similar to: Sieve mails with decoded subject

Hallo, an alle Wine-Freaks, das Programm - bis jetzt Freeware-, ist High-End unter den Hornsimus: http://www.users.bigpond.com/dmcbean/Setup.exe , 400 kB Ich habe wine in etlichen Versionen bis zur 0.9.26 probiert, auf Debian und Mepis (ubuntu), auch Crossover von 4.2 bis zum Beta 6.3a. Es geht nicht, alle Dinge wie native dll,s andere Winvers bringen keine Besserung, IE, Dcom--hilft nicht. Alle

Good Morning! "Was lange w?hrt wird endlich gut" - this is a german saying, meaning "All good things are worth waiting for". And yes, the waiting period on this was was long, very long, maybe too long. But it is nearing: The release of CentOS 4.8 - we "just" had some other things to attend to, first. This means: We are in need of Release Notes again - and in need

Hello, I have two vectors of different lengths. Fx a <- 1:9; b <- c(4, 5). What is the best way to remove the elements in vector b from vector a so that the result would be a vector with elements c(1,2,3,6,7,8,9)? Best regards, Kalle _________________________________________________________________ Find masser af gode tilbud p?? MSN Shopping http://shopping.msn.dk/

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inbound INVITE like: mumble-fratz-sip%3Afoo%40bar.com at asteriskbox.abc.com:5062 The older version of asterisk replies with a 200 OK and a Contact: header that looks like: Contact: <sip:mumble-fratz-sip%3Afoo%40bar.com at

This wav produces a bit of audible static when encoded at the highest bitrate vorbis will allow me to encode at (30.7kbps avg.): http://staff.xmms.org/zinx/misc/5551234.wav.gz An encoded version, with the bitrate set to approximate 128kbps (I think it output to 29 some odd kbps): http://staff.xmms.org/zinx/misc/5551234.ogg This is with the latest CVS tree as of Sat May 12 10:06:17 UTC

[Forwarding to PUBLIC mailing list] On Fri, Nov 25, 2016 at 11:02:29AM +0100, Andreas Schneider wrote: > https://github.com/cryptomilk/parsemscat Andreas wrote a library for reading MSFT catalog files and digital signatures, and the question (for the list) is whether we can use this in virt-v2v to parse the virtio driver catalogs. It's a C library, so that's a good start. Rich. --

Dear all, I encoded and decoded some files using speex codec and I find out with than when using the original raw file and coding it in NB with different bit rate, complexity or with/without enhancement some kind of slight artefact (a strange low level sound appears) is created at t=1.12 s. This is just before an attack which is present in the signal. I did not try to solve the problems but if

Hi, Please start with sampleenc.c/sampledec.c from the manual. Jean-Marc Andre Kirchner a ?crit : > Hi all, > > I have just started playing with speex, and come up with the following code, which just encode a frame of 160 shorts, and the decode it. > > For some reason the decoded sample is completely different than the original one. is my code wrong? If so what? Or is it a

I am encoding and decoding a stream of data. The data comes in in 320 byte packets, these are encoded and I get back 38 from the speex_bits_write method. I am under the impression that this means 38 bytes got written to my output buffer. The data to this point looks fine, but when I put the data through the decoder all the bytes come out as 0. So my first question is why is this happening, and

Hy, can anyone recognize that pixel noise in the playbackfile recorder file: http://www.megafileupload.com/en/file/135429/FMODTestRecording-wav.html playback file: http://www.megafileupload.com/en/file/135431/FMODTestPlayback-wav.html i have no idea what that is anymore. i try everything i know, from changing the way of copying data to different encode/decode algorithms the recorded file is

Hi, in a VoIP application, the endpoint A send speex payload to B. B doesn't know how A acquire audio, it only know that the channel is narrowband so, how can B know the size of the output buffer to pass to the speex_decode()? Thanks, Daniele.

On 14 April 2010 23:50, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > Il 14/04/2010 14:37, Randy Yates wrote: >> >> Usually a buffer is one frame of data, and a frame is 20 milliseconds. >> Since the sample rate is typically 8 kHz in narrowband mode, this >> corresponds to a buffer size of 160 samples. > > Hi Randy, thanks for the reply. >

On 15 April 2010 17:55, Daniele Barzotti <daniele.barzotti at eurocomtel.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Il 15/04/2010 01:30, Conrad Parker wrote: >> >>> But how can I know the size of each speex frame into a multiframe payload? >> >> use speex_bit_read_from() just once on the packet, then call >> speex_decode()

Hi all speex developers, ??? I have encoded 10 seconds buffer of Mic voice (raw) using speex on an embedded device, then transmitted it to a PC where i decoded the voice,in a way similar to the sampleenc.c and sampledec.c on website, ?BUT my problem is that the decoded voice is totally different from the original,what i hear is just noise !! what is possible to be the reason ? Any help is

On Sun, Jun 24, 2001 at 07:45:56PM +0200, kai@adminhell.org wrote: > I also had this verify error encoding a wav I ripped from a CD. I didn't > report this as it happended on flac running on debian linux -current unstable. > This error happened only with one track of a CD I was ripping. > > Another reason for me not reporting this was, that flac 1.0 running on OpenBSD > 2.9

On Mon, Jun 25, 2001 at 01:54:02AM +0200, kai@adminhell.org wrote: > On 24-Jun-2001 Matt Zimmerman wrote: > > Can you try trimming down the WAV file > > (e.g., with a program like xwave) until you have the smallest chunk which > > still triggers the failure? > I tried to clip both ends of the original wav, but the error was only > reproduceable, when I cut of the end of

On Sun, Jun 24, 2001 at 02:30:56PM -0700, Josh Coalson wrote: > There have been reports of -9 using huge amounts of memory. -9 is really > theoretical, but people always seem to want to try the max setting. Anyway, > that's not an excuse but figuring out why -9 is using so much memory is lower > on my list than other stuff. -8 should get within 0.01% of -9 and is pretty >

On Mon, Jun 25, 2001 at 10:54:09AM -0700, Josh Coalson wrote: > > Also, Kai has been kind enough to send me a copy of his file which > > has a > > problem only on -8, which I'll be looking into soon. > > Matt, if you whittle this file down can you make it > available also? Kai indicated that even if he only trimmed a small amount off of each end of the file, the

On Mon, Jul 02, 2001 at 09:50:38PM -0700, Josh Coalson wrote: > After some intense debugging, I found the problem. One block in the file > triggered a very rare bug in the LPC coefficient quantizer caused by > insufficient floating point precision. There is a snippet to compute the > log(base 2) of a number: > > floor(log(cmax) / M_LN2) > > It turns out on at least