Hi,
There are discussions on kde-multimedia about
the future of Linux/Unix multimedia (especially sound).
This is one of the most interesting messages.
/RogerL
---------- Forwarded Message ----------
Subject: CSL Motivation
Date: Tuesday 25 February 2003 16:26
From: Tim Janik <timj(a)gtk.org>
To: KDE Multimedia <kde-multimedia(a)mail.kde.org>
hey all,
due to the recent discussions involving CSL (amongst other
projects) on this list, i've put up a preliminary web presence at:
http://sfk.evilplan.org/csl/
and, more importantly, stefan and i wrote a paper going into the details of
why we think a project like CSL is necessary and what we intend to achieve
with it:
http://sfk.evilplan.org/csl/csl-paper.ps
---
ciaoTJ
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-------------------------------------------------------
--
Roger Larsson
Skellefteå
Sweden
>> BTW, can JACK handle several HW interface using different blocksizes
>> at a time (assuming sample frequencies are coherent) ?
>No, but you could run several instances of JACK.
Acutally, I'm afraid that you can't do that currently.
Taybin
I hacked up a shell script [attached] to valgrind LADSPA .so files. It
will report leaks, bufferoverruns, reads from unitialised variables and
so on, but it doesn't totally exercise the plugin so it may miss some
things.
Obviously it helps if you rebuild with -g on and it works best with gcc3
(better inline debugging).
You can't just "valgrind applyplugin ..." becuase valgrind doesn't chase the
dlloaded objects and applyplugin is full of memory errors and leaks :)
Usage: ladspa-valgrind <plugin.so>+
I dont recommend running it on my current plugin set ;)
Requires gcc and valgrind 1.0+
- Steve
Hi all,
I've got a small problem with my Midisport 2x2 MIDI interface. While it worked
ok before, now for some odd reason I cannot get it to open for MIDI output
(from the computer). I've checked the /var/log/messages and this is what I
got:
Feb 24 20:44:41 mycomp kernel: hub.c: USB new device connect on bus2/1,
assigned device number 2
Feb 24 20:44:41 mycomp kernel: usb.c: USB device 2 (vend/prod 0x763/0x1001) is
not claimed by any active driver.
Feb 24 20:44:44 mycomp /etc/hotplug/usb.agent: Setup ezusbmidi for USB product
763/1001/1
Feb 24 20:44:44 mycomp /etc/hotplug/usb.agent: Module setup ezusbmidi for USB
product 763/1001/1
Feb 24 20:44:44 mycomp /etc/hotplug/usb/ezusbmidi: load
/usr/share/usb/ezusbmidi/ezusbmidi2x2.ihx for 763/1001/1 to
/proc/bus/usb/002/002
Feb 24 20:44:44 mycomp kernel: usb-uhci.c: interrupt, status 2, frame# 1645
Feb 24 20:44:44 mycomp kernel: usb.c: USB disconnect on device 2
Feb 24 20:44:44 mycomp kernel: hub.c: USB new device connect on bus2/1,
assigned device number 3
Feb 24 20:44:44 mycomp kernel: usb.c: USB device 3 (vend/prod 0x763/0x1110) is
not claimed by any active driver.
Feb 24 20:44:48 mycomp /etc/hotplug/usb.agent: Setup snd-usb-audio audio usb
usb-midi for USB product 763/1110/1
Feb 24 20:44:48 mycomp kernel: usb.c: registered new driver snd-usb-audio
Feb 24 20:44:48 mycomp kernel: snd-usb-midi: MIDIStreaming version 01.00
Feb 24 20:44:48 mycomp kernel: snd-usb-midi: EP 01: 2 jack(s)
Feb 24 20:44:48 mycomp kernel: snd-usb-midi: EP 81: 2 jack(s)
Feb 24 20:44:48 mycomp kernel: snd-usb-midi: created 0 output and 4 input
ports
Feb 24 20:44:48 mycomp kernel: usb-uhci.c: ENXIO c0008380, flags 0, urb
cdc59d60, burb cdc59ce0
Feb 24 20:44:48 mycomp kernel: snd-usb-midi: usb_submit_urb: -6
Feb 24 20:44:48 mycomp /etc/hotplug/usb.agent: ... blacklisted module: audio
Feb 24 20:44:48 mycomp kernel: usb.c: registered new driver midi
Feb 24 20:44:48 mycomp /etc/hotplug/usb.agent: ... blacklisted module:
usb-midi
It says that the device was opened with 4 input and 0 output ports. Why is
this the case? I have no problem getting the midi input going, but my output
is mysteriously dead.
I've read somewhere that if the device is reconnected that the output stops
working (might have been CCRMA's Fernando), but this is the log from the
initial connect as soon as the computer boots.
I've also made sure that no MIDI devices were connected to the MIDISPORT upon
starting up.
Any help on this issue is greatly appreciated! Sincerely,
Ico
> we don't care about non-power-of-two periods, i think. however, i do
> consider periods defined in units of times and not frames to be broken
> hardware design. it forces inefficiencies into the software that are
> totally unnecessary.
For what its worth, the SAOL standard has to deal with this problem
too, since the user gets to specify an integer k-rate and an
integer a-rate. Eric went for this solution:
5.8.5.2.2 krate parameter
<global parameter> -> krate <int>;
The krate global parameter specifies the control rate of the
orchestra. [...]
The krate parameter shall be an integer value between 1 and the
sampling rate inclusive, specifying the control rate in Hz. [...]
If the control rate as determined by the previous paragraph is not an
even divisor of the sampling rate, then the control rate is the next
larger integer that does evenly divide the sampling rate. The control
period of the orchestra is the number of samples, or amount of time
represented by these samples, in one control cycle.
---
This has been controversial, since it limits the ability to use
SAOL to emulate existing coding standards that have non-integer
relationships betweens frames and the sample rate. The win has
been decoder implementation simplicity.
-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
Hello,
I just released a new version of tapiir. Tapiir now supports jack.
Tapiir can be found at
http://www.iua/~mdeboer/projects/tapiir/
Tapiir is a simple and flexible audio effects processor, inspired on
the classical magnetic tape delay systems used since the early days
of electro-acoustic music composition. It provides a graphical user
interface consisting of six delay lines, or "taps", which can
introduce an almost arbitrarily big or small delay to their inputs
and can be feed back to each other.
A wide set of effects can be easily achieved by properly configuring
and connecting the delay lines: complex echo patterns, resonances,
filtering, etc. Delays, interconnections and gains can all be
controlled in real time.
Maarten
reposting to the list, my email client had some hick-ups..
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
> if you mean at JACK's level, then it prints out information about
> xruns. i think you mean something else, but i'm not sure what.
OK, but jackd is accessing a DMA buffer - where would that be located for USB devices which doesn't do any real DMA? (OK, maybe that's something for ALSA to figure out)
The debugging I was thinking about would be something more fine-grained than just xruns (as I'm not getting any xruns (well, if I choose a period size of 64 I'm getting lots of them).
I'm not really sure what to monitor either, just thinking out loud here - it might be interesting to find out where the actual playback and recording pointers are pointing when accessing the buffer to find out whether something is out of sync (it kind of sounds like that, that the beginning of the buffer gets overwritten with new data, or that the end of the buffer somehow isn't completely written when jackd grabs it). Not sure whether jackd can actually see that though (?). The two parts of the buffer (when using two periods) is laid out consecutively in memory, right? One possible explanation might be that jackd gets called a bit to early/late when the audio device/driver is switching buffers, but that would be kind of hard to actually see for jackd as a client to ALSA?
(I recognize the audio artifacts from the good old days of DOS demo coding of SoundBlaster DMA, at one point where I had screwed up the logic, writing in the "live" buffer, then the beginning of every period was anything but clean...)
This isn't CPU related, there's plenty of CPU time left, and it's occuring all the time, just curious where to start debugging (it seems like it's a jackd problem because aplay works fine, but then again, I guess that aplay doesn't mmap the buffer)..
/WernerJ
Hi LADers!
I released a realtime Music-visualization-program (my master's thesis)
on http://sonasound.sf.net.
It's probably more of academic use at the moment. Licensed under GPL.
Maybe there is some interest.
Features are:
- Waveform-Display
- short-time-spectrum
- sonogram
- spectrum-generation switchable between lpc and fft
- Major kinds of windowing functions choosable
- Different kinds of windowing functions are directly comparable via
live-switching
- Overlap of 50% can be switched on and off without stopping
- Buffer-Size and FFT-Size freely adjustable
- All Displays work from soundfile while playing or direct live-input
from soundcard
- Comparison of different spectrum-generation-methods via multiple
instances fo SonaSound possible
Libraries needed:
- fftw (www.fftw.org)
- glib >= 1.2
- gtk >= 1.2
- gtkglarea
- GL
- GLU
- portaudio (www.portaudio.com) (Yes, I know, switching to JACK might help)
- sndfile >= 1.0
There are many features I would like to add and a few
not-so-good-features I'd like to clean up.
Have fun*
Niklas
Hi.
I relased ZynAddSubFX 1.0.9 (a very poerfull software
synthesizer for Linux).
News:
- added keylimit to Part
- you can use multiple filter stges in order to make
very steep filter rolloffs (eg. 48 dB/octave)
- ADsynth - added noise mode and you can make fixed
frequencies; added the "Punch" parameter
- added an external program "Controller" which enables
you to use the mouse for MIDI controllers
- other improvements and bugfixes
See more at http://zynaddsubfx.sourceforge.net
Paul.
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Here's another take on that paste-it-together-out-of-C-code "blockless
processing" idea that was discussed a few months back.
Its incomplete and experimental, and I'm not sure how much further its
worth pursuing it (I'll keep it going for a while longer I think) but its
pretty interesting if you like this sort of thing:
http://www.sbibble.pwp.blueyonder.co.uk/amble/amble-0.1.1.tar.gz
Some (not all) of the demos require libsndfile and its headers.
Simon Jenkins
(Bristol, UK)