similar to: Realtime encoding at 44.1khz

There is a big, big difference between cbr and vbr, btw. The machine in question is a Pentium 3 700 oc'ed to 800, 512MB ram, win nt 5.1 (2k and xp being nt) etc In an earlier version the oddcast dsp ate up so much cpu time when set at say 64kbps (for examples sake) 44.1kHz, stereo that the machine became totally unusable and the winamp task had to be (slowly) ended with task manager. Now,

Mark Casey wrote: > I think when the real-time cbr encoding issues are sorted > Vorbis (this is my experience with oddcast dsp, winamp and > icecast 2 win32, big cpu hog) will start gaining more ground, I can't use the managed bitrate option in OddCast on my K6II-500 CPU! (Overclocked to 525, WinXP Pro,192MB RAM). I would like to. Normal VBR uses about 90% CPU. This seems too

80kbps test stream using oddcast DSP and new test win32 port of icecast2 server.... http://www.djlithium.com:8064/djlithium.ogg oddsock --- >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-request@xiph.org' containing only the word 'unsubscribe' in the body. No

hello, I got icecast2 and ices2 going on an old Pen 100 box. It's streaming a lecture given by Richard Stallman at the University of Aberystwyth in June 2002. The files are found on http://www.6809.org.uk/audio/ I had put the sample rate down to 24000, as the Pen 100 seem not to be able to handle anything about that. According ps aux, ices is taking 91 percent cpu. The box is working just

hi, there is a low bitratestream available for testing. it's around 40kbps and it sounds really bad, because it's 11khz stereo :). http://ogg.radiostudio.org:8007/ostream.ogg -- Sent through GMX FreeMail - http://www.gmx.net --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

hi, there is a low bitratestream available for testing. it's around 40kbps and it sounds really bad, because it's 11khz stereo :). http://ogg.radiostudio.org:8007/ostream.ogg -- Sent through GMX FreeMail - http://www.gmx.net --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

Hi: I am a new user of icecast, and I was wondering, wich source software, gives the option of compressing to low bitrates an ogg stream (lower than 48 kbps). Thanks: CC

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Have you tried to send mp3 files with other formats than 44khz stereo? I'm not talking about the encoding quality for send the data to icecast, else about the quality of mp3 files. If you do that, the listeners will hear the music faster and higher, like the little chipmunks in donald duck's, and it will be broken by no-sound intervals. It happens if the files are mono too. This is

On Tuesday 15 July 2003 16:40, Arc wrote: > On Tue, Jul 15, 2003 at 11:06:03AM +1200, Ross Levis wrote: > > Mark Casey wrote: > > > I think when the real-time cbr encoding issues are sorted > > > Vorbis (this is my experience with oddcast dsp, winamp and > > > icecast 2 win32, big cpu hog) will start gaining more ground, > > > > I can't use the

Ok. Now it will run. But when i try to play back my stream(either ogg/vorbis or mp3) it sounds like the chipmunks. Could it be that I'm broadcasting at a sample rate of 22050 in mono? This shouldn't make a difference as the source is also 22050 samplerate and mono(its the line-in plugin for winamp). I'm using Winamp 2.81 with the is the Oddcast DSP for Winamp 2.xx plugin to

Hi. Can anyone tell me the best way to stream Ogg Vorbis to an Icecast2 or compatible server under Windows? I would need something that doesn't rely heavily on graphics, as I have a blind friend who wishes to be able to do this. Thanks, Lorenzo <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

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

Hi, Can anyone point me at a url where i can learn to: 1. display the name of the ogg stream in the winamp window (is this to do with the oddcast config or with the .pls file?) 2. show a url automatically in the winamp minibrowser when the ogg stream plays <p>many thanks adam <p><p><p>Adam Hyde XS4ALL Internet B.V. mobile : + (31) 06 2903 6248 Eekholt 42 1112 XH

hello, I try to listen to the stream at http://ogg.radiostudio.org:8007/ostream.ogg with ogg123, it's a 44100/mono stream. It sounds to me like a mono file played as a stereo file. there are very often silent drop outs and the music is pitched. is this a bug in ogg123? I changed to stereo encoding and know it's better (normal pitch without silence), but I have klicks/scratches in

OK, I've organized this. It's not possible to average 32kb/s stereo with Vorbis using VBR. Quality -2 stereo proeduces roughly 32 to 38kb/s which is too high for many modems. I would have to use ABR, but last I tried that, it utilized too much CPU for my 950Mhz PC, and oddcast doesn't appear to support that option anymore. I've tried AACPlus 32kb/s mono and there is very

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

As the subject implies, my question is: is it possible to use the -b (or -M) flag at non-44K sample rates? I'm working with an application that is trying to optimize for very small audio filesize. I found that downsampling to 11K and then using q0 gives high compression, but won't seem to drop below 64kbps or so. It seems like the combination of downsampling, then reducing to 30kps

I should have provided a link to my low bitrate vorbis stream. It's a mostly easy listening world music format: http://soulfm.stationplaylist.com/listen.m3u Regards, Ross.