Hi all,
I'm working on fixing this issue:
https://github.com/erikd/libsndfile/issues/190
in libsndfile. In order to figure out what's what, I need a WAV file
loop, with an embeded ACID chunks, and the loop needs to be in a time
signature *other* than 4/4.
If you think you may have a loop that would be useful for me, you can
verify it using the sndfile-info program that ships with libsndfile
(its in the sndfile-programs package in Debian/Ubuntu). Running
on candidate WAV files produces something like this:
$ sndfile-info file.wav
========================================
File : file.wav
Length : 146982
RIFF : 146974
WAVE
fmt : 16
Format : 0x1 => WAVE_FORMAT_PCM
Channels : 1
Sample Rate : 44100
Block Align : 2
Bit Width : 16
Bytes/sec : 88200
data : 145408
LIST : 110
INFO
ICOP : Paul Bennet, 1999
ICRD : 1998-09-21
IENG : Paul Bennet
ISFT : Sound Forge 4.0;Sound Forge 4.5
acid : 24
Flags : 0x0002 (Loop,RootNoteValid,StretchOff,RAMBased,??Off)
Root note : 0x34
???? : 0xFFFF8000
???? : 0.000000
Beats : 4
Meter : 4/4
Tempo : 145.576584
strc : 1372
End
What I need is a file with "Meter" of something other than 4/4.
If you have one, please email the sndfile-info output to me at erikd
at mega-nerd.com. I might also get you to email me the file if it looks
like it might help.
Cheers,
Erik
----------------------------------------
Sample Rate : 44100
Frames : 72704
Channels : 1
Format : 0x00010002
Sections : 0
Seekable : FALSE
Duration : 00:00:01.649
Signal Max : 32767 (-0.00 dB)
--
----------------------------------------------------------------------
Erik de Castro Lopo
http://www.mega-nerd.com/
Hi
for the shake of easiness I've pushed a release of screcord.lv2 to github.
screcord.lv2 is a ultra light LV2 recording plugin, which will save your
audio stream to disk.
screcord features stemless recording, means it open a new file when max
file size of wav file is reached.
screcord comes without GUI, but provide a record button, a format
selector, and a clipping indicator.
screcord use libsndfile http://www.mega-nerd.com/libsndfile/
regards
hermann
Please excuse x-posting:
Virginia Tech is seeking graduate student candidates for a competitive
Institute for Creativity, Arts, and Technology (ICAT) graduate research
assistantship starting with the academic year 2017-8. We are
specifically looking for students who are interested in studying in the
newfound Human-Centered Design (HCD) transdisciplinary individualized
PhD and/or Computer Science's Human-Computer Interaction (HCI). We are
also willing to consider Masters level applicants to the CS program with
focus on HCI. Additional assistantships may be available through the HCD
program.
HCD Option
Virginia Tech's new individualized Human-Centered Design (HCD) iPhD
transdisciplinary degree is closely aligned with the newfound Institute
for Creativity, Arts, and Technology (ICAT). iPhD is a degree designed
primarily to support individually tailored PhD experiences, allowing
students to actively participate in shaping their plan of study.
CS HCI Option
The graduate program in the Department of Computer Science at Virginia
Tech is on its way to becoming one of the top programs in the country.
The resources and facilities for research areas such as human-computer
interaction, bioinformatics, and high-end computing are state of the
art, and the growing list of specialized masters and doctoral degrees in
a variety of areas provides opportunities for students to concentrate
their research and study on the specific aspects of computer science in
which they are most interested. At Virginia Tech, students receive
firsthand instruction by award-winning faculty. In the computer science
department, that means that graduate students have the opportunity to
learn from the professors who have helped to define today's computer
industry. The Center for Human-Computer Interaction is a part of ICAT,
offering a diverse infrastructure and human expertise for cutting-edge
research.
The GAship in question seeks candidates with a solid research
background, and a strong sense of independence and self-sufficiency. We
are particularly seeking students interested in sonification, sound
spatialization, and interactive technologies (e.g. mind-body
technologies, laptop/mobile ensembles, hyperinstruments, installations,
mobile/wearable/physical computing, etc.). Knowledge of Max and
Pd/Pd-L2Ork programming environments, C/C++, Java, or equivalent is
strongly preferred, as well as skills in other domains (e.g. interactive
multimedia, K-12 education, 3D animation, etc.).
This is a renewable assistantship with a full tuition waiver and
a stipend. To be eligible candidates need to apply, be accepted, and
enroll in the new HCD iPhD or the Human-Computer Interaction Computer
Science program with primary focus in the computer music domain focusing
on one or more of the aforesaid areas of interest.
The successful candidate will be given an opportunity to participate in
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For questions and application information please contact Dr. Ivica Ico
Bukvic <ico(a)vt.edu>
For more information:
ICAT http://www.icat.vt.edu
DISIS http://disis.icat.vt.edu
L2Ork http://l2ork.icat.vt.edu
HCD http://hcd.icat.vt.edu/
Best,
--
Ivica Ico Bukvic, D.M.A.
Associate Professor
Creative Technologies in Music
ICAT Senior Fellow
DISIS & L2Ork Director
Virginia Tech
School of Performing Arts – 0141
Blacksburg, VA 24061
(540) 231-6139
ico(a)vt.edu
www.performingarts.vt.edudisis.music.vt.edul2ork.music.vt.edu
Hi all,
I was recently thinking about how powerful, yet maybe underestimated,
the JCAK freewheel mode [1] is.
As far as I know this is used by Ardour, Qtractor and MusE for
export/bounce, 'internally'.
Call me old school, but I like to use various (lots) of audio software
on Linux in a modular way. I have stated many times that, in my humble
opinion, modularity plus inter-connectability of applications though
JACK is, from a creative point of view, a killer feature.
So, coming to the freewheel point. Wouldn't it be great if 'all jack
audio' software were freewheel-ready, so that if I activate a switch,
all applications enter freewheel mode and when I activate recording in
my favourite DAW, connected to some of my favourite (standalone!)
synths, connected to my favourite sequencer, everything is recorded
'faster than realtime' in the DAW?
Or also: isn't it startling that in the digital era one should 'record'
stuff internally in real-time??
What kind of work/burden would becoming freewheel-ready pose on existing
applications?
I do can see some cases where this just wouldn't work or might not even
be desirable: for example Pure Data and similar software which by
definition is realtime.
Hopefully this can start some (civilized!) debate on the topic and it
also includes other topics such as (creative) workflow, etc.
Lorenzo.
[1]
http://www.jackaudio.org/api/group__ServerControl.html#gad65c741a1d1d2fb09e…
Hi all,
Just a quick note to say my new album (Patchwork Cacophony, Five of Cups)
was released today.
Once again, almost everything was produced on Linux using Ardour and other
Open Source software (though there are very few soft synths -- it's mostly
olde fashioned drums, guitars, voices and physical keyboards).
Same as last time, if anyone mentions Ardour when buying a copy I'll
collect up £1 per sale for a donation to Ardour development. In truth with
music sales being what they are these days it's unlikely to feed a proper
developer caffeine habit for even a week, but it should be able to eclipse my
all-time revenue from Spotify ;)
The music is prog rock, with a strong emphasis on keyboards and classic-era
sort of sounds rather than more metal-influenced which is seeing a bit of
a rennaissance these days.
You can stream the whole thing for free on Bandcamp:
http://patchworkcacophony.bandcamp.com/
Thanks again to everyone in the Linux Audio world, from the ALSA and FFADO
guys, obviously Ardour, but also things like Scribus and Gimp. It constantly
amazes me how much support there is for producing a thing like this without
recourse to proprietry software. Long may it continue!
Ben
Hi all,
this is the last try I make to solve my problem. I wrote everybody, in
every site/forum, read everything, followed tutorials without any solution.
As written in the title, I have a Fast Track Pro connected via USB on
Ubuntu 12.04 LTS, I have Ardour 5.1.0 and QjackCtl 0.9.3
The problem is after many, many tries, I have the signal coming from the
card arriving in Ardour, I see the v meters move but I cannot hear any
sound in the headphones or from the speakers.
The same happens both in Hydrogen and Rakarrack.
The last thing I did is downloading the fast-track-pro.conf file from
Joe Giampaoli
<https://joegiampaoli.blogspot.it/2011/06/m-audio-fast-track-pro-for-debian-…>
tutorial. Since it's for Debian, I've only downloaded the file and
modified as per his instructions.
Then I modified it again as per instructions in this thread
<https://linuxmusicians.com/viewtopic.php?t=11016> in Linux musicians' site.
I'm also in touch with Joe but until now I haven't found any solutions
for this.
This is the parameters in QjackCtl preferences:
*Device
*hw:0 HDA Intel
hw:0,0 ALC262 Analog
-
hw:5 Fast Track Pro
hw:5,1 USB Audio #1
My laptop characteristics are the following:
* Ubuntu version 12.04 (precise) 32 bit
* Kernel Linux 3.13.0-101-generic
* GNOME 3.4.2
* RAM: 3,9 GiB
* CPU: Intel Core2 Duo CPU P8400 @ 2.26GHz × 2
Relatively to the computer audio system:
*leo@leo-R710:~$* arecord -l && aplay -l
**** Lista di CAPTURE dispositivi hardware ****
scheda 0: Intel [HDA Intel], dispositivo 0: ALC262 Analog [ALC262 Analog]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
scheda 5: Pro [FastTrack Pro], dispositivo 1: USB Audio [USB Audio #1]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
**** Lista di PLAYBACK dispositivi hardware ****
scheda 0: Intel [HDA Intel], dispositivo 0: ALC262 Analog [ALC262 Analog]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
scheda 0: Intel [HDA Intel], dispositivo 3: HDMI 0 [HDMI 0]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
scheda 5: Pro [FastTrack Pro], dispositivo 0: USB Audio [USB Audio]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
scheda 5: Pro [FastTrack Pro], dispositivo 1: USB Audio [USB Audio #1]
Sottoperiferiche: 1/1
Sottoperiferica #0: subdevice #0
*leo@leo-R710:~$* lsusb
Bus 001 Device 002: ID 1058:1100 Western Digital Technologies, Inc.
Bus 001 Device 003: ID 0ac8:c302 Z-Star Microelectronics Corp. Vega USB
2.0 Camera
Bus 006 Device 002: ID 046d:c069 Logitech, Inc.
Bus 007 Device 003: ID 0763:2012 Midiman M-Audio Fast Track Pro
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 003 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
Bus 007 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
Bus 008 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
*leo@leo-R710:~$* cat /proc/asound/modules
0 snd_hda_intel
5 snd_usb_audio
*leo@leo-R710:~$* cat /proc/asound/cards
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xf6200000 irq 47
5 [Pro ]: USB-Audio - FastTrack Pro
M-Audio FastTrack Pro at usb-0000:00:1d.1-1, full
speed
I've downloaded a low latency kernel and installed but then I used
Ubuntu Tweak to clean my computer, as I always do after I made an
update, and it deleted also the kernel because it said it was older than
the actual.
The thing that pisses me more it's the Fast Track works perfectly on
Cubase in Windows, but I don't want to go to that crappy system for any
reason on the earth.
As I wrote, being not expert at all, I have no idea about what to do. I
would like to record with this card but if things go in this way yet,
I'll sell it and get another card suggested directly by Ardour.
There's a second hand Presonus 1818VSL so I might go for that.
The strange thing is the guy at the address I wrote above, Linux
Musicians, seens to have solved in the way he wrote.
This thing is tragic, you have no idea how tired I am.
Anyways.
We'll see if there's a way to solve it.
Thank you all.
Leo
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> >>> Warning. 4GB limit on wav file almost reached.
>
> and the .wav file stops at 4GB.
>
> So it does seem to be a problem with the 32-bit version of jack_capture.
I don't think so. Jeanette had the same problem when using jack_rec,
and I can't think of any reason why jack_capture should behave
differently when running on a 32 bit OS. It's far more likely to be
caused by the filesystem, or something else. I've cc-ed
Erik de Castro Lopo, the author of libsndfile. He probably
knows more about what could cause max file sizes of 2.1GB.