similar to: And the winner is... (C't audio-codec test)

From: Christian Borntraeger <borntraeger at de.ibm.com> This patch implements kvm guest kernel support for paravirtualized devices and contains two parts: o a basic virtio stub using virtio_ring and external interrupts and hypercalls o full hypercall implementation in kvm_para.h Currently we dont have PCI on s390. Making virtio_pci usable for s390 seems more complicated that providing an

From: Christian Borntraeger <borntraeger at de.ibm.com> This patch implements kvm guest kernel support for paravirtualized devices and contains two parts: o a basic virtio stub using virtio_ring and external interrupts and hypercalls o full hypercall implementation in kvm_para.h Currently we dont have PCI on s390. Making virtio_pci usable for s390 seems more complicated that providing an

Salve, till the 28. August midnight the magazin c't has a codec test between AAC, MP3, WMA and Vorbis Ogg online. http://www.heise.de/ct/leserbefragung/hoertest/ <p>Short guide for non German speakers: ---------------------------------------------------------------------------- stepp 1 browse to: http://www.heise.de/ct/leserbefragung/hoertest/ Down on that page is a link with the name

Hi there, I'm new to this list, and in fact rather new to the Ogg/Vorbis codec altogether. I recently downloaded the test files from the test conducted by the German computer magazine c't (this was mentioned by someone here recently). The idea was that you were given seven WAV files each for 64kbps and 128kbps, and had to rate them. The files were randomly chosen for each participant.

Hello developers, I recently participated in a public listening test that was held by German magazine c't. They had two categories, 64 kbps and 128 kbps. I only participated in the 64 kbps test where there were 7 .wav files, with 1 of them the original one and the other 6 being encoded with different encoders. Without knowing which file was encoded with which encoder you had to give each

On Mon, Sep 30, 2002 at 10:48:27AM -0100, fungus wrote: > One the one hand this is a great achievement but on the > other hand normal people aren't going to figure out how > good it is because they'll never push the limits. > Who's going to set "quality 0.5" when they've got a > range from zero up to ten. This is very true. I can easily hear MP3

The last few threads have really been interesting. Technologies that are currently on the market are MP3, RealAudio, and Windows Media (pre-version8). Technologies currently in progress or just shipped are Ogg Vorbis and mp3pro. I don't think anyone has mentioned TwinVQ on this list. While exploring alternatives to MP3 (which is when I also found Vorbis), I noticed TwinVQ. I encoded

I apologize with what may be some very basics questions. I''m trying to go through sections 9 and 15 of the LARTC FAQ, but my head can''t quite grasp what''s going on. I''ll keep going through them to see if something clicks, but meanwhile I was hoping someone might point me in the direction that I should focus my efforts. The server at our church is running

Hello, how good is the quality with very low bitrates like 16kbps mono. 16 kbps files compressed with lame sound awful, realaudio is much better. Will vorbis sound as good as realaudio with low bitrates? How fast is vorbis (in comparison to lame). Smoerk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

Hello. I've made some progress on the issue. I found a tool called TrPlayer which is a text mode front end for real player - initially developed for the use of the visually impaired. The theory is that I can use this and pipe the live stream into vsound which then in turn is passed into ices or another source client. Trouble is, I'm having all sorts of trouble compiling Trplayer on the

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

Hello all, I am a little new to Linux so I would appreciate some assistance with this issue. I have been running a Helix server streaming RealAudio encoded content live and on-demand for a while. For the live stream, I use the Real equivalent to Ices which on the Helix platform is called SLTA. All the content on the server is encoded as RealAudio. I have also successfully setup an Icecast2

Shawn sent me a copy of the non-Direct Sound version of this output plugin, and I'm hosting it on my web site's rare software page: http://members.lycos.co.uk/bhafool1/rarities/ I've found it adds far more smoothness and clarity to various low sample-rate audio files, including an 80-minute speech I mentioned previously, due to my soundcard's approach to D-to-A conversion

At SxSW I ran into the Xiph gang and someone said something in passing that the new version of Ogg Vorbis would have much-improved low-bitrate quality. Wondering - true? Details? ETA? At CD Baby all lo-fi soundclips are still in the SFERA format*. At 20k bitrates, they sound GREAT which is why I stick with it, but I'd much rather switch them ALL over to OGG if there was a way to make

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

My previous message (http://mailman.ds9a.nl/pipermail/lartc/2004q2/012285.html) didn''t generate any replies, so I thought I''d try posting what I''ve done trying to solve my own problem, in hopes that someone will take pity and show me if/where I''m going wrong. Just to recap, I''m trying to accomplish this on my server, which streams RealAudio files

I currently utilize Icecast on my website, but I am looking for the same kind of functionality as I get with realaudio server when I stream static content - namely, the ability to seek, show the time remaining, etc, especially with Vorbis audio. I realize the intent of Icecast is to serve mainly live, streaming content (or at least radio-style content), but I utilize it mostly for streaming

I'm trying to convice a webmaster to start offering Vorbis in addition to his current RealAudio 3(GAH!) web streams. I was just wondering what the status on the RealOne/Helix DNA plugin was so I could include it in a letter I'm going to be sending him today. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

In the notes for 2.3.0 it says that IceCast can function as a RealAudio/Helix server. Does this mean that it does the quality negotiation as well, ratcheting the bitrate up and down if it needs to? Or am I reading this all wrong? Thanks. -- //// Jason M. SULLIVAN jsullivan@nc.rr.com |c-oo http://www.jason0x21.org \_- "That's not music,

Michael Smith wrote: >This should be possible to implement in icecast, but it's not currently >supported. > >I think it'd require: >Sending correct content-length headers for static content (I think we already >do this? It's easy, anyway) >Supporting the Range (or is it Content-Range? I forget) header. This is fairly >complex, but not really that hard. >