Is anyone here aware of a stereo compressor plugin (LADSPA) suitable for
drums and/or bass that reports latency? I've been using SC4, but
realized it adds latency and does not report it! :/
So my mix is basically held back by this... I can hear the delay, it's
definitely there.
So, not having to resort to using an artificial latency plugin would be
nice.
Cheers.
Hi there.
I'm developing a little tool to automatically sync two audio
recordings of the same event made with two different recorders. I've
got the algorithms worked out and can correctly compute the drift and
offset between the files in my test recording. The next step is to
pad, stretch and mix the two files into one perfectly synced stereo
file.
The drift is very small. In my test recording the drift is only 53
milliseconds over the course of an hour. My stretch factor is
0.99998525762821
I've tried to stretch the file with sox:
sox infile.wav outfile.wav stretch 0.99998525762821
When I try to line up the files in a ardour I realise that the stretch
sox makes is hugely inaccurate.
For testing purposes I tried to stretch an hourly long file (44.1kHz)
by some 2000 samples, but the resulting file was some 30000 samples
longer. The stretch was off by more than 0.5 seconds!
Is there anything I can do to increase the accuracy of sox? Is there
any other tool out there that will do a better job at stretching with
very high accuracy?
alex
--
Alex Polite
http://flosspick.org - finding the right open source
On my Ubuntu Dapper system, I'm having trouble with jack clients such as
xmms and jack-rack showing up in the list of clients with their PID
(process ID) attached to their app-names.
My convolver program BruteFIR expects the clients to have fixed names.
Easy fix or workaround for this?
Hello all,
I'm interested in a Kontakt 2 sample library (the Stradivari by
Garritan, see http://www.garritan.com/stradivari.html) which I think
looks absolutely awesome. But there is a slight problem... I don't
have Kontakt 2, and I'm not sure if this type of library can be used
under Linux.
Does anyone here have any experience using Kontakt 2 libraries under
Linux? If so, how did you go about it? I'm not completely opposed to
buying/using commercial software if necessary, but I would much prefer
an open-source solution.
Any suggestions?
Thanks,
-TimH
We have a rivendell workstation which sends sound to a streaming server
via jack.udp.
I'm finishing the setup of our new rivendell workstation. The jack
system (jackd realtime with alsa driver with rivendell) is very robust.
Building the nvidia kernel module doesn't disturb the sound output :o)
Everything is fine except .. jack.udp. jack.udp is clipping on a simple
desktop switch :'( When the system is a bit loaded, the streaming server
receives a very disturbed sound. At the same moment, the alsa ouput or
the jack.backup output (looping jack.record) are perfect.
I tried various buffer size for jack.udp (from 4096 to 20000 ..). At
4096, I have a error message (UDP thread too slow). With bigger buffers,
no message. I reniced jack.udp to -5, -10, -15 .. No changes ..
Anyone experienced the same kind of problem ?
--
Alban Peignier <alban.peignier(a)free.fr>
http://people.tryphon.org/~alban
For all of you who have been waiting for some amazing sounds that are
high quality music first and libre second, I have found just that here:
http://www.chillheimer.com
Some of these sounds simply brought me to other places... they even have
a song called 'Mind Travel'. I can't say exactly in what way but
listening to the stuff is enlightening.
Ambient, Jungle, Deep house genres and a lot of blending a few
completely new styles, 61 tunes.
Carlo
I have a realtime audio/video system that almost works fine. I'm not
too terribly concered with latency, so I'm using 400 samples of
buffering both in and out. For the most part it all works. System load
is about 1.5 according to top, and the 3 processes of my system list
approximately 75% combined CPU usage.
However, I need the system to be network connected. I need it to have
sshd, snmpd, and some web server (not yet decided upon) running. When I
turn them on, any network traffic to either service causes glitches in
the audio, and possibly lost video frames.
I am using Linux 2.6.12 plus bigphysarea patch. I'm using a custom
non-Alsa sound card driver (when I started, I needed all of the AES
metadata, but I no longer do. At this point it is too late for me to
convert to using alsa though). I can provide my kernel .config file if
that would help.
I have CONFIG_PREEMPT and CONFIG_PREEMPT_BKL on. My application runs
with SCHED_FIFO and has an elevated priority, but not realtime
priority. When I tried setting it to realtime priority it grinds the
networking, control panel, and VGA display applications to a halt.
If I set the audio driver to loop back, the network never interupts it.
It would be difficult to move the entire audio portion of my system into
a kernel module though.
I've been watching for awhile and haven't seen to much about how to fix
this. Meanwhile, I've been tangling with this for months.
--
Joshua D. Boyd
jdboyd(a)jdboyd.net
http://www.jdboyd.net/http://www.joshuaboyd.org/
Does anyone else have a problem with Bristol's tuning?
I really like to play with it and love the sounds, but it seems that
it's not in the same tune than other soft synths. Tuning the synths via
gui does not solve the problem. Nick?
My audigy uses samplerate of 48khz, if that helps to answer.
Thanks
--
-----------------------
http://www.emvg.net/esahttp://www.emvg.net
-----------------------
I figured it out!
Applications like xmms and jack-rack show up in the jack connection list
with their PID's appended and this was a bit difficult to reconcile with
brutefir, which wants the user to name the in's and out's in a config file.
The solution was already built into brutefir and I just configured the
qjackctl patchbay to recognize the ports and connect them properly.
The brutefir config file should name it's inputs as follows:
input 0, 1 {
device: "jack" { ports: "", "";};
sample: "AUTO";
channels: 2;};