similar to: low pass filter frequency adjustable

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

> 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

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

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

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

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, 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

Hello, I understand it would be better to post several messages with separate topics but I hope I don't cause too much mess if I put it all in a single message this time. To be clear, recently I've been testing Opus Tools under Windows and these are my questions/observations. ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps with Opusenc and then decoded

-----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

Devin Campbell wrote: > >> http://www.starbeamllc.com/images/feedback.mp3 > I guess my next question is: is there something special about my stream > or my songs that's causing this? I'll have to do some experimentation. Owtch, my ears. :( I find that 16 kHz beep most irritating, it hurts even at low volumes. ;P Anyways, the humming you hear is actually multiple tones,

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

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/

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

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

> > I am talking about the original sweep. > > The original sweep stops pretty close to 24 kHz. I mean the original sweep _as_encoded_, sorry.

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

Dear R-users I am trying to make an adjusted Kaplan-Meier curve (using the Survival package) but I am having difficulty with plotting it so that the plot only shows the curves for the adjusted results. My data come from a randomised controlled trial, and I would like the adjusted Kaplan-Meier curve to only show two curves for the adjusted survival: one for those on treatment (Treatment==1)