Fallowing up the long discussion i'm trying to sort of give the
information you all seem to be missing.
there was a meantion of this, but not too many of you have paid any
here you can find an article about the current status of the protocol mess:
http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28)
obviously eas50 is good to go, but Ethernet AVB is right thing really.
the only thing that it's still work in progress, but many of the
proprietary vendors, which already have their own networking solution
(like Harman with HiQ-net, the one i can name of top of my head)
are involved in AVB stadard deveelopment.
The idea of AVB is to bypass the IP layer, which is right thing really.
you don't need to assign IPs to your audio nodes, really!
in avb you'd just have to select channels that nodes whant to listen to.
there is a fair bit of documentation on the ietf.org AVB group's page.
but XMOS is looking to be the best point of refference:
is think we should forget everything else and crack on with the XS1 AVB
their XS1 chips seem to be really great,
their are basically every innovative and open-source minded.
the official toolchain is LLVM-GCC based.
you can use C, C++ or their own XC.
XC is basically C with some stuff omited (like goto and floats)
and XMOS IO stuff added, don't just say WTF, look at it first!
you should also watch the videos here:
especialy the two about the "XMOS Architecture" and the AVB
some dev-kits are quite expencive, but that's due to low-volume really
there is alos a nice USB Audio kit!
plus there is alittle board that is cheap and has two RJ45's on it
I'm myself studying the XC book at the moment. And geting familiar with
the tool set :)~~
looks very exciting, cause these are the invovative chips!
ok, may be an FPU is really missing on XCore, but how many DSPs have
it anyway? well quite a few, but there was no FPU on dsps for ages! :))
also XC or C/C++ are so much more obvious then the bloody "menthal american military engeneers non-sense" called HDL-whatever!
Hope you will appreciate my excitment :)~ (l0l)
I managed to get SuperCollider and JACK running on my IGEP , on the pre-
configured Ubuntu on the SD, but of course the audio is still bumpy.
So... I'm looking for a RT kernel for this little machine... Anyone have any
I did find this one: http://beagleboard.org/project/omap-rt-patch/
but the project doesn't show much recent activity.
and... I also noticed that JACK didn't want to run with alsa as backend; oss
as backend works though (but gives the bumpy sound).
Anyone willing to share their experiences doing linux audio on ARM processors?
I want to make is a DSP module that improves the sound quality
of the sound coming from the laptop builtin speakers by applying DSP
I want to route all audio played back (from for
instance an offline WAV or MP3 file, from a movie, or streamed from the internet) to
pass through the
DSP plugin whenever headphones are NOT connected. When headphones are connected no processing should be applied.
want to place the DSP module as close to the hardware as possible to
make sure all audio is really routed through the DSP plugin so I always
the DSP improvements made to the sound.
Where should I place this module/DSP code?
- As a plugin to PulseAudio?
- As a plugin to ALSA?
- As its own virtual sound card?
- In the audio driver for my built-in sound card?
- As a kernel module?
I have tried loading the LADSPA-module using the module-ladspa-sink in PulseAudio but I am not sure this is the best solution.
I need to remove some limitations in module-ladspa-sink to get it working properly ( multichannel audio, only DSP processing for internal speakers).
1) Must be able to detect if headphones are connected
2) Must be able to process stereo and multichannel (5.1/7.1) formats. I
need the multichannel formats to perform binaural downmixing to
So it is important that I can receive multichannel audio and mix it down to stereo and output it to a stereo soundcard.
Alex, I hope you don't mind quoting part of your message on the list,
as the subject is of general interest.
Alex stone wrote offlist:
> Fons, i've read in the mailing list in the thread concerning mixers,
> that you have something you're working on.
> More than any other facet of working in software for audio/midi/vid,
> i've been frustrated and tired out by poor navigation options, and i
> don't think i'm on my own.
> The mouse is insufficent for ease, speed, and pain free day in day out
> use, and i ask you to at least consider (or not, as you decide)
> alternatives such as the one above, to make a difference in the way
> users commonly interact with this type of software.
I more than agree with this. Defining the way a user interacts with a
large software mixer is 90% of the work of writing one. Common GUI
methods just don't work for this sort of thing.
The thing that is slowly taking some form here will observe some
GUI guidelines quite strictly, in particular there will be *NO*
repeat *NO* resizing, scrolling or moving of windows, menus will
be used sparingly and be small, and popup dialogs will be used
only for operations that can be assumed to be infrequent (mainly
setting up things, selecting and connecting plugins, etc.).
When setting up a new session, you define the resources:
- number of channel strips
- number of real groups
- number of control groups
- number of visible sections
- number of strips per section
- number of jack/alsa inputs
- number of jack/alsa outputs
These are more or less fixed afterwards, in the sense that you
can't remove any, but probably I can allow to increase these
numbers (the point here is that all snapshots taken for such
a configuration must remain valid without requiring changes
to the resources). Anyway there is no penalty (apart from a
small amount of memory used) for providing more than you need.
A _section_ is a group of strips, typically 8, 10 or 12.
Normally you should select the section size so you can have
two of them side by side, as this facilitates many operations.
For each visible section there is a control strip, which
contains e.g. buttons to select which strips are shown.
You will have default sections, e.g. inputs 1-8, 9-16, 17-24,
etc, same for real groups and control groups. There are also
user-defined sections: these can contain copies of any strip
you want in any order you want. For example you could group
all inputs used for drums in a secion, or have a 'master'
section that contains mainly groups and maybe some solo mics.
There is some 'intelligence' in the section selection buttons,
for example, if you select to see inputs 17-24 in the right
section, you click that button (or use a kb shortcut), then
clicking again restores the previous view. Or selecting a
section that is already visible will swap the two, etc.etc.
Each visible section is vertically divided in two, the bottom
half showing the faders, channel ID, mute, solo etc. most of
the time, while the top half can show parts of the strips, e.g.
EQ, sends or dynamics. Again this layout is configurable to some
extent, you can e.g. select to have some aux sends visible all
the time. The top part of each visible section can also be used
for plugins, metering, 'visual' panning, or whatever extra
functions that would need to be displayed.
All Jack/Alsa ports are generic, each of them can be assigned
to any function inside the mixer. That means e.g. that Jack
connections can remain fixed no matter how you 'rewire' the
mixer, and this in turn means that nothing is lost if you
connect directly to a multichannel Alsa device, e.g. a MADI
The mixer will have its own plugin interface, and it will not
accept any of the existing standards. If someone wants to use
a LADSPA or LV2 I will consider porting it. Condition for this
will be the *quality* of the plugin, and nothing else.
Ciao from very hot Crete (I'll be cooling down 20m below the
water surface in a short time).
There are three of them, and Alleline.
Comments anybody :-)
I have heard from a reliable source, inside Digidesign, that they
actually have Protools running on Linux, and that the port from OSX
isn’t that hard, but are under contract obligation to Microsoft to not
release a Linux port. Otherwise Microsoft can revoke their access to
the Windows SDK.
Anybody have any ideas on
(warning contains large images,
for error text only).
Summary: after successfully writing a spectrogram image in my first
Vala programming http://spekle.googlecode.com/ (a displayless-version
Alexander Kojevnikov's http://www.spek-project.org/ ) I get the
following error, despite not explicitly calling free() out of my Vala
Spekle: saving 'j_s_bach-d_minor_invention.ogg.png'... DONE!
*** glibc detected *** spekle: free(): invalid pointer: 0x00007fa7e05ca010 ***
======= Backtrace: =========
The errors seems to occur intermittently, and more often when using
media that is ogg. Retrieving via HTTP seems to reduce the error
potential, but it still happens that way too...
The relevant libraries and compiler I'm using:
See http://spekle.googlecode.com/svn/trunk/README for instructions on
obtaining by 'svn' and building on Linux.
As I was checking out the new mudita24, I noticed that the peak
line on the meters doesn't go to zero after the signal is
'silence'. I have to manually click "reset peaks".
Is this on purpose?
PS: nice overhaul, great work, thanks.