One p0roblem with two sound cards is that the two frequencies may be out from
each other. Unless they are driven by the same crystal, if the two frequencies
are different by .01%, by a 1000 sec ( 15 min) they will be out by 1/10 of a
sec. Ie, one or the other will be producing sound 1/10 of a
On Tue, 29 Sep 2020, Ralf Mardorf wrote:
On Mon, 28 Sep 2020 22:02:10 -0400, Alan Corey wrote:
On 9/28/20, Zsolt Ero wrote:
I have a few questions related to arecord, which I couldn't find in
the man pages nor anywhere on the internet.
My use case is very simple, I'd like to record stereo
On Mon, 28 Sep 2020, Alan Corey wrote:
I would say you're in danger of overthinking it, try the defaults first.
And read the man pages.
Except when you say 24/192 I'm thinking 192 kbits/sec? My usb sound
I presume he means 24 bits per sample, 192K samples per second per channel
which
Is it clipping? (in audacity you can see that the amplitude goes right to the
top and bottom). 0 dB could well be clipping.
Do you have your laptop plugged in? Trying recording on
battery only.
William G. Unruh __| Canadian Institute for| Tel: +1(604)822-3273
Physics _|___ Advanced Research
On Sun, 22 Oct 2017, John Z. wrote:
I thought this whole thread was about stuff that PA could do and alsa on its
own could not.
Well, that'd be my mistake I guess. I should've been clearer and specify
in my opening post that I'd like to solve this using just alsa.
I assumed that this would
On Sun, 22 Oct 2017, John Z. wrote:
What is so bad about PA that an ALSA plugin would do better?
I wouldn't know, really, if there is anything bad to begin with?
Last two opinions that I read on the subject were that PA used to be bad
in the beginning but its great now, and that in general
And what is the article about, and why should we read it, other than that it
is nice? And why should alsa users be interested in Shorewall?
William G. Unruh __| Canadian Institute for| Tel: +1(604)822-3273
Physics _|___ Advanced Research _| Fax: +1(604)822-5324
UBC, Vancouver,BC _|_
I still do not know what the difference between the UCA222 and UCA 202 is,
except the latter is silver and the former red (and the latter is slightly
more expensive.)
There is also a version of the 202 which has a RIAA preamp built in (UFO202?)
This is for transfering vinyl to digital.
This is
On Fri, 27 May 2016, li...@lazygranch.com wrote:
> Scanner audio has an SNR of about 40dB on a good day. Bandwidth is about 4KHz.
Which means that the Behringer should be far more than adequate for you.
(actually it should even be more than adequate for almost everyone).
>
>
> Bi
On Fri, 27 May 2016, Martin Tarenskeen wrote:
>
>
> On Fri, 27 May 2016, li...@lazygranch.com wrote:
>
>> Is there a reasonably-priced USB card with stereo Line
>> input that works in Linux?
>
> If you want to try a really cheap, but really working one: I have a
> Behringer ACA222. Not good
On Fri, 27 May 2016, Martin McCormick wrote:
> Is there a reasonably-priced USB card with stereo Line
> input that works in Linux?
I am surprized. I would have thought that most sound cards at least with line
input, would do stereo. The microphone inputs will quite possibly be mono (
that if you use -s it only plays
once. No matter what -l says. Ie the man page says that -l only works for two
channel. Butr maybe the man page is wrong? I do not know.
Regards,
Anders
13 sep 2015 kl. 04:05 skrev Bill Unruh <un...@physics.ubc.ca>:
man speaker-test ?
if -s i
man speaker-test ?
if -s is used the program will always only produce a single shot.
SO use -l 0 (loop indefinitely)
and unplug one of the speakers
On Sat, 12 Sep 2015, Alan Bromborsky wrote:
> I am using alsa speaker-test to set my channel gain levels with a spl
> meter and alsamixer one
The cheapest analog hardware method to convert from balanced to
unbalanced requires two conditions: 1) the balanced output must
come from a transformer (coil of wire on a ferrous core); _AND_
2) you are willing to sacrifice a little noise floor in exchange
for economy. That solution is to
He stated that if he ran the signal left and right with the ground as common
ground, he got lots of noise.
If he subtracted L from R the noise disappeared.
Now, that may mean that there is common mode noise, in which case running the
balanced into an unbalanced would be very noisy, or the noise
I have run into a difference between the alsa structure on
git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/
and in the kernel 4.0 rc6.
In the tree on the former there seems to be stucture hdac_bus core, and thus
the patch for the Dell XPS13 ([PATCH] ALSA: hda/realtek - Support Dell headset
Message-
From: Bill Unruh [mailto:un...@physics.ubc.ca]
Sent: Thursday, April 02, 2015 8:24 AM
To: alsa-user@lists.sourceforge.net; alsa-de...@lists.sourceforge.net
Cc: Kailang
Subject: Bug in Dell XPS13
There is a patch in ALSA for the Dell XPS 13 HDA sound system.
http://mailman.alsa
There is a patch in ALSA for the Dell XPS 13 HDA sound system.
http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089789.html
This patch appears to have a bug. It refers to codec-core.version_id
where codec is a hda_codec structure. Unfortunately that structure has no core
member.
William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324
UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca
Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
On Sun, 8
On Sat, 7 Feb 2015, mcmurchy1917-a...@yahoo.co.uk wrote:
Hi all
I seem to have got caught in a bit of an impasse here. My motherboard is the
Asus-Z87-K and I want to record sound using audacity.
I've got this far -
Scenario 1 -
I have an .asoundrc file whereby I can hear sound through
JUst to clarify a bit. pulseaudio sits on top of alsa-- ie it needs alsa (or
oss) to be running properly in order for it to work. It takes audio streams
from the programs and mixes them together and sends them off to the alsa
drivers, etc (at least when it is working properly). This means that if,
, 14 Nov 2014, Alan McConnell wrote:
On Fri, Nov 14, 2014 at 10:29:50AM -0800, Bill Unruh wrote:
JUst to clarify a bit. pulseaudio sits on top of alsa-- ie it needs alsa (or
oss) to be running properly in order for it to work. It takes audio streams
from the programs and mixes them together
, 14 Nov 2014, Alan McConnell wrote:
On Fri, Nov 14, 2014 at 01:41:32PM -0800, Bill Unruh wrote:
Had you told us what it was we might have been able to explain
to you want it was trying to say.
Here it is:
E: [pulseaudio] authkey.c: Failed to truncate cookie file: Invalid argument
W
On Wed, 5 Nov 2014, John Smith wrote:
On Wed, 05 Nov 2014 08:45:16 +0100, Clemens Ladisch
cladi...@googlemail.com wrote:
John Smith wrote:
On Tue, 04 Nov 2014 08:44:17 +0100, Clemens Ladisch wrote:
John Smith wrote:
With outdated, you mean everything in the link
In my kernel libraries I see
kernel/sound/isa/snd-opl3sa2.ko.xz
kernel/sound/drivers/opl3/snd-opl3-lib.ko.xz
kernel/sound/drivers/opl3/snd-opl3-synth.ko.xz
kernel/sound/drivers/opl4/snd-opl4-lib.ko.xz
kernel/sound/drivers/opl4/snd-opl4-synth.ko.xz
but no snd-opl3-sa2
Are you sure you have the
, 5 Nov 2014, John Smith wrote:
On Tue, 04 Nov 2014 22:40:49 +0100, Bill Unruh un...@physics.ubc.ca wrote:
In my kernel libraries I see
kernel/sound/isa/snd-opl3sa2.ko.xz
kernel/sound/drivers/opl3/snd-opl3-lib.ko.xz
kernel/sound/drivers/opl3/snd-opl3-synth.ko.xz
kernel/sound/drivers/opl4/snd-opl4
William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324
UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca
Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
On Sun,
On Mon, 7 Jul 2014, Daniel Mack wrote:
On 07/07/2014 11:26 AM, David W. wrote:
On 07/07/14 07:31, Clemens Ladisch wrote:
Apparently, this device needs some vendor-specific magic to enable
recording.
Ah. Thanks for the info. It is an unconventional sound device . . .
Have any other USB
William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324
UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca
Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
On Sat, 3
On Sat, 3 May 2014, Antonis Kouzoupis wrote:
On Sat, May 03, 2014 at 09:14:34AM -0700, Bill Unruh wrote:
I always thought that was hardware, not software. Ie, the headphone plugin
contains a little switch which diverts the sound. But I could be wrong. Have
you had that same laptop actually
On Wed, 30 Apr 2014, chris hermansen wrote:
Clemens and list,
On Apr 30, 2014 12:07 AM, Clemens Ladisch cladi...@googlemail.com wrote:
Bill Unruh wrote:
On 29/04/14 15:52, Balduin Waldmeister wrote:
So my question is whether it's possible to add the low impedance support
(16
On Tue, 29 Apr 2014, michael norman wrote:
On 29/04/14 15:52, Balduin Waldmeister wrote:
Hello and THX for ALSA - I'm using it on Ubuntu/Linux Mint to get the
best out of my ASUS Xonar Essence STX sound card and it just works great.
The new STX *II* has (on Windows at least) an interesting
On Sun, 16 Feb 2014, Samuele Carcagno wrote:
Hi,
I'm trying to set up the e-mu 0204 for psychoacoustics research purposes on
Debian Wheezy.
When using aplay to play a short (200 ms, or 900 ms) wav file, at the onset
and at the offset of the sound
there are audible pops and clicks. I'm
with a constant that makes it trivial to switch
between the two.)
Yes, I've noticed this problem with PyAudio with other soundcards as well.
In another reply, Bill Unruh suggested routing the output through
the line input of another card. That is a very good idea and may
give you an idea of what's
On Sun, 16 Feb 2014, Samuele Carcagno wrote:
On Sunday 16 Feb 2014 20:47:44 you wrote:
I have never seen that in any of the files I have played. That hints that
that
is your input stream, rather than some problem with the soundcard itself,
although I have also never used your sound card.
On Sun, 2 Feb 2014, Ralf Mardorf wrote:
On Sun, 2014-02-02 at 00:35 +0100, Dominique Michel wrote:
You are a developer. That imply the best way you can get this fixed is
to stop to complain, fix it, and contribute your fix to ALSA.
JFTR this is an ALSA user mailing list and not an OSS or
On Sat, 1 Feb 2014, jon wrote:
...
On a more practical front cant the user just use pulse and the padsp
wrapper. Pulse is just an audio server (with mixing) that sits on top
of alsa, padsp is a wrapper for legacy audio applications that emulates
the original /dev/dsp, mixes down the audio
On Fri, 31 Jan 2014, Beojan Stanislaus wrote:
I am not a developer, just a user who was shocked by the tone of your
email. However I highly doubt that oss will be included in the kernel
again. This its because most applications on Linux have been written using
alsa, sand it appears oss hasn't
Oss never allowed mixing. This is an emulation of oss. It does not allow
mixing. If you want mixing uses alsa with a frontend. Or get the newer oss
implimentations.
On Wed, 29 Jan 2014, ChaosEsque Team wrote:
Why doesn't OSS emulation allow mixing. I got old OSS aps still blocking
/dev/dsp.
On Wed, 15 Jan 2014, David Vincent-Jones wrote:
Takashi Iwai tiwai at suse.de writes:
David Vincent-Jones wrote:
Reference:
http://www.alsa-project.org/db/?f=b5c0d636c3870da14ab5f4e4c6f5f64d7c906507
If this still doesn't work, it can be rather the way of testing.
This digital mic might
On Wed, 18 Dec 2013, Daniel Mack wrote:
What's the exact error that you get? What does
'aplay -Dhw:X -f cd /dev/urandom'
do (you should hear pink noise @0dB, so beware!).
White noise, not pink (equal power per delta f in Hz, not equal power per
octave). But yes, it could be extremely loud
On Mon, 2 Dec 2013, wemp...@gmail.com wrote:
On Mon, Dec 02, 2013 at 11:39:09AM +0100, Clemens Ladisch wrote:
wemp...@gmail.com wrote:
mplayer uses the default device by default, unless you have changed
this in ~/.mplayer/config.
Damn... Why do they do that? Why does some program make
On Sun, 1 Dec 2013, wemp...@gmail.com wrote:
On my way to become an advanced Linux user I started to learn
ALSA. However, it seems that ALSA, and sound system in general on
Linux, is in a pretty bad state at the moment. Here are the examples -
don't get me wrong, fix me if I got things wrong:
On Sun, 1 Dec 2013, wemp...@gmail.com wrote:
On Sun, Dec 01, 2013 at 11:03:09AM -0800, Bill Unruh wrote:
On Sun, 1 Dec 2013, wemp...@gmail.com wrote:
On my way to become an advanced Linux user I started to learn
ALSA. However, it seems that ALSA, and sound system in general on
Linux
On Sun, 1 Dec 2013, wemp...@gmail.com wrote:
True. But your statement is sort of like saying the wheel is engineered
incorrectly, when the wheel merrily missing lug nuts.
(Likely your distribution flavor has a bug, requiring fixing. Quite common,
especially with Debian. ;-)
Maybe, but
On Wed, 13 Nov 2013, Ruben De Smet wrote:
I tried on Ubuntu 13.10, doesn't work either. Any other suggestion?
R
On 10/27/2013 09:11 PM, Ruben De Smet wrote:
Anybody an idea? (aka. bump)
I'm sorry to bother you with this, but I'm not sure if it's hardware
or software related. As I said,
On Fri, 16 Aug 2013, giampaolo ferradini wrote:
*this message was bounced because I did not confirm my subscription, please
disregard if a duplicate *
Hi All,
I was wondering if you guys could help with a scratching audio in my sound
card.
Scratching in your audio? What does that mean?
: giampaolo_f
http://www.facebook.com/giampaolo.ferradinihttps://www.facebook.com/giampaolo.ferradini
On Fri, Aug 16, 2013 at 2:16 PM, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 16 Aug 2013, giampaolo ferradini wrote:
*this message was bounced because I did not confirm my subscription,
please
On Sun, 28 Jul 2013, YuGiOhJCJ Mailing-List wrote:
Yes, I confirm, what you hear is what I mean.
Do you see any way to do this with ffmpeg/alsa ?
The easiest way of course is to simply capture the stream that youare
displaying on your desktop. Ie, something is already sending a digitial
to capture what I hear during the record ?
On Sun, 28 Jul 2013 03:03:02 -0700 (PDT)
Bill Unruh un...@physics.ubc.ca wrote:
On Sun, 28 Jul 2013, YuGiOhJCJ Mailing-List wrote:
Yes, I confirm, what you hear is what I mean.
Do you see any way to do this with ffmpeg/alsa ?
The easiest way of course
On Sun, 28 Jul 2013, YuGiOhJCJ Mailing-List wrote:
Hello,
I am recording my desktop with ffmpeg 1.2 using ALSA 1.0.26.
What does recording my desktop mean? Why would you use ffmpeg to record?
The problem is I am recording audio from microphone whereas I would like to
record audio from
use which channels to use to record the
sound, or tell it to use all 18 stereo channels, or whatever.
As I said I have never seen or used the card, so have no idea how to use it to
best effect.
On Fri, May 3, 2013 at 4:29 PM, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 3 May 2013, Vincent
On Sat, 4 May 2013, Joe Armstrong wrote:
On Fri, May 3, 2013 at 9:24 PM, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 3 May 2013, Joe Armstrong wrote:
On Fri, May 3, 2013 at 7:39 PM, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 3 May 2013, Joe Armstrong wrote:
Try running aplay, etc
it.
all of them will be used at least in the beginning. Is this where the
asoundrc and/or -D hw:.. comes into picture?
Maybe.
You say you have installed the card and have a driver for it. What do you get
when you do
ls /dev/snd?
ls /proc/asound?
/ Vincent
On Thu, May 2, 2013 at 7:52 PM, Bill
On Fri, 3 May 2013, Clemens Ladisch wrote:
Joe Armstrong wrote:
This is what I did:
sudo adduer joe audio
What is adduer? If that is supposed to be adduser, that creates a new user
with that name.
Just edit /etc/group and put the user name into the audio group line. Or
check it now to see
output to see if there is anything there re
the sound card?
On Fri, May 3, 2013 at 6:45 PM, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 3 May 2013, Joe Armstrong wrote:
joe@nuc:~$ grep audio /etc/group
audio:x:29:pulse,joe
joe@nuc:~$ whoami
joe
Good. So now it works?
(Note that I
On Thu, 2 May 2013, Vincent Gulinao wrote:
Hi everyone,
I have a task to setup a system that will continuously capture multiple
stereo signals using a MADI audio card (RME HDSPe MADI) and write them into
files (perhaps in 1 hour chunks). Few checks I've learned while googling on
the topic
I have a Behringer uca222 soundcard, which handles 44100, 48000 and 16 bit
output/input. However if I run arecord, or aplay with illegal rates or bits,
they simply proceed as if nothing is wrong.
Eg
arecord -D default:CARD=CODEC /tmp/t -r96000 -c2 /tmp/t -N -d 2 -t raw -f S24_BE
simply records
On Wed, 24 Apr 2013, Daniel Mack wrote:
On 24.04.2013 19:04, Bill Unruh wrote:
I have a Behringer uca222 soundcard, which handles 44100, 48000 and 16 bit
output/input. However if I run arecord, or aplay with illegal rates or
bits,
they simply proceed as if nothing is wrong.
Eg
arecord -D
Have you looked at the output of
alsamixer?
is there a post dac volume control in that list?
I think that if it is not there, then it has not been implimented in alsa.
I do agree that the alsa documentation is pretty poor. On the other hand, to
expect detailed writeups for every sound card
On Sat, 23 Feb 2013, Alexandru Geana wrote:
Hello everyone,
I have installed a fresh Debian Wheezy on an Asus EEE 1001PX and I cannot
figure out how to get the darn microphone to work. I first noticed this
with Skype, but after reading forums and trying out stuff I stumbled upon
the arecord
On Fri, 22 Feb 2013, Daniel Mack wrote:
Hi Chard,
On 22.02.2013 11:48, Chard wrote:
I need to specify an arbitrary sampling rate via the tlv320aix3x codec.
However, on examination of the driver code, it appears that the code is
hard-wired to only accept rates of 44.1kHz and 48kHz - or
On Sun, 10 Feb 2013, chris hermansen wrote:
Good people;
I believe that I should report my sound card status on the Alsa sound
cards page, but before I do that I have a few questions.
My Audioquest DragonFly does not appear there, and I now believe it
This is somewhat off topic, but I am
On Sun, 27 Jan 2013, Vladimir Snigur wrote:
Hi all!
Seems like Quartet (ice1724 driver) has two stereo inputs, but my software
sees only one from each pair so that I have too choose in alsamixer which
input to use (PCM 1/2 or 3/4). I want though to use both of them. How can I
do that? I'd
On Fri, 30 Nov 2012, Fabio R�mi wrote:
Hey Clemens
Thank you very much for your reply!
On 30.11.2012 17:09, Clemens Ladisch wrote:
Fabio R�mi wrote:
modprobe: FATAL: Module snd-seq-midi not found.
Then it wasn't compiled.
It appears alsa-driver does not get the module configuration
On Thu, 26 Jul 2012, Torquil Macdonald S�rensen wrote:
Hi!
I'm having a problem with my microphone inputs, and the master volume doesn't
seem to be a master volume. In more detail:
Part A of problem (microphone bleed):
1) In my headphones, I can hear the signal picked up by the laptop
On Sun, 22 Jul 2012, Doug wrote:
I've asked a number of places, and haven't seen an answer--at least one
that didn't involve Pulse audio.
I have on-board sound--hw0 HDA Intel and a video card with sound
decoder out to HDMI port with video, hw1 HDA NVidia.
I have AlsaMixer, KMix,
On Tue, 17 Jul 2012, Konstantinos Birkos wrote:
Hello,
Is there a way to record consecutive overlapping tracks via a single
sound card? I mean tracks of constant duration overlapping a couple of
seconds with the next track.
I really have no idea what you are talking about. What kind of
(and track 1 and 4 overlap?)
On 07/17/2012 07:03 PM, Bill Unruh wrote:
On Tue, 17 Jul 2012, Konstantinos Birkos wrote:
Hello,
Is there a way to record consecutive overlapping tracks via a single
sound card? I mean tracks of constant duration overlapping a couple of
seconds with the next
On Tue, 26 Jun 2012, Dominique Michel wrote:
Le Mon, 25 Jun 2012 14:08:07 -0400,
Jerry Geis ge...@pagestation.com a écrit :
The best things to do would be to use something like a dbx, a
feedback suppressor. The problem is than they are expensive pieces
of hardware. The principle is simple:
On Mon, 25 Jun 2012, Doug wrote:
On 06/25/2012 08:34 AM, Jerry Geis wrote:
We are using a USB webcam with integrated USB mic.
When we use the USB microphone the volume level is really good
and we dont have to be right up to the micrphone, we can be some
distance away and still have good
William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
PhysicsAstronomy | Advanced Research | Fax: +1(604)822-5324
UBC, Vancouver,BC | Program in Cosmology | un...@physics.ubc.ca
Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
On Wed,
If you have not tried it, you might find that the Mic in also acts as a line
in as well. Some do.
On Thu, 3 May 2012, Sergei Steshenko wrote:
Mon, 30 Apr 2012 02:26:10 +0400 Vladimir Mosgalin
mosga...@vm10124.spb.edu:
Hi Sergei Steshenko!
On 2012.04.30 at 02:03:34 +0400, Sergei
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
Hello,
I see a number of relatively cheap soundcards on eBay:
http://www.ebay.com/itm/W449-USB-6-Channel-5-1-Audio-Sound-Card-S-PDIF-Exter-/250754613678?pt=LH_DefaultDomain_0hash=item3a6223c1ae
;
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
Sun, 29 Apr 2012 14:35:01 -0700 (PDT) Bill Unruh un...@physics.ubc.ca:
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
[snip]
(but why don't you just use the onboard soundcard on your computer?)
I'll sell you my computer and you'll know
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
Sun, 29 Apr 2012 14:58:35 -0700 (PDT) Bill Unruh un...@physics.ubc.ca:
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
[snip]
And you still have not told us why you donot just use the onboard soundcard.
[snip]
William G. Unruh
) Bill Unruh un...@physics.ubc.ca:
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
Sun, 29 Apr 2012 14:58:35 -0700 (PDT) Bill Unruh un...@physics.ubc.ca:
On Mon, 30 Apr 2012, Sergei Steshenko wrote:
[snip]
And you still have not told us why you donot just use the onboard soundcard
that the Linux people have never heard of it. I think you need to
give far more information before anyone can help you. Who the manufacturer of
the card is what the chipset is, What info you can glean from the windoes
driver, etc.
Best,
Sebastian
2012/4/2 Bill Unruh un...@physics.ubc.ca
On Mon, 2 Apr
On Mon, 16 Jan 2012, Sean M. Pappalardo - D.J. Pegasus wrote:
On 09/22/2011 06:40 PM, Daniel Mack wrote:
The only solution is to contribute to the thousands of
quirk mechanisms the driver already ships with and add yet another one
for this device. Did you try my patch?
Not yet. I'm
On Mon, 9 Jan 2012, Orion wrote:
Hi everybody
I recently upgraded to an Asus Xonar DS sound card from my old Creative
Labs Audigy 2 Value card (which worked very well until it developed a
mic problem which resulted in heavy interference on the input channel).
I haven't had as much luck
( or more
likely that the manufacturer refused to tell the alsa developers what the
various things about the card were all about, and they had to discover them
themselves by guessing and reverse engineering.)
Regards,
Orion
On Mon, 9 Jan 2012 10:23:36 -0800 (PST)
Bill Unruh un
On Sun, 13 Nov 2011, Chen Tao wrote:
Hi,
My sound card generates severe DC offset in recorded audio signal. Is there
a way to remove this offset from audio input, for example, using a
equalizer or filter?
How about a capacitor in the input?
LADSPA plugins can be used to filter audio
sound modules usually start with snd- You have none. Look in
/etc/modules.conf, /etc/modules.d/* -- especially the ones that start with
snd- -- for the sound modules and try reloading
them (eg modprobe snd-hda-intel to load the intel driver-- I am not saying
that is the right one for you, it is
pulseaudio is the new program which takes input from soundcards and delivers
it to the computer, and vice versa. Ie, it is an overall sound control
program. The best thing to do first is to disable pulseaudio and leave just
the bare soundcards. so that you can figure out what is going on. I would
On Wed, 20 Jul 2011, Nikhil Kamath wrote:
Hi ,
I have an iMX35PDK board with Linux2.6.35 running on it. I am getting the
following messages when I use aplay - aplay: pcm_write:1262: write error:
Input/output error.
Boot up log:
usbhid: USB HID core driver
sgtl5000-i2c 0-000a:
On Wed, 13 Jul 2011, Brian Pike wrote:
Hi,
I've got a Compaq Presario 12XL505 laptop with a VT82C686 sound chip.
Playing sound works fine. However, when I record sound using an external
microphone, there is static recorded along with the sound. The microphone
and microphone jack seem to be
/waveform.png
Thanks,
Brian Pike
On Wed, 13 Jul 2011, Bill Unruh wrote:
On Wed, 13 Jul 2011, Brian Pike wrote:
Hi,
I've got a Compaq Presario 12XL505 laptop with a VT82C686 sound chip.
Playing sound works fine. However, when I record sound using an
external
microphone
Of course it could be that some other program is trying ( and failing) to get
a hold of the sound card at that time but disturbing the system in theprocess.
So to be clear you have some external device (radio, amplifier, whatever)
feeding and audio signal to your sound card. YOu have a program
On Mon, 2 May 2011, Sergei Steshenko wrote:
On Mon, 2 May 2011 21:58:46 +0200
owl...@gmail.com owl...@gmail.com wrote:
Thank you Sergei but i don't really hunderstund if it is compatible.
http://www.behringer.com/EN/Products/UCA222.aspx
2011/5/2 Sergei Steshenko steshenko_ser...@list.ru
On Mon, 18 Apr 2011, cong fu wrote:
Hi all:
I want to record what I said in the mic and what I heard from the
speaker at the same time and into one file.
?? Perhaps if you told us what you were trying to do, we might be of more
help. Why in the world would you want to do that. Why can you not
On Mon, 18 Apr 2011, Sergei Steshenko wrote:
On Mon, 18 Apr 2011 00:29:20 -0700 (PDT)
Bill Unruh un...@physics.ubc.ca wrote:
[snip]
Why can you not mix them
afterwards, since the stuff going out your speaker surely originated from
your
computer anyway.
[snip]
Huh ? How about recording
the kind of mixing you are refering to. Again, it
would seem that you could always add in the background music afterwards (or
even if your system is playing the music it will come in through the
microphone with your voice anyway.)
2011/4/18 Bill Unruh un...@physics.ubc.ca
On Mon, 18 Apr 2011
On Fri, 8 Apr 2011, Sergei Steshenko wrote:
Hello,
I've tried to run 'arecord' as part of simultaneous playback + capture rig
(for acoustic measurements) and noticed overruns.
So, even plain single 'record' occasionally produces overruns:
arecord -D hw:0,2,0 -c 2 -r 96000 -d 6 -f S32_LE
On Sat, 9 Apr 2011, Sergei Steshenko wrote:
On Fri, 8 Apr 2011 09:48:04 -0700 (PDT)
Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 8 Apr 2011, Sergei Steshenko wrote:
Hello,
I've tried to run 'arecord' as part of simultaneous playback + capture rig
(for acoustic measurements) and noticed
On Sat, 9 Apr 2011, Sergei Steshenko wrote:
On Fri, 8 Apr 2011 22:34:40 -0500
James Shatto wwwshad...@gmail.com wrote:
Please reread my message in this thread on 'sox' - it contains the complete
command line I've used.
So apples to oranges? since your sox only does 4 seconds (trim 1 5)
On Sat, 9 Apr 2011, Sergei Steshenko wrote:
On Fri, 8 Apr 2011 21:12:52 -0700 (PDT)
Bill Unruh un...@physics.ubc.ca wrote:
[snip]
Sorry, you are feeding the ouput into the input?
[snip]
Yes, in reality this is what I'm doing - the intent is have a whole bunch
of analog electronics
On Sat, 9 Apr 2011, Sergei Steshenko wrote:
On Fri, 8 Apr 2011 21:55:11 -0700 (PDT)
Bill Unruh un...@physics.ubc.ca wrote:
[snip]
The microphones definitely do not have -90dB noise floor unless you are
recording stuff at 120-130dB SPL.
[snip]
The level might be 120-130dB SPL, but neither
On Wed, 2 Mar 2011, Paul Menzel wrote:
No problem. Please also do not top post to make it easier for people to
read up when joining into the thread later. ;-)
A: No.
Q: Should I include quotations after my reply?
Except that any self respecting news reader will produce
A: No.
Q: Should
On Sat, 26 Feb 2011, Friedrich Ewaldt wrote:
Hi Matt,
I didn't use a RME HDSP9632 for quite a long time (also I never used it
with the dmix plugin). However, the dmesg message sounds like a clock
source problem. All I can suggest is to check for the correct rate
settings, e.g. compare what
On Sat, 12 Feb 2011, Marcin Szyniszewski wrote:
On Sat, Feb 12, 2011 at 16:26, James Shatto wwwshad...@gmail.com wrote:
$ sudo depmod -a
$ sudo modprobe snd-hda-intel
WARNING: Error inserting snd_timer
(/lib/modules/2.6.35-25-generic/kernel/sound/acore/snd-timer.ko): Unknown
symbol in
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