Assuming a 7.1/192K/24bit $3:- chip onboard ... What next?
First I will say, it actually will do the planned send and return
to/from the two pieces of stereo outboard equipment I have (that is not
going anywhere,) as well as do separate headphones and monitors. So I
should be a happy camper ...
Suppose one was to face the real world of playing live again? In my
experience this can be a torturous venture into the realms of cheap
lights and the evil thyristors and diacs of this world, all trying to
make their voices heard through my equipment ...
So, how about turning a single unbalanced stereo jack into a single
balanced mono jack by converting a single mono signal to a stereo signal
with inverted phase, would that be a good idea? If so, I believe this
might work in a simalar way on microphone input as well.
If the above holds and we then have two (instead of one) DAC's working
push/pull on the line, would it be possible to take advantage of this?
What I have in mind here is that, the lower you get in the "bittiness",
the more the systenatic errors of distortion will be apparent.
For starters, how about having one the phases at a level slightly below
the other? This should ideally trigger transisitions between absolute
levels at slightly different times for the two phases, giving us an
extra 6dB of useable headroom.
mvh // Jens M Andreasen
Hello all,
I want to know if is there anybody that know anything about open source
implementation of Speaker Recognition[1]
or what kind of tool i can use to develop this kind of tool on linux?
Regards,
Ste
[1]http://en.wikipedia.org/wiki/Speaker_recognition
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Hello everyone!
This is again a bit OT, but still. I have a program with two pthreads. One
pthread is running an audio player on a file repeatedly, the other is waiting
for a signal. If "the other" gets the signal or timeout, then it should tell
the player to stop. Unfortunitely with mplayer kill -15 doesn't have the
desired effect. So I thought:
What if I could make mplayer's stdin available to the program and simply
send character 'q'. But because of pthreads, I left fork and exec* alone and
went with system. Any ideas what to do? If someone knows something nice with
fork and the like, I've seen something with pipes for this. Otherwise the
player-command I call can be called from within a shell script, so I can do
funny things with redirections.
I hope I made myself clear and didn't write too much again... :-(
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
This is totally alphabetagamma = unfinished and unpolished software. Release
early - release often is my guide here ;)
Here's the summary:
ProcessPP is the "realisation" of a rather silly idea: A livecoding
environment for C++. As C++ is not interpreted, but compiled, this indeed
does sound like a really silly idea. But the twist is that while “true”
livecoding cannot be done, it can be at least made as easy as possible to try
out new code snippets.
ProcessPP consists of three distinct parts
- procespp: A host application that can load shared objects and execute the
code contained in them
- qprocesspp: A Qt GUI application which serves as an editor for hacking away
functions and to compile them easily into a shared object. Then it tells
processpp to load and run the shared object
- libprocesspp: A shared library which is the “glue” between processpp and
user generated shared object files. Right now it only containes code for the
shared object to find out stuff like samplerate, buffersize, number of
input/output channels, etc..
For illustrative purposes here are the steps involved in making noise:
- File->Open from Template (choose process.template.cc)
- Add some code in the inner loop to make noise
- Process->Run
- Hear the noise!!!
If the above don’t work, take a look at the log in the lower half of the main
window. Often some slight tweaks to the config file
(~/.config/Ugh!/process++.conf) need to be made.
Get it here: http://tapas.affenbande.org/wordpress/?page_id=91
Regards,
Flo
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Hello!
Sorry, for this, but my skills left me to unskilled... AGAIN... :-(
I'm working with pthreads and now I like to hand over mains argv to the
thread run_me.
int main(int argc, char *argv[]);
void *run_me(void *param);
How to properly use param as a char * [] inside run_me?
Thanks forany help!
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
Download from http://www.notam02.no/arkiv/src/?M=D
jack_capture
============
jack_capture is a program for recording soundfiles with jack. Its default
operation is to capture whatever sound is going out to your speakers into
a file. (But it can do a number of other operations as well...)
Note:
This version includes Hermann Meyer's jack_capture_gui2 program.
"jack_capture_gui2" is a nice graphical frontend
for for jack_capture with lots of options.
Many thanks to Herman for the contribution.
Changes 0.9.23 -> 0.9.30:
*Added Hermann Meyer's jack_capture_gui2 program.
jack_capture_gui2 is a nice graphical frontend
for for jack_capture with lots of options.
Many thanks to Herman for the contribution.
*Don't exit in case port is not found.
*Print runtime warning and error messages on top of the
console to avoid printing the console meter yet another time.
(it's much prettier also)
*Fixed a bug that could cause (and especially after the switch
from calloc to my_calloc apparantely) segfault when specifying
--port more than once. Thanks to Peder Hedlund for spotting
the bug.
*Print error instead of segfaulting when a specified jack port
does not exist.
*Removed -g option and changed -O0 to -O2. (Oops, don't know
how long that's been there)
*Make sure the stop semaphore is initialized before it might
be called.
*Changed the --recording-time / -d option to record exactly the
correct number of frames. (The format for the option is still
in seconds though). This fixes the problem where the wall
clock and the soundcard clock drifts apart.
*Always increase the buffer size with 2 seconds when more than
than half the buffer is used, unless maximum buffer size is reached.
*Added the --maxbufsize / -MB option which sets maximum buffer size.
Default value is 40 seconds.
*Decreased the default buffer size from 20 to 10 seconds.
*Changed internal data representation from lockless ringbuffer to
lockless lifo and fifo stacks. Unmodified lifo/fifo code taken
from midishare. (Copyright Grame 1999-2005)
Rollendurchmesserzeitsammler v0.0.7
------------------------------------
The Audio Rollendurchmesserzeitsammler is a conservative garbage
collector especially made for running inside an audio DSP thread.
0.0.5 -> 0.0.7
* Cleaned up source a bit.
* Fixed a bug in "tar_entering_audio_thread"
which caused it to return false if currently copying a different heap.
* Cleaned up the critical section handling between the DSP thread and
the sweep thread. (it was really messy)
I've been looking around for a library to read and write SFZ files,
which is an open sampler format released by Cakewalk:
http://www.cakewalk.com/DevXchange/sfz.asp
Finding none, I thought I might try my hand at writing a library for
this myself, as there is no embedded wave information like with Gig
files. SFZ is simply a text file to be parsed.
Now, I know about writing a good header file, and its associated class,
and all that, but I have no knowledge of how to write it as a dynamic
library. Google searches on every possible permutation have been
worthless to me as well.
I would prefer to write it in C++, as that's what I know, and even then,
not too well, hence why I thought I'd start with something simple like
parsing a text file. If anyone has any advice, recommendations, or
ideas, I'll happily listen and learn. I have yet to think too much about
how the data will be stored in the class, and what methods to make
available to access it, so if anyone knows any best practices there, I'd
really like to know. Consider this a feeler post.
I'd ultimately want this for a future project, which you can guess at by
now.
Thank you for the help!
Regards,
Darren Landrum
I am doing some preliminary testing of CUDA for audio, Version 2 (final)
has been out for a couple of days, and this is also what I am using.
In order to get anything done, one will always have to do something else
first, here: Get some data transferred to the board. Surprisingly this
appears to be ten times harder than getting the data back, at least for
very small data sets representative of a bunch of on/off events or a
milliseconds worth of samples.
For 1024 bytes the transfer will take about 0.2 ms. That is 20% of the
available time if we use a time granularity of a single millisecond.
OTOH this appears to be mostly an intial constant, (much) more data can
be transferred in the exact same amount of time if that is what is
needed.
Now my card (8400GS) is neither the latest nor greatest and I therefore
wonder if anybody with better equipment is experiencing the same
phenomenon? This card does not support aynchronous transfers, which
otherwise might have been the thing to use here.
mvh // Jens M Andreasen
hello,
i recently purchased a madiface with express card.
trying to compile a debian-source kernel 2.6.26-4, with replaced hdspm.c
and hdspm.h, following instructions on
http://wiki.linuxproaudio.org/index.php/Driver:hdspm
i get following compile error:
ERROR: "__muldf3" [sound/pci/rme9652/snd-hdspm.ko] undefined!
ERROR: "__divdf3" [sound/pci/rme9652/snd-hdspm.ko] undefined!
ERROR: "__fixdfsi" [sound/pci/rme9652/snd-hdspm.ko] undefined!
ERROR: "__adddf3" [sound/pci/rme9652/snd-hdspm.ko] undefined!
ERROR: "__floatsidf" [sound/pci/rme9652/snd-hdspm.ko] undefined!
i tried to get more info on the net about this errors but did not find
usefull hints. any help would be appreciated!
peter
Since returning from holidays (i.e. 10 days) I've been running
patchage as my visual interface to Jack. Never managed to install
it from source (there was always something missing or at least one
compile error), but recently discovered the CCRMA package (thanks
Fernando !).
For the sort of things I'm doing, usually involving a lot of
interconnected Jack clients, it provides a much clearer picture
than qjackctl, and I like that a lot.
Some suggestions / feature requests:
- Include the window scroll offsets in .patchagerc
- Make sure new apps are allways displayed in the
visible part of the window.
- Some way to select and connect groups of adjacent
ports (I'm using 64ch soundcards, and most connections
are multichannel).
- Optionally collapse such sets of ports into a single
visual object.
Long term:
- Multiple tabs, each one controlling a Jackd on a
remote headless machine. This will require splitting
the app into a server and client with a network
connection in between.
Ciao,
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