Hello,
I'm using PortAudio library to try to program a loop-sampler software
under Linux (and perhaps Windows). Actually, I would like to make a
console-based software. I would like to know how could I manage entries
from keyboards. I think that ncurses lib is a good choice. But my
problem is in algorithm term.
I think that making a code in the main() like this:
while (true) {
if (key is pressed) {
switch (value of the key) {
case 'A':
proceed to FFT ;
break;
case 'B':
proceed to time stretch
break;
case 'C:
reduce volume
break;
...
...
}
}
wait for a certain time;
}
is not a good idea. I know that with MFC there is event management with
WM messages. I don't know what is there under Linux, especially in
console-based apps. With GUI libs like GTK, no problems I think. But
what about console-based apps?
Does anybody knows some internet sites, or some other libraries, or some
methods (perhaps multithreads? seems to be heavy for just event
managements...). I have look at ncurses docs but they shows an example
with a kind of loop like the one I just talk about above. Any advice is
welcome
Bye.
hi erik,
i've got a question regarding type combinations in libsndfile:
SF_FORMAT_FLOAT | SF_ENDIAN_CPU works with SF_FORMAT_AU on any
platform, which is great. but it doesn't work with neither aiff nor
wav. i expected that it would work with one of the both formats,
depending on the host endianess.
is there a special reason why it does not?
some other questions regarding string data:
snd does not explicitly allow using string data in the file format.
however, i remember that it was done in the old NeXT days by
increasing the data offset and using the space in between for a
comment. do you think this could be implemented in libsndfile for the
SF_STR_COMMENT thing or is that just too non-standard?
with SF_FORMAT_AIFF, i can't write string data to the file in SFM_RDWR
mode. the call to sf_set_string() succeeds, but the actual data in
the file is not changed. the call to sf_set_string() succeeds even
when the file is in SFM_READ mode.
with SF_FORMAT_WAV, a file with string data can't be opened with
SFM_RDWR anymore. this means there is no way to change the audio data
or the comment of the file without copying it? i guess there is a
special reason for that...
thanks very much for your patience...
bests,
martin
Greetings:
While doing some research on VST/VSTi technology I checked the
Wikipedia page for "VST". It's a good informative page, and there's even
an entry regarding the fst/libfst project. However, there was no mention
of Kjetil's vstserver and its clients. There is now.
It occurred to me that while we're laboring over where & whether to
use wikis, there's already a place for us to get some more information
out to the public: the Wikipedia. I don't know how many other Linux
audio projects are detailed. Ardour, ALSA, Pd, and others are already
there, but I don't know how up-to-date that material is. So, would-be
documenter writers, avail thyselves of this opportunity and start
hitting the wikipedia ! Add new entries, correct the old ones, bring 'em
up to date, but just do it ! ;-)
Thank you for your attention. We now return you to your regularly
scheduled programming.
Best,
dp
Hi.
I recently wrote ALSA PCM and seq support for BRLTTY[1].
However, the fact that libasound.so.2 lives in /usr/lib does
present a little problem for us. BRLTTY needs to be run as
early as possible during startup, most likely before /usr is
even mounted. Linking against -lasound the traditional way therefore is quite
a bug, since the executable now doesn't start anymore if
/usr is missing.
We've solved this now by dlopen() and dlsym()'ing all the
symbols we actually need, and call our private function pointers instead
of the real library symbols. However, this solution is kind
of icky. At the same time, we wrote support for QNX, which
seems to have a early fork of ALSA as the QNX Sound Architecture (also
called libasound). QNX apparently already puts its
libasound.so in /lib, most probably because of this
problem.
I'd like to ask the community what you think about
moving libasound from /usr/lib/ to /lib/? With OSS, that
wasn't a problem at all, since the ioctl calls were simple
enough that there is/was no need for a wrapper lib. But
since ALSA kind of relies on that, wouldn't it make more
sense to have the lib in /lib so that early boot programs
can output sound too?
--
CYa,
Mario
Hi,
I've posted the following message in Rosegarden-devel mailing list, and
Michael pointed me to an old thread from you about the same subject:
http://www.mail-archive.com/linux-audio-dev@music.columbia.edu/msg03322
Attached is a little program summarizing what I have found about the file
format at this moment. I would like to know if you have more information,
ideas or want to collaborate in this task.
Regards,
Pedro
On Sunday 29 August 2004 01:50, Pedro Lopez-Cabanillas wrote:
> Hi,
>
> I have many old songs, stored as Cakewalk "WRK" files, and I don't use
> Cakewalk or Sonar anymore. I'm not planning to do so in the future, 'cause
> I'm sure I will use Rosegarden. Of course.
>
> So, I need to convert my old WRK files to Rosegarden. But the WRK file is a
> closed format, and not publicly documented. It will be necessary to reverse
> engineer it, to rescue my old loved melodies along with the pile of shit.
>
> My plan is to first write a 'cake2rg' standalone utility, to start learning
> the file format and meanwhile to produce some practical results. After
> that, I hope to be able to import the files directly into Rosegarden. This
> won't happen before 1.0-final.
>
> If somebody had the same idea and has the work almost finished, please let
> me know ;-). Let's avoid duplicated work.
>
> Comments, please.
>
> Regards,
> Pedro
It may be that this topic is already covered somewhere, but after much
searching over the past few days I have been unable to come up with a
comprehensive guide to what I am attempting to do.
(Note that I've never written something like this before but know enough
about Linux and how things work that I would probably understand a
higher-level discussion on the topic...maybe. [grins])
I am attempting to write a program that will capture and record digital
audio (in the form of PCM/AC-3 over S/PDIF). What I *do* with that data is
a secondary issue and not something I want to get into now.
The input to the "system" is a USB digital audio capture device. (I
currently have an Edirol UA-1D
(http://www.edirol.com/products/info/ua1d.html) and Creative Labs MP3+
(http://www.creative.com/products/product.asp?prodid=154) at my disposal.)
What I'm not sure of is what I am looking for (or what I should expect to
find). The S/PDIF stream that will be coming into the device may have PCM
or AC-3. My basic assumptions on this topic are that I will have to find
some /dev entry to open and read from to do the capturing itself, but from
everything I've read so far it doesn't look like the standard /dev/audio or
/dev/dsp devices would suffice.
So, there you have it. I'm looking for some pointers to get the ball
rolling and I'm positive I'll be able to pick it up from there...it's always
the first few steps that are daunting. The kind of questions I am asking
are:
1) With the USB audio device plugged in, what device would I need to open
to read the raw digital data?
2) Is there an API already available for reading this information? (I've
looked at OSS, but haven't gotten much into ALSA yet.)
3) Assuming I can get at *a* bitstream, what will it contain? Will it be
the S/PDIF stream from which I will need to extract PCM/AC-3, or will it be
the raw PCM/AC-3 itself?
Thanks.
Paul Braman
Paul dot Braman at NielsenMedia dot com
813-366-5053
Hi,
I'm pleased to announce a project which eventually should support
several Firewire-based breakout-boxes. Following boxes would be
supported:
- ESI QuataFire 610
- M-Audio Firewire Audiophile
- M-Audio Firewire 410
- M-Audio Firewire 1814
- Terratec Aureon 7.1 FireWire
- Edirol FA-101
Each of those boxes is using the same chip produced by the company I'm
working for. I got green lights from my boss to do write a driver for
linux. Of course, this is not a very simple task. Most information
which is needed for writing a driver is available as Standards. Well,
some information is not available as standards...
If someone is interested to participate in this undertaking I would be
very glad, because it is a large task and I have only little resource
left for this project. So it would take rather long to finish it and
I'm sure some of the LAD guys are eager to use something different
than an USB based breakout box.
Any serious hacker who signs a NDA gets access to all information and
will also have for access to hardware (test boards). If s/he is not
signing (for some reason) only the public available Standards most be
enough. Access to hardware can still be arranged (hopefully).
Anyway I have already started to code something but it is still at
very very very early stage and not good for anything. Therefore I
haven't loaded it up somewhere.
cheers,
daniel
1)Does anyone know what format digital audio is stored in for miniDVs?
I know the audio can be 12 or 16 bit and I know that (at least for the
tapes I have) SP is about 60 minutes and LP is about 90 minutes. So, I
figure the sample rates are probably something like 48kHz for SP and
32KHz for LP, but I haven't found a definitive reference, yet.
I ask because I've had a canon zr60 miniDV camcorder for about a year
and have recently started using it to collect sounds. For now I'm
recording from the analog output of the camcorder into my delta66.
Obviously, this is less than ideal, but far superior to the handheld
voice recorder analog cassette tape solution I've been using for the
past 8 years. :-]
2)I've briefly glanced over the web page for kino:
http://kino.schirmacher.de/article/static/2
It appears kino will handle recording the video from a ieee1394
connection and also includes a tool for stripping audio out of the video
to a wav file. Anyone here have any experience to share in using kino,
or any other package, to do something like this?
3)Does it matter what ieee1394 interface I get, or are they all
basically the same as long as there's kernel support for them?
Thanks in advance,
Eric Rz.
PS appologies if this has been discussed recently. I've been away on
vacation and haven't had time to follow the lists closely for 3-4 weeks.
-edrz
Hi Matt:
Thanks for the further clarification. I suppose that the simple fact
is that the Linuxadio community simply doesn't "number up" enough for
Coda to consider us any sort of market. Same old story. Yawn, heavy
sigh... But the truth is that I'm not really interested in Finale, I'm
interested in libre software, and my concern is for the continued
development of NE. It would be nice if someone would step up and take
the reins, and as I said earlier, that is my hope now for NE.
Would I buy Finale for Linux ? Probably, at least to see what the
hubbub is all about. Ditto for Cubase or really any other music app that
would run under my favorite OS, but I would want to make an equal or
larger contribution to the developers of functionally equivalent libre
software.
My two drachmas, of course...
Best,
dp
Polashek, Matthew wrote:
>Hi!
>I met with the president nd founder of Coda, (Now Makemusic!) a few months
>ago at work(Silver Burdett Ginn) and I asked him if they were going to
>publish a Linux version of Finale now that they have completed an OSX
>version. He said that his programming and marketing staff had brought the
>idea up but that they were not going to make a Linux version at this time.
>
>FYI.
>
>Matt.
>
>
Hi my name is Mark and I have been a silent member for two
years - having very little experience with the in's and out's of
Linux. For many years I have blindly believed that Linux is the
way to go - more because of the concept of Open Source which
ties in exactly with my own belief structure. I have supported
linux for years in the form of mail servers but my own knowing
has been lacking. Recently ( 2 months ago ) I reinstalled all my
systems without the comfort of partitioned disks and windows as
a safety net. To date I have re-installed my system close to ten
times. I am only now realizing that I may have a hdd problem or
hardware in general - although no errors are evident. I have a life
long ambition to make sound - to generate ecstacy - to move
energy - to drown people - to open portals of thought and
recreation to fly - to dream - to enlight. In reality I am stuck with
my finger on the reset button.
My setup - P4 - 256 ram - Slackware 10.0 - linux 2.6.7 -
My problem as an end user - is limitless - while the discussion
group handles issues like left or right up or down this or that it
seems as if you all live in Utopia with perfect systems. My
problems as a beginner are endless.
My linux thought is that the main difference between linux and
windows is that under windows everything is packaged i.e. that
one installation will actually get that desired product to work -
while under linux a lot more time is spent on finding the correct
patches mixes matches reading than end time using/producing.
Obviously linux has flexibility power stability way beyond the
confines of windows or mac but it is the tieing up of those 'loose
ends' which make it easier for a new user to get going that make
the ultimate difference. While windows users have licence issues
some linux users have other.
It matters little to me if one program does something in one way
as opposed to a different way in a different program - i.e. each
program is unique unto itself and should be learnt seperately. My
problem in this area has been to get them to work together or
through each other with results.
I return to my silent world of learning and eventually achieving in
linux but that's my two cents.
I believe in Linux and all that it stands for and wish no insult to
any. Thank you for this moment.
bye
Mark
Mark McBride
Cell 08 4414 6809
Tel.: +27 21 462 0044
Fax: +27 21 465 0277
Bitwise Computer systems
EMail : tech-support(a)bitwise.co.za
EMail : sales(a)bitwise.co.za
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