similar to: The GPL and Speex

Speex: I am creating a closed-source projec that implements the Speex library into the application. According to GPL, as freedom in code, my project has to be open-source to use code from open-source project; Speex. Basically, all I ask is special permission into using Speex into a closed-source application. Not to forget, you will, of couse, be credited in the About box. Reply if your

Ok, I figured it out. :) This seems to work: 1) Call speex_bits_read_from() once, specifying the location in memory of the compressed data, and the total length of that data. 2) Keep calling speex_decode() until speex_bits_remaining() returns 0. Then you don't have to keep track of the # of frames per packet, or the size of each compressed frame. It's done magically by the codec.

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

Ohh... Nice! This is new in 1.0.1, isn't it? It doesn't seem to be included in the reference manual yet, though. Thanks! Tom <p>Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > OK, this is how it works: > > The encoder calls speex_encode any number of times and then calls > speex_bits_insert_terminator before sending the bits. > > The

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

Ok here is something I couldn't see on the description page. We need to convert to 8 bit audio right? I'm assuming narrow band is 8 KHz 8 bit. Wide band is 16 KHz 8 bit. UW band is 32 KHz 8 bit audio? George -----Original Message----- From: Tom Grandgent [mailto:tgrand@canvaslink.com] Sent: Tuesday, October 03, 2006 11:42 AM To: Bertie Coopersmith; George Ou Cc: speex-dev@xiph.org

hi everyone, Im implementing a new RBlas dll but just for optimising sake i need to know how and WHERE (which source code) RBlas gets loaded. there is a call to loadlibrary() in dynload.c BUT it seems this is not the function that is loading Rblas. how do i know this? well i added the code :(just after the loadlibrary() call) /*********************************************/ /*this

hi Tom, i downloaded your speex wrappers and they're really what i'm looking for! i really cannot worry about container support at the moment because i'm working with real-time audio, and hopefully toward VoIP when everything else is in order. im using waveform to capture data from the microphone. the raw PCM sample is stored in the WAVEHDR structure's member lpData, and this

Actually, it looks like this is already done in Speex as of revision 7150 in SVN (7/15/2004). As long as HAVE_CONFIG_H is defined in the project settings (which it is in all of the project files in SVN), win32\config.h should be included, and this has #define inline __inline in it. Oleg - you should make sure HAVE_CONFIG_H is defined in your project settings. And, you didn't say which

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

Hi, I newly downloaded rspec, rspec_on_rails and ran the command ruby script/generate rspec from the instructions http://rspec.info/documentation/rails/install.html I then created the directories in spec: controllers, fixtures, helpers, models, views I then created the file spec/models/user_spec.rb as a spec for my model User. (Is the convention to name the model''s spec

Yes, you can use it independently by using speex_preprocess(). This function can do noise removal, AGC, and VAD. I've been using it to do all three and it usually works very well. There can be a train going by outside, producing lots of sound that makes it through the noise filter, and yet the VAD knows it's not speech. However, sometimes the VAD seems to get into a bad state and

Hello R-users, Version 1.8.1 on a 64-bit Solaris 5.8 OS I am trying to instal the Rcmdr package for which the tclk package is required. Once both package installed I issued the command: > library(tcltk) and received the following error message Error in firstlib(which.lib.loc, package) : Tcl/Tk support is not available on this system Error in library(tcltk) : .First.lib failed Upon

Hi there, I use Samba 4.9 since its releaseday and also had some troubles first due to the lack of informations in the wiki. Anyways... after some try & error i got it working and since i run the Cluster (non-production environment) had no errors except CTDBs monitoring of the Samba process did'nt work as i would expect it (RPC-Server was dead but CTDB did'nt complain and "ctdb

That's might be speex_denoise() is trying to do already, based on the fact that if noise is present in speech signal, the former gets suppressed to a certain extend. Also, I noticed after suppression, the short, noisy attack - speex_denoise() have to be doing some kind of profiling. I've looked at a few offline denoise() solutions. They all involve profile building. A real-time

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that