similar to: Encoding 8KHz audio (voice)

Hello all, This is a request for help from somebody who has the Ox interfaces working in R. I am trying to get the Ox interfaces working for Arfima and Garch modelling. However, I am having several problems: 1. The link to download G at rch_v40 does not work. Does anybody have a copy to email to me please? 2. Various guides offer different instructions for installing Ox in the correct place

Hello all, I am trying to write an interface for BQN <https://mlochbaum.github.io/BQN/> and R However, even | dyn.load("/usr/local/lib/libcbqn.so"); .Call("bqn_init")| immediately fails on the BQN side, viz. |Failed to allocate memory for JIT 200 times; stopping trying|| ||CBQN interpreter entered unexpected state, exiting. |This is because BQN tries to allocate

Hello, I try and run mongrel 1.0.1 and ferret 0.11.4 on Windows XP. To break mongrel all I have to do is invoke a class with Ferret in it. Here''s my class: require ''ferret'' include Ferret class SearchIndex def initialize print "SearchIndex:Initialize\n"; end end When I put this in my controller, I break mongrel: searchIndex =

Hello, I''ve several problems with quota on our testcluster: When I set the quota for a person to a given value (e.g. the values which are provided in the operations manual), I''m able to write exact the amount which is set with setquota. But when I delete the files(file) I''m not able to use this space again. Here is what I''ve done in detail: lfs checkquota

I sent this last night but it didn't show up in the archives. Trying again without the attachment. Are messages with attachments blocked on this list? On Tue, Jan 6, 2015 at 11:50 PM, Darby Vicker <darby.vicker at gmail.com> wrote: > Thanks a ton for the suggestions and info everyone. > >> Two things I think you should consider doing... >> 1) in your next post (and

> The cost of down-sampling, if done efficiently, is probably less then > the cost difference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz

Hello, I saw in the CentOS 6.6 release notes that the Xorg ABI has changed but that "This should not concern anyone who uses the default drivers shipped with CentOS." I'm using the ATI drivers in Base but X still fails when I boot a CentOS 6.6 kernel. I was hoping this is a known problem that would get fixed with a kernel update but I'm still having problems with the

On 24 Jun 2015, at 15:33, Martin Jangowski wrote: > I regularly broadcast with 48kBit streaming rate (mp3, mono, 44kHz). > My fallback-file is a 244kBit, stereo, 44kHz, mp3 file. Works like a > charme.... That depends on the player, if it works you are lucky. It might not work for other formats or with some players.

Hi! I'm using the newest (beta2) speex library on PocketPC (Windows powered) and experiencing some problems with noise suppression preprocessor turned on while encoding 44kHz files. You can hear the example here: http://szalik.net/speex (this is a 44kHz, 16, mono file encoded in UWB mode) I tried using speex cross-compiled with gcc (fixed point + arm4 asm) and VS (just fixed point) and it

I had a test machine up and running Samba, joining an AD domain, and set up with winbind... when I ran "getent passwd" it would list all the local users, and all the domain users. I could also log in as a domain user without any problem. So, I made an image of the hard drive, and copied it to 7 other machines. Now, those 7 are having a strange problem. I changed the names and IP's of

constant prediction see this page http://flac.sourceforge.net/format.html oversampling.. i maean digitally change the wave file rate form 44khz to 440 khz it make next sample easyer predictable if i'll found my old work i'll tell you how much compression i could achieve

On Fri, 16 Nov 2001, Jerome Alet wrote: > one thing that would be nice in DarkIce would be to allow the user to pass > specific reencoding options for each server, e.g. DarkIce could acquire > the audio in stereo and send it to a server in mono and in stereo to > another server, which is AFAICT impossible today. I agree! Also, something I've been looking for is a way to pull

What have people found to be the ideal setup for recording asterisk prompts? I'm looking for both the ideal application to record them in, the ideal format, as well as hardware (do I need a fancy studio mic or will a headset mic work?). Thanks, Justin

I know speex is not supposed to do a great job compressing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down

asked once before and didn't get a reply, but it was probably lost in my back-and-forth with Jack.. where is the encrypted password support for icecast 1.3.10? The documentation references a "mkpasswd.c" source file that doesn't exist anywhere in my icecast directory tree.. -------signature file------- PGP Key Fingerprint: 446B 7718 B219 9F1E 43DD 8E4A 6BE9 D739 CCC5 7FD7

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in

Crew, I've run into something a little odd with the vorbis encoder and would like some input. I use SpacialAudio's SAM3 Broadcaster and am having problems with the ogg encoder when I create a mono output stream. It seems as if the signal level of the left and right sides are being combined before the encoding process, leaving the sound muddy and overdriven. As such, if I reduce

Hi, I've tried vorbis on a stream that has ~500Hz tops and it worked quite good, but the raw data was converted to a fake 16bit 44Khz WAV Are there any chances I can take the code and tune it to work with nonPCM data that has info from about 5 up to 500Hz? It would be nice to know where to start. Thanks you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Sat, Nov 17, 2001 at 06:57:46PM +0100, Maroy Akos wrote: > On Fri, 16 Nov 2001, Samuel Hathaway wrote: > > > I agree! Also, something I've been looking for is a way to pull sound from > > the dsp device at 44kHz and then downsample it to 22kHz for one of my two > > streams. Ideas? > > DarkIce does this already. What it doesn't do, is to have mono and