similar to: shoutcast ogg

Hello, I am a student who wants to study the VP3 encoding/decoding process in detail. I downloaded the source code from your site. I was wondering if there is any supporting document which provides details of the entire process - a complete VP3 standards document to aid in the understanding of the code. Any information will be welcome. Thanks Regards, Navneet --- >8 ---- List archives:

Derf's from-scratch alternate implementation of the theora codec is now publicly available from our subversion repository. http://svn.xiph.org/experimental/derf/theora-exp/ This exciting new codebase promises superiour performance, and the encoder is capable of higher quality output than the current represenation, taking advantage of the bitstream extension we've added to the

Apologies if this is a bit of a commercial post (I've not been lurking long), but you might like to know that the world's most listened-to online radio station has just launched an Ogg stream. (As of today!) http://www.virginradio.co.uk/thestation/listen/ogg.html (or alternatively www.virginradio.co.uk/thestation/listen from a non MSIE+PC browser). Any ideas you have to help promote this

This afternoon I've transferred our Ogg Vorbis Icecast2 streams from darkice to the ices2 encoder. Apologies if you were trying to listen; I had to stop and start the streams a few times. They should be stable now as I'm off down the pub now! As a reminder, the URL for the streams is at http://www.virginradio.co.uk/thestation/listen/ogg.html. Ices2 takes slightly different parameters to

I've got my encoder working now... and i put a proeprty page interface to allow the user to set a few encoding paramters to start off with... but changing the paramters doesn't seem to make barely any difference... These are the defaults i'm using... mTheoraInfo.target_bitrate=400000; mTheoraInfo.quality=30; mTheoraInfo.dropframes_p=0; mTheoraInfo.quick_p=1;

On Tue, Sep 06, 2005 at 09:56:06PM -0500, Jay Krivanek wrote: > 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

Another posting in german, due to its subject, which has to do with the german translation of the Samba-3.0-documentation. --- Hallo, deutschsprachige Samba-user, die ?bersetzung der Samba-3-Dokumentation geht weiter. Die Info-Kampagne tr?gt Fr?chte, es gibt neue freiwillige Mitarbeiter! Unsere Projektseite liegt auf Berlios, eine kurze Projektbeschreibung findet man unter

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

Windows 98 is totally obsolete and useless. WindowsXP is a completely different architecture for just about everything except the name Windows. You owe it to yourself to check out WindowsXP, even though it is subsidized by the "eveil empire." :) -g. At 16:00 2005-09-07, Karl Heyes wrote: >On Wed, 2005-09-07 at 23:33, Ross Levis wrote: > > If you have access to a Windows box,

It's technically good at streaming for broadcasters because, if it encodes something simple, it drops the bitrate and therefore our costs. And it sounds great too, but you wanted technical reasons... ! By the way, Virgin Radio UK's broadband streams are now proper stereo, instead of the suspicious mono-sounding version you had for the past few weeks - so if you want your non-technical

Hi all, As per the instructions here <http://llvm.org/docs/DeveloperPolicy.html#obtaining-commit-access>, I contacted Chris Lattner to obtain commit access but haven't received a response (either positive or negative). What's the expected turnaround time for this (I contacted him five days ago)? In case he's currently unavailable, is there someone else I could contact? Thanks,

On Mon, Sep 26, 2005 at 05:30:11PM +0200, Michael Smith wrote: > > Assuming this is correct, the next question that arises is "Is Vorbis > > when used this way truly gapless?" > > Yes, it is. ... and I can confirm this. No gaps were perceivable decoding in ogg123. However, an interesting effect can be had if all the parts have the same serial number. You get a bump

On Fri, 13 Jun 2003 12:04:34 -0400, Brendan Cully wrote > You are trying to use ices 2 with an ices version 1 config file. > <p>ah ha! Is ices2 adventageous to use (other than ogg-vorbis) since I have it installed. I guess I am wondering if this is where most of the development focus will be placed and currently there is a phasing out of ices1 development. I am just needing to

On Mon, Sep 26, 2005 at 12:41:00PM +0200, Michael Smith wrote: > On 9/23/05, Paul Martin <pm@nowster.zetnet.co.uk> wrote: > > Mike... did you look into this? I posted a waveform from a file > > generated by ogg123 dumping to a wav (so that audio driver problems were > > eliminated). > > > > It looks to me like ices2 is restarting the vorbis encoder in a >

Mike... did you look into this? I posted a waveform from a file generated by ogg123 dumping to a wav (so that audio driver problems were eliminated). It looks to me like ices2 is restarting the vorbis encoder in a non-gapless way when it stops and starts the encoding (to change the stream id and update the metadata). Because I'm doing crossfades from one track to another, this shows up.

I'm using ices2's metadata facility to update the name of a track in a vorbis stream: <input> <module>alsa</module> <param name="rate">44100</param> <param name="channels">2</param> <param name="device">hw:1,0</param> <param name="metadata">1</param>

ogg123 seems to have a issue when streaming for long peroids of time. If I have ogg123 up connected to a streaming server for more than 3.5 hours it begins to stutter or add blocks of static. I've verifed this with 2 different soundcards in both alsa oss emulation and regular oss. It's 100% repeatable for me. It does not happen with xmms using libvorbis as the output plugin so it

+thread-sanitizer mailing list On Mon, Sep 7, 2015 at 6:41 PM, Vaivaswatha Nagaraj <vn at compilertree.com> wrote: > Hi, > > I am interested in understand the compiler-rt thread sanitizer tool and have > recently started experimenting with it. In particular, I'm interested in the > deadlock detector. > > I see that deadlock detection currently don't work. (I

On 8/16/05, Paul Martin <pm@nowster.zetnet.co.uk> wrote: > > Caught it yesterday, as I was already running level 4: This line: [2005-08-15 04:10:58] EROR stream/ices_instance_stream Send error: No error (Success) is the one that catches my attention. Unfortunately, ices's error logging is somewhat sub-standard. Can you cross reference this against your icecast error.log to see

Hi, I'm new here. I would like to write 2 simple utilities to maniputale OGG files: 1) oggjoin --output-file=result.ogg file1.ogg file2.ogg ... Merge file1, file2, ... into result.ogg file 2) oggpause --output-file=result.ogg --duration=0.5 Generate result.ogg with silence for 0.5 seconds oggpause --output-file=result.ogg --input-file=source.ogg Generate result.ogg with silence equal