I'm finding that qjackctl-0.3.0, 0.3.1 and 0.3.1a all use up 49% cpu. My
box is a CoreDuo, so that's probably 100% of one cpu.
qjackctl-0.2.22 doesn't do this, so I've downgraded for now.
If it's a bug, what can I do to help?
bye
John
Greetings,
I use the following command to invoke mencoder to compile a series of
TGA image files into an animation :
mencoder -ovc lavc -lavcopts vcodec=mpeg4:vme=1:keyint=30:vbitrate=1000
-vf scale=800:600 -noskip -mf type=tga:fps=30 -o avs-001.avi mf://*.tga
(Sorry about the line breaks.)
The problem starts with the "mf://*.tga" part of the command. When the
AVSynthesis program creates its TGA images it saves and labels them
sequenctially, i.e. 1.tga, 2.tga, 3.tga and so on, as expected. The
problem is that mencoder reads the files as they would be returned by a
plain ls command, i.e. 1.tga, 10.tga, 100.tga, 1000.tga, 1001.tga
...101.tga, 1010.tga, thus interpolating frames out of their correct order.
So, my question is, how do I get mencoder to read the TGA files by their
time of creation ? That should do the trick, yes ? Otherwise I have to
separate the single digit files from the double digit files and so
forth, then I have to create and join separate AVIs. Not terribly
difficult, just really annoying and time-consuming.
Any suggestions ? Any mencoder users out there ?
Also, what do you use in place of mencoder on a 64-bit system ? (It's
not available for 64 Studio.)
Best,
dp
David,
I would consider using a DI box.. You may also wish to consider miking an
amp, even an SM57 sounds great. A nice recording amp with a powerbrake will
sound like a stack at full blow on 2.
as for splitting the signal, it's a decent idea, in a perfect guitar setup
you want to record a DI box and a mike on different tracks so that later you
can go back and 'reamp' or 'remike' your dry signal by running it through an
amp/mike and rerecording it if the original amped track proves troublesome.
if you have noise contraints, I'd suggest at the very least a DI box.
best
bradley newton haug
On Nov 27, 2007 11:51 PM, David Haggett <david(a)haggett.demon.co.uk> wrote:
Should I have something in the chain to match impedance, or is the clipping
> just because my pickup signal is too high? I can eliminate it by turning
> the
> guitar or the pre-amp down but this can lose tone.
>
>
> Second Question
>
> The mic input feeds to Alsa PCM Inputs 5 and 6, but of course is just 2
> identical streams. I have been recording the guitar to a stereo track
> using
> both channels but wondered whether there was any benefit in doing this, or
>
> whether it would be better:
> * take a single input and split it to a stereo track?
> * take a single input and record it as a mono track?
> * take a single input and feed it to a mono-stereo plugin and record
> it as a stereo track?
> __
Hi list
I'm generally happy with this card, but wonder if I'm using it correctly
and/or optimally. I'm a bit of a novice when it comes to audio generally.
If it matters, I use openSUSE 10.2 with 2.6.22.9-ccj56-rt. Record using
Ardour2. Use jack-rack as a ladspa effects host when not recording.
First Question
It (the DMX 6Fire) has a 6.3mm (i.e. 1/4 inch) mono microphone input with
built-in pre-amp which the manual claims to be good for a condenser mic. I
typically plug my electric guitar into this, but is quite a fine balancing
act between guitar and pre-amp volume controls to avoid clipping.
Should I have something in the chain to match impedance, or is the clipping
just because my pickup signal is too high? I can eliminate it by turning the
guitar or the pre-amp down but this can lose tone.
Second Question
The mic input feeds to Alsa PCM Inputs 5 and 6, but of course is just 2
identical streams. I have been recording the guitar to a stereo track using
both channels but wondered whether there was any benefit in doing this, or
whether it would be better:
* take a single input and split it to a stereo track?
* take a single input and record it as a mono track?
* take a single input and feed it to a mono-stereo plugin and record
it as a stereo track?
Thanks in advance
--
David Haggett
Hi folks,
LiLAQ was an attempt to replace the old Audio Quality HOWTO with a
wiki. It's been closed for editing for years, but it's still
available read-only at http://www.slinkp.com/LiLAQ/FrontPage
However, I make no promises after Dec 2007. It may be yanked at any
time, as I need to move my website to a new host before the end of
december, and I want to reduce the amount of stuff to take care of. I
may or may not keep a static copy up at the new site for a while.
If anybody wants to take over or reuse the LiLAQ material in another
form, in whole or in part, as a wiki or in any other format, you are
welcome to do so, under the terms of the LDP license (see
http://www.tldp.org/COPYRIGHT.html). All contributions to the
original AudioQualityHOWTO were made under the terms of the LDP
license, so it still applies to the wiki version. A list of
contributors is at http://www.slinkp.com/LiLAQ/AQHTContributors
This offer does not apply to any material that is not within the LiLAQ
wiki, in particular it does not apply to any other material at any
other part of slinkp.com.
There was some community interest in moving/revamping LiLAQ a couple
years ago, but I don't remember who asked about it and I don't think
anybody ever did anything with it. If I'm wrong, please refresh my
memory :)
I will not be providing technical assistance with exporting the
material, except to note:
- A complete list of pages is at
http://www.slinkp.com/LiLAQ/FrontPage/contents
so if you parse that and extract all relevant links you're on your
way.
- You can get raw source text of any page by appending /text to its
URL. For example, the source of
http://www.slinkp.com/LiLAQ/RealTimeIssues is at
http://www.slinkp.com/LiLAQ/RealTimeIssues/text
- The format used was StructuredText, as described at
http://www.slinkp.com/LiLAQ/StructuredTextRules . It probably wouldn't
be hard to write a script to massage it into some other wiki format,
but I don't have time to help you with that.
I'll post another warning when I'm about to pull the plug.
Fire up those wget scripts :-)
--
Paul Winkler
http://www.slinkp.com
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OK, looks like I'm logged in to the SVN tree that Mark set up (thanks!).
What's the next step? Commit a loop or sample? What format was decided upon?
How will communication occur? IRC channel? Email? This list?
By the way, examples of "killer web apps" for music collaboration:
http://www.ccmixter.orghttp://www.splicemusic.com
The SpliceMusic is particularly nice because it has its own web app to reduce the compatibility issues in converting projects.
Actually, perhaps we should move this whole LAU project onto something that already exists, such as ccmixer or splicemusic?
- -ken
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ok, folks....
i've got something up at http://lau-cb.peterlutek.com
you can browse and download via http. send me personal email to get the ftp user and password for uploads.
i've put up a README with some basic ground-rules -- feel free to make suggestions.
perhaps it would be good for the band to have an independant email list -- i'm happy to set that up.
what do you all think -- should we keep it here, or not?
--
.pltk.
tis 2007-11-20 klockan 12:00 -0500 skrev Philippe Hezaine:
> An HTML attachment was scrubbed...
> URL:
> http://lists.linuxaudio.org/pipermail/linux-audio-user/attachments/20071119…
Please, turn off html in your emails!
To answer your question, I believe we'll (yes, I might chime in) base
work around samples, so you won't be submitting midi-files, as it would
create havoc (sure, you could provide a midi-file so someone else could
recreate what you've done, but please provide a audio sample as well).
Why, you may ask? Well, for starters, you would have to provide us with
presets, instructions for patching in jack and alsa, and some program's
midifiles don't play nice with other's (for example, even if you arrange
a song in seq24, it'll be hell for a Rosegarden-user).
So, well, yes. just fiddle around, and when it sounds good on your
computer, just sample and upload it. That'll work nicest.
Gasten
Charles Linart wrote:
> lately, I'm having a serious issue with JAMin that has prevented me
> from mastering anything: It always wants to output to the sound card
> when it should just output to Ardour.
If you're using qjackctl, it sound to me like you defined (and forgot
about) a connection in the patchbay. It bites me from time to time...
--
peace, love & harmony
Atte
http://atte.dk | http://myspace.com/attejensenhttp://anagrammer.dk | http://modlys.dk