similar to: Quality -1 default low-pass

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

May I suggest that the up coming RC4 or 1.0 needs some quality selection adjustments. It has been said that if you are happy now with a particular quality setting, later releases will produce the same quality but reduce the average bitrate. This is not currently the case. IMHO -q4 in CVS is producing almost perfect results and features audio frequencies up to 21khz -- almost CD quality.

Caleb wrote: > i dont understand you people. >-q0 should be poor quality, only that in vorbis, the poor quality is > actually good! :) Another advantage I see in reducing the nominal bitrate for q0 to 48kb/s with a ~13khz lowpass is a smoother transition in average bitrate and frequency resolution from 22khz to 44khz sampling. Currently there is a jump from 11khz (at 22khz

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:

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

On Aug 07 08:30:31, hans at stare.cz wrote: > 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. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

My wish for 1.1 is to stream my station with VBR at bitrates averaging 28-32kb/s with a minimum 32khz sample rate. Hopefully this will be achievable with the new low bitrate tuning that is planned. This could put it on a par with Real Audio at 32kb/s. Currently a quality of -1, 32khz sample rate produces an average around 34-38kb/s which is too high for modems. I could currently use a managed

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. $

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

Hello. Me again. Have you tried to encode piano solo? Noticed high bitrate Opus gave? And there's also artefact at 15kHz which wasn't in the original audio. Visible with Spek program. Download FLAC and Opus both files, new link: http://www.filedropper.com/example_3 FLAC full: 1084 kbps; FLAC solo: 465 kbps. with --bitrate 160: Opus full: 158 kbps; Opus solo: 190 kbps. Included also Spek

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

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

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

> Why are you using a stereo file > containing the same sweep in both channels > and then downmixing to mono? When I first tried encoding at a higher bitrate, I needed to test the different behavior of the "mid" (l+r) and "side" (l-r) channels. That's why I made the first sweep identical on both the left and the right channel (i.e. "side" is silent)

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/

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, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

Hey guys. I think Vorbis is pretty cool, but since the current OggEnc only offers 44.1kHz, it limits what I wanted to use it for. So I've been using Lame to get 16kHz mono Vorbis files. I'm curious about whether Lame does Vorbis encoding the "right" way for non-44.1k stuff, or whether it just encodes as it would for 44.1k & changes the sample rate on the output, but I'm

Check this clip (it's small, 373kb) http://www.geocities.com/jdxss/udialwav.zip It left oggenc, lame and MP+ encoders choking in dust. -- Vorbis Xtreme | http://solair.eunet.yu/~aldov/ Ogg Vorbis is the free, open source alternative to MP3 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a

On Aug 07 10:08:43, petrparizek2000 at yahoo.com wrote: > > What sine sweep exactly? > > An exponential sweep. It started slightly below 24 Hz and ended almost at 24 > kHz. And it was 50 seconds long. > > > How did you obtain it, > > I used Angelo Farina's "Aurora" modules. One of them is called "Generate > sine sweep". Can you please