I've now finally gotten an Analog Devices ad1988B with six in and ten
out + digital - all in glorious 24bit/192K, which I find mighty
impressive for an onboard three dollar chip. The sources for alsa-1.14
were included on the CD that came with the (Asus) box, and this is also
what I am using now.
The driver has fixes and workarounds for various note-book
configurations with built in speakers and microphones as well as various
numbers of physical multi-purpose ports that connects to the outside
world. Unfortunately nobody thought of the possibility of a vendor wild
enough to actually simultaniously implemement /everything/ this chip has
to offer.
Surround 7.1 on the "six-stack" on the backside is supported by alsa,
but appears to then replicate the "surround front" to the headphone
socket on the physical front panel which may or may not be what you
expected. Getting skype and friends out of the mix - perhaps even on its
own little /dev/dsp1 for closed source legacy? - would in my opinion
have been more like it.
Any ideas how to solve this? There is no such ugly invention as
"surround 9.1" nor "surround 7.1+2" as of yet ... But that could of
course be changed :-)
I have some other minor issues as well, like the headphone amp being
activated on the backpanel port labelled "line" but not on the
frontpanel jack labelled "headphones" ... But those appears to be
comparatively trivial to fix by comparing to how all the other special
case configurations are implemented.
--
Pedro Lopez-Cabanillas wrote:
> Maybe the default sound device is still /dev/dsp using the OSS API. Can you
> test if the OSS emulation is working properly in your system?
For some reason I had not tested HighC on my 64-bit system (64 Studio)
until this morning. It works !
Okay, at this point I think it's a distribution problem, not a problem
with coding practice or ALSA configuration. However, your pointers
certainly got me going in the right direction. I'll contact the
maintainers of JAD 1.0 to see if they've done anything unusual with
OpenSUSE's default Java settings.
Thanks again, Pedro, your assistance has been hugely appreciated. :)
Best,
dp
Hi all,
since the price of fossil fuels is quickly going up to unsustainable levels
the mass production of electric cars
is only a couple years away. (there are already a few nice around but not
yet high volume production and cost an arm and a leg)
http://en.wikipedia.org/wiki/Electric_car
One peculiarity of electric cars is that they emit almost no noise and
sometimes engine noise could come handy.
For example blind people or even distracted pedestrians about to cross a
street cannot hear electric cars coming which
can lead to accidents, injuries and even loss of life.
There are already electric car companies offering artificial engine sound in
their cars.
A linux based embedded system would probably be a nice candidate to provide
such functionality. (ARM cpu, some RAM and audio DAC,
or even a nano-itx x86 board which could do other additional things at the
same time, in-car computer etc)
A nice system would be one which allows pluggable sounds, for example
ferrari or porsche sound :)
My question is what's in your opinion the best method to simulate realistic
car sounds without making the system too complex or too costly.
I was thinking whether multisampling the engine sound at various RPMs
(rotations per minute, frequency proportional to RPM) could be a solution
or if some kind of synthesis would be needed.
The sound generator should in theory take only one input variable, the
motor's RPM (which can expressed as 0..100%) and then generate
a sound with a frequency proportional to the RPM.
A few ideas that come to mind which could provide varying degrees of realism
(I don't know because I did not try).
1) sampling an engine sound at middle RPM and then pitchshifting it with
formant correction up and down.
easy to do but probably does not sound so good at extreme pitchshifting
ranges.
2) sampling an engine sound at various RPMs, the more the better let's say
more times per octave and then using some sort
of sample pitching (standard interpolation or pitchshifting like rubberband)
and then crossfade between samples.
But if the samples to be crossfaded need probably to be phase-aligned
otherwise it could lead to chorus style artifacts etc.
3) sampling a full, very slow RPM sweep both upwards (min to max) and
downwards (max to min), lets say a couple of min per direction.
Then the sound generator figures figures out the RPM direction (by comparing
two subsequent values or some shortterm mean value) and then
plays the sound using the right position in the wave. So for example if you
floor the accelerator you have to play the upwards audio wave
but you have to skip lots of parts so you need some trick to make it sound
smoothly (time-compression comes to mind but I'm not sure if it is ideal)
Not sure how hard it would be to sample those sound sweeps. You need to
accelerate very slowly which can be tricky. Perhaps you need some
servo motor which can precisely move either the accelerator pedal or is
attached directly to the accelerator cable. Not very userfiendly.
4) sample the engine sound at various RPM. feed into a DSP algorithm which
analyzes the timbre and then builds a model for a synthesizer.
any other ways I am missing ?
What's the best trade off between realism and complexity in your opinion ?
Thanks for your comments.
cheers,
Benno
Anyone knows a good vector drawing program for Linux ?
Absolute requirements are:
- Lines, arrows, boxes, circles, etc.
- Linewidths and styles, colors, filling.
- Text
- PDF or PS export.
- PNG and JPEG import (no bitmap editing required).
- Accuracy.
I've been using TGIF for years, but I'm more and
more being blocked by its main flaw which is that
it seems to use a unit of 0.2mm internally (in
metric mode) which is orders of magnitude too big.
Tried QCAD and INKSCAPE, both fail basic
requirements (and have other problems).
Ciao,
--
FA
Laboratorio di Acustica ed Elettroacustica
Parma, Italia
O tu, che porte, correndo si ?
E guerra e morte !
The CLAM team enraptured to announce the 1.3.0 release of CLAM [1], the C++
framework for audio and music, code name ''The Shooting of the Flying Plugins
release''.
Highlights of this release are:
- Automatic binary generation of LADSPA plugins containing the network you are
editing in NetworkEditor.
- Also a new simple API to code CLAM based Ladspa by hand. See [2]
- More FAUST integration into network editor: edit faust code, compile,
reload, view the svg diagrams (Natanael Olaiz GSoC)
- Lots of usability enhancements on the NetworkEditor: cut&paste, context
menus to connect ports, keyboard shortcuts, default double click actions, and
a processing tree filter (Natanael Olaiz GSoC)
- Annotator has also enhanced its functionality (Wang Jun GSoC):
- You can build a project that aggregates content from several extractors
- Extractors may have a config file
- Extractors can write back data (useful if the extractor is a database of
webservice and needs to upload modifications)
- New ProgressControl widget and paired AudioFileMemoryLoader processing to
support seeking (Pawel Bartkiewicz GSoC)
- A bunch of new 3D spatialization processings from CI Barcelona Media[3]
audio research group.
- Scripts [4] and graphical front-end [5] to generate a native CLAM plugin
project from scratch.
- Experimental Python bindings [6]
- TickExtractor example is compiling again (many thanks to Amaury Hazan from
MTG-UPF)
- Development deployment for Windows native compilation using MinGW (Wang Jun
GSoC)
And a lot of small nice features and fixes you will appreciate for sure.
Source and binary packages for different platforms are available at the CLAM
download page [7]. See also: development screenshots [8], the CHANGELOG [9],
and the version migration guide [10].
We are also very excited on what next releases promise us. Some ongoing work:
- Generating other types of network based plugins and programs (LV2, JACK,
VST...),
- Subnetworks (Natanael Olaiz GSoC)
- Improved OSC support, 3D scene descriptors parametrization receivers
processings and Blender exporter to the spatialization processing
choreographer. (Natanael Olaiz GSoC) [11]
- Typed controls (Francisco Tufro GSoC)
- A new musician-oriented standalone chord extraction application (Pawel
Bartkiewicz GSoC)
[1] http://clam.iua.upf.edu
[2] http://iua-share.upf.edu/wikis/clam/index.php/Building_a_LADSPA_plugin
[3] http://www.barcelonamedia.org/
[4]
http://audiores.uint8.com.ar/blog/2008/07/07/clam-processing-generator-scri…
[5]
http://clam.iua.upf.edu/wikis/clam/index.php/Image:ProcessingCodeGenerator.…
[6] http://audiores.uint8.com.ar/blog/2008/08/03/interactive-clam-programming/
[7] http://clam.iua.upf.edu/download.html
[8] http://clam.iua.upf.edu/wikis/clam/index.php/Development_screenshots
[9] http://iua-share.upf.edu/svn/clam/trunk/CLAM/CHANGES
[10] http://iua-share.upf.edu/wikis/clam/index.php/Version_Migration_Guide
[11] http://dadaisonline.blogspot.com/search/label/blender related blogging
--
David García Garzón
(Work) dgarcia at iua dot upf anotherdot es
http://www.iua.upf.edu/~dgarcia
I downloaded alsa-firmware and fxload from medibuntu, it loads the
firmware fine, the lights go on and alsa recognizes it. I get errors in
dmesg and jackd doesn't work.
shawn@ubuntu:~$ cat /proc/asound/cards
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xfebfc000 irq 19
1 [K88 ]: USB-Audio - Keystation Pro 88
Evolution Electronics Ltd. Keystation Pro 88 at usb-0000:00:1d.7-2.4,
full spee 2 [USX2Y ]: USB US-X2Y - TASCAM US-X2Y
TASCAM US-X2Y (1604:8007 if 0 at 007/006)
I get this in dmesg:
[ 95.442460]
ALSA /build/buildd/linux-ubuntu-modules-2.6.24-2.6.24/debian/build/build-rt/sound/alsa-driver/usb/usx2y/usbusx2yaudio.c:313:
Sequence Error!(hcd_frame=345 ep=10out;wait=345,frame=342).
[ 95.442488] Most propably some urb of usb-frame 345 is still missing.
[ 95.442491] Cause could be too long delays in usb-hcd interrupt
handling. [ 95.479404]
ALSA /build/buildd/linux-ubuntu-modules-2.6.24-2.6.24/debian/build/build-rt/sound/alsa-driver/usb/usx2y/usbusx2yaudio.c:313:
Sequence Error!(hcd_frame=382 ep=10out;wait=382,frame=379).
[ 95.479429] Most propably some urb of usb-frame 382 is still missing.
[ 95.479431] Cause could be too long delays in usb-hcd interrupt
handling. [ 96.012293]
ALSA /build/buildd/linux-ubuntu-modules-2.6.24-2.6.24/debian/build/build-rt/sound/alsa-driver/usb/usx2y/usbusx2yaudio.c:313:
Sequence Error!(hcd_frame=5 ep=8in;wait=1026,frame=2). [ 96.012311]
Most propably some urb of usb-frame 1026 is still missing. [ 96.012314]
Cause could be too long delays in usb-hcd interrupt handling.
I get this with jackd:
23:01:43.405 JACK is starting...
23:01:43.406 /usr/bin/jackd -R -dalsa -dhw:2 -r96000 -p1024 -n3
23:01:43.434 JACK was started with PID=22944.
jackd 0.109.2
Copyright 2001-2005 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
JACK compiled with System V SHM support.
cannot lock down memory for jackd (Cannot allocate memory)
loading driver ..
apparent rate = 96000
creating alsa driver ...
hw:2|hw:2|1024|3|96000|0|0|nomon|swmeter|-|32bit control device hw:2
configuring for 96000Hz, period = 1024 frames (10.7 ms), buffer = 3
periods ALSA: final selected sample format for capture: 24bit
little-endian ALSA: use 3 periods for capture
ALSA: final selected sample format for playback: 24bit little-endian
ALSA: use 3 periods for playback
ALSA: cannot set hardware parameters for playback
ALSA: cannot configure playback channel
cannot load driver module alsa
no message buffer overruns
23:01:44.454 JACK was stopped successfully.
23:01:44.455 Post-shutdown script...
23:01:44.455 killall jackd
jackd: no process killed
23:01:45.053 Post-shutdown script terminated with exit status=256.
23:01:46.681 Could not connect to JACK server as client. - Overall
operation failed. - Unable to connect to server. Please check the
messages window for more info.
Help please!
Miesco is online now Report Post Edit/Delete Message
Hi dear LAU and LAD users,
I read somewhere that it was possible to change the USB bus sample
rate. I tried to do this with the 'setpci' command by changing the
'latency_timer' PCI register of my USB controllers, but I don't think
I am on the right way. Does anyone knows if that is possible and makes
sense, and if so, how to do it ?
Thanks.
Regards,
Adrien
Dear Linux-Audio-Users
Denemo is a music notation program for Linux and Windows that lets you rapidly enter notation for typesetting and lets you produce beautiful output via the LilyPond music engraver.
Now version 0.7.9 is released. Is has many great features and improvements which make Denemo unique as a notationeditor in Linux/Win/Mac. It can be downloaded as source, Debian-package, Fedora-Package or Windows-installer on http://www.denemo.org or on our Savannah page: https://savannah.gnu.org/projects/denemo/ Please feel free to tell us your opionion, comments, bugs or feature-wishes in a way you like it: http://www.denemo.org/?q=node/1 .
We also need help!
While Denemo itself is developing very fast and well there is still no JACK- and/or ALSA- midi support. This is needed that you can use Denemo as a Sequencer on Notation-base. Currently Denemo uses CSound for direct play or exports a Midi first and plays it afterwards. If you are willing to help us in this point (or in any other!) please contact our mailinglist or, if you like because its shorter, in this list or directly to my adress.
This release includes:
Allow LilyPond editing within Denemo.
Gallery of examples - Ossia, Multi-measure rests, cues, cautionary accidentals, reminder accidentals. Rehearsal Marks ...
Printing of excerpts as images (e.g. for inclusion in texts).
LilyPond import improved
Midi import improved.
Better handling of keyboard shortcuts.
Contexts (Piano context, choir context etc)
http://www.denemo.org
Community Links: http://www.denemo.org/?q=node/1
Hi everyone,
what would be your suggestions as to the best setup (parameters/patches)
for customising 2.6.25 regarding scheduling and realtime audio?
Thanks
Victor
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Hey LAD's,
With Ico out of the country, Burkhard graduating, Thorwil overloaded and
France on summer holiday, linuxaudio.org is running critically low on
maintainer(s). Would some of you guys mind to keep an open eye?
I've just posted a "wanted" sign on http://linuxaudio.org asking to join
LAO for the good cause. - I know you LADs are always busy, but trust you
to welcome and guide volunteers.
There'll be an official announcement after the summer-break. We have
been preparing a report on linuxaudio.org services where we also address
future perspectives. currently pending in the comittee. - Nothing to
worry about; we'll continue to manage linuxaudio.org services
unobtrusively; I suggest to save some of the entropy-reducing ideas for
upcoming www consolidation tough.
linuxaudio.org is a drupal CMS to reflect the consortium.
I will not argue about wiki, forum & admin-permissions etc until Ico is
back but if you're interested I can arrange editor accounts for news and
articles.
Greetings,
#robin
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