Reiner wrote:
>> - there's patch for PD that provides JACK-support
>> ->
>> http://sourceforge.net/mailarchive/message.php?msg_id=1169519
>Actually, this is no longer current. Check the pd list archive for
>newer approaches to pd jackification. Personally, I now use Günther
>Geiger's patch applied to pd 0.35 and it works quite well. Together
>with the plugin~ object for LADSPA plugins the patched pd can be used
>as effect rack for ardour.
This would make an excellent quicktoot.
If you have the time to send me a brain dump I will happily format it.
Please consider the benefits that this would provide to pd, ardour and
Linux audio users.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
Hi,
I am a musician and also a programmer. I have many years experience of
coding but not in the audio field. Any tuts around? How do I get into
it?
Cheers for any info.
-Lea.
>> You can calculate your tranformation for the input signal S(x_i) once
>> And the same transformation for S(x_i)+1 again.
>
>Won't that just give you the gradient at point x_i, ie. d/dt(S)?
yes.
>We are talking about frequency domain aliasing here,
oh yeah i see. sorry.
> which is when you
>generate partials that would be above the nyquist frequencyi, so they get
>reflected down into low frequencies. It is not directly related to the
>differential of the signal, though a high differential is often indicative
>of an aliasing problem.
>
>Typically you prevent audio aliasing by generating the waveform in a way
>so that it contains no partials above nyquist, or by generating it at a
>sufficiently high sample rate that there are none, then decimating down.
but you can do the pretty same trick for the frequencies:
take two different numbers with no common divisors
(or just a pair of prime numbers) as sampling rates
and see how the output signal changes.
The difference give you some linear combination of aliased frequencies.
__________________________________________________
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>> my feeling is that oversampling alone will not do, although
>> you're probably right in that it will get rid of most of
>> the aliasing.
>
>Its how most people tackle to problem. I think I can get the valve cheap
>enough that it will be practical.
I'm not really familiar with what are the techniques of anialiasing
in use. But I can see the following way.
You can calculate your tranformation for the input signal S(x_i) once
And the same transformation for S(x_i)+1 again.
I mean two similar input signals differing in one bit of value.
So when you have two output saignals you can calculate
the error due to aliasing: abs( F(S(x_i)) - F(S(x_i)+1) )
And apply some techniques to eliminate this aliasing
ranging from qubic splines to plain dithering applied to
the output signal.
It seems to me that this is much better than oversampling.
Especially for nonlinear transformations.
(Or is it just a trivial thing and it doesnt work and
this is not what you discuss?)
nikodimka
>
>> i'm thinking about how to apply the technique to produce
>> square clipping, or better a [0 .. 1] range of clipping,
>> but the integration approach is awkward.
>
>I doubt that the electronics ever produces hard square clips.
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>Well, I just added myself and discovered the first problem -
>you can't change anything after you hit that final submit :-\
>I can guarantee that I will be relocating in the next
>4 months.
No worries. I/we'll have it more configurable by then.
>also, I didn't realize that the "Company Name" is necessary
>to get the web URL to show up.
I haven't taken this scenario into account. I'll look at fixing it. The
data is still available I just have to add an if statement. Actually I'm
surprised it didn't show up on the preview.
>Good idea, I think.
Thanks. This is definitely a work in progress.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
> this is what's on the page:
>Creative Labs | Soundblaster Live Platinum | EMU10K1 | Install |
>(4)[A][B]
> bottom of the page:
>
>(4) Hardware mixing supported
>
>...
>
>NOTE: Just because an I/O is listed does NOT mean it is guaranteed to
>be supported. Please check the mailing
>list archives before making purchase decisions based on requirements
>forIEC958 or ADAT I/O.
>[A] IEC958 RCA Input
>[B] IEC958 RCA Output
>which does not have a lot of informational value (I know what it does
>not mean but what does it mean?) and it definitely does not imply
there >is a problem of getting the docs to support certain functionality.
Hmm. It has a certain informational value although I will concede that
when I started getting into alsa these comments were cryptic for me
then. They assume a certain level of knowledge is known when referring
to the notes.
I'll try to think of a way to word it for users who may not have the
requisite knowledge.
Apart from that at the top of the page it's explicitely stated that some
companies have not provided the required docs to enable support for a
device. I have always noticed those comments however it may just be my
angle on things. I have made some changes to the matrix to be more
obvious that even if we have the docs we may not have all the necessary
parts. Although I would've thought that most people already know that
about Linux hardware support.
As you know the known bugs section in the driver notes gives explicit
information about things that we know do not work. I'll look making a
form for adding specifically to this file.
I would like to do that and also make it easier for people to post info
about supported devices. I have most of the work done but not finished
yet. As it is I currently stay up to 6:00 in the morning many nights of
the month. I'm lucky though as I only have to work 16 hours a week in my
paying job. I could be working many more hours but have made the
decision to sacrifice some of my current earnings in order to build
Boost Hardware and DJCJ. Along the way I have also picked up the alsa
website :)
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
I'm pleased to anounce that a database for people willing to provide
Tech Suppport to their local community has been setup. The service is
free of charge and hosted at
http://www.djcj.org/
The purpose of this database is to promote the professional arm of the
Linux Audio Developers community. It is intended to be of use to
potential clients who may be interested in getting a Linux Audio system
working but don't have the time or background knowledge to do the
installation or system maintainance?
The people and businesses presented in the database are not endorsed or
guaranteed by DJCJ.org but they are active members of the Linux Audio
Community. Payment for services received is encouraged. Rates are
decided by the parties involved.
This is intended to be a database for professional tech support. Please
let people know about it so that we can show the world we are more than
just a bunch of amateurs hacking in our spare time.
The database currently provides a very simple interface for adding your
contact details and there is also a contact form provided for potential
clients to easily get hold of you. Special thanks to Steve Harris and
Antti Boman for assistance with the internal code. It is guaranteed that
the interface will become much more user configurable over time.
Any feedback is welcome and appreciated.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
Just so people don't think the code has fallen off of the face of the planet ;-)
Currently I'm working on two things with the wine jack driver , one of which needs to be complete before I submit the patch, the other can wait until some ongoing jack support is complete.
1. The driver currently closes and opens jack connections synchronously with wodOpen and wodClose. The problem is that jack doesn't actually close the clients right away, sometimes it can take many seconds for this to occur. Apparently this is a known jack bug but I'm also wondering if this is a problem in wines pthread implementation of conditions(there is no support for them right now). Either way I've implemented a workaround that basically consists of opening jack connections and marking them as in-use or available as apps open or close audio devices. This behavior can be enabled/disabled via a define in the driver. This isn't complete yet but it should be soon and would prevent the lockups and delays that currently make things a pain.
2. I don't know of an easy way to do resampling so the input audio matches the sample rate that jack expects. What I want is to be able to do something along the lines of:
set_sample_rates(input_rate, jack_output_rate);
set_resample_algorithm(x); /* pick a reasonable algorithm based on the current cpu and latency requirements */
And then in the callback routine something like:
resample(audio_data_in, *audio_data_out);
I haven't found anything even close to this simple as of yet, if anyone has any suggestions feel free to mail me about them. There is someone on the jack mailing list working on a resampling library that is supposed to be easy to use and flexable. Either way, this isn't critical although your music sounds funny if your sample rates don't line up very well ;-)
Otherwise I'm also looking into a crash with using the jack driver and directsound that might be due to the client process and memory allocation.
After working on the arts driver I realized that the future of audio servers really is callback based. It is quite frustrating to see so little use of jack and so much use of poor solutions like direct hardware drivers(alsa/oss) or high latency servers like artsd. No offense intended to people that work on or use either of those audio solutions, it just doesn't make sense when you want to have multiple apps outputting audio at the same time or if you need low latency(high end audio, games?, etc). I'll see what I can do to get things cleaned up and a patch submitted in the next week, conditional of course upon Alexandre accepting jack support into wine. At least now I have someone other than myself with an interest in using jack and wine together ;-)
Thanks,
Chris
>
> From: "Kjetil S. Matheussen" <k.s.matheussen(a)notam02.no>
> Date: 2002/10/23 Wed PM 12:18:13 EDT
> To: linux-audio-dev(a)music.columbia.edu
> CC: wine-devel(a)winehq.com
> Subject: Re: [linux-audio-dev] Fwd: Opinions on running VST or DirectX plugins
> on Linux in real time
>
>
>
> On Wed, 23 Oct 2002, Taybin Rutkin wrote:
>
> > On Wed, 23 Oct 2002, Kjetil S. Matheussen wrote:
> >
> > > Could it be possible to overcome all problems mentioned just by simply
> > > adding jack-support to the wine-server? Then someone could make a simple
> > > windows-program for chaining directx and vst plug-ins.
> >
> > Someone has done this, I believe:
> > http://kt.zork.net/wine/wn20021011_139.html#2
> >
> Wow, it allready exist. Hey, wine-people, this driver is really needed for
> real-time audio-work, and for programs as sound-editors, multitrackers
> and interactive dsp programs as pd, jmax, and a lot of others. Artsd
> (f.ex) is not an alternative in its current form, and probably never will
> be either. Its something else. I guess Paul or some other competant person
> can explain a bit more why jack is so important.
>
>
> --
>
>
>
>
>
> -----Original Message-----
> From: Patrick Shirkey [mailto:pshirkey@boosthardware.com]
> Sent: Wednesday, October 23, 2002 11:38 AM
>
> >> Plus the sound matrix at
> >> http://www.alsa-project.org/alsa-doc/ doesn't say there
> are problems
> >>getting docs from manufacturer.
>
> >Perhaps someone should add a note.
>
> I thin there is right at the bottom of the page. But I will
> make a more
> noticible note.
this is what's on the page:
Creative Labs | Soundblaster Live Platinum | EMU10K1 | Install | (4)[A][B]
bottom of the page:
(4) Hardware mixing supported
...
NOTE: Just because an I/O is listed does NOT mean it is guaranteed to be
supported. Please check the mailing
list archives before making purchase decisions based on requirements for
IEC958 or ADAT I/O.
[A] IEC958 RCA Input
[B] IEC958 RCA Output
which does not have a lot of informational value (I know what it does not
mean but what does it mean?) and it definitely does not imply there is a
problem of getting the docs to support certain functionality.
> > professional system or not, I would expect it to be
> capable running
> >browser and mp3 player at the same time (even though I would not
> expect >this capability to be used in studio work).
>
> AFAIK this is possible if you use kde or gnome as your
> desktop and don't
> mind having artsd block the interface. Many other desktops
> support artsd
? it's not desktops but applications that need to support sound server.
you mean that the sound server is not installed as part of that dektop
environment?
> also but they are usually not installed by default. Hence
> anyone who can
> install an alternative desktop should also be willing to
> get the sound
> server running if they desire that functionality. This has
> been the case
> for at least the past three years.
yes, sound servers can be used with certain application, but what is
needed is something that supports ALL applications (or more precisely, that
is supported by all applications), or at least does not prevent other
applications from working (jack might turn out to be the solution - once
it's as good as planned it might become the only player in town)
erik
>> Plus the sound matrix at
>> http://www.alsa-project.org/alsa-doc/ doesn't say there are problems
>>getting docs from manufacturer.
>Perhaps someone should add a note.
I thin there is right at the bottom of the page. But I will make a more
noticible note.
> professional system or not, I would expect it to be capable running
>browser and mp3 player at the same time (even though I would not
expect >this capability to be used in studio work).
AFAIK this is possible if you use kde or gnome as your desktop and don't
mind having artsd block the interface. Many other desktops support artsd
also but they are usually not installed by default. Hence anyone who can
install an alternative desktop should also be willing to get the sound
server running if they desire that functionality. This has been the case
for at least the past three years.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem