search for: 20khz

...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 frequency tone is being played). Here is what opusinfo outputs: Processing file "D:\Work\Ulrich\Musik\Vega\Opus\Audacity\Test-Sweeps\02 Sweep (0 -20kHz at 96kHz) log.opus"... New logical stream (#1, serial: 000028fd): type opus Encoded with libopus 1.3, libopusenc 0.2.1 User comments section follows... ENCODER=opusenc from opus-tools 0.2-3-gf5f571b ENCODER_OPTIONS=--bitrate 56 --vbr --comp 5 ALBUM=Test-Sweeps...

...misch sweep. Is that a fixed number of Hertz per second (SoX calls that linear)? Or a fixed number of semitones per second (SoX calls that exponantional)? (The ogg comment says 5s=10Hz, 10s=20Hz, 15s=39Hz, 20s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, 50s=5kHz, 55s=10kHz, 60s=20kHz so it seems the frequency rises logarithmically.) $ sox -c 1 -r 96k -b 16 -n /tmp/sweep.wav synth 60 sin create 1-20 gain -3 > With Opus I noticed that the file size for 48kHz and 48 kbps > compared to 96kHz Vorbis at 31kbps is about double the size Your opusenc line says "--bitrate 5...

...did (and also reduced the 24bit to 16bit). >> > >> >> Why using 96kHz in the original: AFAIK Vorbis and Opus both use >> >> frequency components to encode the file. >> > >> > What frequency components are those, >> > in a sweep from 0 to 20kHz? >> >> The frequency component at any time should be more or less exactly one >> (THE frequency). > > Exactly. So what frequencies above 48kHz are there > to be sampled at 96kHz? None. Do we want to discuss the Nyquist theorem? (AFAIR it's for pure sinus waves and...

...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 testing or adjusting streams for special purposes would be good the possibility of change the frequency both above 20Khz and below 20Khz (or other). I guess that the cut-off frequency is specified somewhere in the code, so should not be hard to add this optionvia command line. I hope that the developers take this into account in future versions. --- Mike

...misch sweep. Is that a fixed number of Hertz per second (SoX calls that linear)? Or a fixed number of semitones per second (SoX calls that exponantional)? (The ogg comment says 5s=10Hz, 10s=20Hz, 15s=39Hz, 20s=78Hz, 25s=156Hz, 30s=312Hz, 35s=625Hz, 40s=1.25kHz, 45s=2.5kHz, 50s=5kHz, 55s=10kHz, 60s=20kHz so it seems the frequency rises logarithmically.) $ sox -c 1 -r 96k -b 16 -n /tmp/sweep.wav synth 60 sin create 1-20 gain -3 > With Opus I noticed that the file size for 48kHz and 48 kbps > compared to 96kHz Vorbis at 31kbps is about double the size Your opusenc line says "--bitrate 5...

.... 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 rather than the usual 20Hz to 20KHz. Bending the code out of spec is acceptable in this particular application because it is normal for radio mic transmitters to be paired with matching receivers from the same manufacturer. I would be interested to hear the opinion of the developers to get an...

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,

> 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 just because the Opus format itself doesn't have "room&q...

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 frequencies are missing, and you can easily perceive that the file is a low bitrate file. By harmonically transposing frequenc...

...My current thought is to filter the input down to a bandwidth of 7KHz or 4KHz (traditional values for high and low quality speech), decimate the samples so that the sound is sampled at, say, 44/3=14.6KHz or 44/5=8.8KHz, then run it through the standard Vorbis encoder. Vorbis then sees an ordinary 20KHz bandwidth stream that sounds like a tape recording running at 3 to 5 times normal speed and encodes it as usual. I checked the mailing list archives, and found an old thread about low bit-rate encoding that quickly degenerated into a highly bogus discussion of the proper way to decimate the sample...

In x64 SMP Vista HVM guest (vcpus=2 in the configuration file), the wall clock time is 50% slower than that in the real world. The attached patch fixes the issue. -- Dexuan Signed-off-by: Dexuan Cui <dexuan.cui@intel.com> _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

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

...w 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? > SBR exploits a limitation of your ears. At high frequencies (like over 10kHz) you cannot determine pitch with any accuracy. You hear up to 15kHz to 20kHz (depending on age and other factors), but you really cannot identify pitch at these frequencies. You cannot even determine if content above about 10kHz is properly harmonically related to the lower pitched fundamentals which usually give rise to them. I don't know of any voice specific code...

...trparizek2000 at yahoo.com> > > Cc: opus at xiph.org > > Subject: [EXT] Re: [opus] Re: Opus Tools -- low bitrates, new features in 1.5, > > "expect-loss" > > > > [Windl, Ulrich] > [...] > > > > Also if you look at the samples for (e.g.) a 20kHz sine samples at 44kHz, > > > > the samples hardly resemble a sine wave very mch, and seen reversely: > > > > It's not obvious that it once was a pure sine wave. > > > > On the contrary, a sine wave of 20 kHz > > can be perfectly reconstructed from samp...

...ver going to be great, but I should be able to at least make it bearable. What's the prevailing opinion on using high and low pass filters? One would assume a phone handset is expected to provide frequency response in human speech zones, and not really much outside that (certainly not the 20hz-20khz one might expect of a CD). Suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons

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 settings? -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net <p&...

...e 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 trying to obtain 20khz audio frequencies at -q0. I could use a 3rd party filter but I want to encode on the fly and this makes it impossible. Ross. <p>--- >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 '...

...all I'd like to suggest having a test-suite with Vorbis files (tag tests, sound tests, etc) on the web-site for vendors to check compliance against. I see the issue with copyrighted music, so we need a good idea (or musicials willing to donate material). I once made a sinus sweep from 20 to 20kHz (for fun), and I'd like to contribute that if anyone likes (the file was recoded from WAV, but it would probably much more perfect if the sweep were done in Vorbis' frequency domain. (i.e. oggenc would guess the frequency from discrete samples, so it would be better to directly feed the...

...>quality with speech codecs as it does for music codecs. Do you know if there > >is any open source variant of SBR? > > > SBR exploits a limitation of your ears. At high frequencies (like over > 10kHz) you cannot determine pitch with any accuracy. You hear up to > 15kHz to 20kHz (depending on age and other factors), but you really > cannot identify pitch at these frequencies. You cannot even determine if > content above about 10kHz is properly harmonically related to the lower > pitched fundamentals which usually give rise to them. > > I don't know o...

? Do you have any references for me to investigate, I am trying to understand how noise is reduced by introducing higher sampling rates. (I tried to search, but maybe it is so obvious that nobody even explains it) This is not very obvious. It requires you to understand basic signal processing theory. I will give some pointers below. Any physical signal (e.g. audio coming out of speaker, current