Hi,
I'm using my laptop to make some music. At home I planning to buy a
computer with enough power to host a lot of music software. But when
I'm away from home I'd like to use my laptop. The thing is that
according to the specification on my laptop it would be enough power
to run lots of demanding software. But there is, for example,
Rosegarden and I have problem with xruns when I try to run that. I
have an ACER Ferrari 3000 (a damned hot bastard that makes me wonder
if the ACER community doesn't have a fertility problem if they really
have this thing in their lap ;-) that has an Athlon-XP-M 1800+. I had
a lousy HD in it that i changed but that only boosted the memory-to-HD
performance. I seem to have other problems. The graphic is one
thing. I have done all the things for real time performance and other
stuff to get down the latency but still there is performance
problem. It seems to be connected to jack but I'm not really sure. It
would be interesting to here about other people who have set up
laptops for music production.
If there's already a site or compiled list of systems and how they
perform I would be delighted to be pointed to it. But if it's not it
would be nice if we could start a thread with how laptops are
performing as music stations.
Here's my system
Model: ACER Ferrari 3000
Soundcard: VT8233/A/8235/8237 (yeah I know. crap)
OS: Linux/Gentoo
Kernel: 2.6.10-gentoo-r4
Audio: ALSA
Audio-server: Jack-0.99.0-r1
Measurements
Latency: Haven't had time to test this yet.
Disk capacity with a Hitachi 5400, HTS548080M9AT00
Output: 19.6Mbit/s (measured with bonnie++)
Input: 34.7 Mbit/s (measured with bonnie++)
Music related software and their versions
ardour-0.9_beta23
ecasound-2.3.3
rosegarden-4.0.9.91
seq24-0.6.0
audacity-1.2.2
timidity++-2.13.2
fluidsynth-1.0.5
hydrogen-0.9.1
museseq-0.7.1
zynaddsubfx-2.1.1-r1
- Bengan --------------------------------------------------------------
- KTHNOC/SUNET/NORDUnet | http://www.sunet.se/~bengan | +46 8 7906586 -
Hi all,
what is the best and easiest way to convert a w64-file (from timemachine) into
a wav-file? I am trying "sndfile <w64-file> <wav-filename>" but except 100%
CPU-usage for _really_ long time nothing happens. No new file, no changes, no
stopping after a time.
I know that one of [audacity|sweep] opens w64-files but loads them into memory
fully and the file to convert is 2.5GB.
Can anybody give me some hints?
Arnold
--
There is a theory which states that if ever anyone discovers exactly what the
Universe is for and why it is here, it will instantly disappear and be
replaced by something even more bizarre and inexplicable.
There is another theory which states that this has already happened.
-- Douglas Adams, The Restaurant at the End of the Universe
i've been using mixxx (http://mixxx.sourceforge.net) with no problems
for some time, but recently - *around* the time i patched my alsa driver
with emu10k1 multi-channel, and updated qjackctl to the latest - it's
being weird:
* it starts without any drama the first time in a session (by that, i
mean the first time i start up mixxx after having booted my computer),
but if i close it and try to start it again, it's ouputs show up in
qjackctl, but there is no gui ... even if i stop jackd and start again,
the same problem - outputs, but no gui ...
* if i delete the .mixxcfg in my home directory, then it starts again,
but this requires me to each time reconfigure mixxx :(
* if i reboot the computer (without deleting mixxx's config file), then
it starts properly with all configurations intact :)
if i start mixxx from the command line when it doesn't show a gui,
here's what output i get:
> [mrmachine@localhost <mailto:mrmachine@localhost> mrmachine]$ mixxx
> Debug: Starting up...
> Debug: type signal
> Debug: type marks
> Debug: type signal
> Debug: type marks
> Debug: playlist name Default
i haven't been able to discover any more detailed debug output ...
if it's any help, this is how i start jack (from qjackctl):
#jackd -T -R -P60 -u -dalsa -dhw:0,3 -r48000 -p256 -n2 -s -S -P -o16
any help appreciated.
shayne
Hi all,
I have been asked to give a mini-seminar and live demonstration later
this year on Ardour and Linux audio applications to the audio
engineering students at the university that I work for. I plan to do
a general overview of Linux with some diagrams and web links for
further reading, some info about applications available, and supported
hardware.
Next, I will give a live demo of a session in the recording studio at
the university. I'll bring in my computer, and set it up next to the
ProTools machine that they are used to using, connect it to the
projector so they all can see what I'm doing, and basically do a
simple recording and mixing session with Ardour.
The point is these students work with nice, expensive hardware and
ProTools on a nice fast Macintosh computer in the studio, but when the
class ends, many will want to continue their work in the home studio
environment, and the director there wants me to present Linux as an
affordable and stable alternative for pro audio work (as opposed to
students having to buy ProTools, Sonar, Reason, or whatever) to get
some decent work done.
I plan on having someone videorecord the presentation and demo, and if
it comes out any good, I'll make up DVD's and distribute them to
whomever wants one for the cost of the media and shipping.
Any ideas on stuff to include, or how to make this a killer
presentation? BTW, sorry to cross-post, but I felt that this
pertained to both the Ardour group and the LAU folks (many who are on
both lists). I'm just overjoyed that I will get to show this stuff
and hopefully convert a few users :) I'm also thinking about burning
off a bunch of copies of the various audio-focused distros to hand out
at the seminar.
Jon M.
ok ill bite
whilst playing with alsamixergui i worked out how to change what input I
use for capture but it seems i cant record from cd in, mic and line in
all at once?
in fact only from one at a time!
SB type on motherboard audio.
as soon as i set line to capture it disables mic etc
Your all going to tell me this is normall right ;(
And yes, -s doesn clear TRAM. I think, what is needed is to implement
"zero reading from TRAM" - this is emu10k1 feature, for now not used
in ld10k1.
Peter Zubaj
____________________________________
>>> POZOR ZMENA - KONCERT PRESUNUTY NA 28.2.2005
RAMMSTEIN, 28.02.2005 o 20,00, Bratislava Incheba,
Info: 0904 666 363, http://www.xl.sk
> maybe my old sblive has only 512 M
TRAM is not in sb live, it is in system memory. I don't know how is
allocated memory inside linux, but maximum TRAM size needs 2 MB
continuous memory. Maybe memory is fragmented and there is not such
big fragment.
Peter Zubaj
____________________________________
>>> POZOR ZMENA - KONCERT PRESUNUTY NA 28.2.2005
RAMMSTEIN, 28.02.2005 o 20,00, Bratislava Incheba,
Info: 0904 666 363, http://www.xl.sk
Hi,
>Anyway the TRAM is not unlimited, i dont know why, but in ld10k1
1.0.8 version
>i was need to reduce the chorus delay line because ld10k1 returns no
free
Did you try
ld10k1 -t 8
this should give you max tram size. Yu can check actual tram size in
/proc file - there is hex size in samples (1024 Mega is maximum)
In alsa emu10k1 driver was bug in setting TRAM size.
Previous -t 7 means 1024 M samples - now it means 512 M samples.
This was fixed in 1.0.8 or litle sooner - I don't know exactly.
Peter Zubaj
____________________________________
>>> POZOR ZMENA - KONCERT PRESUNUTY NA 28.2.2005
RAMMSTEIN, 28.02.2005 o 20,00, Bratislava Incheba,
Info: 0904 666 363, http://www.xl.sk
I'm new to this whole linux sound thing, and I'd like
to get my M-Audio Quattro USB working. sorry if this
email is a little long, i just want to go ahead and
put most of my information up now, so you all aren't
asking for it later.
so far, i can play from XMMS using it's ALSA driver, I
can send/receive midi data. i can refer to it as
quattro1, quattro2, etc, as outlined in the standard
.asoundrc (which i'll put at the end). i haven't even
touched recording yet
there's two problems i'm having.
1) when i do a
$ aplay --device quattro1 test.wav
I can hear it, but it's choppy and it get this:
Playing WAVE 'test.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Stereo
underrun!!! (at least 0.079 ms long)
underrun!!! (at least 0.038 ms long)
underrun!!! (at least 0.046 ms long)
underrun!!! (at least 0.039 ms long)
...
same if i just use aplay --device hw:1
when I try --device quattro I get
aplay: set_params:837: Channels count non available
the second problem I'm having, is I can't get the
Quattro to work under JACK.
If I try duplex mode, I get this in my message window:
22:07:32.554 /usr/bin/jackstart -R -dalsa -dhw:1
-r48000 -p1024 -n2 -i2 -o2
22:07:32.627 JACK was started with PID=3974 (0xf86).
back from read, ret = 1 errno == Success
jackd 0.94.0
Copyright 2001-2003 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to
redistribute it
under certain conditions; see the file COPYING for
details
loading driver ..
apparent rate = 48000
creating alsa driver ...
hw:1|hw:1|1024|2|48000|2|2|nomon|swmeter|rt|32bit
control device hw:1
configuring for 48000Hz, period = 1024 frames, buffer
= 2 periods
Couldn't open hw:1 for 32bit samples trying 24bit
instead
Couldn't open hw:1 for 32bit samples trying 24bit
instead
could not start playback (Broken pipe)
DRIVER NT: could not start driver
cannot start driver
22:07:33.937 JACK was stopped successfully.
22:07:35.861 Could not connect to JACK server as
client.
if I only set for playback, then I get a bunch of
clicking and a bunch of XRUN callbacks.
if anyone could help me out, I'd really appreciate it.
thanks!
here's some info:
$ more /proc/asound/cards
0 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ENS1371 at
0x1080, irq 11
1 [Quattro ]: USB-Audio - USB Audio Quattro
M Audio USB Audio Quattro at
usb-00:07.2-2, full speed
$ more .asoundrc
# quattro1 is pcm0 which has a maximum sample rate of
44100 and 16
# bit stereo
pcm.quattro1 {
type hw
card 1
device 0
}
ctl.quattro1 {
type hw
card 1
}
# quattro2 is pcm1 which has a maximum sample rate of
96000 and 24
# bit stereo
pcm.quattro2 {
type hw
card 1
device 1
}
ctl.quattro2 {
type hw
card 1
}
#----
#
# compose 4 channels from two channel x two devices,
hw:2,1 and
# hw:2,2
# assuming that hw:2,1 and hw:2,2 give the same
condition, 24_3LE/96k
#
pcm.quattro {
type multi;
slaves.a.pcm "hw:1,0";
slaves.a.channels 2;
slaves.b.pcm "hw:1,1";
slaves.b.channels 2;
bindings.0.slave a;
bindings.0.channel 0;
bindings.1.slave a;
bindings.1.channel 1;
bindings.2.slave b;
bindings.2.channel 0;
bindings.3.slave b;
bindings.3.channel 1;
}
ctl.quattro {
type hw;
card 1;
}
#
# Remap 4 channels as interleaved.
# Use plug instead of route here, since 24_3LE is
unlikely supported
# by applications.
#
# arecord -r 44100 -c 4 -f s16_le -D q4 -d 5
/home/xxx/q4.wav
pcm.q4 {
type plug;
slave.pcm "quattro";
ttable.0.0 1;
ttable.1.1 1;
ttable.2.2 1;
ttable.3.3 1;
}
ctl.q4 {
type hw;
card 1;
}
#
# Use route plugin for applications that do support
24_3LE
# This lowers latency which the plug plugin introduces
due to
# resampling.
#
# arecord -r 44100 -c 4 -f s16_le -D q4b -d 5
/home/xxx/q41.wav
pcm.q4b {
type route;
slave.pcm "quattro";
ttable.0.0 1;
ttable.1.1 1;
ttable.2.2 1;
ttable.3.3 1;
}
ctl.q4b {
type hw;
card 1;
}
Hi,
QjackCtl 0.2.15a has been released!
This is just a fix release, update highly recommended.
As from the change log:
- Regression from 0.2.13, of the not so stupid pseudo-mutex guards on the
connections management framework, after fixing some crash reports from
Fernando Pablo Lopez-Lezcano and Dave Phillips (thanks!); it pays to be
such a paranoid after all :).
Check it out from the usual place:
http://qjackctl.sourceforge.net
Enjoy.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org