similar to: frequency cutoff?

I was wondering, is there a way to encode Oggs with EAC on-the-fly? I'm looking for a way to shorten the whole ripping and encoding process, but so far I've only seen support for the commandline encoder. -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Hi all ! I use cdparanaia for which I would like to get information. Does anyone know if there is a mailing list concerning this ripping application. In fact, I would like to know if cdparanoia can support scsi drive .... if someone can help me on this matter ... it is welcome. ;-) <p>Thanks for your help ! <p>Ce message contient des informations confidentielles ou appartenant

This may be unrealistic and useless, but I've been thinking about an interesting file-sharing solution. In my system, only Ogg Vorbis files may be shared, and they should all be directly ripped from a CD. Nothing that someone would convert from MP3, etc. While that cannot be enforceable, it should be made clear. Also, track listings should be not filenames, but the Artist and Title of the

can someone email me a list of special spots in quality settings, or point me to a website that tells these settings? i know i may be confusing, i'm talking about like the jump between -q4.99 and -q5 because of the whole lossy/lossless channel coupling, and filesize jump...i also know somewhere above there there is different high and low frequency cutoffs, and a point where there IS NO

Important (imho)! I found this site in the internet: http://www.plusv.org/ PlusV is an audio enhancement algorithm similar to SRB of Mp3Pro, but 1) better than SBR 2) Fully open-source It seems to me wise to include PlusV into new versions of Ogg Vorbis. P.S. what about some long-awaited features in Ogg Vorbis? 1) ability to turn off the frequency filter in the encoder (especially for high

Hello, I was wondering about the lowpass filter applied at the different quality levels. It seems that the 16kHz cut-off is still there at quality 3. I didn't really abx it, so maybe my mind plays tricks on me. :) Please can someone enlighten me what is used for -q0, -q1 and so on. (lame tells me whats used when encoding tracks) I know I shouldn't judge by bandwidth but I would like to

Don't know if anybody is still missing the lame oggenc features for resampling, lowpass/highpass filters etc, but I wrote a little script that uses sox to do all the stuff I need to real-time encode oggenc from the radio, or any input device. #!/bin/bash DATE=`date '+%m-%d-%Y-(%H.%M)'` DESTIN=/video/music/perftoday export DATE=$DATE'-PerformanceToday.ogg' sox -V -r 44100 -c

Hi, recently I made some test with the opus tools (enc and dec) and I'm very (and positively) surprised about the resultant quality. But the only think that I miss is the ability to change the low pass filter frequency via "--lowpass" option or similar. For example at a quality or 96 kbps the cut off of the filter starts at 16Khz and is completely cut at 20 Khz. But in case of

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on

Will oggenc have a lowpass switch? I would prefer to lowpass at 15-16khz at -q3 for use with FM broadcasting. The additional frequencies would be chopped off anyway by the transmiters hardware lowpass filter so the encoder could use the addition bits for other purposes. It could be enforced that the lowpass can only be reduced and not increased from the default. This would stop people

I would very much like a lowpass option because for FM radio broadcasting I don't want to encode frequencies above 15khz. I'm waiting for this option before switching to ogg from mp3(lame). Ross. > -----Original Message----- > From: owner-vorbis@xiph.org [mailto:owner-vorbis@xiph.org]On Behalf Of > Gian-Carlo Pascutto > Sent: Tuesday, 4 September 2001 01:46 > To:

I have a a simple function that generates a time series square wave: genseq <- function() { x <- numeric(4*365) s <- seq(as.Date("2005-01-01"), as.Date("2008-12-31"), by="month") ob <- as.vector(s[c(10,22,34,46)] - as.Date("2005-01-01")) oe <- as.vector(s[c(11,23,35,47)] - as.Date("2005-01-01")) for(.index in 1:length(ob))

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

I mentioned many months back that I intend to switch to Ogg/Vorbis from MP3/LAME once the final version is out but I need to have an adjustable lowpass filter option similar to LAME's. If this isn't done yet, could you please add it to your list for OggEnc. Specifically I need to cut all frequencies above 17khz. Thanks, Ross. --- >8 ---- List archives: http://www.xiph.org/archives/

Hace poco me puse al tema del big data y la verdad es que een este campo la decisión parece clara. Python ahora mismo está un paso por delante de R, aunque sparklyR puede igualar la contienda. Pero lo que me empieza a preocupar es que parece que si nos alejamos del Big Data, Python tb está ganando adeptos a pasos agigantados. ¿Está perdiendo R la batalla? [[alternative HTML version deleted]]

Hey, I've compiled DarkIce 0.6 dynamically linked to LAME 3.89. I'm running Slackware 8 and using gcc 2.95.3. Running DarkIce yields the following output: DarkIce 0.5 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: darkice.cfg Using POSIX real-time scheduling, priority 98 LAME version 3.89 (beta

Hi, does anybody have any ideas regarding the easiest and most efficient way of implementing a bandpass filter in R ? any help would be truly appreciated. cheers, Pat Johnston -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or

Is there a list somewhere of "standard" encoding rates? I know, for example, CDs are encoded at 44100, as is a lot of digital sound, but I've seen programs that specify different levels of quality (like radio, phone, tape, CD) and I'd like to know if there are some encoding rates that are accepted as standardized for recording at different levels of quality. If so, is there

Dr. Williams, I ran across your inquiry on one of the R-help mailing lists regarding digital filter design and implementation. I found no response to your email in the archives and was wondering if you were able to find anything. Thanks, Israel -- Israel Christie, Ph.D. Email: ichristie at gmail.com Phone: 865.766.0214 Mobile: 865.406.4615