Hello.
I have started developing a simple speech synth starting at
National Instruments SP0256 chip. Currently Allo includes
a sampled audio recorded from the chip; that should be replaced
with a "noise/pulse + format filter" speech generator soon.
I have not yet written a code for such generator, so, if you
know a short-cut, please help me.
Code is at "ftp://ftp.funet.fi/pub/sci/audio/devel/speechsynth/".
Read the USAGE first. Plenty of examples are provided, from
the chip's documents, in WORDS.
The files eye*wav in the directory are excerpts from latest Yello
album, The Eye (www.yello.ch). That is the target for Allo project.
Regards,
Juhana
Salem Radio Labs, the Research and Development arm of Salem
Communications, is pleased to announce the initial beta release of
the Rivendell Radio Automation System.
Rivendell aims to be a complete radio broadcast automation solution,
with facilities for the acquisition, management, scheduling and
playout of audio content. At present, the following components have
been implemented:
RDAdmin - System configuration and management tool
RDLogEdit - Program log editing tool
RDLibrary - Audio acquisition and management tool, for maintaining a
library of audio content
RDAirPlay - On-air audio playout application
RDCatch - Automatic audio recorder specifically optimized for
simultaneous capture of multiple long-form audio programs
Rivendell is released under the GNU Public License. Further
information, source code and RPMs for select Linux distributions are
available at the Salem Radio Labs web site at
http://www.salemradiolabs.com/rivendell/
Cheers!
|-------------------------------------------------------------------------|
| Frederick F. Gleason, Jr. | Director of Broadcast Software Development |
| | Salem Radio Labs |
|-------------------------------------------------------------------------|
| "You see, wire telegraph is a kind of very, very long cat. You pull |
| his tail in New York and his head is meowing in Los Angeles. Do you |
| understand this? And radio operates exactly the same way: you send |
| signals here, they receive them there. The only difference is that |
| there is no cat." |
| |
| -- Albert Einstein, upon being asked to describe radio |
|-------------------------------------------------------------------------|
Hello,
DRC 2.3.1 has been released. Changes:
Some minor corrections to the program have been performed and the
documentation has been restructured. A new option to automatically count
the number of lines in the target function and microphone compensation
files has been added. A new optimized sample configuration file has been
added.
For details see: http://freshmeat.net/projects/drc/
Regards,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it
Hello,
together with the latest release of MCP/REV/VCO-Plugins by Fons Adriaensen,
AlsaModularSynth-1.7.1 can be considered as a major step forward in realistic
virtual analogue modular synthesis. There are some new sound examples (.ogg)
on alsamodular.sourceforge.net that give an impression of this.
For a detailed list of what is new, check out the "News" section on the
project page.
Have fun !
Matthias
--
Dr. Matthias Nagorni
SuSE Linux AG
Maxfeldstr. 5 phone: +49 911 74053375
D - 90409 Nuernberg fax : +49 911 74053483
Hi all.
I've been trolling and testing various sync methods and I've noticed some
questions popping up. Is it just me or is the linux audio situation a bit
incomplete at this point with regard to information on that subject?
It may just be that most of the programs are still struggling to develope
sync techniques (MTC, MMC, ADAT sync, SMPTE, jack transport, etc. etc.), but
also is there a relative lack of info/tutorials/etc?
The reason I ask, is because not only for my sake, but for the sake of the
community, I'd sure be glad to try and make a dent in that, maybe document
some toots, or whatnot. of course, this all has to be in sync (no pun
intended ;) ) with the development of the capabilities...
I wish I had time to sort of type out something more detailed - I'll get to
that sometime here (running short on time right this minute :) )
one basic hole I see at least in my understanding, is what the HECK is
jack_transport, how does it fit in, etc.? that's ONE - there are lots of
holes for me *laugh* another curious one is ecasound? it's the veteren
around here in a lot of ways - does it sync to anything?
ok well I'll wait for thoughts and then as soon as I can get a chance, I'll
write something longer and sort of put out to y'all what I've found out, and
see what holes can be filled - cheers (sorry for any unclarity in this
email, unclarity is why I'm starting this thread! :) )
- Aaron
www.nquit.com
Hello.
Anyone would like to develop cdparanoia further?
I ripped a CD two times with the following results:
(== PROGRESS == [ !-------| 159332 00 ] == :^D * ==)
(== PROGRESS == [ + + + | 159332 00 ] == :^D * ==)
The errors are explaned this way:
- Jitter correction required
+ Unreported loss of streaming/other error in read
! Errors are getting through stage 1 but corrected in stage2
Which error is more severe: "+" or "!"?
Is "+" corrected properly?
The problem is that the channels of other ripping are swapped at the
point where "!" is located (but "!" may not have caused it). I don't
know which one of the rippings is incorrect (they both could be
incorrect). The channels also have one sample offset after the
swap.
The second ripping has "+" before the "!" point, but both rippings
are indentical up to the "!" point. Would that mean that "+" is
a harmless error and that thus the second ripping is correct? And
that we should rewrite only the algorithm handling the "!" errors?
In addition to rewriting "!" handling (or "+" handling if that caused
the channel swap), I would like get out a printed list of errors.
The error list could then be used in an audio editor for marking
the errors with red color (say).
Also an audiofile comparer program would be great to have: a program
which finds matching regions. Now it was easy to compare the regions
up to the "!" point with md5sum program, but it was difficult to compare
the end part due misaligning. I compared them visually at a few random
points. In such a comparer program each matching sample would match
bitwise for this application, but they could match with an error tolerance
if the application is to compare an original audio and an edited audio
which is affected by dither noise (say). So, such a comparer would
reveal what edits were done in an audio editor -- I have needed such
a program a few times earlier.
Regards,
Juhana
Hi All,
I'm still new to LAD so sorry if this has been asked before - is there
anywhere online that has an introductory tutorial to creating custom
pixmap sliders? I'm looking for a tutorial similar to the gtk dial
tutorial on the gtk.org site. If anyone has a link or info I'd
appreciate it.
Cheers,
Adam
Hello,
I am looking for a deconvolver, that is able to produce impulse
responses from sinus sweeps (and especially the exponentially
sweeping sine wave introduced by Farina).
Do you have any suggestions or at least tips to start an
implementation by myself?
Recently I managed to use the mls tools from nwfiir to produce
an IR of my microverb.
I had to learn the hard way, that simple soundcards are not able
to be used as MLS source because of the non linearities. Even a
simple DA-AD loop gives a result wave that mls2imp cannot cope
with. But an empty loop with an US-122 (unfortunately not with
linux for now) gives something very near to a dirac impulse!
The hunt for the linux convolution reverb has started ;-)
Uwe
--
voiceINTERconnect www.voiceinterconnect.de
... smart speech applications from germany