search for: 11khz

Boink wrote: > They're streaming only in 22050/11khz mono. > I'm doing my stream around 30 kbps/11khz in *stereo*. Just my opinion but I've found streaming with quality -1 22khz Mono produces the best sound quality for 32kb/s. 11khz stereo sounds like crap in comparison. I'm just waiting (impatiently) for OddCastDSP(v1) to support 2...

...a lot of them about) or will 24 be good enough fo that? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of the lame defaults. For example, 16kbps 11khz mono sounds very reasonable. At 24kbps, either 22khz mono or 11khz stereo sound pretty good, though the latter is a bit iffy under lame. At least with 24kbps however, these aren't the lame defaults. Mono will give you 16khz and stereo gives you a lowly 8khz (yuck!). You can get lame to do h...

...ity, only that in vorbis, the poor quality is > actually good! :) Another advantage I see in reducing the nominal bitrate for q0 to 48kb/s with a ~13khz lowpass is a smoother transition in average bitrate and frequency resolution from 22khz to 44khz sampling. Currently there is a jump from 11khz (at 22khz sampling) to 15khz (at 44khz sampling -q0). But then I suppose there is 33khz sampling but I don't think too many devices support it. Something to think about: One global quality scale from 0 to 10. 0 = resample to 11khz, 8kb/s, through to 10 = original sample rate, high kb/s....

hello all! After having read http://www.xiph.org/ogg/vorbis/listen.html and listened to the examples on that very educational page, I decided to augment the info. This example is more simple, it involves the comparision between two streams of the same radio station, FranceInter (a station of Radio France in Paris). The ogg stream is running at around 30 kbps/11 kHz in stereo. The WMA stream

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey Gang, I figured this is a frequently asked question, however it is not in the faq. i am wondering the lowest possible bit-rate to stream ogg at. is this an ogg limitation or is this dependent on the streaming server. Thanks in advance, (please reply to me too as i am not on the list.) <p>- -------------------------------------- Jeff

...ough, except the machine only has a 2GB harddrive). Currently I am archiving my radio show using the Fraunhofer MP3 codec with some software called AudioGrabber. I just have Audiograbber to record from the line-in on my soundcard. I set it to use Fraunhofer to record directly as MP3 mono, 16kbps, 11kHz. I believe speex could produce better quality/smaller file size. The problem I have is not being able to choose the speex codec in Audiograbber instead of Fraunhofer MP3. Just wondering if there's a way to make speex a selectable encoding codec. I'd like to use speex in real-time to record/...

...having speex as a selectable codec in windows would help out alot of people (or at least get alot more people interested). Anyone up to the task of coding it to do this?;) Please, please. :) I was just thinking Speex could beat Fraunhofer MP3 for what I do. My current MP3 settings are 16kbps, 11Khz, mono. A five hour recording is approximately 35MB. Speex sounds like it would be a perfect application for this and achieve smaller/better quality recordings. What do you think? Ben <p><p><p>>In AudioGrabber, you have the option to set up an external encoder, you >don...

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 'icecast-request@xiph.or...

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 'icecast-request@xiph.or...

Hi, I just wanted to know if there are any recommendations for pre-/post-processing (processing power isn't a question). I'm aiming at 16kHz and tried an lowpass-filter at 11kHz before encoding but this didn't improve the results... <p>Regards, Thomas --- >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 'speex-dev-request@xiph.org' containing only the...

...io expert, but here's my take on the subject. 22KHz will take more space, but it is higher quality, and would probably be better if you ever decide to put them on CD (44.1KHz, stereo is the required format for CD Audio). There was a noticeable difference in the two when I tried recording at 11KHz vs 22KHz; I ended up just recording at 44.1 KHz even though the tape wasn't high quality, because I'm putting mine on CD, and that makes it easier (less conversion steps = better audio). If you decide to go higher than 11KHz, be prepared, because the better the quality, the more space yo...

--- boink <boink@tetter.xs4all.nl> wrote: > The ogg stream is running at around 30 kbps/11 kHz in stereo. The > WMA > stream (provided by yacast.fr) is running at 16 kbps/22 kHz in > mono. Is stereo vs. mono, and 22khz vs. 11khz even a fair comparison? 22khz vs. 11khz: 22khz, by definition, gives the 22khz stream much more frequency range and *opportunity* to sound clearer. Yes, there are more audible artifacts in the .wma than in the Vorbis stream, but IMHO the sampling rate should match anyway. Stereo vs. mono is less...

I'm streaming a police scanner online. What would be the best encoding to use the ogg or mp3 or mp3 pro Right now i'm using MP3 16KBPS 11KHZ and it uses about 3.1KB/sec of bandwidth. Is there something I can use that has the same quality with lower bandwith. I've tried MP3pro using the lowest profile which I think was 20KBPS at 16KHZ and it uses 3.7KB/sec --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project...

...month or two if significant updates are imminent. With the start of summer vacation in Denmark the whole country pretty much grinds to a halt anyway, so I might as well wait 'till august if there's a sufficiently good reason ;) Specifically I'd be interested in improved support for 11KHz sampled source files. Not that I'm dissatisfied with the quality, but the warnings from speexenc sound pretty dire... TTYL Anders --- >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 '...

On Wed, May 22, 2002 at 10:11:21PM +1000, Geoff Shang wrote: <p> > WMP 7 and above will play shoutcast streams, but don't ask me for specifics WMP 6.4 and above will play icecaststreams. -- Venlig hilsen/Kind regards Thomas Kirk ARKENA thomas(at)arkena(dot)com http://www.arkena.com <p>BOFH excuse #342: HTTPD Error 4004 : very old Intel cpu - insufficient processing power

Can anybody tell me where I can find some info on the new books? I noticed there are now A-E. What exactly does this mean? How does it translate to higher and lower sampling frequencies and bit rates? Ideally I'd like something in the 11kHZ, 8bit mono area... I also noticed this struct: /* CD quality stereo, no channel coupling */ vorbis_info info_B={ /* channels, sample rate, upperkbps, nominalkbps, lowerkbps */ 0, 2, 44100, 0,0,0, /* smallblock, largeblock */ {256, 2048}, ..... Does this mean I can just start tweaking away...

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >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

...n 8-bit sample resolution audio file (8-bit .wav -> .flac), the audio upon playback is severely clipped; heard, as well as seen with visualization plugins. The problem persists even with --no-replay-gain given to the encoder, and is present in both mono and stereo audio files. Sample rate (eg, 11khz, 22khz, 44khz) seems to be irrelevant. 16-bit flacs are played back properly. flac -d (to decode) writes a properly-scaled .wav as expected. steps in brief: assuming "snd_8.wav" is, for example, a 44100 stereo 8-bit Microsoft WAV (PCM) file... 1. flac snd_8.wav produces snd_8.flac...

On Mon, 2003-08-25 at 17:04, W. Kevin Pedigo wrote: > But if your problem is serving more bandwidth than you've got, you gotta > serve less (narrower or fewer streams) or get more bandwidth. It's that > simple. Tell us what you want to do about it, and we'll try to help. OK. I've gotten everything running with one problem. I'd like to downsample a live stream.

hi all I use ices0.2.3/icecast1.3.12 on a linux box. I stream mp3 . I would like to make my listeners to discover ogg . My tracks are all in mp3 format. i reencode from 128/96 kbs and 44100Hz => 16kbs and 11kHz ( mp3 ) Is it possible , ( if yes tell me how ) to feed icecast with both mp3 and ogg stream , at different mount points . how to configure ices to play ogg ? i know i'll have to run 2 different ices so how to manage ? by indicating in the command line the conf_file to read for each launched...