similar to: ogg@48kb/s ~ mp3@96

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

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.

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

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

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

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 have found the following opinion on Vorbis' talk page at wikipedia. It might be something interesting to consider, I suppose. ~*~ "Quality settings run from -1 to 10 and are an arbitrary metric; files encoded at -q5, for example, should have the same quality of sound in all versions of the encoder, but newer versions should be able to achieve that quality with a lower

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

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:

The merge of the format changes and new psychoacoustics is complete. There's still tuning work left to do (most notably there's a somewhat higher-than-acceptable level of background low-midrange/upper bass noise; this is due mostly to incomplete psychoacoustic modelling data in the 100Hz-500Hz range) however, at this point, pending bugfixes, the Vorbis 1.0 stream format is frozen.

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:

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/

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

I just tried rc4-cvs and I can't believe how good it sounds! I can hear very little difference between the wav and the q0-ogg on my above-average stereo... It really makes MP3 look (sound) *pathetic* in comparision! I did some comparisions on a few songs, from noisy 77-punk to Jean-Michel Jarre to EBM/Industrial, and q0/~64kbit RC4-vorbis sounds as good or better than V9/~128kbit LAME-mp3

Hi, I'm no DSP or audio expert by any means, but I can share what works for me. People in the know, I would appreciate tips on whether this stuff is ok. You could sample at 32000Hz (or 48000Hz, any AC97 card will support this), run a 8000Hz lowpass filter over the data (16000Hz sample rate can only represent frequencies up to 8000Hz) and then drop every second (or 2 out of 3 for

The following behavior began about a day ago. NO changes have been made to the Fedora 20 (Heizenbug) system on which these things run. For no obvious or accountable reason, EZStream has stopped streaming. Here's a copy of the output from the ezstream command itself: # ezstream -c "/home/admin/ezstream/dn.xml" ezstream: Connected to http://localhost:8000/broadband ezstream: Streaming

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

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

On 20030610: Gregory Maxwell wrote: (in reply to Ralph Giles) >> I assume you're aware of the technical documentation on dolby's site? >> (http://www.dolby.com/pro/) In particular the surround mixing guide has >> a lot of detailed guidelines. I don't have any practical experience >> with it though, so I can't vouch for it. >Yes I am. I've done a

I will be out of the office starting 06/30/2005 and will not return until 07/06/2005. Please call Ken Milberg at 718-312-5070 or Steve Goldman at 718-312-5075 if you have a midrange issue. Otherwise, I will respond to your note upon my return. "WellChoice, Inc." made the following annotations on 06/30/05 08:02:11