was generated against 2.6.26 and
should appl
On 19-07-08 17:02, Media Fan wrote:
> Thanks a lot for your help, Stan. Sorry for my late reply. I'll pay
> attention to that thread.
That one just ended. I wasn't paying attention to this one due to the
fact that halfway in you reported that it worked but I see it's the
exact same issue. Also
On 18-07-08 22:11, Landis McGauhey wrote:
> > From: [EMAIL PROTECTED]
> > Were you in the correct subdirectory when running 'patch' ?
> Yes, I was. Thanks for confirming this.
No you weren't. But yes, I'll leave this issue be. It can't be debugged
this way. Please note that when/if you upgr
On 16-07-08 20:50, Rene Herman wrote:
Moreover, the AC97 codec seems integrated in the EV1938. Just found a
patch for FreeBSD following up a similar report for your card. Will
look at it tomorrow.
Nothing interesting. Ready to give up on this. There is a communication
problem with the AC97
On 16-07-08 22:30, Sergei Steshenko wrote:
> Maybe Rene meant '-R' (reverse patch) rather than '-r rejects_file'.
Hey, I said I made at least one error per post -- there wouldn't be any
logic in that all of a sudden not being so for the 3rd correction to
another post since that, obviously, is a
On 16-07-08 20:00, Landis McGauhey wrote:
> OK, this is interesting:
>
> # patch -p1 -r < ens1371-ac97_reset_hack.diff=
> bash:ens1371-ac97_reset_hack.diff: No such file or directory
>
> # patch -p1 -r < ens1371-ac97.diff
> bash: ens1371-ac97.diff: No such file or directory
>
> Go figure.
Of c
On 16-07-08 16:44, Rene Herman wrote:
> On 16-07-08 16:40, Rene Herman wrote:
>
>> # moodprobe -r ens1371-ac97_reset_hack.diff
>> # moodprobe -r ens1371-ac97.diff
>
> I'm getting really sick of the fact that I seem to need to make at least
> one typo or other
On 16-07-08 16:40, Rene Herman wrote:
> # moodprobe -r ens1371-ac97_reset_hack.diff
> # moodprobe -r ens1371-ac97.diff
I'm getting really sick of the fact that I seem to need to make at least
one typo or other small error per post. "modprobe&
On 16-07-08 15:38, Landis McGauhey wrote:
> "pop" in speaker as alsaconf loads snd-ens1371
>
> running alsamixer-- no mute toggle underneath mic, that's a first
>
> and just so you know, there's no mute toggle under "master",
> "master-m", and "PCM", either, but that's nominal. There was one
On 15-07-08 22:28, Landis McGauhey wrote:
Yes, I did delete '-dry-run'.
In fact, just to be doubly certain, I just re-ran the whole process and
I still worry a little bit, since if all's well, you should have seen
the "patch -p1" command fail this time (it commenting that the patch
seemed a
On 15-07-08 16:42, Rene Herman wrote:
> On 15-07-08 16:19, Takashi Iwai wrote:
>
>> The below is a patch to improve the codec access routines in a bit
>> more robust way (and clean-ups, too). Give it a try.
>
> Thank you for taking this...
>
> Landis, if it&#
On 15-07-08 18:45, Landis McGauhey wrote:
> # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0=
> 0-0/0: 0x76058384 F�S
[ ... ]
> 0:7c =
> 0:7e = 8384
>
>
> Will do blacklist as recommended in earlier message, then reboot and see what
> happens.
Nah, don't bother. If you are sure the displa
On 15-07-08 17:25, Landis McGauhey wrote:
>
>
> > Date: Tue, 15 Jul 2008 16:42:05 +0200
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > CC: [EMAIL PROTECTED]; alsa-user@lists.sourceforge.net;
> [EMAIL PROTECTED]
> > Subject: Re: [Alsa-user] is this card supported by ALSA?
>
>
>
On 15-07-08 17:01, Sergei Steshenko wrote:
> And that's why something equivalent to 'ndiswrapper' for audio is
> necessary.
Please note -- if you're _trying_ to be a troll, it's too obvious to
have the fun last longer than this message.
Rene.
--
On 15-07-08 16:52, Landis McGauhey wrote:
> the result as soon as possible. I really appreciate this help-- this is
> the type of community that makes me a Linux user-- never would any help
> like this be available in the WinTel world.
Nor would the bug though... ;-/
Rene.
--
On 15-07-08 16:19, Takashi Iwai wrote:
The below is a patch to improve the codec access routines in a bit
more robust way (and clean-ups, too). Give it a try.
Thank you for taking this...
Landis, if it's easier for you due to webmail stuff, I'm attaching the
patch to this message so that it
On 15-07-08 01:36, Landis McGauhey wrote:
It seems there's just a bit too much oddness going on. Takashi, you know
more about ac97. Also bringing in alsa-devel...
> # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0=
>
> 0-0/0: 0x76058384 F�S
Eep? A 0x83847605 would be a SigmaTel STAC9704. And:
[
On 11-07-08 22:26, Landis McGauhey wrote:
> Thanks again, Rene. I'm really glad to hear your prognosis that we
> should be able to get ALSA going. Looking forward to your guidance
> so we can bring that about.
Just now looking. I'm actually new to the AC97 code (way too modern for
me...) so w
On 12-07-08 20:19, stan wrote:
> Rene Herman wrote:
>
> -snip-
>> At that point, you are running "real" OSS, not the ALSA OSS
>> emulation, and the "mplayer -ao oss foo.wav" should work again
>> (aplay nor alsamixer would anymore). This stands
On 11-07-08 03:17, Landis McGauhey wrote:
> Thanks! Actually it was "Southern Cross" (one of my favorites even
> though I'm 52 years old, old-enough to remember their vintage stuff
> from 1969 and the early seventies) though I don't know if that's on
> "Daylight Again".
Heh, yes, it is. Third
On 11-07-08 03:01, Landis McGauhey wrote:
> Success! I'm listening to Crosby, Stills & Nash on xmms. Thanks,
> Rene!
Hope it's "Daylight Again". Love that album...
But, you're not done yet! This was basically just debugging. The fact
that you now hear sound means that we should be able to get t
On 11-07-08 02:07, Landis McGauhey wrote:
> Success!
> # cd /usr/src/linux-2.6.25.9
> bezdomny:/usr/src/linux-2.6.25.9# patch -p1 --dry-run <
> /home/bezdomny/0001-OSS-resurrect-ES1371-driver.patch
> patching file sound/oss/Kconfig
> patching file sound/oss/Makefile
> patching file sound/oss/es13
On 11-07-08 01:44, Landis McGauhey wrote:
>>> Ready to proceed with the patch. Where is the root of the source tree?
>> That's something only you know. You are running a self-compiled 2.6.25.9
>> currently, right? Can't find the forum on which I saw you say so earlier
>> again...
>>
>> You dow
On 11-07-08 01:25, Landis McGauhey wrote:
>>> The attached is in the usual Linux format for sending kernel changes, "a
>>> patch". To apply it, you save the attached file somewhere, then change
>>> to the root of your 2.6.25.x source tree and type:
>>>
>>> $ patch -p1 --dry-run < 0001-OSS-resurr
On 10-07-08 23:20, Landis McGauhey wrote:
> Please bear with me, Rene-- I really want to fix ALSA in this
> installation and the expertise of ALSA mailing list is my last, best
> hope.
Bearing... an old bug report I found implied that this problem might be
in the ac97 code but before I go chas
On 10-07-08 16:00, Arthur Marsh wrote:
> By running the "Compaq Setup for Portables" boot floppy (delayed by my
> original floppy going bad /-:), I could reset the ESS1869 to use DMA
> channels 0 and 1, which results in an automatically working set-up (-:.
>
> Thanks for all the helpful suggest
On 10-07-08 14:12, Landis McGauhey wrote:
> [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or
> directory
This should not be if you made sure you loaded the snd-pcm-oss and
snd-mixer-oss modules (udev should've created the /dev/dsp node).
In any case -- I am afraid I'll h
On 10-07-08 05:59, Landis McGauhey wrote:
> sox /usr/share/sounds/startup3.wav -t ossdsp /dev/dsp=
> sox soxio: Can't open output file `/dev/dsp': unknown file type `ossdsp'
Please mutilate your distribution's "sox" packager for compiling without
OSS support.
mplayer -ao oss foo.wav
Rene
On 10-07-08 03:01, Arthur Marsh wrote:
> Sound is now working with the following setting:
>
> $ cat /proc/asound/cards
> 0 [ES1869 ]: ES1869 - ESS AudioDrive ES1869
>ESS AudioDrive ES1869 at 0x220, irq 5, dma1 1, dma2 5
>
> but there is some spurious clicking af
On 10-07-08 01:46, Landis McGauhey wrote:
> Rene Herman keyaccess.nl> writes:
>
>> Off for now so I'll hear it some of this worked later, but "oh", where
>> did that ">" come from? Just:
>>
>> aplay -D front foo.wav
>>
>>
On 10-07-08 01:24, Arthur Marsh wrote:
> Hi, I am having trouble getting sound working in an old Compaq Armada
> 1750 with an ESS AudioDrive ES1869.
>
> Even when it's working (verified by aplay some-sound.wav), KDE 3.5.9
> sound doesn't always work )-:.
KDE 3.5.9 uses aRts which, I'm sorry to
On 08-07-08 22:40, Rene Herman wrote:
> On 08-07-08 22:33, Landis McGauhey wrote:
>
>> Thanks, Rene. Yes, I'm willing to try recompiling the kernel-- with
>> guidance.
>>
>> uname -r= 2.6.25.9
>>
>> modprobe snd-pcm-oss snd-mixer-oss= FATAL: Error
On 08-07-08 22:33, Landis McGauhey wrote:
> Thanks, Rene. Yes, I'm willing to try recompiling the kernel-- with guidance.
>
> uname -r= 2.6.25.9
>
> modprobe snd-pcm-oss snd-mixer-oss= FATAL: Error inserting snd_pcm_oss
> (/lib/modules/2.6.25.9/kernel/sound/acore/oss/snd-pcm-oss.ko): Unknown sy
On 08-07-08 22:32, Rene Herman wrote:
> On 08-07-08 20:00, Rene Herman wrote:
>
>> On 08-07-08 16:22, Landis McGauhey wrote:
>>
>>> Yes, ran alsamixer and thoroughly checked the settings-- levels all
>>> raised, no muting, made sure the correct parameter for t
On 08-07-08 16:54, stan wrote:
> Owen.yan wrote:
>> And another question: can alsa mix its playback and capture data? If
>> can, which tool does it use?
>>
> You have to tell it to do this through some external means. In a recent
> post, one suggestion was
>
> arecord -f | aplay
>
> arecord
On 08-07-08 16:22, Landis McGauhey wrote:
> Yes, ran alsamixer and thoroughly checked the settings-- levels all
> raised, no muting, made sure the correct parameter for the right card
> was used according to cat /proc/asound/cards. No sound.
>
> Played a file with aplay. No sound.
Unfortunatel
On 08-07-08 01:12, Landis McGauhey wrote:
> PS to Rene & Sergei-- your comments about amplified speaker set are
> well-taken, so I set up my kids amplified speaker set during this
> latest round of testing-- no sound except for the occasional loud
> squeal at shutdown mentioned already in my reply
On 06-07-08 00:49, Landis McGauhey wrote:
> Windows Device Manager calls it Creative SB audioPCI.
>
> cat /proc/asound/cards calls it ENS1371 - Ensoniq AudioPCI Ensoniq
> AudioPCI ENS1371.
>
> ALSACONF calls it ens1371 Creative Labs Ectiva EV1938.
>
> I'm confused! Can anyone tell me if this c
On 25-06-08 09:24, Norbert van Bolhuis wrote:
> I would like to test an ALSA driver by capturing audio
> and immediately play the captured audio (loopback).
>
> It's an embedded driver and the still to be developed audio app
> will interface to the driver via ALSA-LIB.
> therefore I would like to
On 23-06-08 01:01, Jerry Geis wrote:
> I am user root at this time when trying to use aplay.
>
> drwxr-xr-x 2 root root 160 Jun 22 15:49 /dev/snd
> ls -l /dev/snd
> total 0
> crw-rw 1 root root 116, 0 Jun 22 15:49 controlC0
> crw-rw 1 root root 116, 24 Jun 22 15:49 pcmC0D0c
> crw-rw--
On 22-06-08 22:10, Jerry Geis wrote:
> play wavefile.wav works fine
"play" is using the OSS interface ...
> aplay wavefile.wav shows this:
>
> ALSA lib pcm_dmix.c:1046:(snd_pcm_dmix_open) unable to open slave
> aplay: main:564: audio open error: Invalid argument
... and this might indicate a p
On 12-06-08 17:09, Rene Herman wrote:
> On 12-06-08 16:14, Florian Winter wrote:
>
>> I have some more questions:
>> Is there a way to find out which soundcards support hardware mixing (and
>> have support for it implemented in their corresponding ALSA drivers)?
>
On 12-06-08 16:14, Florian Winter wrote:
> I have some more questions:
> Is there a way to find out which soundcards support hardware mixing (and
> have support for it implemented in their corresponding ALSA drivers)?
Not all that easily from the code. The ALSA soundcard matrix at:
http://www.a
On 12-06-08 08:28, Demian Martin wrote:
> I don't think we are in disagreement in substance. I was trying to
> give a larger framework for all of those not as familiar with the
> general workings and decisions behind computer audio as currently
> implemented. "Just works" is very important to most
On 12-06-08 05:52, Rene Herman wrote:
Oh, by the way, upon rereading:
> On 11-06-08 17:41, Dominique Michel wrote:
>> You must add some device definitions in /etc/modules.d/alsa (or
>> whatever file
>> your distribution is using):
>>
>> ## ALSA portion
>&
On 12-06-08 07:17, Pete Black wrote:
> dmix *is* an application, conceptually it is not different [ ... ]
Let's call it a conceptlication then. As said, your basic description
was correct.
Rene.
-
Check out the new SourceF
On 12-06-08 07:13, Demian Martin wrote:
> Computer audio and sample rate issues are popping up everywhere, driven
> by the desire for high quality audio on PC's finally. On Windows and
> Mac's its even harder to get it right.
>
> In Alsa (and PC audio architecture in general) the system has a d
On 12-06-08 06:53, Pete Black wrote:
> Its very simple.
>
> Most sound devices support a number of sample rates. Common ones
> include 16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc.
>
> Only one application has exclusive control over the sound hardware at
> any time.
>
> Whatever rate t
On 12-06-08 06:30, Sergei Steshenko wrote:
> Yes again - to me ALSA's sample rate implementation looks quite
> illogical - IMO it should be the other way round - user first
> mandates sample rate, and then playback sources adapt through
> resampling if necessary.
Great setup once we have infinite
On 12-06-08 05:52, Sergei Steshenko wrote:
> From: Rene Herman <[EMAIL PROTECTED]>
> The sampling rate is a property inherent to the data.
> Are trying to tell me that sample rate is inherent to analog source connected
> to microphone or line input ?
Oh, not again... pleas
On 11-06-08 17:41, Dominique Michel wrote:
> Le Tue, 10 Jun 2008 17:47:58 +0300,
> "alexander merkulov" <[EMAIL PROTECTED]> a écrit :
>
>> need to setup 2nd dummy card
>> how to do it?
>
> You must add some device definitions in /etc/modules.d/alsa (or whatever file
> your distribution is using)
On 12-06-08 02:17, Sergei Steshenko wrote:
> From: Grant <[EMAIL PROTECTED]>
>> Where is my output sample rate defined? I'm trying to make sure mpd
>> isn't resampling my music before it's sent to the USB DAC.
> I initiated a similar thread recently.
>
> The short answer - nowhere.
>
> As I w
On 12-06-08 00:35, Grant wrote:
> Where is my output sample rate defined? I'm trying to make sure mpd
> isn't resampling my music before it's sent to the USB DAC.
Check /proc/asound/card0/pcm0p/sub0/hw_params while mpd is playing (for
a suitable value of (0,0,0) ofcourse).
Rene.
-
On 10-06-08 16:47, alexander merkulov wrote:
> need to setup 2nd dummy card
> how to do it?
modprobe snd-dummy enable=1,1
Rene.
-
Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell services fo
On 06-06-08 12:47, Juan A Fuentes Bermudez wrote:
> hi, sorry for my englihs
>
> how can set permanent order of this devices:
>
> AD1988 (SND_HDA_INTEL)
> HDMI, integrated in my graphic card ATI (SND_HDA_INTEL)
> Waveterminal 192m (SND_ICE1724)
> EWX24/96 (SND_ICE1712)
>
> now in ubuntu hard
On 22-05-08 14:01, Arthur Marsh wrote:
> under Debian unstable, by overwriting the unpacked Debian source tarball.
I'm afraid you lost me here. I have little clue about Debian.
Rene.
-
This SF.net email is sponsored by: Mic
On 14-05-08 23:55, Eliot Blennerhassett wrote:
> When you said
> "The linux kernel is no longer accepting closed firmware blobs"
> did you mean to imply that this is new policy?
> If so, can you give a reference to where this is documented eg LKML
> archive reference.
>
>
> thanks and regards
>
On 13-05-08 13:32, Rene Herman wrote:
> On 13-05-08 13:20, Takashi Iwai wrote:
>> Anyway, the problem of beep on Dell XPS is a different. The PC beep
>> isn't implemented (initialized) in the sound driver side. I have no
>> interest in fixing it as I hate PC beep
On 13-05-08 13:20, Takashi Iwai wrote:
>> Yes, that would seem to be amazingly clumsily done. Perhaps there
>> was a reason (adding alsa-devel).
>
> Since snd-pcsp itself provides the input pcspkr functionality, it
> replaces the input pcspkr driver.
Ah. Yes, then it starts to make sense.
> An
On 13-05-08 12:44, Armin ranjbar wrote:
>> Device Drivers ->
>> Input device support ->
>> Miscelaneous devices
>>
>> With current mainline, it's:
>>
>> General setup ->
>> Configure standard kernel features (for small systems)
>> Enable PC-Speaker support
>
> Thank you very much for
On 13-05-08 08:33, Armin ranjbar wrote:
> this is my lspci of soundcard on dell xps m1330 :
>
> 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio
> Controller (rev 02) Subsystem: Dell Unknown device 0209
> Flags: bus master, fast devsel, latency 0, IRQ 21
> Mem
On 27-04-08 22:47, Sergei Steshenko wrote:
> On Sun, 27 Apr 2008 22:30:46 +0200
> Rene Herman <[EMAIL PROTECTED]> wrote:
>> Makes no sense. Setting the sampling rate has no meaning outside of the
>> action of playing or recording.
>
> ALSA is needed only to playba
On 27-04-08 12:55, Sergei Steshenko wrote:
> On Sun, 27 Apr 2008 02:21:23 +0200 Rene Herman <[EMAIL PROTECTED]>
> wrote:
>> (the card = the M-Audio Revolution. No control is the expected
>> situation)
>
> ???
>
> For me it's the opposite - if a car
On 27-04-08 02:19, Rene Herman wrote:
>> Why mixers for M-Audio Revolution have sample rate control and mixers for
>> Intel HDA (on my two machines) do not have sample rate control ?
>
> No idea what that's about. Maybe the card resamples in hardcware or
> something
On 27-04-08 02:08, Sergei Steshenko wrote:
>>> How do I change sample rate ?
>> By opening the device with the desired paramters. aplay/arecord -r ,
>> specifically for the reference utilities.
> Suppose I want to run, say, 'mplayer' or 'vlc'.
>
> How do I change sample rate (the physical one,
On 27-04-08 01:51, Sergei Steshenko wrote:
> How do I change sample rate ?
By opening the device with the desired paramters. aplay/arecord -r ,
specifically for the reference utilities.
Rene
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On 24-04-08 21:13, Matt Garman wrote:
> In short, I'm looking for the best stereo sound I can get from my Linux
> system---the main use will be listening to my CD collection (ripped as
> FLAC).
>
> A specific question: if using digital output, is there a difference
> (sound quality-wise) between
On 17-04-08 20:41, Helge Fredriksen wrote:
> On Wed, Apr 16, 2008 at 8:59 PM, Rene Herman <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
> Set the desired default settings for the card on one system and then
> do a "alsactl store" once. Thi
On 16-04-08 09:07, Helge Fredriksen wrote:
> We're making our own specially tailored Linux based upon the Ubuntu
> package repository. The application running on the distro is a
> sound-enabled one, and it will always be setup with the same sound card.
> However, we cannot use the default settin
On 14-04-08 11:06, Rene Herman wrote:
> On 14-04-08 01:46, Sebastián Tobías Castro wrote:
>
>> I will try: ;)
>>
>> When GRUB is installed in MBR, alsa no found or recognize the pci
>> card... But when grub is not installed and boot with super grub di
On 14-04-08 01:46, Sebastián Tobías Castro wrote:
> I will try: ;)
>
> When GRUB is installed in MBR, alsa no found or recognize the pci
> card... But when grub is not installed and boot with super grub disk,
> TATA! Sound ok.
> why?
>
> Thaks all Sorry for the simple of my explanation.
>
On 13-04-08 21:12, Sebastián Tobías Castro wrote:
> Muchas gracias Alexander por la respuesa... y tan rápido!
Guys, only English on this list please.
Rene.
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On 26-03-08 23:57, Bill Unruh wrote:
> On Wed, 26 Mar 2008, Helge Fredriksen wrote:
>
>> Ok, here's what I found out:
>>
>> Since these cards seem to lack the support for doing frequency conversion in
>> HW, you simply need to use the alsa plugin
>> plughw instead of hw when referring to the card
On 18-03-08 14:02, Sergei Steshenko wrote:
> On Tue, 18 Mar 2008 13:44:59 +0100
> Rene Herman <[EMAIL PROTECTED]> wrote:
>
>> On 18-03-08 05:28, Alexander Indenbaum wrote:
>>
>>> OK, it works on Linux 2.6 but silently fails on Linux 2.4.
>>> U
On 18-03-08 05:28, Alexander Indenbaum wrote:
> OK, it works on Linux 2.6 but silently fails on Linux 2.4.
> Unfortunately, I need it to work on Linux 2.4 :)
> So what can be done?
Sorry, but I just don't care about 2.4. It's obsolete. This was a proof of
concept for you -- a final solution mig
On 18-03-08 01:27, Alexander Indenbaum wrote:
> Still no PCM in /tmp/pcm.out. I get following error messages though:
> "(snd_determine_driver) could not open control for card 0"
> "(_snd_config_evaluate) function snd_func_concat returned error: No such
> file or directory"
> "(snd_func_concat) er
On 17-03-08 16:07, Takashi Iwai wrote:
>>> Recently I added a new option "truncate" to file plugin. As default,
>>> it's set to true (for compatibility reason), and the plugin overwrite
>>> the existing file if reopened. When it's set to false, a new file is
>>> created with a different suffix (
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