I think I am missing some essential understanding of Ardour. I have
Ardour set up in default manner (Fedora 13), one Master bus, and have
added one track. The input of the one track has been successfully set
to Linux Sampler; when I get sound out of LS, the meter in the track
moves nicely. However, when I change the metering point in the track
from "input" to either "pre" or "post", the meter doesn't move at all.
(Obviously there are more consequences too, but this is the boil-down.)
There are no plugins in all. Recording does work, the wave data is
clearly visible. What am I doing wrong?
J.E.B.
Drumstick is a C++ wrapper around the ALSA library sequencer interface using
Qt4 objects, idioms and style. ALSA sequencer provides software support for
MIDI technology on Linux. Complementary classes for SMF, WRK and OVE file
processing are also included. This library is used in KMetronome, KMidimon and
KMid2, and was formerly known as "aseqmm".
Changes:
* OVE file format support, contributed by Rui Fan
* Optional RealtimeKit support for MIDI input thread
* guiplayer simplified and optimized, with OVE format playback
* Build system fixes: using visibility=hidden if it is available,
exceptions, static build.
Copyright (C) 2009-2010, Pedro Lopez-Cabanillas
License: GPL v2 or later
Project web site
http://sourceforge.net/projects/drumstick
Online documentation
http://drumstick.sourceforge.net/docs/
Downloads
http://sourceforge.net/projects/drumstick/files/
'Owdo again!
Now that I've icecast running with ices as a feeder, I wonder, if I can turn
down latency. In icecast I set the burst-size to 0 and in ices I set realtime.
Still the delay is considereable. Is there any chance of tuning this setup
further to reduce latency?
If it helps, I will post my ices.xml and my icecast.xml.
Kindly yours
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
Excited to see
http://googleblog.blogspot.com/2010/08/call-phones-from-gmail.html , I was
quickly disappointed to find the plugin only supported debian and was 32
bit. However, I persevered and got it running on Fedora 12 x86_64 anyways.
Solution:
http://www.google.com/support/forum/p/chat/thread?tid=10ffe01c3a4779f5&hl=e…
Google ought to hire someone that knows about App Development and Linux
Audio and Fedora packaging (me?) to make the audio/video experience nicer
for it's Linux users. In particular, not just blindly using ALSA devices for
input that cannot possibly support audio capture, although I guess I'm a
fringe-case since I don't use pulseaudio, since I've got KDE's phonon setup
to do the right thing w/r/t all my audio devices, including the ones talking
via Jackd. (
http://ccrma-mail.stanford.edu/pipermail/planetccrma/2010-May/016886.html ).
Also, not assuming every Linux user runs a debian/ubuntu distro would be
helpful as well :-)
---------- Forwarded message ----------
From: Google Help <noreply(a)google.com>
Date: Wed, Aug 25, 2010 at 5:13 PM
Subject: Re: [Google Chat Help] Linux version now available!
To: nielsmayer(a)gmail.com
NielsMayer has posted an answer to the question "Linux version now
available!":
FYI, adventures in Installing google talk on fedora 12 x86_64
Linux installation doesn't distinguish between Fedora/OpenSUSE and
Debian/Ubuntu linux systems, so you get a download for the wrong distro:
-rw-r--r-- 1 npm npm 6926676 2010-08-25 15:47
google-talkplugin_current_amd64.deb
Which can be unpacked via ark(1) or file-roller(1) and contains a file
data.tar.gz
I installed the plugin for chrome/mozilla 64 bit by doing
132 16:01 cd /
134 16:02 sudo tar xvzf ~/Download/data.tar.gz ./opt/google/talkplugin
151 16:11 cd /usr/lib64/mozilla/plugins
## (or /usr/lib ... for 32 bit)
152 16:11 sudo ln -s /opt/google/talkplugin/libnpgoogletalk64.so
/opt/google/talkplugin/libnpgtpo3dautoplugin.so .
now, about:/plugins shows the existence of the plugin:
Google Talk Plugin (2 files)
Version: 1.4.1.0
Name: Google Talk Plugin
Description: Version: 1.4.1.0
Version:
Priority: 1
Location: /opt/google/talkplugin/libnpgoogletalk64.so
Disable
MIME types:
MIME type Description File extensions
application/googletalk Google Talk Plugin
.googletalk
Name: Google Talk Plugin Video Accelerator
Description: Google Talk Plugin Video Accelerator version:0.1.43.3
Version:
Priority: 2
Location: /opt/google/talkplugin/libnpgtpo3dautoplugin.so
Disable
MIME types:
MIME type Description File extensions
application/vnd.gtpo3d.auto O3D MIME
.
However, back in gmail. clicking on the "call phone" doesn't work....
Issue:
gnulem-230-~/Download> /opt/google/talkplugin/GoogleTalkPlugin
/opt/google/talkplugin/GoogleTalkPlugin: error while loading shared
libraries: libssl.so.0.9.8: cannot open shared object file: No such file or
directory
Solving:
gnulem-236-/usr/lib> sudo ln -s libssl.so.1.0.0a libssl.so.0.9.8
## note the plugin seems to rely on 32 bit libraries being installed, which
fortunately,
## they are even though I have x86_64 system as i've needed to run other 32
bit binaries...
Issue:
gnulem-245-~> /opt/google/talkplugin/GoogleTalkPlugin
/opt/google/talkplugin/GoogleTalkPlugin: error while loading shared
libraries: libcrypto.so.0.9.8: cannot open shared object file: No such file
or directory
Solving:
gnulem-247-~> sudo ln -s /usr/lib/libcrypto.so.1.0.0a
/usr/lib/libcrypto.so.0.9.8
gnulem-248-~> /opt/google/talkplugin/GoogleTalkPlugin
/opt/google/talkplugin/GoogleTalkPlugin: /usr/lib/libcrypto.so.0.9.8: no
version information available (required by
/opt/google/talkplugin/GoogleTalkPlugin)
/opt/google/talkplugin/GoogleTalkPlugin: /usr/lib/libssl.so.0.9.8: no
version information available (required by
/opt/google/talkplugin/GoogleTalkPlugin)
socket(): Address family not supported by protocol
socket(): Address family not supported by protocol
restarting chrome... and it ...
!!!!!!!!!!!!!!!!!!!!!!!! WORKS !!!!!!!!!!!!!!!!!!!!!!!!
testing in a call, I'm using an ALSA device that doesn't have input by
default
gnulem-253-~> /opt/google/talkplugin/GoogleTalkPlugin
/opt/google/talkplugin/GoogleTalkPlugin: /usr/lib/libcrypto.so.0.9.8: no
version information available (required by
/opt/google/talkplugin/GoogleTalkPlugin)
/opt/google/talkplugin/GoogleTalkPlugin: /usr/lib/libssl.so.0.9.8: no
version information available (required by
/opt/google/talkplugin/GoogleTalkPlugin)
socket(): Address family not supported by protocol
socket(): Address family not supported by protocol
socket(): Address family not supported by protocol
ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only
playback stream
Issue:
Given my default ALSA device is a dmix device, I now need to totally rework
http://nielsmayer.com/npm/dot-asoundrc.txt to make this all useful....
-- Nielsl
http://nielsmayer.com
A new Early Music CD has been produced using FLOSS on Linux, mainly Ardour:
Sound out my voice - Italian madrigals and bastarda music for viol consort
The CD is played by the viol [1] consort "ORLANDOviols". It is
commercially available at http://www.orlandoviols.de/, and soon also via
amazon (Europe at least). A non-commercial publishing is planned after a
while.
A video, more or less the making of the CD, is available on youtube:
http://www.youtube.com/watch?v=PUx_UIcht2Y
The video is completely made with Blender on Linux.
Thanks to all the creative people around the Linux Audio World!
Giso
[1] http://en.wikipedia.org/wiki/Viol
Hi,
I need a new workstation | DAW pc, would be nice to get some advise.
Silent
Energy friendly (relative...)
Good for audio work
Price: 300 - 500 euro
1. Which components are the most important? And which components might be
good second hand as well?
2. How many cores, is a dualcore enough, or do I need a quatro core? It
would be nice to be able to enjoy the computer in the coming 3 years (I
don't know how heavy computing will be in 3 years...)
3. Which computer case?
4. Motherboard
5. Monitor
6. Fans?
When I look around, Phenom processor and MSI motherboard are mentioned...
Thanks in advance,
\r
Greetings,
What the Subject says:
http://www.youtube.com/watch?v=5BkSx2lH3NE
A Blur study. Soundtrack made by Noise and FM3Random audio generators
with Waveguide and Waveset filters.
Best,
dp
Kevin Cosgrove:
>> > raise the heat of all the rest of the things in the case above them.
>> >
>>
>> I would rephrase that to avoid putting things which are sensitive
>> to heat at the top, and avoiding putting things which are hot
>> close to things which are sensitive to heat.
>>
>> Normally you only have to worry about the temperature of the CPU
>> the PSU. The other stuff inside an audio-computer
>> usually works fine in higher temperature without making
>> lots of noise or failing. (at least to a certain degree of corse).
>
>My seemingly hidden point was that when the hot stuff is at the top,
>and the air exit is at the top, then the whole case will run cooler.
>A cooler case means the fans can run more slowly. Slower fans mean a
>quieter system, and that is good for audio work when the system is in
>the room where audio is created or enjoyed.
Your point was not hidden, it was wrong. Some of the hot stuff is
also the stuff which is sensitive to heat. If you put the CPU
and the PSU at the top of the case, they will be heaten up by
hard drive, motherboard, etc., and they will cause more noise
than if you had putten them at the bottom.
Kevin Cosgrove:
>
>
> On 7 September 2010 at 19:55, david <gnome(a)hawaii.rr.com> wrote:
>
>> Design a case like a chimney and let the hot air rise on it's own. ;-)
>
> ... and avoid putting the hottest things at the bottom where they
> raise the heat of all the rest of the things in the case above them.
>
I would rephrase that to avoid putting things which are sensitive
to heat at the top, and avoiding putting things which are hot
close to things which are sensitive to heat.
Normally you only have to worry about the temperature of the CPU
the PSU. The other stuff inside an audio-computer
usually works fine in higher temperature without making
lots of noise or failing. (at least to a certain degree of corse).
Hello list, Niels
first of all, your redesigned version of Envy24 Control looks pleasing to the eye, thanks for this update!
I have a couple of questions that come to mind
- i remember with the old version 0.6 i never succeeded in getting any signal from my DAT recorder using S/PDIF in. I must admit i haven't tried it with this version (1.03) - it takes a lot of physical effort from my part to make a connection with cables due to too many hardware fighting for space, but did you do any work in that area as well? Is it worth the try?
- looking at the tab Hardware Settings i see a message : IEC958 Input Status, unable to read IEC958 Input Status: No such file or directory
What does it need? What can i do?
- is it possible to scale the mixer to other dimensions? i would like to have the mixer in sight all the time while doing my work but it would be nicer to be able to resize it to smaller proportions and still seeing all the levels.
thank you
Menno