snd-hda-intel enable_msi=1
Yes my microphone does actually work, yes I have tested it, yes it does
work in another O/S and in my older copy of fedora.
I also have a mixer channel center I don't want and have lost the stereo
ouput I used to drive the headphones with but this is less of a proble
pcm_recover)
underrun occurred
OK, managed recover !
framesleft=5160
framesleft=5160
Any ideas ?
And for many tedious reasons I cant simply use the callback, I am
compiling multiple versions of the application, writing data to 2 other
sound systems and at least one of the
On Fri, 2013-04-05 at 09:28 +0200, Clemens Ladisch wrote:
> jon wrote:
> > every so often the process underruns, freezes and does "weird shit".
>
> Known bugs:
> http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/058136.html
>
Thanks for that, u
longer time to stop playing),
yet the latency values i'm reading always seem to be the same.
How do I read a true latency in milliseconds ?
Thanks,
Jon
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On Tue, 2013-04-09 at 21:59 +0200, Clemens Ladisch wrote:
> jon wrote:
> > I am reading the latency like this :
> >
> > int get_playback_latency_ms()
> > {
> > int framesleft;
> > int ms_in_buffer;
> >
On Wed, 2013-04-10 at 08:46 +0200, Clemens Ladisch wrote:
> jon wrote:
> > On Tue, 2013-04-09 at 21:59 +0200, Clemens Ladisch wrote:
> >> jon wrote:
> >>> If I increase the bufchunks value I seem to get larger latency
> >>> yet the latency val
is
zero. As a result the code only moves the volume by the top 20% ish of
its range.
Is the min value supposed to be negative or is the most common code
everyone is using broken ?
Thanks,
Jon
Example output
min=-10239 max=400 writing 0
Code i've crudely hacked together from va
On Mon, 2013-04-15 at 21:20 +0200, Clemens Ladisch wrote:
> jon wrote:
> > Is the min value supposed to be negative or is the most common code
> > everyone is using broken ?
>
> > snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
> > newvol
v/dsp, mixes down the audio and passes it to alsa for
playback (well that is my understanding of it anyway).
Jon
--
WatchGuard Dimension instantly turns raw network data into actionable
security intelligence. It gi
Hello, I recently installed a new mobo, Asus M4N78 PRO, and I am not able to
use the line input, which I have connected to a TV card (I checked it works
with a speaker).
I have tried all kind of combinations, muting and not muting, changing
volume, etc with alsamixer, but I fail. Can somebody help
I have an Asus X99-DELUXE motherboard and nVidia GTX960 graphics card.
The graphics card is connected to a Dell U3415W monitor via DisplayPort.
The problem is that there is no sound coming out of the display's
speakers. When I plug headphones into the case headphone jack, there is
sound. I'm runnin
t happen with two back to
back DD tracks.
Another data point, under Edgy, I saw similar behavior when I
directly used ALSA:SPDIF. The problem went away when I used the
above asound.conf. Not so lucky under Feisty. :-(
Thanks Michael,
Where do I look next to solve the problem.
dmix
build the lastest alsa from source
submit a bug to the mythtv team
What is my next step?
Thanks,
Jon
On May 5, 2007, at 2:31 AM, Michael Bourgeous wrote:
> You're probably hearing Dolby Digital data being played as t
s up. How to I completely and safely do this?
Thanks,
Jon
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sa install to /usr or /usr/local?
If it installs to /usr this should not be a problem.
Thanks,
Jon
-Original Message-----
From: Jon Boehm [mailto:[EMAIL PROTECTED]
Sent: Friday, June 08, 2007 12:34 PM
To: 'alsa-user@lists.sourceforge.net'
Subject: Install / Uninstall Howto??
Hi
oto __err_val;
}
in usbaudio.c like in the original patch, but with my device id, but it
didn't work.
I would appreciate any advice. Thanks,
Jon
By the way, I just noticed that the change I made to the driver seemed
to cause the following in the system log. Hmmm... mayb
amples (capture)
That -L output doesn't look right to me. It used to have pages of
output. This makes me think something in my alsa configuration got hosed.
Please help ... I'm panicking a little,
Jon
# ~/.asoundrc or /etc/asound.conf
# ALSA configuration file
# USAGE #
# Save
0=0x6,AES1=0x82,AES2=0x0,AES3=0x2
I wonder if that hosed something up at a really low level.
Thanks for helping,
Jon
Jon Boehm wrote:
Hi,
Everything been working great for quite a while. I use a spdif
connection between my htpc and my receiver. I used dmix as my
alsa:default thus pas
backup copy of /etc/asound.state placed it in
/etc/asound.state.safe and did the following:
alsactl -f /etc/asound.state.safe restore
rebooted to make sure it stuck and I was good to go.
Thanks for the help,
Jon
PS. bad xine ... bad
FYI. Since I have a well thought out asound.conf I changed
know. TIA.
Jon
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using ALSA 1.0.2 as recommended for
the EPIA boards, where I used to have 0.9.7. Are there any issues with
1.0.2c and EPIA?
I really don't know where to start troubleshooting this. Any ideas would
be most welcome.
Cheers,
Jon
---
SF.Net
> For starters, one may start measuring average frequency of interrupt
> requests generated by both cards provided both of the play the same
> length buffer.
>
> The measure of the average frequency will be the computers RTC.
>
> I am not saying that it is simple, and yes, prior knowledge, like NTP
> I would've thought though that if you had a software buffer of X
> milliseconds, then you should be sending that buffer to the card every X
> milliseconds (according to some other timer in the system.) If it takes
> longer before the card is ready for the next buffer then you need to
> drop some
> Hi. The only solution I saw to this problem was in a HOW-TO that a guy posted.
> Sadly I can't find it, but it was a soldering iron job on the soundcards.
Right here:
http://www.djcj.org/LAU/quicktoots/toots/el-cheapo/
You're just running both devices off the same crystal. That's
certainly on
Hi, I have a Behringer X32 mixer. The default X-USB interface works great, it
appears as a S32_LE multichannel and has no problems up to 32 channels I/O. Im
using RaspberryPi4 as the host.
Generally I will be using mpd to play background and interval music from disk
or streaming.
I installed
bapwnt.lib (2,400,081 bytes)
I have performed this installation under Windows ME & can confirm that it
worked.
The above files are PURELY drivers. There is NO application S/W included.
Hope this info helps.
Regards, Jon Amblin.
___
Alsa-user maili
erently (it took me forever
to find that "input gain" in snd was "capture" in alsamixer!), and the
"Select" options (Mono Output Select and Mic Select) don't seem to me
adjustable at all in alsamixergui, just the console version.
It's all rather confusing to me. Is
ight: 32767 [100%] Capture [off]
This is probably the issue though - this is the Analog-Digital chip, and
if its capture is set to off, nothing will be recordable except stuff
that's already digital.
--
Jon-o Addleman
Curious about the PGP attachments in my e-mail?
see http://www.st-andrews.ac.uk/scotmid/pgpweb/pgpintro.shtml
to find out why they're there!
pgp0.pgp
Description: PGP signature
anished - I don't know if that's related, but I imagine so.
Anyone have any idea what could have happened? Will I just have to use
the dmix plugin and give up hope on getting the rear channel working?
I sure hope not...
Thanks!
--
Jon-o Addleman
-
I had already tried that, but I forgot to run lilo
again afterwards, and was still using the old kernel. That perennial
silliness... Maybe it's time for Grub. :)
Everything works fine now though.
--
Jon-o Addleman
---
This SF.Net ema
ssarily, IMHO) is
extremely well-documented all over the place.
I guess I'll just keep lurking here, and hope that I can answer a few
questions in the meantime.
--
Jon-o Addleman
---
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