similar to: PlusV algorithm

Hi, I haven't seen this mentioned here before. http://www.plusv.org/ "With traditional MP3, a typical Near CD Quality audio file has been encoded with a data rate of 128 kbits/s. While this is ok for people with big hard disks and fast Internet connections, this data speed has clearly been a bottleneck for people using modems or storing their music into 32 or 64 MB portable player

>> Do you (or the ogg vorbis community) understand how SBR works? There is >> info on http://www.codingtechnologies.de/technology/sbr.htm >> bnut I am guessing this infop is not enough to explain the technology. Robert Voigt: >I haven't heard about SBR before. After reading that webpage I can say the >following: [ . . . ] I don't think SBR will give an

Sort of off topic, but I was wondering about how the SBR data factors into an mp3pro bitrate. If you encode a "64kbps" mp3pro stream, I'm guessing that 64kbps would include the SBR data. So, when played in a normal MP3 player, would a 64kbps mp3pro stream actually sound worse than a 64kbps MP3 stream? Tony Arcieri --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg

Just a reminder to keep testing. The latest builds should be rock solid. The url is: http://i.cantcode.com:8888/stats.xml You can now get stats ouptut by pointing a browser to: http://i.cantcode.com:8888/stats.xml We're looking for more win32 freeamp reports, and any other clients besdies the standard ones (sonique, winamp, xmms, etc). Oh, and I stuck some silly station ids in between

Just a reminder to keep testing. The latest builds should be rock solid. The url is: http://i.cantcode.com:8888/stats.xml You can now get stats ouptut by pointing a browser to: http://i.cantcode.com:8888/stats.xml We're looking for more win32 freeamp reports, and any other clients besdies the standard ones (sonique, winamp, xmms, etc). Oh, and I stuck some silly station ids in between

But the source computer runs the other os, so I don't have too much choice as to what I can do and the timeout problems remain with mp3 as well. The persons involved don't want to use .ogg for now so that is also not an option. Is this an icecast bug, or some other problem? on Thursday 08/14/2003 Arc(arc@indymedia.org) wrote > On Thu, Aug 14, 2003 at 12:01:11PM -0400, John Covici

Hello, Are there any open source project that is similar to SBR from Coding Technologies that possibly could be used together with Ogg Vorbis? /Pontus <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 'vorbis-dev-request@xiph.org' containing only the word

Fredag, 31 januar 2003, skrev du: <p>>Next thing is, that I would like to clarify the difference between a codec's >latency (which I would die to know in exact figures for the vorbis codec one >day - as well as block sizes, but that's another story) and the (bandwidth) >"smoothing window" latency - call it buffering or whatever. One thing is the codec

Craig Dickson (crdic@yahoo.com) wrote : > Jack Moffitt wrote: > > > I agree that this might be the case. But what happens with the > > violation is by the author? :) > > > > Ie, in the case of PlusV releasing a patented codec under the GPL or in > > the case of LAME or FreeType's authors doing similar things. What > > state is that? > >

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same

Jan, I'm forwarding this message to the rest of the developers. Hopefully one of them will have a good answer to your question. --------- Does someone want to take this one? Just make sure you copy him on the answer, as he's not on the list. jack. ----- Forwarded message from Jan.Tangring@et.se ----- From: Jan.Tangring@et.se To: jack@icecast.org Subject: spectral band replication

fwiw, there are definitely applications that would use Vorbis but can't until there is a _strict_ CBR mode. Meaning, architecturally the datarate simply cannot exceed X no matter what happens to the quality. Send silence if you have to, but limit the datarate. -dbm -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef

I have been using libvorbis for the past few weeks and have been asked to summarise what I have discovered about the codec. There is an early draft of the document at http://www.geocities.com/gatewaystation/vorbis/vorbis.htm - Please forgive the dodgy formatting (it was formerly a MS word document that got converted with their 'save as html' feature). I still have some additions to

I was a bit confused by that statement as well Lee. I've done a bit of work with mp3PRO using a source developed from scratch and a version of the original Icecast modified only trivially to deal with specific metadata requirements. There is a very high level explanation of mp3PRO at the http://www.mp3prozone.com/basics.htm site. At the risk of being somewhat incomplete i will add: 1) The

If anyone is interested, mp3PRO demo is available on Coding Technologies web site: http://www.codingtechnologies.de/ Demo is a Win32 encoder/player, and it is limited to 64kbps only. I did a quick test, and it really sounds good at 64kbps (but not as good as 128kbps - we all know that was hype). SBR (Spectral Band Replication) seems to be working pretty well. Of course SBR is only used if you

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

Will oggenc have a lowpass switch? I would prefer to lowpass at 15-16khz at -q3 for use with FM broadcasting. The additional frequencies would be chopped off anyway by the transmiters hardware lowpass filter so the encoder could use the addition bits for other purposes. It could be enforced that the lowpass can only be reduced and not increased from the default. This would stop people

I mentioned many months back that I intend to switch to Ogg/Vorbis from MP3/LAME once the final version is out but I need to have an adjustable lowpass filter option similar to LAME's. If this isn't done yet, could you please add it to your list for OggEnc. Specifically I need to cut all frequencies above 17khz. Thanks, Ross. --- >8 ---- List archives: http://www.xiph.org/archives/

Reading Monty's message, I better re-post this one :-). In another thread the following was said: > It might help if you also mentioned that the sound quality of Ogg is > significantly better than MP3 (assuming that you use regular MP3, not > MP3Pro), especially at 32kbps. Speaking of which, would there be anything to stop Vorbis, maybe not v1 but in the future, to use similar