search for: 22050hz

Hi! I'd like to encode from a 22050 Hz source wav file (only in that case), but i get the following error message: "Invalid floating point operation" I use delphi: vorbis_encode_init_vbr(vi,2,22050,0.5) . . . While (i < readCount div 4) Do Begin buffer[0][i] := smallInt((pArray(@readbuffer)[i shl 2 + 1] shl 8) or pArray(@readbuffer)[i shl 2 + 0]) / 32768;

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $ sox -V War\ -\ Low\ Rider.wav -r 22050 fo...

...ht direction. I'm developing a dictation application on the Dell Axim (Windows Mobile 5.0 Pocket PC). A key requirement of the application is the best possible sampling rate as the audio goes into a speech reco system. So, I've set up my wrapper around libspeex to capture audio (1 channel, 22050Hz, 16 bits/sample) in blocks of the frame size. However, the encoder lags significantly - typically upwards of 2000 frames, at complexity <= 1 and quality <= 3. I've a On the decode side, the decoder is not able to do real time at these sampling rates (16kHz, 22kHz and 32kHz). The only sam...

Hi! We have a little problem. We have to get 4 channels through a Dual-ISDN Line (128k). MP3 sounds awful (tested with lame 24k). Real sounds good, but I cannot encode 4 mono streams from a 4 channel soundcard, only 2 stereo streams. So I tested Vorbis. I encoded a 22050Hz mono file with -b 112, but this has a average bitrate of 59kbps. the 44100 mono file is 94kbps in average and the stereo 44100 108kbps. It was the same song converted with sox. The stereo file is only 14kbps bigger than the stereo file. How could I achive lower bit rates without reducing the sampl...

Hi! We have a little problem. We have to get 4 channels through a Dual-ISDN Line (128k). MP3 sounds awful (tested with lame 24k). Real sounds good, but I cannot encode 4 mono streams from a 4 channel soundcard, only 2 stereo streams. So I tested Vorbis. I encoded a 22050Hz mono file with -b 112, but this has a average bitrate of 59kbps. the 44100 mono file is 94kbps in average and the stereo 44100 108kbps. It was the same song converted with sox. The stereo file is only 14kbps bigger than the stereo file. How could I achive lower bit rates without reducing the sampl...

I am writing a long-run sound recording application using OggVorbis. The user can adjust sound quality parameter to make balance between storage space and sound quality. The input sound is 22050Hz, mono. To maximize the recording capability on a given storage volume, we want the result bitrate as low as possible. I use -q0 parameter and it produces a 32.0kbps stream. It is too high (our competitor using Microsoft WMA can produce 16kbps streams at the lowest quality setting). I tried -...

...what to change (I can access the IceCast admin window without a problem). I have included what happens when I run IceCast and LiveIce below.... Any help would be greatly appreciated... Best regards, Bryan <p>[root@live liveice]# ./liveice playlist playlist 0 Initialising Soundcard 16Bit 22050Hz Stereo opening connection to localhost 80 Attempting to Contact Server connection successful: forking process opening pipe!... writing password Setting up Interface Soundcard Reopened For Encoding Input Format: 16Bit 22050Hz Stereo Output Format: 32000 Bps Mpeg Audio IceCast Server: localhost:80 Mo...

...eam live audio. I am getting an error message that I can't seem to find an answer to. FIrst of all, I start Icecast (Version 2.2.0) and it seems to come up fine.Then I start liveice, and I get the following error message: liveice -c /usr/local/bin/liveice.conf 0 Initialising Soundcard 16Bit 22050Hz opening connection to localhost 8000 Attempting to Contact Server connection successful: forking process server didn't send OK response I am running Fedore core 3. My config files are below: icecast.xml +++++++++++++++ <icecast> <limits> <clients>100</client...

At 4:37 PM -0400 7/6/01, Brendan Cully wrote: >On Friday, 06 July 2001 at 13:21, tom erbe wrote: >> At 1:48 PM -0400 7/6/01, Brendan Cully wrote: >> > > 0.2 doesn't segfault, but it seems to overrun the server (if that >> >> makes any sense), it plays a 3 minute file in seconds. >> > >> >I'd like to know more about this. Are you

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

Hi Folks, I'm having the weirdest trouble with liveice. When I start it up with the correct servername, it grinds to a halt with these messages: playlist 0 Initialising Soundcard 16Bit 22050Hz Stereo opening connection to localhost 8000 Attempting to Contact Server connection successful: forking process opening pipe!... writing password Then it waits there. Now, when I tried it with a bogus servername, liveice starts working. Of course, it's not sending me stream anywhere! Why...

...what to change (I can access the IceCast admin window without a problem). I have included what happens when I run IceCast and LiveIce below.... Any help would be greatly appreciated... Best regards, Bryan <p>[root@live liveice]# ./liveice playlist playlist 0 Initialising Soundcard 16Bit 22050Hz Stereo opening connection to localhost 80 Attempting to Contact Server connection successful: forking process opening pipe!... writing password Setting up Interface Soundcard Reopened For Encoding Input Format: 16Bit 22050Hz Stereo Output Format: 32000 Bps Mpeg Audio IceCast Server: localhost:80 Mo...

...admittedly it was a long time ago, Oddcast presented some issues with regard to screen reader accessibility. I should really run it up the stick again to see what it's like nowadays. 3. Quite a lot of our broadcasters are very fond of the FHG encoder. At least for now, we broadcast at 56kbps 22050Hz stereo, and whilst LAME has made some great strides at this bitrate in recent times (I use it myself under Linux and am quite happy with it), the FHG encoder does sound good at this rate and I know some would be loathed to give it up. At least as far as I'm aware, there's no way for Oddcas...

...as in some cases the wrong mime type will be returned from the server and so causes the relay url fail to be recognised by sc_trans. This depends on the server. What is wrong with this scenario. 1. Currently running icecast server 2.3.2 and ices0.4 using mp3 (32kbps/22050Hz) and custom perl script to serve up the content. (AirProgressive.org). 2. Install shoutcast dnas and sc_trans 3. use calendaring system to add a relay from icecast, <relay url="http://airprogressive.org:8000/stream" priority="1"/> Anyone tried this yet? From just readin...

> > >> >> > >That line relates to this >struct ovectl_ratemanage_arg ai; > >which is defined in vorbis/vorbisenc.h. Is one of those other lines >stating that certain headers files could not be found? If so then make >sure that your system has the ogg and vorbis packages installed. You may >have to build libogg and libvorbis from source if you

...I'm developing a dictation application on the Dell Axim (Windows Mobile > 5.0 Pocket PC). A key requirement of the application is the best possible > sampling rate as the audio goes into a speech reco system. So, I've set up > my wrapper around libspeex to capture audio (1 channel, 22050Hz, 16 > bits/sample) in blocks of the frame size. However, the encoder lags > significantly - typically upwards of 2000 frames, at complexity <= 1 and > quality <= 3. I've a > > On the decode side, the decoder is not able to do real time at these > sampling rates (16kHz, 2...

...Dell > Axim (Windows > > Mobile 5.0 Pocket PC). A key requirement of the > application is the > > best possible sampling rate as the audio goes into > a > speech reco > > system. So, I've set up my wrapper around libspeex > to capture audio > > (1 channel, 22050Hz, 16 bits/sample) in blocks of > the frame size. > > However, the encoder lags significantly - > typically > upwards of 2000 > > frames, at complexity <= 1 and quality <= 3. I've > a > > Just a detail here. 22.05 kHz is not recommended for > Speex. I sugge...

I have been tyring to use the preprocessor standalone by feeding it pcm data from a headerless pcm file (mono, 22050hz) and then write the result to another file. Unfortunately, all I get back is noise (first frame is set to 0, rest is noise). Frame size is set to 440. I just modified testdenoise.c slightly to read/write from a file (see below) but I can't get useful results. What am I doing wrong? Do I nee...

...e_frame was setting status to > FLAC__FILE_DECODER_STREAM_ERROR. As I said, the 'flac' decoder could decode > the file just fine, and unless I'm mistaken, that uses the file > decoder also. > > The only odd thing about the input wave file was that it was 8bit mono > (22050Hz). oN a hunch, I converted the wave file to CD audio format > (16bit/stereo/44100Hz) with sox, and re-encoded it. The plugin played the > resulting file fine. I tried a few tests (converting to stereo and leaving > everything else the same, converting to 16 bit and leaving everything els...

...s setting status to > FLAC__FILE_DECODER_STREAM_ERROR. As I said, the 'flac' decoder could > decode > the file just fine, and unless I'm mistaken, that uses the file > decoder also. > > The only odd thing about the input wave file was that it was 8bit > mono > (22050Hz). oN a hunch, I converted the wave file to CD audio format > (16bit/stereo/44100Hz) with sox, and re-encoded it. The plugin > played the > resulting file fine. I tried a few tests (converting to stereo and > leaving > everything else the same, converting to 16 bit and leaving ever...