similar to: winamp bitrate (kbps) display wrong

I've just spotted that the latest version of Winamp (v2.80) has support built in (not requiring the plugin to be installed) at last. At least according the the "new features" section. Regards, Tone. <p><p><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

> You can ask. A bug report with priority 'enhancement' is probably the > best way to do that. I'd consider it. Thanks for your help, Michael. I have entered this as an enhancement request. <p>Also, for the record, I originally made this comment: > I am actually trying to get OggEnc up and running nicely with > AudioGrabber, but so far I no joy I

Hi All I am having 2 very long character strings (550chars) and I want to put them as expressions together with c(). The problem is that I also get these double-quotes, as seen below in 'fct'. How can I remove these double-quotes? I tried as.name() but it did not work (because of size?). These are creating trouble with subsequent programs, which I tested with strings that for some

Hi, When I encode a file, with the command line (for example): oggenc h:\wavs\abc.wav -a sunshine as expected, it creates a file h:\wavs\abc.ogg. However, adding the '-t' command line option: oggenc h:\wavs\abc.wav -a sunshine -t MrSunny causes it to now create a file called MrSunny.ogg This behaviour could be expected if I had used the '-n' option, but I have not, so

Hola de nuevo, Esta es la información de mi sesion: R version 2.10.1 (2009-12-14) i386-pc-mingw32 locale: [1] LC_COLLATE=Spanish_Spain.1252 LC_CTYPE=Spanish_Spain.1252 [3] LC_MONETARY=Spanish_Spain.1252 LC_NUMERIC=C [5] LC_TIME=Spanish_Spain.1252 attached base packages: [1] stats graphics grDevices utils datasets methods base Lo que yo prentendo es

I am trying to run rTest example (XP, Eclipse, ) available with code. JRI etc are used from the rJava package (i.e. Haven't built myself) The code dosent runs properly - it shows ( i tried passing argument as --save). what else is possible. 'Creating Rengine (with arguments)' and then gets terminated. When I am trying to use run.bat >run.bat examples\rtest.java it says >

I am fitting models to the responses to a questionnaire that has seven yes/no questions (Item). For each combination of Subject and Item, the variable Response is coded as 0 or 1. I want to include random effects for both Subject and Item. While I understand that the datasets are fairly small, and there are a lot of invariant subjects, I do not understand something that is happening

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

I recently ripped a bunch of tracks to ~192 VBR OGG using OGGDROP XPd v. 1.1 running on Win98 SE and they worked fine and sounded great in Winamp 2.80. I then edited the tags (also with Winamp 2.80) and copied the files over to to my Mac G4 running OS 9.0.4, where they sat untouched for about a month. Now, having copied them back to my Win98 SE machine, they won't play in Winamp,

I want to limit pps (packets per second) not just bandwidth on the ingress side. I can do this using IP tables but I''m curious if there is a way to do this with TC. Thanks. Jon Flechsenhaar Boeing WNW Team Network Services (714)-762-1231 202-E7

Hi there ! I checked Vorbis' performace for mono files at approx. 64-80 kbps (it does a good job) and I'm wondering why the current OggEnc still uses Point-Stereo (>10kHz) for -q4 and -q5 I know, we're usually unable to percieve those phase correlations above 10 kHz, But a Dolby Prologic Decoder isn't. I think a future version of OggEnc which is able to use a user-selectable

I halved the volume of the sample before encoding with `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. So the artifact I heard was just clipping when encoding or decoding. Opus remains unbeaten for me at 133 kbps. That's totally awesome. Cheers On 10 April 2017 at 17:14, Agustín Dall'Alba <agustin at dallalba.com.ar> wrote: > Hello! I found a

Hi to everyone. I'm using speex in an application at the minimum bit rate (2.15 kbps), setting the encoder quality at 0, but obviously the quality is terrible. My question is: there are some signal processing that I can do to improve the quality of the signal at very low bit rate? Or can you tell me other codec that works well at that rate? Thank you.

I'm not sure, but I think that Microsoft have the license now.. anyone know if there is the source code to download? Or others codecs.. thank you very much ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Mon, 19 Jun 2006 10:15:59 -0700 >Do a search on "Voxware MetaVoice". I'm not sure who licenses

I am very happy to announce that the aoTuV tuning of Ogg Vorbis has tied with Musepack at first place in the 128 kbps listening test. It has beat iTunes AAC, Lame MP3, ATRAC3, and WMA standard. :) http://www.rjamorim.com/test/multiformat128/results.html -- ------------------------------------------------- Stephen So PhD Student Signal Processing Laboratory School of Microelectronic

Vorbis, using the AoTuV beta4.51 encoder, come first in a "Public, Multiformat Listening Test @ 128 kbps" organized on Hydrogenaudio forum. Hydrogenaudio thread: http://www.hydrogenaudio.org/forums/index.php?showtopic=40607 Presentation: http://www.maresweb.de/listening-tests/mf-128-1/ Results: http://www.maresweb.de/listening-tests/mf-128-1/results.htm

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > I halved the volume of the sample before encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding

Hello! I found a sample I can ABX successfully when encoded at 133.333 kbps. I was targetting 1 MB/min. https://drive.google.com/drive/folders/0B8KWShoIrA1kQzR1Z0FFRUlfcEU floex.wav is 4:54–5:04 of a lossless copy of 'Forget-me-not' by Floex, downloaded from http://store.floex.cz/album/zorya floex-133.opus was created with `opusenc --bitrate 133.333333 floex.wav floex-133.opus`,

Hello. The page http://www.vorbis.com/faq.psp (last faq) and http://cs.leander.isd.tenet.edu/~mitchell/vorbis_intro.html both state that it's possible to lower the bitrate of an existant ogg enc file without reencoding. Where does this feature exists? oggenc only likes wav files as input, how can I feed it an .ogg file and get a lower bitrate in a new one? -- Grzegorz Adam Hankiewicz

Hi: A friend asked me a question that I was unable to answer, so I'm posting it here. Can files created by the beta4 encoder be peeled down when that becomes available, or is the ability to encode peelable files also not implemented? My friend wants to know if they should switch over now or hold off until this becomes possible. Geoff. --- >8 ---- List archives: