On Sat, 2007-03-31 at 13:17 +0200, Dennis Schulmeister wrote:
Thanks Paul and Pau for pointing me to GStreamer. I took a glimpse on
some tutorials yesterday and my impression is that GStreamer is easier
to handle than I thought.
whatever you do, don't forget to start using the playbin element
On Fri, 2007-03-30 at 16:28 +0200, Dennis Schulmeister wrote:
Hi,
My question is if there's any high-level sound API for python. I don't
want to process audio. Instead I'm looking for an API (or a combination
thereof) which could be used to implement a small media player feeding
several
On Thu, 2007-03-15 at 13:12 +0100, Malte Steiner wrote:
I wonder what are the alternatives, if there are any I would jump
instantly. Windows Vista in which nothing works and wastes CPU cycles to
spy and torture you? OSX which comes with a hefty price, limits your
choice (and money) and
On Wed, 2007-03-14 at 08:56 -0400, Paul Coccoli wrote:
Besides, what you want is probably impossible. You can't have
pre-comiled, binary-only drivers *and* a custom kernel.
in theory, you certainly can. but the kernel development team, and linus
in particular, are not interested in an
On Wed, 2007-03-14 at 10:21 -0400, Lee Revell wrote:
Why should Linux sacrifice stability just so vendors can keep their
hardware interfaces secret?
although i broadly agree with lee on most things, i think that this way
of approaching this issue is unnecessarily confrontational. just flip it
On Mon, 2007-02-19 at 14:18 +0100, Stefano D'Angelo wrote:
How often are more than one plugin with the same control inputs used in
paralel? I was rather thinking of colapsing (or swapping) plugins in
series. They'd have to be linear and time invariant, of course.
Or maybe plugins could
On Mon, 2007-02-19 at 18:10 +0100, Stefano D'Angelo wrote:
nope. thats not a linear arrangement of the two mono plugins, but a
parallel arrangement. the signal going to each instance of the mono
plugin is different.
I'm obscure even in Italian, I can just imagine how it can sound like
On Mon, 2007-02-19 at 13:33 -0800, Jonathan Ryshpan wrote:
On Mon, 2007-02-19 at 13:18 -0800, vreuzon wrote:
Jonathan Ryshpan a écrit :
The above recording session was done while jack was Stopped. Would
jack work better if it were Rolling?
This play button refers to jack transport
On Mon, 2007-02-19 at 19:49 -0300, Camilo Polyméris wrote:
Julien Claassen wrote:
Hi!
I'm sorry to ask that here, but it seems I can't get an anser anywhere
else.
Does the libstdc++ support UTF-8 strings? Or is there some simple example
code snippet somewhere to derive/modify
On Wed, 2007-01-31 at 21:35 +0100, David Olofson wrote:
On Wednesday 31 January 2007 21:02, Michael Ost wrote:
[...]
We have a 32 sample setting (.7 msecs) in Receptor which I have yet
to see in a Windows driver. And it actually works with some plugins,
even a large sampler like Synthogy
On Sat, 2007-01-27 at 16:05 +1100, Fraser wrote:
Hi All,
I've been converting my old VST plugins over to LADSPA and have come
across something in the api which I really miss - the inability separate
the algorithmic to the displayed value of a parameter.
I'm finding this inability is leading
On Wed, 2007-01-24 at 16:06 +0100, Jay Vaughan wrote:
At 20:08 +0100 22/1/07, Stefano D'Angelo wrote:
What I'd like to work on is a sound processing architecture (LADSPA,
VST, DSSI, etc.) wrapper, which hides the details of a particular
implementation to audio program developers.
Nice idea.
http://www.freshpatents.com/Low-latency-real-time-audio-streaming-dt20060406ptan20060074637.php?type=description
in which Microsoft patents designs partially implemented by OSS 10 years
ago and fully implemented by ALSA 5 years ago. Wrapping this up in
windows API nonsense obscures the basic fact
On Tue, 2007-01-23 at 14:31 +, John Rigg wrote:
On Tue, Jan 23, 2007 at 08:53:13AM +1100, Erik de Castro Lopo wrote:
Hi all,
SecretRabbitCode was recently included in a test of a number of
commercially available sample rate converters and while it wasn't
the best, it certainly
On Tue, 2007-01-23 at 08:53 +1100, Erik de Castro Lopo wrote:
Hi all,
SecretRabbitCode was recently included in a test of a number of
commercially available sample rate converters and while it wasn't
the best, it certainly didn't disgrace itself either.
congrats Erik. as you said, not the
On Fri, 2007-01-19 at 11:54 +0100, Marc-Olivier Barre wrote:
Ok, Apparently forum addicts have decided to hijack this thread
though, the original question was : do you want LA* _lists_ (LIST is
the keyword here, can't stress it enough) to be moved to
linuxaudio.org.
I have noting
On Tue, 2007-01-16 at 07:30 -0500, Dave Phillips wrote:
Damon Chaplin wrote:
What are the recommended books to read for people new to audio
development? (Covering things like synthesis techniques, effects
processing and basic acoustics stuff.)
On the bottom of the Documentation section of
On Mon, 2007-01-15 at 20:00 +0100, Richard Spindler wrote:
2007/1/15, Dave Phillips [EMAIL PROTECTED]:
This link may have been posted here before now, but it bears repetition :
http://www.cs.auckland.ac.nz/~pgut001/pubs/vista_cost.txt
Nasty stuff planned in Redmond.
It seems as
On Mon, 2007-01-08 at 23:23 +0100, Milan Mimica wrote:
Hello!
We're designing a new sound subsystem for allegro game programming
library, and we would like to take advantages of multiple hardware voice
capabilities.
very few audio interfaces have this feature anymore. most h/w makers
seem
On Wed, 2007-01-10 at 18:02 +0200, Jussi Laako wrote:
Paul Davis wrote:
in general, you should forget about the h/w capabilities of an audio
interface. for every user that has a device with some interesting
qualities, there will be 10 who do not.
welcome to winmodem for audio
On Wed, 2007-01-03 at 16:28 -0500, kind king knight wrote:
I need a framework of a sequenced sample player. This is becouse I want
to start my own project, and don't want to invent everything from
scratch. Ofcourse there is lot of this kind opensource applications, but
I need the simplest.
-
On Wed, 2006-12-20 at 15:43 -0800, Anthony Green wrote:
I understand that LADSPA and friends specifically exclude any
functionality around how to find and load plugins, but it seems that a
lot can be gained by introducing some standards in this area.
As a package of audio apps/plugins for
On Thu, 2006-12-21 at 02:02 +0100, Leonard Ritter wrote:
Hi Anthony,
I guess most of us use the sample enumeration c code included with the
LADSPA sources as starting point. This code expects a LADSPA_PATH
variable to be set. As a fallback, I suppose most programmers
added
On Sun, 2006-12-17 at 19:43 -0800, oscar si wrote:
Hello:
Could anyone please tell me if there are sample driver codes for PCI
based sound card that are not using either alsa or oss?
why would you want such a thing?
On Wed, 2006-11-22 at 15:40 +0100, conrad berhörster wrote:
Hello all,
is there a way to restart jack out of an application, or is this the reason,
why ardour need a running jackd.
i want to write an application with a reinit app button and need a way to
restart jackd, if it has been
On Thu, 2006-11-16 at 18:50 +0100, Takashi Iwai wrote:
Hi,
is anyone interested in participating in the third Desktop Architects
Meething (DAM) on December 7-8 in Beaverton near Portland, Oregon?
I'm not able to attend it (I'll be in vacation in Japan exactly at
that time), and hope someone
On Sun, 2006-11-12 at 22:02 +0700, Patrick Shirkey wrote:
They might pay Stephane Letz some money to port JACK though ;-)
Stephane has already ported JACK to windows.
On Wed, 2006-11-08 at 23:19 +0100, Elthariel wrote:
Thk you for testing it.
You build problem looks quite strange,
is there any error message about other EV_XXX identifier missing.
EV_SW is a macro defined in linux/input.h
can you send me your version of the kernel and your
On Mon, 2006-11-06 at 16:38 -0500, Paul Winkler wrote:
On Mon, Nov 06, 2006 at 01:39:43PM -0500, Lee Revell wrote:
Real linux drivers reside in the mainline kernel. Out of tree stuff is
is irrelevant.
I don't think that's a fair blanket statement given that drivers often
begin life
On Thu, 2006-11-02 at 20:58 +0100, Jens M Andreasen wrote:
On Thu, 2006-11-02 at 10:33 +0100, lemmel wrote:
Lots of people have been wondering, but this is the meat I think:
and even randomly truncated,
The difference in behaviour /might/ arise from differences in philosophy
of
On Wed, 2006-11-01 at 22:32 +0100, lemmel wrote:
Well, all the files, that I will play, will have the same charactistics, do I
really need to bother with an hi-level API [2] ?
[1] I noticed that a lot of applications still use OSS, and I thought it was
because the migration to ALSA was
On Mon, 2006-10-30 at 18:52 +0100, Fons Adriaensen wrote:
- hardware presents itself as 2 * 96 kHz
- user wants to see a device with 4 * 48 kHz.
interestingly, ADAT devices do the opposite to get to SR's above 48kHZ:
- hardware runs as N * 48 kHz channels
- data is
On Fri, 2006-10-20 at 22:44 +0200, Tim Goetze wrote:
[Fons Adriaensen]
Input the vertical video sync signal via the audio card and analyse
its timing in terms of audio samples (e.g. using a DLL). This will
enable you to predict where the next sync will be in the audio input.
Back in the
On Tue, 2006-10-17 at 11:56 +0200, Fons Adriaensen wrote:
'THE SAMPLES ARE NOT THE SIGNAL'. The real peak level of a
signal when converted to the analog domain can be several
dB above that of the highest sample.
indeed. there are people who are coming to believe that this error is
responsible
On Wed, 2006-09-06 at 22:11 +0100, John Rigg wrote:
Hmm. The manufacturer's web page describes it as a 32 channel interface
but I could only count 16 :(
On Wed, 2006-09-06 at 22:11 +0100, John Rigg wrote:
Hmm. The manufacturer's web page describes it as a 32 channel interface
but I could only count 16 :(
many companies count input + output separately. RME, for example.
several years ago, i wrote an system to control an ICube MIDI Sensor
interface, described here:
http://equalarea.com/paul/icube
unfortunately, the actual software has gone missing, even google cannot
find it.
if anyone has a copy of the software, i'd like to get a copy of it.
On Fri, 2006-08-25 at 17:02 +, carmen wrote:
i guess everyone has to pay the rent somehow...but do a indeed/simplyhired
search for linux audio, or similar. and check out the names of the top 10
entriesSony, Avid, Qualcomm. id rather work at starbucks than give them
more intellectual
On Sat, 2006-08-19 at 05:16 -0700, Stephen Cameron wrote:
On Fri, 2006-08-18 at 20:39 -0400, Stephen Sinclair wrote:
Audio doesn't use setitimer()-driven sleeping. It's interrupt-driven,
not timer-driven.
Yes, the driver is interrupt driven, but the driver interrupt handler
is only
The db inside jack are dbFS with a maximum possible signal of 0 db.
Now, both jacqeq and jackmix give you a maximum conrol level at +6dB.
It mean at +6dB in those EQ is equal to 0dbFS in jack.
there are no dB units inside of JACK. some JACK applications use dBFS,
that much is true. however, it
Someone called illiac wrote:
Linux as it is ordinarily distributed is not a small-footprint real-
time operating system. You will notice that your cell phone does not
run Linux. There is a reason for that.
i am sure nokia will be interested in your reason, since they don't seem
to have
On Fri, 2006-07-14 at 13:27 +0200, conrad berhörster wrote:
Hello maarten and dmitry and the rest,
thanks for the quick answers.
Faster means, that the workerthread is called more often than the jackthread
no, the workerthread should *not* be called more often than the
jackthread. either
On Thu, 2006-07-13 at 21:11 +0200, Dirk Jagdmann wrote:
Libsndsfile is plain C, but will do what you want without any fuss.
You could write a WAV specific C++ wrapper on top of this in a few minutes.
libsndfile is superb, but sometimes you don't want to link against
external libraries in
On Fri, 2006-07-14 at 00:56 +0400, Dmitry Baikov wrote:
If you need to stream a file, mmap'ed variant will eat memory up to
file size. Given large enough file, it will eat all you memory and
then will begin to page out unused portions of the file.
Of course, details on when and where will vary
On Fri, 2006-07-14 at 06:48 +1000, Erik de Castro Lopo wrote:
but I am not a fan nor a great user of C++. The wrapper should
really be written by someone with a love for the language.
LOL! that's pretty great. not a fan translates in real world terms
into one of LAD's most persistent critics of
On Tue, 2006-07-11 at 17:06 -0400, Dave Robillard wrote:
Semaphores seem about perfect for this to me.. am I missing something?
Why doesn't anyone ever recommend them?
i think mostly because in 2000-2001, they were very slow.
On Sat, 2006-07-08 at 13:34 +0100, James Courtier-Dutton wrote:
Hi,
Is there a standard way of converting a 24bit sample to 16bit?
I ask because I think that in different scenarios, one would want a
different result.
1) scale a 24bit value to a 16bit by simple multiplication by a fraction.
On Tue, 2006-07-04 at 07:48 +0200, Andreas Kuckartz wrote:
I have a simple question:
Which companies are (or have been) distributing LinuxSampler as part of
a package also including hardware and/or proprietary software?
as noted liontracs do, and that means that is incumbent upon me to
On Mon, 2006-07-03 at 02:26 +0700, Patrick Shirkey wrote:
If they really want to get people to give money then they should just
make it so that you have to pay or contribute code/time for a while to
get access to the newest downloads from their site. Keep the stable
version far enough
On Tue, 2006-07-04 at 10:22 +1000, Ryan Heise wrote:
On Mon, Jul 03, 2006 at 05:55:11PM -0400, Paul Davis wrote:
both guesses are wrong. i think it will be precise enough to say that a
company expressed what appeared to be a serious interest in leveraging
the existence of LS for its own
On Thu, 2006-06-29 at 18:35 -0400, Forest Bond wrote:
I've been looking at fst, and was going to package it for Ubuntu.
you cannot legally package FST. please do not do this. its not likely
that steinberg will come after you, and neither torben nor I are likely
to either, but its a violation of
On Sat, 2006-06-24 at 23:26 +0100, [EMAIL PROTECTED] wrote:
Hi peeps.
I've just been running an app through valgrind and I'm getting a few
of these:
==11955== Syscall param write(buf) points to uninitialised byte(s)
==11955==at 0x4D51BDB: (within /lib64/libpthread-2.4.so)
==11955==
On Tue, 2006-06-20 at 15:26 +0100, Steve Harris wrote:
On Tue, Jun 20, 2006 at 09:39:30 -0400, Dave Robillard wrote:
I can make the plugin validating host check the latency primitively (eg
run a single sample through the buffer) and fail if it isn't reported
correctly, so we're sure the
svn should once again have working (better than before the libsndfile
changes) support for tape tracks. what a total pain this has been, but i
think the end result is worth it - a standard library shared with other
apps. let me know if you have issues with it. it may not work with
existing
On Tue, 2006-06-20 at 00:57 +0200, Fons Adriaensen wrote:
It's not beyond the realms of the possible to describe the mathematical
relationship between the octave pitch unit and Hz, but it's probably
excessive.
A well-designed set of tags like the ones you show above would
probably solve
On Thu, 2006-06-15 at 16:32 +0900, David Cournapeau wrote:
I am in no way as experienced as most people on this list for audio
programming, but I don't see why C/C++ should be the only way to write
software to handle audio stream, neither do I see why GC would be the
only useful feature. For
On Thu, 2006-06-15 at 10:49 -0700, Michael Ost wrote:
As we looked over the Jack docs, it seems like a natural for supporting
this kind of architecture. We would break out our VST support into a
separate app and connect them to our Host app via Jack. This seems to be
how FST is implemented and
On Wed, 2006-06-07 at 11:12 +0300, Jussi Laako wrote:
Paul Davis wrote:
writing to a pipe is not 100% RT safe, but if the pipe is created in a
shm filesystem, its as close to it as you will get without ...
Nowadays, there's also available a very good interface from POSIX RT
extensions
On Mon, 2006-06-05 at 01:21 +0200, Stefan Westerfeld wrote:
Hi!
I am trying to notify a high priority (nice -20 or nice -19) thread from
a realtime thread (from a jack callback to be precise). Of course I want
the realtime thread to not block, but I want the high priority thread to
react
On Tue, 2006-05-30 at 09:23 +0200, Clemens Ladisch wrote:
If you want to avoid the syscall overhead, you can try to call mmap() on
/dev/hpet and read the timer directly. In that case, you don't need to
use any header.
and as a reminder, there is code within JACK now to do this, if you want
an
On Thu, 2006-05-25 at 02:44 +0200, Jens M Andreasen wrote:
On Thu, 2006-05-25 at 02:24 +0200, Jens M Andreasen wrote:
I am looking for a cross-platform implementation of an atomic
integer.
sizeof(int) is your friend
int speedy; as well, since int is defined to be the fastest
On Tue, 2006-05-23 at 13:54 +0200, Alexandre DENIS wrote:
3. dlopen libjack from within the program and continue if it fails
I haven't tried that; is it a reasonable option?
It would require to dlsym() every libjack symbol. A better option seems
to be:
i don't believe that is true.
On Sat, 2006-05-20 at 02:31 -0500, Gene Heskett wrote:
I should add another program which is crippled by this same problem Lee,
is grip. It can rip a cd, but cannot play the cd even when not
ripping it. And the ripped sound isn't up to the usual quality, often
sounding as if the
On Sat, 2006-05-20 at 09:27 -0500, Gene Heskett wrote:
Paul Davis wrote:
On Sat, 2006-05-20 at 02:31 -0500, Gene Heskett wrote:
I should add another program which is crippled by this same problem Lee,
is grip. It can rip a cd, but cannot play the cd even when not
ripping
On Fri, 2006-05-19 at 15:27 +0100, Chris Cannam wrote:
Announcing Sonic Visualiser, an application for viewing and analysing
the contents of music audio files.
http://www.sonicvisualiser.org/
massive props!
err, does that make me sound a like 40-something father of 3 trying to
be cool?
On Tue, 2006-05-02 at 17:57 +0200, Alfons Adriaensen wrote:
I can't imagine any sane interface standard for audio controls without a
way to say that the natural way to represent a port's range is exponential.
saying that the port range is exponential doesn't pin it down very much.
it still
On Wed, 2006-04-26 at 11:51 +0100, Steve Harris wrote:
I've written a first cut at an ontology/schema for the plugin RDF:
http://plugin.org.uk/ladspa2/ladspa-2.ttl
The term schema is a bit misleading, as it doesn't really enforce
anything, it really just gives you some hints about what to
On Wed, 2006-04-26 at 12:22 +0100, Steve Harris wrote:
Paul, while I've got your attention, are you OK with droppping runAdding?
You're the person who most likly to have implemented it I think.
99% of my plugins support it, but it would have made my life much easier
if they didn't.
On Tue, 2006-04-25 at 22:05 -0400, Phil Frost wrote:
You are not alone on this one. I think it's great to have as much data
as possible in a place that need not be dlopened to access. However, if
I have to learn to use some whizz-bang library to read yet another
markup language, spend an hour
On Fri, 2006-04-14 at 16:47 +0200, [EMAIL PROTECTED] wrote:
P == Paul Davis [EMAIL PROTECTED] writes:
P
P On Thu, 2006-04-13 at 17:45 -0700, Kjetil S. Matheussen wrote:
The biggest thing about this release of Snd-ls is probably that the
rt-player is enabled by default. The rt-player
On Thu, 2006-04-13 at 17:45 -0700, Kjetil S. Matheussen wrote:
The biggest thing about this release of Snd-ls is probably that the
rt-player is enabled by default. The rt-player is an alternative player
engine for SND that plays soundfiles using the rt-extension and reads data
from disk
On Wed, 2006-04-12 at 08:40 -0400, Dave Robillard wrote:
On Tue, 2006-04-11 at 12:10 -0700, Kjetil Svalastog Matheussen wrote:
Paul Davis:
As an interface designer, the first thing I look for on an engine's
project site is some sort of asynchronous API - I should never concern
myself
On Mon, 2006-04-10 at 12:26 -0400, Dave Robillard wrote:
If you want this in C++, you definitely want to use libsigc++.
(Keyword definitely)
actually, its worse than that. if you do this in C++ and you do not use
libsigc++, then a team of ninja open source developers will enter your
house
On Mon, 2006-04-10 at 21:02 +0200, Julien Claassen wrote:
In Apr 10 A.D. 2006 Paul Davis scripsit:
On Mon, 2006-04-10 at 12:26 -0400, Dave Robillard wrote:
If you want this in C++, you definitely want to use libsigc++.
(Keyword definitely)
actually, its worse than that. if you
On Mon, 2006-04-10 at 14:58 -0800, Patrick Stinson wrote:
Paul, Everyone,
What kind of assumptions can we make about the performance of the
various FIFO pipe/socket/whatever mechanisms on a POSIX machine? This
is by no means my area of expertise, but I currently hold the
assumption that a
On Fri, 2006-04-07 at 14:23 -0800, Patrick Stinson wrote:
As an added note to my previous comments, I really like the app
interface that mpd uses. Writing ascii events using some spec or
another to a file descriptor (socket in mpd's case) seems to be a
terrific way to communicated with apps
On Thu, 2006-04-06 at 11:48 -0400, Lee Revell wrote:
On Thu, 2006-04-06 at 04:14 -0800, Patrick Stinson wrote:
I've been looking for a high-performance music engine. It must have an
asynchronous control (socket, pipe?) mechanism to seperate the
application from the audio thread.
I'm
On Mon, 2006-04-03 at 22:58 -0400, Lee Revell wrote:
On Mon, 2006-04-03 at 21:33 -0500, Richard Smith wrote:
The following post came across the the Open graphics list from one of
the founders of the project looking for other things to do with the
technology they have developed.
Heh... I
On Mon, 2006-03-27 at 01:08 +0200, Florian Schmidt wrote:
On Mon, 27 Mar 2006 00:49:04 +0200
Florian Schmidt [EMAIL PROTECTED] wrote:
http://tapas.affenbande.org/?page_id=42
Oh and i forgot to ask a question:
What is the canonical way for a gtk app to receive hotkey press events
On Wed, 2006-03-15 at 10:11 -0800, muzak24h wrote:
The DSP can be from port 0x210 to 0x260. (there's code to find it)
The IRQ number can be found too. (there's code there to find it too)
controlling a DSP requires writing to registers, not just memory-mapped
ports. it also typically requires an
On Mon, 2006-03-13 at 00:34 -0500, Lee Revell wrote:
On Mon, 2006-03-13 at 00:23 -0500, Paul wrote:
Are these the threads you are refering to?
2.6.X, NPTL, SCHED_FIFO and JACK :
http://lkml.org/lkml/2004/6/30/104
I was looking for the thread(s) referenced in the above message:
On Mon, 2006-03-13 at 16:45 -0500, Lee Revell wrote:
On Mon, 2006-03-13 at 22:21 +0100, [EMAIL PROTECTED] wrote:
On Sun, Mar 12, 2006 at 03:50:26PM -0500, Lee Revell wrote:
On Sun, 2006-03-12 at 14:06 +0100, [EMAIL PROTECTED] wrote:
netjack-0.9rc1 is here...
endianess issues fixed.
On Mon, 2006-03-13 at 23:10 +0100, fons adriaensen wrote:
On Mon, Mar 13, 2006 at 10:21:39PM +0100, [EMAIL PROTECTED] wrote:
On Sun, Mar 12, 2006 at 03:50:26PM -0500, Lee Revell wrote:
Why do you use big-endian on the wire, requiring a double swap for x86
- x86? Wouldn't LE make more
On Tue, 2006-03-14 at 00:21 +0100, fons adriaensen wrote:
On Mon, Mar 13, 2006 at 11:59:15PM +0100, stefan kersten wrote:
as paul stated, network byte order is defined to be
big-endian, so yes, you have to convert 32 bit floats (and
doubles, for that matter) on intel, because they are
On Thu, 2006-03-09 at 16:28 +0100, Tobias Scharnberg wrote:
Hello List,
I'm new to audio programming and have problems finding the right
solutions for the development I work on right now:
I need to use the left and the right audio channel seperately. The
device is an ARM based board with
On Thu, 2006-03-09 at 17:01 +0100, Tobias Scharnberg wrote:
What do you mean - there are many ways when using ALSA or are there
also ways when using OSS. I know - this is RTFM worthy - but I have no
time for that, so a pointer would be very helpful!
if you use OSS the kernel will print a
no, it would provide names like
MOTU 828 mkII channel 1+2
RME HDSP (#1)
Builtin Audio
to the user.
it would also fix a myriad of other problems in ALSA, such as its
reliance on interrupts that occur at regular sample-based intervals,
Can you suggest
On Thu, 2006-03-02 at 00:09 +0100, David Kastrup wrote:
Lee Revell [EMAIL PROTECTED] writes:
On Wed, 2006-03-01 at 22:32 +0100, David Kastrup wrote:
What happens now if I do
aplay -D spdif something.wav
? Most certainly not the soundcard with the S/PDIF output gets used.
Instead some
On Wed, 2006-03-01 at 17:28 -0500, Lee Revell wrote:
On Wed, 2006-03-01 at 17:18 -0500, Paul Davis wrote:
now that's what i call a sick joke.
if only i had more time, i'd be writing CoreAudio for linux right this
very second.
Which would magically make 5 zillion different sound
On Tue, 2006-02-28 at 21:49 +, Julian Storer wrote:
Hi folks
A while ago there was some talk on the newsgroup about my Juce library,
and people were asking if/when I'd add support for audio under Linux..
well it's taken me a while to get round to it, but I finally battled
through the
On Mon, 2006-02-27 at 19:24 +0100, Christoph Eckert wrote:
* There are some old towns nearby worth a visit like Worms, Speyer,
Freiburg and Strasbourg. The last one really is worth a visit, and you
have even been to France/Alsace :)
Ahem. Hiedelberg? The jewel of the Neckar ?
Also, the
On Mon, 2006-02-20 at 14:55 -0500, Pete Bessman wrote:
So let's hear it!
WHAT is your NAME?
WHAT is your QUEST?
i can hack ardour part time while i work. i can hack ardour full time
while i live the rest of my life. but i know for sure that i definitely
can't get into philosophical debates
On Sat, 2006-02-11 at 02:29 +0100, [EMAIL PROTECTED] wrote:
Does anyone know a good way to write code that renders synth
knobs/potis/controllers?
I was looking around to rotate an image which only worked in opengl...
Creating a circle took too much time in sdl.
the gimp has/had an animation
On Mon, 2006-01-30 at 11:13 +, cdr wrote:
i know design by committee can be horrible but these situations usually
utilize vastly similar yet incompatible formats, so its sort of biting off
something small, i hope.. :)
(1) Peak Files
some of my favorite wav files have 10 metafiles
On Mon, 2006-01-30 at 11:29 +0100, Benno Senoner wrote:
Hi LADers,
During the last months the LAD website (
http://www.linuxdj.com/audio/lad ) was hosted on the lionstracs.com server.
Domenico from Lionstracs told me that he does not want do host the LAD
site anymore since it consumes
Screenshot:
http://www.kvraudio.com/forum/viewtopic.php?
t=114488postdays=0postorder=aschighlight=linuxstart=105
i will be interested to see just how good its audio handling
capabilities are.
--p
On Fri, 2006-01-27 at 14:13 +, Chris Cannam wrote:
On Friday 27 Jan 2006 14:03, Michael Bohle wrote:
I don't like dssi-vst so much, too buggy
Your bug reports and fixes have been invaluable.
It dosn't work with fst 1.7 because of a graphical prob.
I notice jacklab.net offers a
fer cryin out loud, will you guys *look* at the screenshot!
it already has JACK working!
Hmm, must have something on my eyes, see only screenshots with dssi-vst
wrapping eXT. OK, dssi-vst working as a Jackclient... but where for hell
is the screenshot of a native build of a jackified eXT?
On Fri, 2006-01-27 at 20:43 +0100, Richard Spindler wrote:
Writing against the native ALSA API is a pain, because it requires much
more low level stuff, that I don't really care for. And on top of that it'd
also
require me to setup my own thread for the callback mechanism.
i thought this too
On Fri, 2006-01-27 at 18:21 -0500, Lee Revell wrote:
On Fri, 2006-01-27 at 23:04 +, Chris Cannam wrote:
Lee Revell:
Won't help if the code is to be part of a GPL'd
application.
The Linux kernel is a GPL'ed application yet Nvidia
gets away with linking into it.
1 - 100 of 1290 matches
Mail list logo
