search for: lowpass

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 b...

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 P...

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/ Ogg project homepage: http://www.xiph.org/ogg/ To...

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 kno...

...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 cuttoff) PLS forward them. I'd like a 19kHz 'brick wall' cuttoff with no ripple in the passband. : ) >From my undergrad engineering work I vaguely remember about 'too steep a cutoff gives you a lot of ripple in the passband', and other such design homilies abo...

...-20110914-163803.wav|bW(2)|/usr/bin/lame "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.wav" "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.mp3" -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1 && rm -f "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-1 0001-20110914-163803.wav"") in new stack This is part of the log with Asterisk 1.8.4 -- Executing [08129981925 at speedy:7] MixMonitor("SIP/...

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 settings? -- GMX - Die Kommunikationspla...

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 -V9 which averaged 99kb/s. Lame defaults to a 13khz lowpass at this quali...

...Unless you have an extreme latency constraint, that will likely result in the best quality per bit. > > The only issue I can think of is any of the perceptual stuff will be off by -8%, e.g. crossover/mask frequencies, etc. Or is that true? > > Yep, that's true ... and since Opus lowpasses at 20 kHz, if you run it at 44.1 you will always be lowpassing at 18 kHz, and you will never get grade-A quality no matter how many bits you spend. Ah yes, it low passes below Nyquist, so that's a thing. Though, 18KHz might be acceptable. > > And with Opus_custom is all that stuff...

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: 'speex-dev@xiph.org' > Subject: RE: [Speex-dev] Sampling Rate > > Hi, > > I'm no DSP or...

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 testing or adjusting streams for special purposes would be good the possibility of change the frequency both above 20Khz and below 20Khz (or oth...

.../www.angelfire.com/rock3/roleypup/inst.tar Apologies for the huge file size (1026KB). I wanted it to be as close to lossless as it could get. Encoded with Oggenc 1.0 Shawn Riley PS- More info Encoding command strings below- "oggenc -q-0.93 --downmix" and "lame -a -q0 -v -V9 -B32 --lowpass 4980 --lowpass_width 0" produced the same size. For those unfamiliar with Lame, it's downmixing, using slowest/best compression, selecting VBR q9 level (lowest qual), limiting frame size to 32k/sec frames, & setting lowpass in Hz. Lame was more vicious in its lowpass filtering at this...

...ut: 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 1, Oct 2 2001) (http://www.mp3dev.org/) Using polyphase lowpass filter, transition band: 8002 Hz - 8535 Hz DarkIce: LameLibEncoder.cpp:194: lame lib opening underlying sink error [0] This also happens with DarkIce 0.5. Any ideas on how to fix this? Thanks in advance. -samuel --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project ho...

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...

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 from this list, send a message to 'speex-dev-request@xiph.org' con...

...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 98 LAME version 3.89 (beta 1, Oct 5 2001) (http://www.mp3dev.org/) Using polyphase lowpass filter, transition band: 3645 Hz - 3912 Hz DarkIce: TcpSocket.cpp:212: connect error [111] [root@jabez etc]# --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph....