search for: lowpassed

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 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/

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

A while ago someone asked about a low-pass filter for oggenc and was told to get AFsp and filter outside of Oggenc. Well, I got it, and am totally lost (It's way more complicated than SOX) so now can anyone briefly describe what type of filter I should set up (FIR, IIR, all-pole), why one is better than the other, and if you have filter coefficient files lying around (lowpass, 19 or 20 kHz

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

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))

Back in the day, when I was still using LAME, I was aware of the fact that the program would cut off frequencies above a certain level (lowpass). With 192 kbps this was usually around 20Khz (which is the highest frequency a human can hear, as far as I know), and at 128 something like 16Khz. Does Vorbis do something similar? If so, does someone have a chart of the cutoffs at the different quality

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 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"

As a matter of interest I tested the quality at -1 which has a nominal bitrate of 45 but the 2 tracks I encoded both averaged out to exactly 48kb/s. I noticed the lowpass defaulted to 15.1khz like -q0 so that may need addressing at some stage. I used a lowpass of 13khz which overall sounded better with less artifacts. I found it sharper & clearer than a lame encoded mp3 encoded with

Thanks for the answers Benjamin? On Jun 14, 2013, at 8:05 PMEDT, Benjamin Schwartz wrote: > I have flexibility in the frame sizes of the unencoded audio, and packet sizes on the RF link. > > This implies that you don't have a very tight latency constraint, so you can afford to run a resampler. > I assume the resample costs CPU cycles? the RX is battery powered, I'd just as

Oops, CTRL+Enter send strikes again ... At the other end for playback you can convert it back to 48000 (or whatever) by repeating each sample 3 times (48/16 == 3), then running a 8000Hz lowpass over the result to remove any aliasing artifacts. Cheers, David Hogan > -----Original Message----- > From: David Hogan > Sent: Tuesday, 12 December 2006 10:44 AM > To:

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

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

Hi everyone. I did a rough recording of an instrumental (electric piano sound) & e-mailed it to a friend in Vorbis 11025 Hz / mono. I was seeking a bitrate in the range 8-16k/sec. The song is 2:55 in length. q=-1 happily achieved a 12.6k/sec bitrate. All file sizes I mention in this are for files without informational tags. And I hope this isn't interpreted as trying to plug my music or

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

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, I just wanted to know if there are any recommendations for pre-/post-processing (processing power isn't a question). I'm aiming at 16kHz and tried an lowpass-filter at 11kHz before encoding but this didn't improve the results... <p>Regards, Thomas --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

Yea, I figured it out...brain fart on my part. Got it working, but getting some sort of TCPSocket error now... still trying. :) [root@jabez etc]# darkice -c /usr/local/etc/darkice.cfg DarkIce 0.6 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: /usr/local/etc/darkice.cfg Using POSIX real-time scheduling, priority