similar to: Tuning Vorbis for Low Freq

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

Dear R-ians, I'm looking for a computational simplified formula to calculate a measure for heterogeneity (let's say H ): H = sqrt [ (Si (Sj (Xi - Xj)?? ) ) /n ] where: sqrt = square root Si = summation over i (= 0 to n) Sj = summation over j (= 0 to n) Xi = element of X with index i Xj = element of X with index j I can simplify the formula to: H = sqrt [ ( 2 * n * Si (Xi) - 2 Si (Sj

Hello, I have seen some odd behaviour of my Web Services code plus some ''similar'' postings to this list, so I believe this may be a bug in Rails. But I''m failry new to RoR, so maybe I''m doing something wrong. I''ll explain: I have a WS API defined, communicating with a remote system that expects a stucture: When defined like this (all

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Folks, If I have a series mm of, say, monthly observations, and a series dd of daily dates, what's a good way of expanding mm such that corresponding to each day in dd within the corresponding month in mm, the values of mm are repeated? So e.g., if I have mm: mm <- c(15, 10, 12, 13, 11) names(mm)<-c("Nov 2008", "Dec 2008", "Jan 2009", "Feb

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 use Sony (previously Sonic Foundry) Sound Forge, which allows me to save audio files in .w64 (Wave 64) format to get around the 2GB .wav file limitation. W64 was invented by Sonic Foundry, and is an open format as far as I know. The only programs I know about using the .w64 format at the moment are Sound Forge and Steinberg Nuendo, although there may be others out there. With increasing

The following modification is needed in os.h to compile under C++Builder. It has the alloca prototype in malloc.h. #ifndef __GNUC__ #ifdef _WIN32 # ifdef __BORLANDC__ # include <malloc.h> # else # define alloca(x) (_alloca(x)) # endif # define rint(x) (floor((x)+0.5)) #endif #endif Encoding a full range pure tone (500Hz) seems to cause a domain error in sqrt(). Can anyone

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

Package: logcheck-database Version: 1.2.37 Severity: normal Hello again, openntpd gives messages like these failry often: Apr 7 14:25:55 terminus ntpd[673]: peer 204.17.42.202 now invalid Apr 7 14:26:10 terminus ntpd[673]: peer 204.17.42.202 now valid I am not sure if this is something that an admin may find relevant but they happen fairly often and they do not offer a lot of info for me.

Hi All, I am plotting vertical lines using xyplot in lattice and type="h". It works well, but the problem is that the tops of the lines are convex and the bottoms are concave. Is there a way to flatten the tops and bottoms? Here's my code: Source<-matrix(1:30,10,3) colnames(Source)<-c("x","y1","y2") Source<-data.frame(Source) xyplot(y2+y1~x,

Hello. > x_c(1,2,5,7) > plot(x,type="h",lwd=10) produces for me a plot where the tops of the high-density lines are rounded. Is there a way to get the high-density lines so that their tops are flat? Thanks, Michael -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send

Hi all, I am attempting to tune a (RedHat 7.3) Linux box running samba 2.2.8a. This application is serving files to PC clients via samba, and the PC clients are uploading files to the server via ftp. Now for the specifics... These files are mpeg2 encoded video with bit rates ranging from 3Mbit/s - 20Mbit/s. When there is zero write activity against the disks I can kick off multiple concurrent

> On Nov 3, 2018, at 06:26, Pete Biggs <pete at biggs.org.uk> wrote: > I don't think dump has changed anything in 20 years or so! And I > certainly can't see it changing such that it can't read old files - > that is, sort of, it's raison d'etre. A bit off topic but... Funny story, at a previous job, my boss was able to extract the data off tapes made in the

This came out of some debugging I was doing to figure out how BE mode works on this hardware. Among other things, it came out that we're not exposing 16-bpp mode support and that the ARGB8888 mode that we do expose is broken. Also the overlay logic was pretty broken, I must have only tested with very "normal" overlay buffer sizes with modetest before. That said, this code has only

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

Is there a way to increase the "sensitivity" of the color palette in order to more clearly represent certain sections of data? For example I am wanting to clearly differentiate between height data for a rolling landscape but because of the extremes of the dataset (sea and mountain tops), the bulk of the landscape is shaded in closely approximating green - i have attempted to do this by

> 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'm using an apache server to host 8 virtual hosts. even though this server is local.. 7 out of these 8 virtual hosts open extremly slow.. it takes around 10 seconds to open a page.. though the 8th (which is a completely different site) it opens fairly fast in around 1 or 2 seconds tops.. i tried tailing the error_log and i found nothing .. is there a way i could monitor wht each