> You're not specifying the price range. However, I'm very happy with my
> KRK Rokit 6 active speakers, and I've heard that many others are happy
> with them too. They're about £450 in Norway.
That's way too much for me right now, and probably a whole lot
more than I actually need (both the number of speakers and probably
the quality, too). 400 EUR is the **hard** upper limit for me.
> To me I prefer active speakers because:
>
> [...]
Thanks for pointing that out, that would have been my next question.
The Alesis speakers mentioned by Arnold do already hit my budget hard,
but I think 333 EUR is still pretty cheap for them, considering the
exclusively positive reviews I found.
So maybe I'll go for them, although I'm still unsure because an external
amp seems more flexible to me.
In any case, I'd still like to collect some more opinions.
Do I really need an amp? Doesn't the signal level of the sound card suffice?
Consider that I don't want to crank up the volume substantially.
Leslie
Hi!
I just set up an icecast2-server and have an ices2 source-client working
perfectly.
But I'd like to do some realtime-streaming with jack. So I installed darkice
0.18.1 and get this error:
DarkIce: VorbisLibEncoder.cpp:142: vorbis lib opening underlying sink error
[0]
I hear it arises quite a lot, because darkice can't connect to the server. I
checked all my ports and compared the config with that of ices. Everything
seems to be the same.
I wonder: I know that at least on the linux-audio conferences Joern and Eric
used icecast and some jack-input. Is there another solution to do it?
I'd need a text-based or no I/O client, for not being able to use the GUI.
Or does anyone have any idea on the darkice error?
Here's some system info and the darkice config:
Linux 2.6.21-mm2 # PREEMPT
Debian etch (4.0)
Self-installed jack and ogg-vorbis tools (linked into darkice properly!)
jack 0.107.2
icecast2.3.1 (deb-package)
Darkice config:
[input]
device = jack # OSS DSP soundcard device for the audio input
sampleRate = 48000 # sample rate in Hz. try 11025, 22050 or 44100
bitsPerSample = 16 # bits per sample. try 16
channel = 2 # channels. 1 = mono, 2 = stereo
# this section describes a streaming connection to an IceCast2 server
# there may be up to 8 of these sections, named [icecast2-0] ... [icecast2-7]
# these can be mixed with [icecast-x] and [shoutcast-x] sections
[icecast2-0]
bitrateMode = abr # average bit rate
format = vorbis # format of the stream: ogg vorbis
bitrate = 128 # bitrate of the stream sent to the server
server = localhost
# host name of the server
port = 8000 # port of the IceCast2 server, usually 8000
password = I12oveBS # source password to the IceCast2 server
mountPoint = /progrock.ogg # mount point of this stream on the IceCast2
se
rver
name = Progressive Rock stream
# name of the stream
description = Progrock from the 70s to 90s
# description of the stream
url = http://juliencoder.de
# URL related to the stream
genre = Progressive Rock
public = yes # advertise this stream?
I skipeed [general], for it is unchanged to the standard.
Thanks for any help!
Kindest regards
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
All recent 0.9 variants and the 1.0 preview.
I noticed that much of its compiles are being done with -O3. I believe other
programs had problems with this as well, that recent gcc/g++ et al are
untested with optimization levels other than 1. Lmms-0.3 is compiling with
-O2. See if that works.
Anybody had the problem? Fixed it?
I just got my new copy of computer music magazine in the mail (September
2007, #117) and there on page 065 is an article "Pursuing the Penguin",
four pages about making music with Linux.
It's good to see Linux getting a little main-stream exposure.
vic
I am having trouble with qjackctl and Qsampler. I don't know how to
work the connections in qjackctl, and I am not sure if Qsampler is
setup correctly. Any help is very much appreciated. Here is what I get
from Qsampler:
---------------------------------------------------------------------------------------------------
20:17:24.628 Client connecting...
20:17:24.630 Server is starting...
20:17:24.630 linuxsampler
20:17:24.655 Server was started with PID=3623.
lscp_client_create: cmd: connect: Connection refused
LinuxSampler 0.4.0
Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck
Copyright (C) 2005, 2006 Christian Schoenebeck
Detected features: disabled at compile time
Creating Sampler...OK
Registered sampler engines: 'GIG'
Registered MIDI input drivers: ALSA
Registered audio output drivers: ALSA,JACK
Starting LSCP network server (0.0.0.0:8888)...OK
20:17:27.871 Client connecting...
20:17:27.873 Client receive timeout is set to 1000 msec.
20:17:27.877 Client connected.
20:17:27.879 New session: "Untitled1".
LinuxSampler initialization completed. :-)
LSCPServer: Client connection established on socket:4.
LSCPServer: Client connection established on socket:5.
20:17:30.756 New Channel setup...
20:17:44.549 Channel 0 added.
20:17:44.570 Channel 0 Audio driver: JACK.
20:17:44.597 Channel 0 MIDI driver: ALSA.
20:17:44.599 Channel 0 MIDI port: 0.
20:17:44.601 Channel 0 MIDI channel: 0.
20:17:44.852 Channel 0 Engine: GIG.
20:17:44.873 Channel 0 Instrument: "/home/Preston/Desktop/gig
files/S1000 CDRom SteinwayD/PARTITION A/VOLUME 001/32 MEG PIANO.gig"
(0).
JACK tmpdir identified as [/dev/shm]
SSE2 detected
Starting disk thread...OK
Loading gig file '/home/Preston/Desktop/gig files/S1000 CDRom
SteinwayD/PARTITION A/VOLUME 001/32 MEG PIANO.gig'...OK
Loading gig instrument...
OK
Caching initial samples...
OK
--------------------------------------------------------------------------------------------
I'd like to adjust the sound levels coming out of my computer in a
manner similar to an equalizer. I'm using a Sound Blaster Live and
alsamixer only has the standard bass/treble adjustment available. Is
there a way to get a more fine-grained control over the sound before
it hits the sound card?
- Grant
Hi,
I am an amateur radio operator, and I'm using the aoss plugin with jack
to route my audio. I have a software defined receiver that takes audio
input from the sound card. I route the audio out of that to gmfsk's
audio input port. Gmfsk is a radio teletype decoder/encoder. When I
change from the receive mode to the transmit mode, the ports for the
receiver for gmfsk disappear in qjackctl, and the ones for transmit
appear. When I go back, the opposite happens, which is all good. The
problem is that the default connections are not what I need, so I have
to change them back and forth each time I change from transmit to
receive in qjackctrl. How do I set up the default connections the way I
want them?
I have my .asoundrc file set up as directed here:
http://www.mail-archive.com/discuss-gnuradio@gnu.org/msg07215.html
Thanks,
Rob
Hi folks,
The following has nothing to do with audio, or even with Linux per se,
but I think of you as the right sort of people to be interested in
something like this :-) Feel free to write me off-list and we can take
the discussion elsewhere.
I've been programming professionally for about 8 years. I'm
self-taught. Lately I've been wishing I had a better grasp of computer
science fundamentals, but I can't afford the time or money to attend a
degree program. But hey, MIT has most of their academic materials
online for free - including textbooks, class notes, and lecture
videos!
The problem with doing it alone in my spare time is that there are too
many other things competing for my attention. I think it would help a
lot to have an online discussion group - a group that commits to
finish* a given course in a certain number of weeks, and can help each
other stay motivated, and help each other with difficult concepts,
etc. By "finish" I mean: watch every lecture, read the texts, and
complete every assignment.
Is anybody else interested in something like that?
I'd propose to start at the beginning, with the introductory scheme
course:
http://ocw.mit.edu/OcwWeb/Electrical-Engineering-and-Computer-Science/6-001…
(btw, I got this idea from hearing about some people doing something similar
with Knuth's "Art of Computer Programming" books. but of course now I
can't find a relevant link.)
--
Paul Winkler
http://www.slinkp.com
Hi there!
Ok... I was actually considering posting this anonymously, but since I
have no shame and laugh so hard it hurts looking at it myself I thought...
"Gubbar med bluetoothheadset" translates roughly to "Old men with
bluetooth headset".
Mainly for the swedish speaking folks out there. A modern punk protest
song with black metal "theme" about corporate bullshit. Written and
performed by Gävel.
Of course everything is made with Linux!
http://www.youtube.com/watch?v=vGc_4DeKBN4
Regards,
Robert
Hi folks,
I have a video cd. Somehow, KDE was giving me problems with it, but I ripped
it using K3b and now I have the mpeg file. I want to rip out a certain song
from the video and convert it to mp3/ogg etc. How would I be able to do this?
TIA.
--
----------------------------------------
Mrugesh Karnik
GPG Key 0xBA6F1DA8
Public key on http://wwwkeys.pgp.net
----------------------------------------