I am looking for a Linux application developed in java, php or C + + that
allows me to test in a network with different musical instruments. End
resulting
in the possibility of uniting the different instruments in a single file or
midi wab
Jorge
Hi,
Not that I am a drummer, but not a real fan of midi editing either ;)
So I need a hardware interface for the drums (something like the Roland
octapad). Moreover I have thought about it and this is how I like to use
the computer, as server of sounds, midi/audio record and mixing. For
workflow an hardware interface is better I think.
So searching for a drum pad now to use with LinuxSampler and probably
Hydrogen (if it finally get stable with JACK).
Any recommendations as it comes to touch, features, connection
possibilities, brands etc.?
Like I said, I am not a drummer, so it's gonna be pretty low budget I think.
Regards,
\r
Hi all,
Gathering by the short shrift the last piece of Indie pop received
that was aired here, I guess many of you won't like this, but thought
I would share it anyway ;) (mainly because it's the first cross-OS
piece I have been involved in, despite thinking this would be a really
good way of working for several years now).
It's a demo for the band I play in at the moment ("River Fury"), and
still in the process of polishing/improving. The song was written by
the lead singer/songwriter, and he recorded the vocals and guitars in
Garageband on his Mac. I recorded the drums and bass (and mixing) in
qtractor under Linux. Drums were played live and recorded thru midi
driving linuxsampler. Bass was played direct in via a zoom B2.1U.
Apologies for the mistakes in playing, but as I said, it does need
more work. I personally feel the vocals are a little too clean at the
moment for the style of music - but that is Dan's call - not much I
can do about that, and the drums are a bit on the boring side (which I
can do a bit about - up to the rather low limits of my ability). Would
appreciate feedback in its current state and thoughts on improvements
to the composition and mix.
My experience has been pretty positive from the Linux side this time.
Qtractor behaved nicely, but I was getting some xruns for some
reason.. Just changed video drivers to Nouveau, and various other
changes to the system.. Maybe expecting too much in terms of latency
from a really quite ageing system now (and an almost broken 10 year
old SBLive!). Only thing I had to do was convert the m4a files to mp3
before importing to qtractor. qtractor impressively handled the
different samplerates (44.1 vs 48k on my system) without complaint!
Anyway, enough waffling. The link is here:
http://www.box.com/s/rrcvn668gqfq0t1mqc12
Let me know if there are any problems accessing this.
Thanks,
James
Hello all,
Now available on
<http://kokkinizita.linuxaudio.org:/linuxaudio/downloads>
Zita-lrx - First release 05/01/2012
------------------------------------
Zita-lrx is a command line jack application providing 2, 3, or 4-band,
4th order crossover filters. The filter type is continuously variable
between Linkwitz-Riley (-6dB at the xover frequency) and Butterworth
(-3 dB at the xover frequency). Outputs are exactly phase matched in
the crossover regions.
The application supports up to 16 channels. This is a compile time
limit and easily changed, but if you have many channels it may be
a better idea to use two or more instances in order to spread the
load in an SMP system. Not that it would matter much - on my old
2 GHz P4, CPU load is around 0.6% per channel for four bands.
Configuration is by a text file using 'OSC' style syntax (similar
to Ambdec and Jconvolver). Apart from the basic filter parameters,
the following can be set:
- Channel labels (used for naming Jack ports).
- Frequency band names (used in output port names).
- Optional output autoconnections.
- For each channel: gain and delay (in ms).
- For each frequency band: gain and delay.
Enjoy !
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
chino-0.10 is out, another re-write.
Chino is a session management framework written in Bash.
Pretty much alpha, I suspect there might be some bugs left
occurring in corner cases I haven't encountered yet.
http://chino.tuxfamily.org/
Previous versions generated session scripts for a very limited variety
of setups. The current version differs in three main points, in that:
(1) no script is written, sessions can now be recovered from a
'session definition file';
(2) the possible variety of setups is much larger since the main
script is separated from implementation, i.e. a user can quite
freely define applications and their behaviour;
(3) it provides a runtime user interface, for adding and removing
applications while the session is running.
The purpose is not to add another session management system
to the LASH/LADISH/JackSession ecosystem, chino is
somewhat different in scope. I neither claim that it's exactly
simple nor that it is of any particular elegance, but it serves my
needs very well -- I can hopefully now start making music. :)
An excerpt from the features list on the website:
a 'load session' command and a 'save session' keybinding
(the latter saves only the session, the state of any involved
application—if applicable—needs to be manually saved to
the appropriate file from within the application);
presets and template sessions for simple creation or forking
of sessions;
a text user interface to add or remove applications while
running, with no manual connection-making involved;
adding support for an application amounts to adding a file
containing some variables and functions, thus no support on
the application's side is required (apart from the ability to
recover a state via command line and/or file loading, without
manual user interaction);
mono-stereo-agnostic audio connecting and Jack-Alsa-
agnostic Midi connecting;
a hierarchical session layout with dependency check,
suggestions and the option to view an image displaying the
session graph (the passive "GUI").
best,
d
> To really make sure, you could try another slot, but I predict that
> you
> will have to try another card.
PC connections and issues with those:
Assumed it's a PCI or similar slot card, just remount the card several
times ... "fidgeting" isn't good, please ask somebody able to see to do
it. Mounting cards even for "seeing" people is a PITA, but the
poltergeists like to disconnect RAMs, PCI and ESPECIALLY SATA
connections without clips ;).
I have a USB DAC (Fiio E-10) that I'm using in Linux to play music (ogg files, etc). It works fine except that at the start of every file, I get no sound until about 0.1 seconds into the file--the music before that is cut off. I don't have this problem using the built-in sound card on my computer, and the USB DAC works fine in Windows.
Any ideas on how I can fix this? I didn't do anything special to set up the DAC--just plugged it in and set AUDIODEV="hw:1,0". I'm using ALSA, have tried a few different programs to play the files, and tried using JACK (with ALSA), but it doesn't make any difference.
Thanks,
Phil Carter
A 03/02/2012, às 22:50, Florian Faber escreveu:
> Miguel,
>
>>> Once you have hdspmixer up and running, you should see the levels on the
>>> input row. If not, check if you have chosen the correct input (optic/coax).
>> Indeed, the input was not set to “optical”. Changing that setting to
>> optical made the inputs work. That saved my day !
>
> The ALSA init script should save that setting when shutting down the
> system. If not, make a custom boot script where you set everything to
> your needs.
Yes, it saves the setting.
>
>> Btw, currently the hdspm doesn’t work out in ubuntu 11.10. This is
>> fixed in the kernel git, so compiling the kernel from there, makes the
>> card functional again.
>
> I believe Adrian accidentally released a version some time ago that
> didn't recognize the PCI version of the card. I fixed exactly that about
> two years ago.
Well, all I can say is that in ubuntu 11.10 the card is not recognized. I was under the impression that it was this commit that fixed it :
commit c09403dcc5698abf214329fbbf3cf8dbb5558bea
Author: Adrian Knoth <adi(a)drcomp.erfurt.thur.de>
Date: Thu Oct 27 21:57:54 2011 +0200
ALSA: hdspm - Enable all firmware ranges for PCI MADI/AES cards
From the Windows INF file, we know the firmware ranges for all RME
cards. For PCIe, a single revision ID per device (RayDAT, MADI, AIO,
AES) is used. Contrary, the older PCI versions use ranges, that is,
one revision ID per firmware version.
Instead of listing all possible revisions individually, match the range.
This commit enables all MADI and AES PCI versions ever shipped.
Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai(a)suse.de>
>
>> Also hdspmixer is not present in the
>> alsa-tools-gui package of ubuntu, it has to be compiled from source.
>
> I am no Ubuntu user, so no idea.
There is already a bug report on this.
>
>> thanks !
>
> You don't have to. It's always nice to meet other linux MADI users :)
Well, it’s always nice to get help when we desperately need it ! :-) There will be many happy users of two MADI soundcards in the Den Haag conservatory.
Btw, because we use the cards for live input processing in supercollider, I would like to use a small period in jack without getting xruns. Right now I have it set to 128samples. I have rebuilt the kernel setting the preemption model to low latency desktop (I didn’t see a setting for Real-Time) and the timer frequency to 1000Hz. I’ve configured jackd for realtime with 'dpkg-reconfigure -p high jackd’. I’m starting jackd with priority 70 (i.e.jackd -R -P 70 ... ). It is working mostly fine, I don’t hear any dropouts, and I don’t see XRUNs most of the time. If I start many synths in suprecollider at the same time, such that the overall cpu usage goes to 80% I do get some XRUNs just at the instance of starting the synths, but then no more. Are my settings correct or is there anything else I could do better ?
best,
Miguel Negrão
Hi,
I have two M-Audio Delta1010LT cards on Ubuntustudio 10.04 with FalkTx's
repos. I double checked, this is happening also on Kubuntu 11.10. Also
it's not a matter of the kernel, currently using 2.6.33-29-realtime.
I use Jack2, where can I find the exact version?
Only thing I know, it's jackdmp 1.9.7.
The cards are hardware synced via S-PDIF.
In envy24control, a mixer program that knows about these features, hw:0
is set to internal clock and hw:1 is set to S-PDIF.
I know of three options to get both cards working simultaneously, but
none is working quite perfectly.
a)use .asoundrc
I have a .asoundrc from the web.
After a few seconds, it crashes saying this:
Unknown request 4294967295
Destination port in attempted (dis)connection of system:monitor_6 and
system:monitor_6 is not an input port
Unknown request 4294967295
Unknown request 0
Unknown request 0
Unknown request 0
Unknown request 4294967295
Unknown request 0
Unknown request 4294967295
Unknown request 4294967295
jackd: ../common/JackGraphManager.cpp:45: void
Jack::JackGraphManager::AssertPort(jack_port_id_t): Assertion
`port_index < fPortMax' failed.
While running, it produces tons of xruns, each of them with -v option
generates a message like
Jack: fPollTable i = 1 fd = 11
Jack: JackRequest::Notification
Jack: JackEngine::NotifyClient: no callback for event = 3
Jack: JackEngine::NotifyClient: no callback for event = 3
The last thing I get before crashing is:
Jack: JackRequest::ConnectPorts
Jack: fPollTable i = 1 fd = 11
Jack: fPollTable i = 1 fd = 11
Jack: fPollTable i = 1 fd = 11
Jack: fPollTable i = 1 fd = 11
Jack: JackRequest::ConnectPorts
Jack: JackEngine::PortConnect src = -1 dst = 3
Jack: JackGraphManager::AssertPort port_index = 4294967295
But there is no delay of the second card.
b) use jack_load
this is working best atm, only it introduces a delay to the channels of
the second card when recording with ardour. I measured the delay... it's
exactly (periods/buffer)*(frames/period) frames.
However, I'm not able to delay the first card with the -O or -I options.
Probably the second is delayed as well, or this option is not working.
In ardour I can move the regions to be in sync, but it gets quite
confusing in bigger sessions.
Btw does jack_load resample?
c)use alsa_in
produces the same delay but additionally uses a lot of cpu. Resampling
is not what I want, and also shouldn't be necessary.
Does anyone have an idea on how to fix this?
I'd prefer a solution without resampling, as these cards are already
synced. Also there should be no delay and it should keep running for hours.
Thx!
Any help is appreciated.
/mn0
Hi wizards and wizardesses,
Trying to compile sndfile-tools from git, the author suggests
I have a library problem.
Cairo appears to be present during the ./configure
phase, however some dependent libraries are not
found during make.
One of these libraries is libxcb-shm.so.0,
which is present:
/usr/lib/x86_64-linux-gnu/libxcb-shm.so.0
Can someone help me solve this:
On Fri, Feb 03, 2012 at 09:58:31AM +1100, Erik de Castro Lopo wrote:
> Joel Roth wrote:
>
> > My naive attempt to compile (adding various libraries along
> > the way) got stuck.
> >
>
> <snip>
>
> > Found CAIRO ................... yes
>
> That means that Cairo thinks it is installed correctly.
>
> > Found JACK .................... yes
> >
> > Installation directories :
> >
> > Program directory ............. /usr/local/bin
> >
> >
> > (master) ~/build/sndfile-tools $ make
> > CCLD src/sndfile-spectrogram
> > /usr/bin/ld: warning: libxcb-shm.so.0, needed by /usr/lib/libcairo.so, not found (try using -rpath or -rpath-link)
>
> But the linker disagrees.
>
> You didn't say whay distro you're on, so my best advice is to make
> sure you install stuff like Cairo from your distro, making sure to
> install the -devel packages as well and then retry.
>
> Erik
> --
> ----------------------------------------------------------------------
> Erik de Castro Lopo
> http://www.mega-nerd.com/
--
Joel Roth