[linux-audio-dev] running x programs from the command line
On Tue, 20 Nov 2001, Joe Pfeiffer wrote:
so, DISPLAY=0:0 programname options will start any x program (should be
useful in init scripts, as i'd like to have pd to be up-and-running on a
(re)boot when in a performance mode).
What that's doing is setting an environment variable to tell realplay
that there is an X server, becasue apparently it will refuse if you
try to run it when X isn't running. But then it apparently is able to
get by without communicating with the server, otherwise it would
crash. This approach will not work for a program that actually
requires its GUI, because it would indeed crash when it was unable to
communicate with it. And I'd expect a program that could run without
its GUI to be able to see it didn't have an environment variable set
for the server, and to proceed on that basis. Hence my comment. I'm
not quite sure what the framebuffer has to do with it...
i'm not sure of the rules on this with x, but i did some testing, and i'm
keeping my position. DISPLAY=0:0 programname options only works for me
*if* x is running. what this does is tells the program not to try to run
from the current display (which is not x), but that x is located at 0:0
and to run there. the x framebuffer is for (among other things) when you
don't have an x-capable display because of really old video hardware,
unsupported video hardware, or no video hardware. for example, i can run
realplay from the normal command line without x running with the
following:
(Xvfb -pixdepths 15 DISPLAY=0:0 /usr/lib/RealPlayer8/realplay wava.ram)
i can also get pd, gimp, and netscape to run the same way (although i see
no point in running netscape that way). and i can do this on a computer
that has no video card. this is good for any program that can be
controlled (at least enough) with the command-line, but pointless for any
x app that cannot do anything useful on startup (as this is the only
control easily available).
-dave
--
perl -e'@email=split(//,.tenmhd\@nosbud);foreach$letter(@email){$
email=$letter.$email;}$email=~s/(m|net\.)/a\1\1/g;print$email\n;'
Re: [linux-audio-dev] [OT] job postings on LAD ?
On Tue, 20 Nov 2001, Dave Phillips wrote: Greetings: Very rarely I receive mail from companies looking for Linux audio programmers. Normally I just contact one of you directly, but I wondered if I should post them here. Do we have a job-postings policy here ? not that i know. If not, should we ? definitely. would be a sorry omen for LAD if getting paid for a change was off-topic. :-D
Re: [linux-audio-dev] embedded sound api.
Hi, [EMAIL PROTECTED] wrote: (This isn't really on-topic for the audio-dev list, but maybe other people are doing embedded audio apps or appliance-like Linux systems for audio, so it's at least somewhat relevant...) (ditto for this mail..) [..] Then check out Busybox, which is incredibly useful for putting together a small embedded system. It is quite easy to put together a Linux system which boots and runs from 4 MB of flash memory and runs a useful application. With 8 MB you can include a full version of the Gnu C library, and with 16 MB you can have X Windows - well, a stripped down version anyway. An alternative could be to try MicroWindows (http://www.microwindows.org) whose authors state its basic memory footprint is around 100 KByte. I admit I never tried it (just read some short articles about it), and it certainly has the disadvantage of not being X but a different system that requires apps to be adapted to it. Anyway, some might find it useful for their projects.. Frank
RE: [linux-audio-dev] embedded sound api.
The idea is to have a dual boot system where the default boot is into the embedded version and the secondary boot is to a full working version of linux. For those who are interested what we are doing. We are going to supply a fully functional portable digital recorder. It will have a multiple input peamp with gain and panning, a large battery and a notebook HDD. I have sourced a very tidy little Motherboard which will run 2 sdram sticks and upto a 1 ghz cerelon. I am going tomorrow to meet with cns sytems who make a refrigerated cooling unit to replace the fans and hopefully will be able to use that technology in the design. The only draw back is that it will only have 2 tracks.ie stereo. The sound card is an onboard cmipci. The target audience will be the movie and film industry but I'm guessing that a lot of other people will find uses for it once we start shipping. So, I'll check out busybox as that seems the most likely route to take. Obviously this is more than powerful enough to do a stereo recording but we don't intend to stop there. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com - For the discerning hardware connoisseur. http://www.boosthardware.com/LAU/Linux_Audio_Users_Guide/ ===
Re: [linux-audio-dev] new gdam (ladspa skins and presets)
On Tue, Nov 20, 2001 at 11:28:34AM -0800, Lance Blisters wrote: * custom ladspa skins can override default autogenerated skins, even a filter as volumous as 'hermes' is rendered silky and manageable. (see screenshot) Impressive! Now I will have to make it work properly ;) - Steve
Re: [linux-audio-dev] latencytest results webpage
[snip] TestVersion Latencytest version TestFrags # of audio fragments during test TestFragSize Size of each audio fragment TestFileSize Size of file for disk tests TestList List of tests performed (example: X, proc, write, read, copy) my 2 p. i wonder: we hab some different latency test programs, latencytest from Benno and the latency from alsa, and maybe others? - maybe we should categorize them and/or choose only one which will do the job. If there is a bug in the latencytest program you discover after some weeks, you'd have lots of unusable graphs/data. - Also some people did additional stressing by doing a find . . - Either a new option for that or say please exclude other stresses during test. [snip] XWinVersion X windows version - are framebuffer still in discussion, i.e. do not switch fb consoles during test? - maybe a flag, if you did it under X, fb console, or old text ? VideoDriver Video driver string (NVIDIA-1.0-1541, X, etc) [snip] - what about laptops. Because people explicitely asked for it, maybe we should have a different category for them? My preferred web development platform is probably PHP/MySQL. I probably have some time to work on something like this so lets start discussing details :) Would be nice to have a list to discuss this project, should personal since I still shiver when people use PHP, MySQL, Wiki, you-name-it , because I've no clue what's that all about, I'd like to know what you did, when you did it :) /personal Josh Green Smurf Sound Font Editor (http://smurf.sourceforge.net) Hope you had a nice time around the world after the LinuxTag, Josh. See you, Tobias.
Re: [linux-audio-dev] embedded sound api.
On Wed, Nov 21, 2001 at 10:10:51PM +0900, Patrick Shirkey wrote: [...] quality. Also I would want to incorporate a sound conditioner (is that the right word?). This would be able to take a sample of the standard input and then attempt to equalise or delete it from the track before writing to disk. In my opinion this will be one of the most important features because it will allow the user to target sounds they want to capture by getting rid of the unwanted peripheral noise. It will also be good for people who have cheap mics or happen to be recording in a high solar wind period or a lightening storm etc... ;-] [...] What I would like to hear from people who are interested in sharing their knowledge is what drawbacks they can see in the software design/setup and ideas on how to fully open the internal power with the [...] I know nothing about audio, but I have spent a lot of time processing images (in astronomy), and I suspect some of the lessons carry across. One of the hard lessons is that you don't get something for nothing - you can't get a good quality signal out of a bad one. So I'm a bit worried about the details of your sound conditioner. More precisely, you usually need to know the details of both noise and underlying signal very well before you can do anything clever with noise removal (very well could mean, for example, that you already knew the notes you are recording, and when they would occur, or that you separate record the noise of the lightning away from your main source *at the same time* - in parallel - as your main recording). So I suspect the best that you will be able to do in practice is low pass filtering (you could use the pre-sampled noise to adjust the filter parameters to some optimal values). I doubt this would sound much better than a simple electronic filter (something with a couple of knobs the user can twiddle to adjust performance by ear). Do any packages exist now that take noisy recordings and make them not-noisy? If that (low pass filtering) is what you're aiming at, or there's some clever processing in audio that I don't know about (perhaps because you can sample at a much higher frequency than the data your are interested in?) then fine. This is just a warning in case you were hoping for spectacular improvements in sound with no real idea how. Of course, I may simply be wrong - it's happened before. The rest of it sounds great (although I wonder if people are becoming so computer-literate that they are prepared to use PCs rather than specialised devices). Good luck, Andrew -- http://www.acooke.org
Re: [linux-audio-dev] embedded sound api.
how about a trackball and a miniature lcd monitor? then you'll be getting closer to some of the portable digital recorder/ mixer units built by the likes of roland and akai. otherwise, i don't see the point of having a glorified tape recorder with a 1ghz cpu?? best matthew For those who are interested what we are doing. We are going to supply a fully functional portable digital recorder. On the hardware side it will have a multiple input preamp with gain and panning, a large battery and a notebook HDD.
RE: [linux-audio-dev] suggestion for developers: real quad-channel output on an SBlive
-Original Message- From: Josh Whiting [mailto:[EMAIL PROTECTED]] i am a list-lurker, primarily a windows multimedia user, but with an interest in audio software development and hence an interest in linux. anyway, i have and idea that has been provoked by my recent explorations into quad-channel audio output. Basically, the idea is to write a driver to interface with the SB-Live EAX bus that converts the SB live into a more-or-less 'true' four channel output device. How? 1 - setup two stereo virtual audio input ports, 2 - assign each channel a status in the EAX bus as a sound source with a specific spatial position so as to emulate the direct dispatch of each channel to its respective output in the dual-output SB live DACs. You may not even need two virtual ports, as you could probably assign the main output (sb live wave out) to the front stereo outputs and then just send the second (virtual) stereo stream to the rear outputs. and bam - the SB live is now a (low cost) four channel output device, overcoming the (idiotic, painful) limitations imposed by creative's engineering of the board only support a single stereo output even though the card has two of them... ? sb live provides 5.1 output under windows (you can set output under speakers in eax control center (headphones, two speakers, 4 speakers, 5.1 speakers), there's also a demo for that somewhere there). Not sure how it works under linux (I don't have linux installed on that computer yet). erik
Re: [linux-audio-dev] suggestion for developers: real quad-channel output on an SBlive
STEFFL, ERIK *Internet* (SBCSI) wrote: -Original Message- From: Josh Whiting [mailto:[EMAIL PROTECTED]] i am a list-lurker, primarily a windows multimedia user, but with an interest in audio software development and hence an interest in linux. anyway, i have and idea that has been provoked by my recent explorations into quad-channel audio output. Basically, the idea is to write a driver to interface with the SB-Live EAX bus that converts the SB live into a more-or-less 'true' four channel output device. How? 1 - setup two stereo virtual audio input ports, 2 - assign each channel a status in the EAX bus as a sound source with a specific spatial position so as to emulate the direct dispatch of each channel to its respective output in the dual-output SB live DACs. You may not even need two virtual ports, as you could probably assign the main output (sb live wave out) to the front stereo outputs and then just send the second (virtual) stereo stream to the rear outputs. and bam - the SB live is now a (low cost) four channel output device, overcoming the (idiotic, painful) limitations imposed by creative's engineering of the board only support a single stereo output even though the card has two of them... ? sb live provides 5.1 output under windows (you can set output under speakers in eax control center (headphones, two speakers, 4 speakers, 5.1 speakers), there's also a demo for that somewhere there). Not sure how it works under linux (I don't have linux installed on that computer yet). EAX is proprietary. It isn't available under Linux, and unlikely it ever will. The OpenAL code is probably the closest you'll see in the near future, but something of a similar idea could probably be used there (just not EAX). D. Stimits, [EMAIL PROTECTED] erik
Re: [linux-audio-dev] latest Snd screenshot
Dave Phillips wrote: Juhana Sadeharju wrote: It looks messy. Could you make it look simpler? Just like it would look if I make basic edits and effects to audiofiles. Sorry about that, I wanted an overkill display shot. I'll post some other shots. Anyway, what is the best/most usable audiowave editor in Linux at the moment? I like Ecawave. I think it's the best. Luis Pablo Gasparotto _ Do You Yahoo!? Get your free @yahoo.com address at http://mail.yahoo.com
Re: [linux-audio-dev] problems creating/managing pthread stuff (audio-app related)
[ pthread cry for help ... ]
1) buy a good book on programming with threads. i happen to like
Programming with Threads by Kleiman, Shah Smaalders, but there
are others. You'll need it.
2) general method for using C++ objects when starting threads:
class Foo {
public:
static void *function_called_by_thread (void *arg) {
((Foo *) arg)-actual_function_for_thread_to_run ();
}
private:
void *actual_function_for_thread_to_run ();
};
pthread_t thread_id;
Foo *myfoo;
if (pthread_create (thread_id, 0,
Foo::function_called_by_thread, myfoo) != 0) {
... error: cannot create thread ...
}
At this point, there is a new thread executing
Foo::function_called_by_thread(), which in turn will call
Foo::actual_function_for_thread_to_run().
Hope this helps.
--p
RE: [linux-audio-dev] suggestion for developers: real quad-channel output on an SBlive
-Original Message- From: D. Stimits [mailto:[EMAIL PROTECTED]] Sent: Wednesday, November 21, 2001 11:17 AM To: [EMAIL PROTECTED] Subject: Re: [linux-audio-dev] suggestion for developers: real quad-channel output on an SBlive STEFFL, ERIK *Internet* (SBCSI) wrote: -Original Message- From: Josh Whiting [mailto:[EMAIL PROTECTED]] i am a list-lurker, primarily a windows multimedia user, but with an interest in audio software development and hence an interest in linux. anyway, i have and idea that has been provoked by my recent explorations into quad-channel audio output. Basically, the idea is to write a driver to interface with the SB-Live EAX bus that converts the SB live into a more-or-less 'true' four channel output device. How? 1 - setup two stereo virtual audio input ports, 2 - assign each channel a status in the EAX bus as a sound source with a specific spatial position so as to emulate the direct dispatch of each channel to its respective output in the dual-output SB live DACs. You may not even need two virtual ports, as you could probably assign the main output (sb live wave out) to the front stereo outputs and then just send the second (virtual) stereo stream to the rear outputs. and bam - the SB live is now a (low cost) four channel output device, overcoming the (idiotic, painful) limitations imposed by creative's engineering of the board only support a single stereo output even though the card has two of them... ? sb live provides 5.1 output under windows (you can set output under speakers in eax control center (headphones, two speakers, 4 speakers, 5.1 speakers), there's also a demo for that somewhere there). Not sure how it works under linux (I don't have linux installed on that computer yet). EAX is proprietary. It isn't available under Linux, and unlikely it ever will. The OpenAL code is probably the closest you'll see in the near future, but something of a similar idea could probably be used there (just not EAX). aha. but if eax can do it then there has to be hw that supports it, right? otherwise eax would be able to output 5 separate channels (and it does, even though I don't know how well separated they are). the original post was talking about more-or-less true 4 channel and I was saying that 5 channel output is possible (using sb live HW, there might be software issues involved that are not easy to solve). or are you saying that using information available to opensource developers it is not possible to use the same HW that eax uses? erik
Re: [linux-audio-dev] jack question
I am attempting to write a piece of code that does output via jackd. I'm playing a bit with the jack_simple_client program. From what I understand this client connects ALSA I/O:Input 1 to ALSA I/O:Output 1, not quite. it connects *its* input port to ALSA I/O:Input 1 and *its* output port to ALSA I/O:Output 1. the effect is the same, but the concept is a little different. you can't actually do what you describe (only the ALSA client/driver could do that). i.e. recording from a source and immediately outputting it again, right? However, I get no output when running jack_simple_client, even though the recording source is set and generating data. Can anyone confirm that the current jackd actually outputs audio? yep. myself as well as several other people have run it and heard it as well. are you sure you have the capture input set properly in ALSA? do you have the mixer stuff properly set up for the PCM output stream? BTW, if jackd is run with real-time priority, shouldn't all the clients be running in real-time too? The process() call runs inside the client after all!? yes. if you call jackd with the -R (realtime) flag, then all clients must all necessarily be realtime. if they are not, they will not be able to connect to the server, since they stand an (increased) chance of messing up the timing of the overall JACK system. this, like many other things needs to be made more clear. --p
Re: [linux-audio-dev] Looking for a specific developer
On Tue, 2001-11-20 at 15:20, Vincent Touquet wrote: snip And also because he wrote some great software to interface with the A3000 and also to access Yamaha specific disk images (proprietary format ...) and I think he wouldn't mind helping me and other people out, to develop some free software for doing these things. snip Just wanted to mention that I'm planning on supporting other MIDI sample based patch formats (in the distant future) for the Smurf Sound Font Editor. Of course I would probably be changing the name of the program then, as well :) I'm currently creating a libsoundfont library which will handle sound fonts, but will be attempting a libmidipatch (name not decided on yet) which will supersede this. It seems like many of these sample based patch formats have many things in common, it would be nice to abstract these into a library (and provide interfaces to each formats more unique parameters and such). I could really use some help on the project, so if anyone is interested.. Lates Thanks vini -- Josh Green Smurf Sound Font Editor (http://smurf.sourceforge.net)
Re: [linux-audio-dev] latencytest results webpage
On Wed, 2001-11-21 at 07:09, Tobias Ulbricht wrote:
snip
i wonder: we hab some different latency test programs, latencytest from
Benno and the latency from alsa, and maybe others?
- maybe we should categorize them and/or choose only one which will do the
job. If there is a bug in the latencytest program you discover after some
weeks, you'd have lots of unusable graphs/data.
- Also some people did additional stressing by doing a find . .
- Either a new option for that or say please exclude other stresses
during test.
I think we should create a specific test program that is used always
(based on latencytest). A version of the test suite could be supplied in
any data submitted to the database. If someone needs a test for some
other aspect of latency (nic smashing test, cdrom, etc) we could always
add it if its something commonly wanted by users.
I've created the database in MySQL for this kind of functionality.
Basically there is a table for tests and then another table for the
results for each stress test called results:
CREATE TABLE results (
ID INT UNSIGNED NOT NULL AUTO_INCREMENT,
TestID INT UNSIGNED NOT NULL,
HasData ENUM ('FALSE', 'TRUE'),
TestTypeENUM ('X', 'proc', 'diskread', 'diskwrite', 'diskcopy')
NOT NULL,
OverrunsINT UNSIGNED NOT NULL,
MaxLatency FLOAT NOT NULL,
CPULatency FLOAT NOT NULL,
FragLatency FLOAT NOT NULL,
Between2ms FLOAT NOT NULL,
Between1ms FLOAT NOT NULL,
Between2tms FLOAT NOT NULL,
Between1tms FLOAT NOT NULL,
INDEX(ID)
);
So tests could easily be added to the database simply by adding another
ENUM to the TestType field. Something that I should probably add to the
results table is fields for the parameters of the test. For instance
with the disk tests the file size in MB. So perhaps:
Param1INT INT,
Param2INT INT,
Param3INT INT,
Param4STR VARCHAR (255),
Would cover most tests and any we might add.
[snip]
XWinVersion X windows version
- are framebuffer still in discussion, i.e. do not switch fb consoles
during test?
- maybe a flag, if you did it under X, fb console, or old text ?
Yeah that might be nice to have. I guess if you didn't run it in X, it
simply wouldn't run the X test. It might be nice to have a latencytest
config script that asks questions for which tests should be run, etc.
This could then be saved in a file for further use.
VideoDriver Video driver string (NVIDIA-1.0-1541, X, etc)
[snip]
- what about laptops. Because people explicitely asked for it, maybe we
should have a different category for them?
Thats probably a good idea. Perhaps just a flag would be sufficient.
My preferred web development platform is probably PHP/MySQL. I probably
have some time to work on something like this so lets start discussing
details :) Would be nice to have a list to discuss this project, should
personal since I still shiver when people use PHP, MySQL, Wiki,
you-name-it , because I've no clue what's that all about, I'd like to know
what you did, when you did it :)
/personal
Sure. I'm just starting to set it up. Hopefully I can go through with
the project, but I have so many other projects too. It gets overwhelming
sometimes. I'm going to try to get something useful going. I'll be sure
to post to this list if and when I have something.
Josh Green
Smurf Sound Font Editor (http://smurf.sourceforge.net)
Hope you had a nice time around the world after the LinuxTag, Josh.
I did, it was an awesome trip through Europe. I hope to return again
someday :) Where are those LAD LinuxTag pictures that Frank Neumann
posted? Perhaps it was never mentioned on the LAD lists? I can't find
them in any archives at least.
See you,
Tobias.
--
Josh Green
Smurf Sound Font Editor (http://smurf.sourceforge.net)
Re: [linux-audio-dev] jack question
Sounds like the sound card would need to support full duplex too, no? correct. the only supplied driver for JACK, which is written around ALSA 0.9, requires full duplex from the audio interface its connected to. it also requires mmap access and the ability to use either S16_LE or S32_LE format samples. this is not a stipulation of JACK - as I've said many times before, the JACK server has no clue that there even *is* an audio interface. its quite possible to drive a JACK system from a video interface, an interval timer, or anything else that will regularly wake up the server with a frame count. if somebody wants to hack the ALSA driver for JACK so that it doesn't require full duplex, i'll happily apply patches. --p
