similar to: HTML emails (was: New Member seeks advice.)

Sorry about that! I realized I had made the mistake of just changing the subject title minutes after I had posted. If I broke a second rule I am not sure what it was but if two out of three is all I managed to break, with almost no effort on my part I am sure I could bat 1000! I will check the Speex cite you recommended. I will likely have some questions. Thanks, Michael

Okay, my responsibilities at school have recently eased up a bit, and I might have some actual free time over the coming months. So I'd like to volunteer my services to write some needed documentation, tutorials, FAQs, etc. I am "fluent" in C and C++ and am not bad at Perl. I have been told I'm good at explaining things. I am anal-retentive about grammar, syntax, and

Okay, my responsibilities at school have recently eased up a bit, and I might have some actual free time over the coming months. So I'd like to volunteer my services to write some needed documentation, tutorials, FAQs, etc. I am "fluent" in C and C++ and am not bad at Perl. I have been told I'm good at explaining things. I am anal-retentive about grammar, syntax, and

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

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

I was wondering, is there a way to encode Oggs with EAC on-the-fly? I'm looking for a way to shorten the whole ripping and encoding process, but so far I've only seen support for the commandline encoder. -- GMX - Die Kommunikationsplattform im Internet. http://www.gmx.net <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

I would like to see the following information added to the FAQ at vorbis.com, appended to the "Audio Quality" section. "Uiver," "boa," and "jonI" from OPN's #Vorbis channel contributed and reviewed it. "What does the 'quality' setting mean?" Beginning with vorbislib 1.0-rc3, audio quality is no longer measured in kilobits per

Hello, I noticed that kbps jumps quite high in -q 10 mode and is nearing lossless compressors. Would it be a difficult task to make OGG a lossless compressor at -q 10? I don't know much about lossy/lossless compression internal logic, so excuse me if my question is out of place. But this would be a really nice feature that would make OGG an "all in one" tool. Cheers, Bostjan

I've been running through a collection of hard to encode samples that the lame developers use, and others, doing ABX testing at various quality levels looking for my personal sweet spot. Last time I did this for RC3 I found that most samples where pretty transparent around the 3.5-4 region (my ears aren't that well trained, and I don't want them to be :-), but this time I noticed

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello vorbis, I know that hipzip/Dadio support was planned - does anyone have any idea what the status of this is due to iomega's discontinuing of the Hipzip product line? Is there any way we can add ogg support to our hipzip's in the future if Interactive Objects adds it to Dadio ? - -- Best regards, Brendan

Anyone know of any vorbis players more or less the size of an MP3 player - something that will fit in a pocket? Or some projects to make one? I'd be willing to work on putting one together and port vorbis to some embedded processor. Any rough idea of what processor capabilities should be needed for the number crunching (is an FPU necessary, 16/32 bit, speed)? I realize the dev list might be a

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

Hi all ! I use cdparanaia for which I would like to get information. Does anyone know if there is a mailing list concerning this ripping application. In fact, I would like to know if cdparanoia can support scsi drive .... if someone can help me on this matter ... it is welcome. ;-) <p>Thanks for your help ! <p>Ce message contient des informations confidentielles ou appartenant

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

The Vorbis audio format, uses something called the Modified Discrete Cosine Transform (MDCT). I was interested in how this worked, but did not know where to start. Is this the name of a particular algorithm, or does it just mean something like a cosine transform, but not quite right? So I asked the Vorbis mailing list. Nobody answered. It was not a high priority, so I did other things for

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

SUMMARY I'm impressed. RC3 withstood anything I could throw at it. "-q 3" is really Vorbis' sweet spot, almost perfect, and "-q 0" is eminently usable, with only marginal defects for normal usage. Already sent money, will do that again. * * * I spent at least 40 hours in the last week testing RC3. I selected a few fragments from pop, classic and jazz CDs in my

On Jun 17, 2012, at 7:23 PM, Óscar Fuentes wrote: > OTOH it's my personal > opinion that you may be explaining too much: it is not necessary to > explain, nor even mention, the different available build tools on > Windows. As LLVM and Clang are developer tools, one can suppose that > the > user knows how to run a build. IMO, Mikael's efforts are on-target and very

This may be unrealistic and useless, but I've been thinking about an interesting file-sharing solution. In my system, only Ogg Vorbis files may be shared, and they should all be directly ripped from a CD. Nothing that someone would convert from MP3, etc. While that cannot be enforceable, it should be made clear. Also, track listings should be not filenames, but the Artist and Title of the

From: Peter Arremann <loony at loonybin.org> > Compiles aren't a great benchmark for a box since its 100% cpu and > neglects memory or disk performance but I had the numbers handy > for that :-) BTW, it is 100% ALU and a major strain on the ALU LOAD. In other words, it's not a good benchmark for even CPU. That's why the 3-issue ALU in the Nx586 on-ward blows the