similar to: lowpass option (Was: RE: channel coupling in rc2)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've been off this list for a while now so I lost touch with ogg vorbis development. Could somebody please bring me up to date? As I take, the current RC2 release contains full 1.0 feature support (incl. channel coupling). Does this mean I can now start encoding all of my music as a ~128kbit/s ogg bitstream without fear of reduced

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >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

Well, this may not be unique to RC2. I believe this is the first time I have noticed it. It is definitely the first time I have vorbis encoded this song. Try encoding They Might Be Giants - Istanbul. Go to offset around 50 seconds. It should say "Maybe people just like it better that way." However, the maybe is pretty bassy (low pitch) and is just gone. Anyway, RC2 is so much

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

In case you missed it, RC2 is on slashdot.org too: http://slashdot.org/article.pl?sid=01/08/13/1811241 BTW Gian-Carlo Pascutto wrote on /. that RC3 should appear very soon (in a week or so)? Also, it looks like Wall Street Journal article is not available on-line (at least to non-registered persons). If someone is registered or have a paper edition, please post this article somewhere so that we

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

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

Hi, maybe this is of interest for someone out there. I found that taking the difference between an original sample and its decoded Vorbis counterpart is a cool way to mangle sounds (especially voice samples get a nice weird touch :] ). A side effect is that the result is not only exactly what information the encoder omits, it also contains things that it adds to the sample (artifacts). Leaving

And thats only a player. What about an (Open Source) encoder? Nah, I'll stick with OGG for the time being. Regards, Mark -- Mark de Bokx Internet Service Engineer ICT Communication and Media Services PlanetMediaGroup the Netherlands "They that can give up essential liberty to obtain a little temporary safety deserve neither liberty not safety." --Benjamin Franklin

Hi! I would like to share with you an interesting test result. I re-encoded a song 10 times (the 2. made from the 1., the 3. made from the 2.,... the 10. made from the 9.), with a beta4 (CVS 20010620) and with an RC2 (CVS 20010817) library version, 256 kbit/s mode, channel coupling disabled at RC2. The beta4 is very good at the 10. encoding too, but the RC2 has some interesting quality bugs. So,

This question has probably been asked a zillion times, but I'm either too dumb or too lazy to sift through the archives, so I'll ask it here again: Which of the bitrates in RC2 utilize channel coupling and which don't? And when can we expect to choose for ourselves whether we want to use channel coupling at any particular bitrate? Thanks for your patience, mlf -- Michael Flegel,

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

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

I really don't know if this is the same problem that was reported by Ingo Saitz (I really couldn't say which one was 128kbit and which was the original wav when blind-testing. The original had some distortion that perhaps does somehting with my cheapo soundcard) but here's a description of what i've found: When doing some sample encoding with rc2 (rebuilt rpm with latest redhat

Hello -- I desperately want to start using Ogg compression on music I release on the internet, but I am encountering a fair amount of artifacts. I noticed some in beta4 and was hoping they would disappear with RC1. However, I've been playing with RC2 and I've still been banging my head into a number of problems. A majority of the music I make has long drones with a fair amount of spectral

Try this: make encoder invert polarity of right (or left) channel on input and listen to output files (at about 128 kbps). I know that I am being nasty, but some software uses this kind of trick to produce 'surround'-like effects; this stuff causes channel coupling to puke (works fine with RC2 when channel coupling is disabled, tried with my Winamp output plugin which has an option to do

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

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

Hello all, I'm glad to announce to you that Vorbis now has full ReplayGain support. If you're not familiar with ReplayGain, take a look at www.replaygain.org. The main features are: a) all songs play back with equal loudness b) removes the need for normalization c) allows for clipping prevention Using it is very simple. Get a compatible decoder (ogg123, XMMS and WinAmp all support it