Hi all,
I have played a little bit around with ardour3 and the import function of
freessound.org. For mixing and mastering I have use a lot of linuxdsp and
calf plugings. I hope the mix is ok.
http://soundcloud.com/ch-bunge/free-sound-lab44
thanks
chris
Dear list,
allow me to post a question here that I already sent to alsa-user, but
without any response so far, thank you!
----------8<-----------
Hi all, now here is one of the weirdest problems I have come across in
the past years. Actually it is halting my current production even, but
I am primarily interested how this error is caused (and how it can be
solved):
The sampling rate of my RME HDSP Card is set from "Internal 44.1k" to
"Internal 48k" upon connecting a Yamaha DM1000 mixing console via usb!
The console does not provide a USB audio interface, but several midi
ports over USB, using the snd_usb_audio module.
cat /proc/asound/cards:
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xf2520000 irq 43
1 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall DSP + Multiface at 0xf0000000,
irq 19
2 [DM1000 ]: USB-Audio - DM1000
Yamaha DM1000 at usb-0000:00:1d.0-1.2, full
speed
29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control
ThinkPad Console Audio Control at EC reg 0x30,
fw 6QHT26WW-1.07
>From HDSP's alsamixer I can see that upon connecting the DM1000 the
control "Sample Clock Source Locking" gets set to [Off], and the
sample rate is set to 48k as described above.
The change is also reflected in the 'hdspconf' software that
configures some properties of the card.
cat /proc/asound/pcm does not list the DM1000:
00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1
00-03: HDMI 0 : HDMI 0 : playback 1
01-00: RME Hammerfall DSP + Multiface : RME Hammerfall DSP + Multiface : playback 1 : capture 1
This is on a Debian box runnig kernel 3.2.0-4-rt-amd64
Let me add here, that upon starting any audio application using the
same card under alsa can (re)set the sampling rate to the correct
value.
If the mixer is re-connected, after that audio application was
launched, the sampling rate of the HDSP card is changed to 48k,
causing an audible transposition.
Can anyone imagine what causes this strange behavior?
Thank you so much for all hints
best, P
Already sent to the Rivendell users list several days ago but so far nobody
seems to be doing exactly this so I thought the wider linux audio community
might have some valuable insight.
Looking for ideas / thoughts / discussion / advice on:
a small box to put in remote locations.
it will:
have a tuner card(s) or an external radio(s)
record the input from the radio via rotter (other?)
send to an icecast instance for remote listening.
be low power and inexpensive.
Hopefully be able to handle more than one station on the same box.
Links for those who may not know of rotter or some of the other bits I want to
run.:
Rotter - http://www.aelius.com/njh/rotter/
Liquidsoap - http://savonet.sourceforge.net/
Darkice - http://code.google.com/p/darkice/
Icecast - http://www.icecast.org/
all the best,
drew
On Thu, Jan 30, 2014 at 12:05 AM, Clemens Ladisch <clemens(a)ladisch.de>wrote:
> Romain Michon wrote:
> > I'm trying to use two audio interfaces (Guitar Link UC6102) in
> > parallel in Alsa on a UDOO board that runs Ubuntu. The two interfaces
> > are connected on the two USB 2.0 port available on the board: I'm not
> > using any hub. In 20% of the cases, this works fine and I'm able to
> > use the two interface as one single "virtual" interface with 4 inputs
> > and 4 outputs (see my .asoundrc file bellow).
>
> So the hardware actually supports the required bandwidth.
>
Yep...
>
> > However, in 80% of the cases, this doesn't work and dmesg says:
> > "cannot submit datapipe for urb 0, error -28: not enough bandwidth".
>
> Sounds like a bug in the USB controller driver of your board.
>
You're right. I'll send an e-mail to the UDOO community, may be someone
had a similar issue... I'll keep you posted.
Thanks.
Romain
Hi Everyone,
I'm trying to use two audio interfaces (Guitar Link UC6102) in parallel in
Alsa on a UDOO board that runs Ubuntu. The two interfaces are connected on
the two USB 2.0 port available on the board: I'm not using any hub. In 20%
of the cases, this works fine and I'm able to use the two interface as one
single "virtual" interface with 4 inputs and 4 outputs (see my .asoundrc
file bellow). However, in 80% of the cases, this doesn't work and dmesg
says: "cannot submit datapipe for urb 0, error -28: not enough bandwidth".
Any idea of where this problem comes from? I can use 4 of these interfaces
on the same USB port with a hub on my laptop without any problem...
Thanks for your help :)
Cheers,
Romain
.asoundrc:
pcm.myMAIN {
type route;
slave.pcm {
type multi;
slaves.a.pcm "myOUT0";
slaves.b.pcm "myOUT1";
slaves.a.channels 2;
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;
}
ttable.0.0 1;
ttable.1.1 1;
ttable.2.2 1;
ttable.3.3 1;
#ttable.0.2 1; # front left
#ttable.1.3 1; # front right
#ttable.0.4 1; # copy front left to rear left
#ttable.1.5 1; # copy front right to rear right
}
ctl.myMAIN {
type hw;
card CODEC;
}
pcm.builtIn {
type hw
card vt1613audio
}
ctl.builtIn {
type hw
card vt1613audio
}
pcm.myOUT0 {
type hw
card CODEC
}
ctl.myOUT0 {
type hw
card CODEC
}
pcm.myOUT1 {
type hw
card CODEC_1
}
ctl.myOUT1 {
type hw
card CODEC_1
}
--
Romain Michon
PhD Candidate
Center for Computer Research in Music and Acoustics
Stanford Universityhttp://ccrma.stanford.edu/~rmichon
Before "modern" computers as the Atari ST were able to sync to tape by
SMPTE, computers as the Commodore 64 synced by some kind of click,
without a time code. IOW you couldn't cue at a wanted position, you
always needed to start from the beginning. This sync was not less good
than sync between MIDI and audio is nowadays, in my experiences it was
much more accurate, than the supposedly "frame accuracy" between MIDI
and audio of Linux DAWs is nowadays.
YMMV!
Regards,
Ralf
I'm trying to use midi-alsa-latency-test 0.0.3
https://github.com/koppi/alsa-midi-latency-test/
with an ESI Romio II USB - MIDI interface
http://www.esi-audio.com/products/romio2/
and it's not working reliably.
The Romio has two MIDI inputs and two MIDI outputs. It is
recognised by my system :
$ alsa-midi-latency-test -l
Port Client name Port name
24:0 RoMIO II RoMIO II MIDI 1
24:1 RoMIO II RoMIO II MIDI 2
But no matter what combination I try :
$ alsa-midi-latency-test -o 24:0 -i 24:0
$ alsa-midi-latency-test -o 24:0 -i 24:1
$ alsa-midi-latency-test -o 24:1 -i 24:0
$ alsa-midi-latency-test -o 24:1 -i 24:1
I get :
timeout: there seems to be no connection between ports 24:x and 24:x
The transmit LED for the MIDI output lights up once but the
receive LED for the MIDI input never does.
No, it's not a bad connection. Running alsa-midi-latency-test
between one output (or input) of the Romio II and one input (or
output) of another MIDI interface works reliably. What doesn't
work is loopback between the Romio II and itself.
Is this a known problem ?
Thanks in advance.
--
André Majorel http://www.teaser.fr/~amajorel/
Dear list,
did anyone have success using jackd with the RME HDSP card and
settings
jackd -dalsa -r44100 -p32 -n2 -D -Chw:DSP -Phw:DSP -i18 -o18
?
Here is the post from the messages window:
Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben
Hohn 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
JACK compiled with System V SHM support.
15:31:07.074 JACK was started with PID=24668.
loading driver ..
apparent rate = 44100
creating alsa driver ...
hw:DSP|hw:DSP|32|2|44100|18|18|nomon|swmeter|-|32bit
configuring for 44100Hz, period = 32 frames (0.7 ms), buffer = 2
periods
ALSA: final selected sample format for capture: 32bit integer
little-endian
ALSA: cannot set period size to 32 frames for capture
ALSA: cannot configure capture channel
cannot load driver module alsa
15:31:07.095 JACK was stopped successfully.
15:31:09.097 Could not connect to JACK server as client. - Overall
operation failed. - Unable to connect to server. Please check the
messages window for more info.
Thanks for sharing your experiences.
best, P
[Sorry for cross-posting, please distribute]
We are happy to announce the next issue of the Linux Audio Conference
(LAC), May 1-4, 2014 @ ZKM | Institute for Music and Acoustics, in
Karlsruhe, Germnany.
http://lac.linuxaudio.org/2014/
The Linux Audio Conference is an international conference that brings
together musicians, sound artists, software developers and researchers,
working with Linux as an open, stable, professional platform for audio
and media research and music production. LAC includes paper sessions,
workshops, and a diverse program of electronic music.
*Call for Papers, Workshops, Music and Installations*
We invite submissions of papers addressing all areas of audio processing
and media creation based on Linux. Papers can focus on technical,
artistic and scientific issues and should target developers or users. In
our call for music, we are looking for works that have been produced or
composed entirely/mostly using Linux.
The online submission of papers, workshops, music and installations is
now open at http://lac.linuxaudio.org/2014/participation
The Deadline for all submissions is January 27th, 2014 (23:59 HAST).
You are invited to register for participation on our conference website.
There you will find up-to-date instructions, as well as important
information about dates, travel, lodging, and so on.
This year's conference is hosted by the ZKM | Institute for Music und
Acoustics (IMA). The IMA is a forum for international discourse and
exchange and combines artistic work with research and development in the
context of electroacoustic music. By holding concerts, symposia and
festivals on a regular basis it brings together composers, musicians,
musicologists, music software developers and listeners interested in
contemporary music. Artists in Residence and software developers work on
their productions in studios at the institute. With digital sound
synthesis, algorithmic composition, live-electronics up to radio plays,
interactive sound installations and audiovisual productions their
creations cover a broad range of what digital technology can inspire the
musical fantasy to.
The ZKM is proud to be the place of the LAC for the fifth time after
having initiated the conference in 2003.
http://www.zkm.de/musik
We look forward to seeing you in Karlsruhe in May!
Sincerely,
The LAC 2014 Organizing Team
