Hi.
I'm an old hand at linux but I'm just starting out with audio editing.
I have an interview that was recorded with the "recording level" set
just at tad to high. When the speaker is talking loud the sound gets
distrorted.
I realize that there is no way of completely getting rid of this after
the fact (next time I'll set the recording level a bit lower). But
maybe there's some digital magic that will reduce the distortion a
bit.
alex
--
Alex Polite
http://flosspick.org - finding the right open source
Hi.
I am wondering, there is a way to either save FLAC data as "native"
FLAC format, or in a OGG container. Is there any advantage
to using OGG over the native format? Why would one want to
use ogg containers instead?
And, a somewhat unrelated question, I am looking for a reliable way
to automatically remove leading and trailing silents from
FLAC files. Does such a tool exist? It would be able to be used
in a batch mode, I am talking several thousand files here, which is
nothing I'd like to do with anything GUIish...
--
CYa,
Mario
I've been offered a slightly used TASCAM US-224. I see in the sound
card matrix that it should be fully supported by ALSA.
Is anyone here using one of these with multitracking software under
Linux, and what do you think about it? Is it stable, are all its
features supported, and so forth?
As an alternative, I'm considering getting a new US-122. Would there
be any reason at all to go for that one instead, or are they equally
well supported?
Hello everyone,
I'm trying to install fst-1.6 on my Ubuntu machine. I've installed
the latest Wine from an Ubuntu package from the WineHQ site, and did
not compile it from source. When I run the configure program, I get
this error:
checking for winedef.h header...configure: error: Could not include
the Wine headers (winedef.h)
I'm curious as to why this is happening. Could it be that, because I
didn't compile it from source, that winedef.h is somehow not included
in my package? Or do I simply need to "point" the configure program
to where winedef.h is located? I'm a little lost here, and so I don't
really know where to start. I'm running a slow laptop (PIII 700 Mhz),
so if I can get away with not having to compile Wine from source, I
would be relieved. :)
If I've neglected to give you any information that you need, please
let me know. Thanks for your help.
Josh
--
Josh Lawrence
http://www.hardbop200.com
Hi everyone
I've been lurking around for a while but this is my first post -- I had hoped
that that would be to post some music, but I've encountered a bizarre problem
during mixdown. Well, it seems bizarre to a relative newbie like me! I hope
this is the correct place to be posting as the problem seems to be a general
sound/music one rather than being Linux specific.
I have a piece that I am mixing in Ardour. None of the tracks are going over 0
dB. The master channel is not going over 0 dB.
The kick drum is on a track on its own; it was recorded from Hydrogen using the
Natural Studio kit free samples. It is a mono file in a mono track. I have the
SC1 compressor and the three band parametric with shelves pre-fader. It is
panned absolutely dead centre and when playing back in Ardour everything sounds
fine and the kick sounds central through my nice new Wharfedale studio monitors
:-).
However, when I exported to wav there was terrible clipping. A quick look in
ReZound showed the problem: there was a mass of clips in the left hand track.
Closer visual and aural inspection showed that the clipping is from the kick
drum.
In the quieter sections I can see spikes in the right hand channel, too, when
the kick er... kicks, but they are a lot smaller and nowhere near clipping. I
have tried running the kick with a limiter in Ardour and bouncing down again --
that gets rid of the clipping but the transients are still so much bigger in the
left channel that the kick just sounds to be panned left.
I've played around with applying limiters in ReZound but I'd have to really
crank them up (or should that be down?) to get the kick transients to the same
size in both channels and would end up taking the tops off quite a lot else in
the left channel in the process.
I don't understand why a centrally panned, mono file should only clip in one
channel -- I would expect it to clip both sides more-or-less evenly. What's
causing this problem? Is it a problem or have I just misunderstood something?
And most importantly, how do I get my kick drum back dead centre without
applying a limiter to the entire left channel?
Thanks in advance. Oh, and sorry for the long post!
Q
(responding to three messages here ...)
On Tue, 24 Jan 2006, Cesare Marilungo wrote:
> I don't have Rosegarden listed in the "readable clients / output
> ports" but If I play my midi keyboard through Rosegarden and then
> out to ZynAddSubFx it works.
Thanks. That confirms that what I'm seeing in JACK isn't unusual or
unexpected.
On Tue, 24 Jan 2006, Mike Taht wrote:
> It's probably not finding the midi device... do an strace on
> rosegarden to see what midi devices it's trying to open and see if
> they are in /dev
Checked first with lsof:
Once running, with the device selected that I expect to see as the
physical "MIDI out" connector on the CS46xx interface:
rosegarde 25807 syl 14u CHR 116,1 125373 /dev/snd/seq
For reference, aplaymidi (which _does_ produce MIDI output) also holds
the same device file open:
aplaymidi 25846 syl 3u CHR 116,1 125373 /dev/snd/seq
so, alright, a trace of the rosegardensequencer process is in order ...
In response to Markus Herhoffer's suggestion, though, I ran the strace
test with JACK not running, and sure enough, I got the MIDI output, tried
again without strace (still no JACK), and again MIDI output was working!
I won't bore you with the strace output! :-)
On Tue, 24 Jan 2006, Markus Herhoffer wrote:
> Midi has actually nothing to do with Jack. Jack cares only about
> audio streams.
I'm not convinced of that. For example, I do need to "connect" (via
qjackctl's "connect" screen, "MIDI" tab) "80:CS46xx" to "129:FLUID Synth"
to be able to control that software with my MIDI controller.
> As far as I know there are some problems in the compatibility of
> qjackctl and Rosegarden (described in "Rosegarden Companion").
I would love to get a copy of that. Anyone know of its availability in
Canadian bookstores, or should I be looking to purchase a copy online?
> So you should use only Rosegarden for linking the MIDI devices.
Ok, as mentioned above, I tested with JACK not running, and sure enough
I'm now able to get MIDI output from Rosegarden. Thanks to all who
replied about this! The beautiful sounds of my music theory final from a
couple of years ago (ok, well, it's still beautiful to MY ears!) emanating
from various MIDI synthesizers and a piano sample! :-)
I continued testing and found that if I start JACK _after_ starting
Rosegarden, it seems to do the right thing. Should I perhaps simply have
Rosegarden start Jack, or am I asking for different trouble if I do that?
(I do still want to use JACK because some of the applications I expect
to be making heavy use of use it or require it ...)
This software (all of it!) is such a beautiful thing! Thanks once again,
and good night! :-)
--
----------------------------------------------------------------------
Sylvain Robitaille syl(a)alcor.concordia.ca
Major in Electroacoustic Studies Concordia University
Faculty of Fine Arts / Music Department Montreal, Quebec, Canada
----------------------------------------------------------------------
I have been working on improving the midi import functions of denemo. I have the current cvs version importing polyphonic type 1 files. I want to
know if there is a website that anyone knows of that has a collection of midi format 0 and format 2 files. I would like to add support for importing
and exporting type 0 and 2 files. I want some files to test with.
Thanks,
Jeremiah
hi... the favicon.ico file for the linux-audio-user (and dev, and announce) webpages is an Apple logo. any chance of changing it to a Penguin wearing a pair of headphones or something?
thanks,
concerned listie
Hi all,
Does anyone use an M-audio device like MobilePre or FastTrack with success?
I've read in some archives that some peoples have try to use them, but i have
not find any clear feedback.
Some talks about firmware loading, others talks about usb compliance problem...
I plan to use it with a laptop shipped with Debian/DeMudi with custom recent
kernel (>2.6.9).
So if there is anybody outhere that is happy with its M-audio MobilePre or
FastTrack, please let me know.
regards,
Christian
PS: I am looking for a "cheap" USB audio interface in which i can plug a guitar
or a bass and a stereo microphone (for percussion or acoustic guitar).
I can easily buy m-audio devices, but the vendor doesn't have (many) edirol
devices...
Hi,
I have fixed the two preceding problems discussed on alsa-user mailing
list (1: "advanced mode" switch on lead to card not recognized. 2: lot
of clip/distortion on output).
These problems was solved by updating alsa kernel modules and alsa lib
to 1.0.10.
I'm am using ardour with jamin as insert on the output master track.
And now, i'm facing a new issue, i get sometimes (often) lot of xruns,
leading jack to stop and ardour/jamin to crash!
jack says:
Lot of messages like:
**** alsa_pcm: xrun of at least 0.029 msecs
and finally:
jackd watchdog: timeout - killing jackd
Jamin says:
zombified - calling shutdown handler
Ardour says nothing.
I get only this behaviour when puting jamin as an insert on out master
track in ardour. When removing the insert i can get all the stuff
working well.
Before switching to this external USB sound card, i was able to play my
ardour project (with the jamin insert) without problems... (built in
ugly PCI sound card)
I guess, jack get lot of overruns mainly because of the USB, but i'm not
sure at all...
Does anyone know some nice jack parameters that suits the use of such an
external sound card?
Regards,
Christian