search for: 32000hz

Hi! I would really appreciate if someone could try to help me out with this... i've been struggling with it for quite a while now. :( Anyway, I was trying to use ices to broadcast mp3s with icecast2 (a recent cvs snapshot of all 3 components from brendan's site). I have it working fine for some of my mp3s (encoded with lame -b 32) but with the other mp3s which are actually the ones that

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 16000hz with 40kbps stereo, so I'm guessing > this is the problem. > You can eithe...

Hi everyone, I'm about to release version 1.2-beta1 (which I could have called 1.1.13), which includes many, many changes. It would help if everyone could give the svn version (http://svn.xiph.org/trunk/speex/) a try and see if it works fine. I'll check my email next week when I'm back from some vacations and if nothing bad has been reported, I'll make the release. Have fun,

...unk r11792, here are my findings: * Builds ok for Symbian, with 1 warning (patch to fix this this attached.) - not tested on hardware * Builds ok for Ubuntu 6.06 (AMD64), although there are some issues with 'make -j4' (try it). Runs fine on AMD64 (64-bit mode), tested 8000Hz and 32000Hz * Stereo encoding/decode goes at half the frequency, but this problem is also present in 1.1.11 and 1.1.2 from Debian, so it might be a problem in my application.. in general it works as expected, no major issues found.. /alfred Jean-Marc Valin wrote: > Hi everyone, > > I'...

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 48000->16000) sample. The result being, 16000Hz sampled audio. If you omit the filter the resul...

...To: 'speex-dev@xiph.org' > Subject: RE: [Speex-dev] 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 > 48000->16000) sample. The result being, 16000Hz sampled > audio. If...

...clients for source 1 -> [13/Mar/2001:22:03:33] [Bandwidth: 0.000000MB/s] [Sources: 0] [Clients: 0] [Admins: 1] [Uptime: 8 minutes and 1 seconds] -> After a while, the connection is dropped due to lack of data. And: [root@audio liveice]# ./liveice playlist 0 Initialising Soundcard 16Bit 32000Hz Stereo opening connection to localhost 8001 Attempting to Contact Server connection successful: forking process Liveice does not give the nice screen with the level indicators when I start it with icecast running. And: [root@audio /root]# netstat -t Active Internet connections (w/o servers) P...

Hi everyone, Just to let you know that version 1.2beta3 is out. The main change is the split into libspeex and libspeexdsp. Basically, libspeex keeps the codec (just like it was for 1.0.x) and libspeexdsp has all the new components. Other changes include a new jitter buffer algorithm and resampler improvements/fixes. This is also the first release where libspeex can be built without any floating

Hello, In one of my recent assignments, I was asked to develop a ogg player. I am not able to find the right repository of the source code. There are few repositories on vorbis but I am not clear which one is the right one for Windows environment. If somebody has successfully compiled any ogg player ever, his ideas will be helpful to me. In one of the code set, I was able to compile the

Hi everyone, I've recently been making various changes to the way the modes work and the supported frame size. On new feature that may be of interest to some is that CELT should soon support changing the frame size dynamically within a stream. By that I mean varying the amount of audio (in time) transmitted at once, not the compressed size -- which has always been variable. That would