similar to: Sampling Rate

Hi, I'm no DSP or audio expert by any means, but I can share what works for me. People in the know, I would appreciate tips on whether this stuff is ok. You could sample at 32000Hz (or 48000Hz, any AC97 card will support this), run a 8000Hz lowpass filter over the data (16000Hz sample rate can only represent frequencies up to 8000Hz) and then drop every second (or 2 out of 3 for

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

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

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

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

-----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

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:

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi, I'm not anywhere near an expert, but I had successfully used Darkice, TwoLame, and Icast231 to netcast a mp2 stream. If it will help, here is a snip from the related area of my darkice.cfg: [icecast2-1] format = mp2 bitrateMode = cbr bitrate = 384 quality = 1.0 server = 127.0.0.1 port = 32710 password = (duh!) sampleRate =

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

Ben, This is what I have in my liveice.cfg file. #SERVER xx.xx.xx.xx SERVER xxxxx.xxxx.xxx PORT 8000 PASSWORD xxxx USE_LAME3 /usr/local/bin/lame ENCODER_ARGS --lowpass 6 --silent --noshort -q2 SOUNDCARD HALF_DUPLEX SAMPLE_RATE 48000 BITRATE 64000 STEREO X_AUDIOCAST_LOGIN NAME KSL Radio 1160 GENRE KSL's Testing Internet Radio URL http://xxxxxx.xxx.xxxx/ DESCRIPTION Utah's Home for News,

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

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

> I believe there is new resampling functionality in the speex svn head, > although I haven't tested it yet. You might also want to check out > 'Secret Rabbit Code' for your resampling. Yes, I've just been working on a resampler recently. Its changing a lot, but it's now usable. I'd actually be quite happy to have some feedback on it. Jean-Marc > Hope this

In what fashion does the encoder 'blowup' when passed a frame of zeroes to encode? What version of speex are you using? ________________________________ From: Chris Cowden [mailto:ccowden@ncsoft.com] Sent: Thursday, 15 February 2007 7:57 AM To: David Hogan; speex-dev@xiph.org Subject: RE: [Speex-dev] frame of silence Okay, you've answered part of my question, which is