similar to: Possible bug in Opus 1.3 (opus-tools-0.2-opus-1.3)?

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

(Please wrap your lines.) On Oct 26 01:38:34, Ulrich.Windl at rz.uni-regensburg.de wrote: > 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. Can you please post the original wav? I am not sure what Audacity means by a logarithmisch sweep. Is that a fixed number of Hertz per

>>> Jan Stary <hans at stare.cz> schrieb am 05.11.2018 um 11:05 in Nachricht <20181105100534.GB44329 at www.stare.cz>: > (Are we off‑list now by intention?) No, just fooled by the list defaults (some need just reply, others need reply to all) > >> Did you also try to listen at the beginning, shortly before the real tone > appears in the audible spectrum?

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

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

Hi people, looking around for a new audiocard, my eye fell on the M-audio audiophile 2496. It has 4 digital in/out and is 24bit, 96kHz. The sound quality is very good, if I can believe the reviews. <p>My question is: can vorbis do 24bit, 96kHz ? -- --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

Hi I used R for the first time yesterday. I wanted to plot the aliasing effect of sampling a 5.5KHz sinusoid at only 8KHz (below the Nyquist limit). So I wrote a small R script that a) plots 1msec worth of a 5.5KHz sin wave b) plots 1msec of the resulting 2.5KHz alias and c) plots the 8 sampling points on the 5.5KHz source wave. I think I have found a bug. The script is as follows:

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

Hi, Live audio (especially music) applications usually need a total latency of less than 5ms. Correct me if I am wrong but the minimum latency of the Opus codec is 2.5ms (encoder) + 2.5ms (decoder) = 5ms? Is there any way to go down to, say 1.25ms frame size? One characteristic of wireless radio mic links, if that helps, is that they usually only need a frequency response from 50Hz to 20KHz

I finally got RC3 built on my home machine (had to sneakernet in a tarball of libcurl to get it to build :-( ) and started testing it out on the range of quality levels, side by side with lame using it's range of vbr quailty modes. Overall, it sounds pretty good, but I haven't really tested a good range yet, as I ended up starting on udial.wav... vorbis handles it ok up to quality 3,

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 just came across the following page: http://r3mix.net In the "News" section there is a fairly negative critique of Vorbis; especially it is mentioned that Vorbis "has terrible masking problems". There is also a Vorbis-encoded frequency sweep which shows strong deficiencies at high frequencies, but I suppose this is due to the quite low bitrate of the distributed

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

An idea would be like for WMA 9 lower bitrates (32-42-48Kbps) to use a 'crystallizer'; which is basically a harmonics booster focussed at transposing sharp tones some octaves higher. Eg: A file has been recorded @ 20khz computer (or 10khz real life) to preserve space. While playing back the file sounds a bit mushy, almost as if someone was speaking through a cardboard wall. The higher

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

Submitted for perusal, comment, improvements, and/or critique. The presentation is in 3 sections: motivation, code, and comment. Motivation: As a new-comer to R from matrix oriented Gauss and Mata, I miss the tools for using a vector (and operator) to ‘sweep’ across a matrix. Here is how these work. If M is I rows by J columns, then one entry corresponding to

This patch series includes a series of bugs related to p2m, ept, and PoD code which were found as part of our XenServer product testing. Each of these fixes actual bugs, and the 3.4-based version of the patch has been tested thoroughly. (There may be bugs in porting the patches, but most of them are simple enough as to make it unlikely.) Each patch is conceptually independent, so they can each

> 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

Hi, I'm trying to get ffmpeg (linked with libvorbis 1.2.3) to generate a 96kHz ogg file, but I seem to be limited to 50kHz . The command is ffmpeg -i t16bit96kHz.wav -acodec libvorbis test.ogg and I get the error message [libvorbis @ 0x2051460]oggvorbis_encode_init: init_encoder failedError while opening codec for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc