similar to: Vorbis vs Speex

Hi, Once upon a time, I can access the admin information such as client connected, streaming title/artist, etc, from web browser by browsing: http://servername:port/admin I used to had this just worked when I installed icecast and ices on the same machine. I then switch machines and have icecast in machine A, ices in machine B, connecting to icecast using mountpoint <mount> to machine A.

We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the "Transfer" button on the phone (these phones are Grandstream BudgeTone 100s) or using the "#" key (the T/t flag must be set in the Dial command in asterisk for

hi I wonder if you would like to join my localplayer.com network? I operate on the same philosophy. All my artists are unsigned and I have their consent to stream their music. on my station. I envision a drop down menu that helps me navigate to the city of my choice. Raymond wrote: > Hello: > > I'm starting a radio station on the LPBN. The music will be from all > local Los

Alright folks, here's the solution. 1) Keep extensions to 3 letters for audio & video. Except for special situations where the user might be doing a codec specific name. Since the official extensions are 3 letters, those can always be used on any 8.3 device. 2) introduce a new extension .OGV for ogg container video. With a strong preference for Xiph only codecs. (If you want 3rd

Sorry for my ignorance but what is the LPBN???? I don't think my main stream was running after I mailed you earlier today. It is running right now. Do you have an URL for LPBN??? Raymond wrote: > Yes, I'm listening to Aardvark Recorders Present: Local Players - one.mp3 > > So you want to syndicate? Or you want both localplayers and the LPBN > to just share stuff. Anyway I

what do want the tool to do? you could look at a tool in development called the Frequency Clock -> http://frequencyclock.montevideo.nl its a dev server but you can get the idea, an example of the player output by the system is linked from http://www.radioqualia.net <p>adam On Mon, 28 Oct 2002, Raymond wrote: > Hello: > > I'm starting a radio station on the LPBN. The music

Marco, I have some preliminary HowTos and software available at http://quasi.ksl.com/icecast/ It's not as nice as http://www.6809.org.uk/kja3/ices2-howto.shtml, but I'm working to make it better as I become more familiar. Hope it helps. The ices2 program is designed for .ogg streaming while icecast-1.3.12 and related streamers, MuSE, liveice and DarkIce will still stream .mp3 quite nicely.

Hey everybody I have worked quite to near the end of The RSpec Book, but now I''m having some troubles. I''m trying to get the last few Cucumber steps to work, which are: Then /^Caddyshack should be in the Comedy genre$/ do visit genres_path click_link "Comedy" save_and_open_page response.should have_content("1 movie") response.should

Hi, there. While waiting for the encoder RC1, I will report some minor problems I experienced with Peter's Winamp decoder. First, I am using Nullsoft Vorbis Decoder v.1.11a with Winamp v.2.76 on Windows 98 SE. 1. When the buttons of Playlist Editor are extended, the part extended in playlist screen seems to be flickering when an Ogg file is being played. 2. I check the option of

>Stephen Bannasch wrote: > >> I'm getting this error trying to build speex on MacOS X 10.5.6: >> >> ./configure: line 21256: syntax error near unexpected token `FFT,' >> ./configure: line 21256: ` PKG_CHECK_MODULES(FFT, fftw3f)' > >Why are you compling from Git? If you grab a release tarball you should >never see an error like that.

> Good point. File extensions normally represent groups of related > formats. I don't propose differentiating everything (e.g. standalone > FLAC from Ogg FLAC). I do want as a minimum to tell apart these > categories: > > - Lossy audio: Vorbis, Speex. But speech is useful to distinguish > from music, so making Speex separate is not a bad idea. > - Lossless audio:

Hi, I have one doubt again, that is Vorbis use DCT/MDCT based algorithm and also use psychoacoustic model so this is lossy codec. And I dont think it ca regenerate a better matching waveform than speex. Then there comes FLAC which is the perfect answer to my question, I suppose. But my concern is this that FLAC use simple prediction algorithm and doesnt use any CELP based algo which could have

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

OGG vorbis vs speex. What should I use for comedy albums. Would speex be just as good for straight standup comedy and the likes. How do they compare when you might have sound effects, music in them, etc. <p><p>http://mobile.yahoo.com.au - Yahoo! Mobile - Check & compose your email via SMS on your Telstra or Vodafone mobile. --- >8 ---- List archives:

Hello kind developers, I am planning on creating a steganographic extension for speex/CELP. Since speex/CELP is lossy, there should be quite many bits that one can use for hiding data. I'm familiar with the principles of CELP (two filter loops, one for pitch, the other for formants, let the raw speech frame run through those filter loops, take residue signal, map it on a codebook entry,

Hello everyone, I'm a student at the Universtiy of Delaware, and will be soon starting some research on the effects of lossy audio compression on speech sounds. I will be preforming test with both mp3 and vorbis. First of all, if I use the '--ogg' switch to lame, does lame use GPSYCHO to encode the wave, or some other psychoacoustic model (perhaps one designed for

Hi, recently I made some test with the opus tools (enc and dec) and I'm very (and positively) surprised about the resultant quality. But the only think that I miss is the ability to change the low pass filter frequency via "--lowpass" option or similar. For example at a quality or 96 kbps the cut off of the filter starts at 16Khz and is completely cut at 20 Khz. But in case of

Hi, I chose speex initially because i had some work in VQ on speex i.e. modifying split VQ to GMM based parametric VQ and I thought If I train the GMM based VQ codebooks with audio signal and then do audio coding with speex, I probably get a better(smaller) residual signal even with speex. But I couldnt get that. I was trying to get a lossless bitstream by MUXing the speex-bitstream and the

Hello I had an idea about judging the quality of ogg vorbis (or any other lossy codec) I took a wave-file and encoded it to ogg. Then decoded ogg to wav and inverted it's phase. When mixing the original wav with the phase-inverted decoded ogg-file, any identic parts of compressed and uncompressed audio should be eliminated. Of course there's always a "rest" of sound because

Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu ("press 1 to leave a message, press 0 to speak to a representative") and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a