similar to: Sampling Rate

Oops, CTRL+Enter send strikes again ... At the other end for playback you can convert it back to 48000 (or whatever) by repeating each sample 3 times (48/16 == 3), then running a 8000Hz lowpass over the result to remove any aliasing artifacts. Cheers, David Hogan > -----Original Message----- > From: David Hogan > Sent: Tuesday, 12 December 2006 10:44 AM > To:

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling

What would be speex configuration recomended for Telefone/Voip quality voice? With a quality just a little better/similar then G.729? or GSM? is there a comparison chart somewhere, but telephone quality oriented? Thanks, Alain Tom Grandgent escreveu: > Kirk, > > Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you > don't use one of these sample rates,

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for voice chat, try 16kHz with VBR quality 6. You should see

It seems that I only have the following values available for sampling from the mic. "The value must be 8000, 11025, 22050, 32000, 44100, or 48000" Which leaves 8000 and 32000 for use with speex. I think since this is a game and not a voice application, I'm stuck using the 8kHz rate. What speex setting would you recommend I use for the best quality/performace, what frame size

Frame size of 320 means 320 samples, which is 640 bytes of data if the samples are shorts. Speex can work with samples as shorts or floats - for example, see speex_encode_int vs. speex_encode. In both cases the values should be signed ranging from -32768 to 32767. I suggest trying the sampleenc.c and sampledec.c programs in the doc directory. If those work, then maybe you can spot what

Hi everyone. I did a rough recording of an instrumental (electric piano sound) & e-mailed it to a friend in Vorbis 11025 Hz / mono. I was seeking a bitrate in the range 8-16k/sec. The song is 2:55 in length. q=-1 happily achieved a 12.6k/sec bitrate. All file sizes I mention in this are for files without informational tags. And I hope this isn't interpreted as trying to plug my music or

I have a time series of data from an electroencephalogram (EEG). I wish to filter the data to get rid of 50Hz mains 'hum'. I have 'designed' a combination bandpass and notch filter using a web- site. The site returns the filter in "ANSI C" source code. It is:- /* Digital filter designed by mkfilter/mkshape/gencode A.J. Fisher Command line:

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 would very much like a lowpass option because for FM radio broadcasting I don't want to encode frequencies above 15khz. I'm waiting for this option before switching to ogg from mp3(lame). Ross. > -----Original Message----- > From: owner-vorbis@xiph.org [mailto:owner-vorbis@xiph.org]On Behalf Of > Gian-Carlo Pascutto > Sent: Tuesday, 4 September 2001 01:46 > To:

So there is no way around having to re-encode them all? --Pritpaul --- Geoff Shang <gshang@uq.net.au> wrote: > Hi: > > My guess is that ices can't handle 16000 or 32000hz > sampling rates. The > 32kbps file is 22050hz and the 80kbps file is > 32000hz. I couldn't get > mpg123 to play test2 remotely, but lame defaults to > 32000hz with 40kbps > mono and

i have error in install module of tor2. but it compile good. what happen ? ivr2:/usr/src/zaptel # make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o

As a matter of interest I tested the quality at -1 which has a nominal bitrate of 45 but the 2 tracks I encoded both averaged out to exactly 48kb/s. I noticed the lowpass defaulted to 15.1khz like -q0 so that may need addressing at some stage. I used a lowpass of 13khz which overall sounded better with less artifacts. I found it sharper & clearer than a lame encoded mp3 encoded with

Dr. Williams, I ran across your inquiry on one of the R-help mailing lists regarding digital filter design and implementation. I found no response to your email in the archives and was wondering if you were able to find anything. Thanks, Israel -- Israel Christie, Ph.D. Email: ichristie at gmail.com Phone: 865.766.0214 Mobile: 865.406.4615

On Tue, Apr 05, 2005 at 02:26:03AM -0500, Hal Vaughan wrote: > Is there a list somewhere of "standard" encoding rates? I know, for example, > CDs are encoded at 44100, as is a lot of digital sound, but I've seen > programs that specify different levels of quality (like radio, phone, tape, > CD) and I'd like to know if there are some encoding rates that are

On Fri, 6 Apr 2001, John Griffiths wrote: > ok so 24kbps for 56k modems... > > can i go any lower and get the 28 k modems? (still 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

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/

Hello, I was wondering about the lowpass filter applied at the different quality levels. It seems that the 16kHz cut-off is still there at quality 3. I didn't really abx it, so maybe my mind plays tricks on me. :) Please can someone enlighten me what is used for -q0, -q1 and so on. (lame tells me whats used when encoding tracks) I know I shouldn't judge by bandwidth but I would like to

A while ago someone asked about a low-pass filter for oggenc and was told to get AFsp and filter outside of Oggenc. Well, I got it, and am totally lost (It's way more complicated than SOX) so now can anyone briefly describe what type of filter I should set up (FIR, IIR, all-pole), why one is better than the other, and if you have filter coefficient files lying around (lowpass, 19 or 20 kHz

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched