search for: 15khz

I cant make the nouveau work with an arcade monitor, problably because the EDID. I can make modeline work fine to reduce the horizontal to 15khz, tested in a normal pc monitor (with ugly visual because the 15khz in horizontal). But when i try in the arcade monitor, dont show any image after xorg start. In official nvidia driver has an option NOEDID to force the driver to ignore information from monitor, so no matter what type of monitor is...

I brought this up 12 months ago or more. I believe the low-pass filter defaults too high with quality -1 in Vorbis 1.0. Is this going to change in the future? I think it should at least cut off at 15khz not 16khz and perhaps even 14khz. I believe it's important when streaming. Removing some of the higher frequencies compared to removing more of the audible sound makes more sense to me. Sampling at 33075 is an option but this produces a cut-off at 13khz which is too low in my opinion, and 32...

Hi, I am developing a voip phone that uses speex v1.2b1 narrowband mode. Libspeex decodes the 21-byte packets I recieve into 160 bytes and I can play them fine, but when I encode, the result is 38-bytes long. I have read that 38 bytes means that it is encoding at 15KHz, but I have set it to nb-mode and when stepping through the libspeex encoding sourcecode, I see that st->sampling_rate is 8000. Any idea why I speex-compress 160 bytes to 38 and not 21? Thanks in advance. y -------------- next part -------------- An HTML attachment was scrubbed... URL: http:/...

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 spectrogram PNG file for O...

...> Yea, don't get me wrong, ogg is definitely an awesome format but aacplus v2 > is currently the leader in high fidelity quality at low bitrates. 24kbps > aacplus even kicks ass in my view. Is that 24kbps AACPLUS a full 44.1kHz stereo signal with frequencies preserved up to at least 15kHz? (Just curious, not criticising.) As a datapoint, Vorbis aotuv b4 gives acceptable quality with q=-1.001 44.1kHz stereo at approximately 44kbps. -- Paul Martin <pm@zetnet.net> (work) <pm@nowster.zetnet.co.uk> (home)

...oping a voip phone that uses speex v1.2b1 narrowband mode. > > Libspeex decodes the 21-byte packets I recieve into 160 bytes and I can > > play > > them fine, but when I encode, the result is 38-bytes long. > > > > I have read that 38 bytes means that it is encoding at 15KHz, but I have > > set > > it to nb-mode and when stepping through the libspeex encoding > sourcecode, I > > see that st->sampling_rate is 8000. > > > > Any idea why I speex-compress 160 bytes to 38 and not 21? Thanks in > > advance. > -------------- next p...

...up to 21khz -- almost CD quality. In my opinion -q1 (80kb/s) is producing results similar or better than mp3@128. Why not reduce the nominal bitrate down a couple of steps. Of course some additional tuning will be required. q 0.0 = 48kb/s (13khz lowpass) q 1.0 = 56 (14khz) q 2.0 = 64 (15khz) q 2.5 = 72 q 3.0 = 80 q 4.0 = 96 (default) q 5.0 = 112 q 6.0 = 128 q 7.0 = 160 q 8.0 = 192 q 9.0 = 256 q 10 = 320 This is still assuming 44.1khz sampling. I presume 22khz has it's own table of quality/nominal bitrates. Just thinking out loud. Good work Monty. Ross Levis. http://winvorbi...

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: vorbis@xiph.org > Subject: Re...

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

Hi all ! If someone could give me a hint on how to proceed with the following i'd be very happy: I have a test setup on an nrf52832 (Cortex M4) in which I receive audio from a PDM microphone (64 sample frame) and pass it directly to an I2S device i.e. from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The

...#39;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? > 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 spec...

...> Yea, don't get me wrong, ogg is definitely an awesome format but aacplus v2 > is currently the leader in high fidelity quality at low bitrates. 24kbps > aacplus even kicks ass in my view. Is that 24kbps AACPLUS a full 44.1kHz stereo signal with frequencies preserved up to at least 15kHz? (Just curious, not criticising.) As a datapoint, Vorbis aotuv b4 gives acceptable quality with q=-1.001 44.1kHz stereo at approximately 44kbps. -- Paul Martin <pm@zetnet.net> (work) <pm@nowster.zetnet.co.uk> (home) _______________________________________________ Icecast mailing l...

...: > Opus full: 158 kbps; > Opus solo: 190 kbps. The two recordings are totaly unrelated. The fact that it's the same song has nothing to do with it. Why would a recording of a solo piano necessarily use a smaller bitrate than a recording of a band? > And there's also artefact at 15kHz > which wasn't in the original audio. > Visible with Spek program. > Included also Spek spectrogram PNG file > for Opus solo to see the artefacts. It is already present in the FLAC file (see attached spectrogram, obtained with SoX). -------------- next part -------------- A non-te...

..., even though the full mix has less entropy if you retain every bit. On Sat, Nov 17, 2018 at 5:37 AM ongaku zettai <sergeinakamoto at gmail.com> wrote: > 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 k...

http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio _codec.html?tk=rss_news any thoughts? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ood > >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'...

...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 sampling) to 15khz (at 44khz sampling -q0). But then I suppose there is 33khz sampling but I don't think too many devices support it. Something to think about: One global quality scale from 0 to 10. 0 = resample to 11khz, 8kb/s, through to 10 = original sample rate, high kb/s. Ross. <p>--- >8 ---...

I'm am currently using autuvb4 at q-2 mono 44100. This produces roughly 28 to 34kb/s. It's ok but I've heard AACPlus at 32kb/s stereo and it's definately better, unfortunately. Regards, Ross. >> I'm hoping that Monty and others can improve Vorbis to start >> competing >> with AAC+ at low bitrates. Monty said he had some ideas but I wonder >> if

...e surprised if that's where > those are coming from. If your encoding path can't handle full 10 > complexity, dial it down to 9, 8, etc until it runs ok. > > opus_custom_decode doesn't downsample from 48kHz, so your playback code > must still be treating it as if it's 15kHz, thus the pitch down. If you can > swing it, play as 48kHz, otherwise you have to insert another CPU-hogging > resampler. If you do need it and don't have a resampler handy, Opus has > silk_resampler_init & silk_resampler available for you. > > Emily Bowman > > > &gt...

Here is a patch for icecast 2.3.2 that adds support for listening to mp3 and aac+ streams on the iphone. A quick background on the technical aspects of the patch and why it is needed : With the advent of 3g on the iphone, lots of people have been jumping on the bandwagon of providing internet radio streams that work on the iPhone. The biggest problem is that (without having to install an