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Is someone working on the infamous high frequency boost effect ? S.Gesemann published an interesting paper three months ago: http://www.xiph.org/archives/vorbis-dev/200403/att-0005/01-sq.pdf <p> -- Giuliano. --- >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 ...

...vol.ASSP-34,No.5,October 1986 <p>If you know the reason why,please teach me. Regards, Kato ------------------------------------------------------------------------------------------- >Vorbis Development List, November 2002 >[[vorbis-dev] MDCT-Window still wrong] >From: Sebastian Gesemann (sgeseman@uni-paderborn.de) >Date: Wed 13 Nov 2002 - 14:56:39 EST > >In vorbis-spec-intro.html, the MDCT-Window is still wrong. > >Should be sin(1/2*pi*sin^2((x+0.5)/n*pi)) >for 0 <= x < n > >bye, >Sebastian <p>--- >8 ---- List archives: http://www.xiph....

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

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

Hi, there ! I'm pretty sure that the following text (and attached picture) explains the reason for the gain of high frequencies. when quantizing a vector scalar by scalar by simply rounding each scalar to the nearest level, the quantization-error-vector and the original signal-vector can be assumed to be orthogonal (average case) This is a problem when we want to preserve the energy level

I'm looking to trim down the size of the buffers i'm using... and at the moment, i've set each "media sample" (ie packet) to the size of the maximum reasonably possible for any codec (ie video)... about half a meg each... and there's quite a few of them... now i'd like to set the buffer size based on the codec, as obviously the audio codecs need much less space

Hi ! <p>I previously posted a mail in the vorbis-dev mailing list because I'm trying to implement a vorbis decoder in java from scratch in my sparetime and have some questions about the bitstream spec. As I did not get any replies, I got kinda dissapointed and demotivated - But now.... I'm trying to take my chances here. :-) I'm NOT a newbie to source coding (I successfully

Dropping the other topic as requested, I'd like to follow something up in a new thread. I want to throw a question or two out there and see what responses people give (if any ;-). Assuming that you think Vorbis could have developed faster than it has done in the past year, what in your opinion has been holding it back and what remedies do you suggest? Speaking for myself, I think that it

No offense, Segher, but the output quality of this thing is awful. =) I'll disregard the fact that, at least with *my* compiler, the source tarball I downloaded reduces every packet to zero bytes, which isn't terribly interesting. I decided to set the byte reduction to something constant: I started by dividing each packet's size by 2 just to see what would happen. The resulting ogg

It obviously would be nice to have such a mode available, for e.g. DVD audio compression. Apparently, the list doesn''t tell me too much about it. My questions are: 1. What is the current status of the 5.1 channel coupling in Vorbis? 2. If I''ll be interested in participation in its development, what is the recommended reading? -------------- next part -------------- An HTML

>Yesterday I have finished writing the ambisonic pan filter for oggenc. May I ask what this "pan filter" is? I made some tentative suggestions for coupling Ambisonic B-format in a post "Vorbis Ambisonic coupling" on 4feb07 I gather from the last monthly meeting, that some of you, including Monty, had problems with the phase behaviour of B-format. Would anyone like a

Hello! I just noticed how the LPC prediction is quantized by browsing the source code of the java port jFlac: : int sum = 0; for (int j = 0; j < order; j++) sum += qlpCoeff[j] * data[startAt+i-j-1]; data[startAt + i] = residual[i] + (sum >> lpQuantization); : I checked the C version (libFLAC) and it matches. However, it was a surprise because this way of quantizing the

Hi there ! I'm kinda "yet another guy who tries to implement a vorbis decoder from scratch only by using the vorbis spec". (pure Java, OO-style) (I downloaded the long-awaited-all-in-one-PDF) The data structures are quite ready, I can successfully load the codec setup packet and all its components it's containing. <p><p>So far so good.... Here's my first note

128 kbps! In-Reply-To: <E77B14D5-B72A-11D8-91A8-000A95A4DC02@kernel.crashing.org> Message-ID: <Pine.LNX.4.44.0406081019270.6447-100000@gorlois.cs.upb.de> On Sat, 5 Jun 2004, Segher Boessenkool wrote: > There are quite a few models of how the ear works. All the good > ones are computationally expensive, and not usable at all > mathematically. Your paper uses just one simple