similar to: joint stereo implementation

Hello, I've been postponing some of my encoding for when joint stereo gets implemented. The reason I've been doing this, is that I am under the impression this is the largest step in the quality/bitrate ratio that's left. Now I'm wondering if I am correct in thinking this. lame's documentation seems to imply it doesn't make much of a difference at higher than 128kbps (I

Hi, I've just read this on Vorbis Xtreme site: "11. YOU SAY THAT OGG VORBIS IS PATENT-FREE, BUT I SAW A PATENT NAMED 'JOINT-STEREO' ON FRAUNHOFER'S PATENT LIST? SO OGG VORBIS ACTUALLY ISN'T PATENT-FREE SINCE IT ALSO USES JOINT-STEREO? No. You can't judge on a patent just by looking at its name - what's 'inside' is what matters. So if the name of the

Hi everybody: Here is this week's Ogg Traffic with all new updates on recent activities in the world of Xiph.org. The HTML version is up at http://www.vorbis.com/ot/20030415.html. Enjoy! -Carsten <p>Ogg Traffic for Tuesday, April 15, 2003 [1]Carsten "Purple" Haese April 15, 2003 _________________________________________________________________ Table of

Hi! What are the assumed listening conditions in vorbis ? For example there are big differences in perceived sound when listening with headphones compared to listening with speakers. ( "super stereo effect" , different frequency response due to HRTF etc ... ). Is this stuff worth considering in a vorbis class encoder ? David Balazic --- >8 ---- List archives:

Hi! >From http://www.vorbis.com/stereo.psp : "Oggenc's default choice varies by bitrate and each mode is selectable by the user" So, how am I supposed to do that? E.g. what do I have to do when I want an 128kbps .OGG using lossless stereo, or does this "selectable by the user" mean only that I can choose the stereo mode indirectly by choosing an appropriate bitrate?

Hi! I'm sending this message to both lame and vorbis developers, since my concerns apply to both ( and in case of vorbis it probably applies to the more advanced ambisonic modes the X, Y and Z parts ) Recently I did some thinking about M/S encoding and wondered if the same psychoaccoustic model is applied to the S ( and M ) channel as to normal L and R channels. My concern is that different

Hello all, I created today a fake stereo piece of music (the left and right channels were completely equal), and encoded it with the all the modes the RC2 encoder supports, and found that: * * dual * waste * waste * * mode * joint * stereo * in * in % * * * stereo * /2 * kbits * of JS *

Hi, If I want to encode two audio channels that are not related, audio from two different sources not stereo, is it best to create two instances of a single channel encoder or use one encoder with two channels? thank you, Paul.

Hi! Is there any Java Vorbis player that works on MIDP 2.0 platforms (mobile phones) ? I know the pure Java players, but I'm looking for a mobile version. Regards, David --------------------------------------------------------------------- http://noepatents.eu.org/ Innovation, not litigation ! --- David Balazic mailto:david.balazic@hermes.si HERMES Softlab

I am resending this ( for the third time, mispelled vorbis-dev-@xiph.org ), because the last time it didn't make it ( it didn't appear on the web archive at www.xiph.org , the archive of vorbis-list for the current month is broken BTW ) -------- Original Message -------- Subject: Re: [vorbis-dev] WAVE loader for oggenc and oggdrop Date: Wed, 01 Aug 2001 13:44:47 +0200 From: David Balazic

At 4:37 PM -0400 7/6/01, Brendan Cully wrote: >On Friday, 06 July 2001 at 13:21, tom erbe wrote: >> At 1:48 PM -0400 7/6/01, Brendan Cully wrote: >> > > 0.2 doesn't segfault, but it seems to overrun the server (if that >> >> makes any sense), it plays a 3 minute file in seconds. >> > >> >I'd like to know more about this. Are you

I've got an idea for joint multichannel coding. Here is my idea: My idea is a process for coding 2,3,4 or 5 combined channels, in a compatible way for decoders unable to deal with combined channels. part1: All channels to be coded are summed, each one effected by a coefficient related to its sound power importance for the listener. The sum of all channels is devided by the square root of the

On Wed, 6 Feb 2002, Brendan Cully wrote: > Is the sample rate changing? The decoder is currently very fussy about > having the audio format stay consistent. Yeah, that's what I thought it might be... most of the MP3s I'm playing are 160, but some are 128 and some are 192... I should convert them all to one bitrate, but I'm just too lazy. And I'm only doing this for fun,

CALL FOR PAPERS The 11th International Joint Conference on Computer Vision, Imaging and Computer Graphics Theory and Applications ? VISIGRAPP 2016 Website: http://www.visigrapp.org/ February 27 ? 29, 2016 Rome, Italy Regular Papers Paper Submission: September 17, 2015 Authors Notification: November 12, 2015 Camera Ready and Registration: November 27, 2015 Position Papers Paper

CALL FOR PAPERS The 11th International Joint Conference on Computer Vision, Imaging and Computer Graphics Theory and Applications ? VISIGRAPP 2016 Website: http://www.visigrapp.org/ February 27 ? 29, 2016 Rome, Italy Regular Papers Paper Submission: September 17, 2015 Authors Notification: November 12, 2015 Camera Ready and Registration: November 27, 2015 Position Papers Paper

Hi, I've read the Vorbis stereo documentation on square polar mapping and currently reading the source code to understand it. But there are some things which I don't quite understand and hope I can get some guidance on. I understand the decoding/decoupling part as it is the same as the one described in the stereo docs: From mapping0.c: /* channel coupling */

Hi guys, I am on a C55x stereo implementation of CELT 0.7.1; I am using 100 sample frame size and writing 58 Bytes per Packet. At the output of the decoder I see one channel reconstructed perfectly, but the other channel has random corruption that appears to be a sinusoid with a period of the frame size. The artifact is non-periodic and appears to happen randomly during the audio stream. Anyone

Hi All, Please find below an updated Vorbis-RTP Internet Draft document for review and discussion at the Xiph IRC meeting on Saturday. The changes in this version have been: Codebook caching mechanism Expanded SDP parameters Expanded MIME section Expanded introduction Packet loss section Minor tweaks and clarity changes to text There are probably some minor tweaks to the formatting needed

Thank you. Very helpful. > On Jan 27, 2017, at 12:40 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > > On 27/01/17 12:29 PM, Jon Lederman wrote: >> Thank you. Yes, we do need both channels independent. So, if we >> encode each channel separately, we will be sacrificing the >> compression ratio we would achieve with stereo encoding, correct? > > Not

Hi Jean-Marc, Thank you. Yes, we do need both channels independent. So, if we encode each channel separately, we will be sacrificing the compression ratio we would achieve with stereo encoding, correct? So, based on what you say here is my understanding. Please confirm this is correct or not: 1) If we use fec, we can reduce cross-talk but increasing bitrate. However, that should result in