similar to: Re-encoding ogg-files at lower bitrates

G'day, doing a spot of reencoding on my 64k ogg files, trying to create a stream for those poor sods who are still on dial up. The syntax below works - gives me a 28k stream. Seems like the bitrate has little effect on the reencoding. Anyone know all the commands I can use - still no documentation for this ver2 of ices and icecast hey? <p> <encode>

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

I'm trying to get a bitrate scaled live ogg stream working, but I've not found out how to get this running with the current tools. I've downloaded the CVS icecast2, and darkice, but the only option seems to be if the same stream is actually encoded multiple times at different bitrates. This is not good for my cpu. Can I get bitrate scaled ogg streams working with the current tools?

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

Hi! Inspired by the discussion about mixing Vorbis streams and so on I came to the question whether Vorbisstreams can be, just like MP3, reencoded without further loss. As I heard, and I am pretty sure that it's true, MP3 files can be decoded and reencoded unlimited times without further quality loss (of course the same encoder with the same preferences has to be used all the time). Is this

Hi: Actually, if the stream were to peal to the best quality dependent on a listener's bandwidth, the capability would have to be in icecast, surely. Geoff. <p> -- Geoff Shang <gshang@uq.net.au> ICQ number 43634701 Make sure your E-mail can be read by everyone! http://www.betips.net/etc/evilmail.html Please avoid sending me Word or PowerPoint attachments. See

Hi. I'm confused about one of the statements on http://www.vorbis.com/faq.psp#unique I have a server using Ices2 as the source. I provide it with Ogg's encoded at quality 5, and I get it to provide lower bitrate streams like so: ices.xml --snip-- <instance> --snip-- <encode> <samplerate>44100</samplerate>

Hi all, I recently started to study the Ogg Vorbis codec and found an interesting feature called bitrate peeling [1]. Do you know where I can find more information on this topic? E.g., how is the actual peeling being implemented, what are the performance results in terms of resulting quality, what are the requirements for bitrate peeling (can it be applied to any ogg vorbis stream), ...? Also,

Hello. The page http://www.vorbis.com/faq.psp (last faq) and http://cs.leander.isd.tenet.edu/~mitchell/vorbis_intro.html both state that it's possible to lower the bitrate of an existant ogg enc file without reencoding. Where does this feature exists? oggenc only likes wav files as input, how can I feed it an .ogg file and get a lower bitrate in a new one? -- Grzegorz Adam Hankiewicz

A couple pf questions regarding reencoding... I've got ogg, mp3, and aac files to stream. a) What are the restrictions in the current version of ices (0.4?), is it that channels and samplerate must be equal in all files? b) will they be reencoded to ogg or to mp3, or is there a choice? Thanks. -- Guy

Thomas was gracious enough to provide a second stream of his beats.dk vorbis broadcast. http://vorbis.arkena.com:8004/test.ogg is 128kbps http://vorbis.arkena.com:8004/test_64.ogg is 64kbps Now even us lowspeed dsl people can enjoy :) Everyone should give a listen to the 64kbps and see what they think of the quality (it's being reencoded from 128kbsp Ogg). jack. --- >8 ---- List

I'd like to use the Bitrate and Reencode features of ices and understand I must recompile with lamelib. However, it's Very Unclear where to begin here. This is noted in the ices-0.2.3 INSTALL file: I installed lame as a debian package so lame.h is in /usr/include/lame/lame.h and libmp3lame.a is in /usr/lib. configure finds everything automatically. <p>I assume we

I have some OGG files which would benefit from the following simple corrections: a) removing left channel (which is empty) and changing right channel to the mono channel (mono sound stereo-recorded by mistake) b) volume change Of course I can open the files in audacity and do those corrections but that would require re-encoding. And I am afraid that it will result in additional loss of the

On Aug 07 10:08:43, petrparizek2000 at yahoo.com wrote: > > What sine sweep exactly? > > An exponential sweep. It started slightly below 24 Hz and ended almost at 24 > kHz. And it was 50 seconds long. > > > How did you obtain it, > > I used Angelo Farina's "Aurora" modules. One of them is called "Generate > sine sweep". Can you please

Greetings to all, It's been rumored that subj is possible, even with no decoding/encoding the .ogg file, but simply throwing out some information from it. I just wonder: is it really possible, and if yes, if there some software capable of doing this? -- [dan@shinestar ~]$ ls -l /dev/brains ls: /dev/brains: No such file or directory --- >8 ---- List archives:

Hi, is there a good concept to lower the quality of a Theora stream without completely reencoding it? For example to make a lower bitrate stream out of a higher bitrate stream on the fly on some sort of proxy server. I am thinking about something like requantization. If you think requantization is also the best (or the only) option for Theora please let me know. Kind regards, Franz

Greetings Fellow Vorbis Gurus! How can I encode, using CDex, at a CBR (constant bit rate) rather than the VBR? Also, can one take a 128kbs MP3 file and re-encode it to ogg? If so, please explain the procedure. By the way, I'm using a Windows based system. Thank you! /jp/ dojpaul@coffey.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

I've written up a nice streaming jukebox so I can listen to my mp3's and ogg's from anywhere with a net connection. I even got it set up so you can re-encode mp3's on the fly by calling lame and using stdout as the output. I've found a program ( http://www.inf.ufpr.br/~rja00/ogg.html ) called ogg2ogg which basically is oggenc combined with oggdec, allowing you to

What is the best way to set up multiple streams, at multiple bitrates, from the same line-input audio source? I currently offer a 96 Kbps stream of the line-input of a D-Link DSB-R100 radio tuner. I'd like to offer a 32 Kbps stream as well, for 56K modem users. Should I run multiple copies of liveice on the machine with the soundcard? Or should I have ices reencode the 96 Kbps stream down