similar to: Library split (poll)

Hi Jean Marc, On Wed, 28 Jun 2006, Alfred E. Heggestad wrote: > good idea to split the libraries > > If e.g. AEC/Jitterbuffer can be used with other codecs such as G711, > without linking to the speexcodec lib, I think people will reuse the > Speex jitterbuffer implentation and avoid writing their own. > > what about: > > libspeexpreproc Preprocessor >

I have sent the visual studio project files that compile the speex client to jean-marc directly (as an attachment), sometime this week, I will also upload the binary builds of static library files to my site www.phonestack.com. I am porting our LTP (lightweight telephony protocol) to some embedded systems. I require to run speex in an environment that doesn't support memory allocations. How

Hi everyone, Judging from the lack of complaints about the split, I think I can go safely forward with that. The only thing to decide is the name of the new library that will contain the preprocessor, the AEC, jitter buffer, and resampler. So far, I've got a few potential names: - libspeaks - libspeexproc - libspeexvoip - libspeech - libspoke Let me know (in private so as to not flood the

Hi, A while ago I mentioned I was considering splitting libspeex into two parts. I'm attaching a patch that does exactly that, but I haven't applied it to svn because I first want to have some feedback. That patch leaves the codec in libspeex and moves the other stuff (jitter buffer, AEC, preprocessor, ...) to libspeexvoip (I'm open to suggestions on better names). Some of the things

Hi, I am an user of speex codec.Can I ask some question about the speexcodec 1.2beta2 on installed to linux(fedora 6) thanks! After I downloaded the latest version of codec then I type the commands to install : configure -prefix=/home/...../test -enable-shared -enable-static make make install then I try to compile my encode.c here is the encode.c of mine: =====================================

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

On Wed, 2006-06-28 at 11:34 +0530, Ashhar Farhan wrote: > I have sent the visual studio project files that compile the speex > client to jean-marc directly (as an attachment), sometime this week, I > will also upload the binary builds of static library files to my site > www.phonestack.com. Actually, Zen (illi on IRC) is the one handling the Windows stuff. I can't do anything with

I vote for splitting 'em off and documenting the great features of each. I've been using the codec for a while now and would be more likely to dig in to the additional features if they were offered as separate components so I could use them with other audio projects (as well as being more likely to use those features in speex, simply because they would be easier to dig in to at

Hi Aymeric, > 2 questions: I count 4 :-) > * Do you "feel" that the current svn is enough stable Yes. I'm planning on releasing it as 1.2beta3 this week. Even in normal time, I try to keep svn in good shape so it is almost always better than the last release. And even when it breaks, Speex bugs tend to be very obvious and easy to reproduce (no segfaults that happen once a

Hi All, I am trying to cross compile speex-1.1.12 to powerpc-405, i get a error after the make, speexec.lo error, please help me how to get rid of this error. On 2/9/07, speex-dev-request@xiph.org <speex-dev-request@xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev@xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

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openh323 speexcodec.cxx has: BOOL SpeexCodec::EncodeFrame(BYTE * buffer, unsigned & length) { // convert PCM to float float floatData[SAMPLES_PER_FRAME]; PINDEX i; for (i = 0; i < SAMPLES_PER_FRAME; i++) floatData[i] = sampleBuffer[i]; // encode PCM data in sampleBuffer to buffer speex_bits_reset(bits); speex_encode(coder_state, floatData, bits);

If more than one speaker receives the *same* signal, it doesn't matter the number of speakers. It only gets tricky when the speakers are playing slightly different signals (e.g. from a stereo song). Jean-Marc Quoting Greger Burman <greger at mobile-robotics.com>: > One thing I can think of is if you are using two or more speakers. If the > speakers are not at the exact same

Greetings. I just upgraded to the latest 1.2beta3 and now I get the following link error: unresolved external symbol speex_wb_mode. Using VS2005. If I put back the old libspeex.lib it compiles just fine, but I would like to upgrade to use the new AEC stuff if I can. My code looks like: enc_state = speex_encoder_init(&speex_wb_mode) ; Worked before the upgrade so did the API change?

Hi all, I've taken preglows ARM versions of qmf_synth and iir_mem16 from rockboxes speex codec, and tweaked them a bit further for some more speed. I attach them here so you can review and take on any changes you want. Please let me know if you have questions etc. Thanks, Robin -- Robin Watts, Email: <mailto:Robin.Watts@wss.co.uk> Warm Silence Software, WWW:

Thanks Jean-Marc. Is there a Speex trunk for the C64xx I can use? Or do I have to figure what options to turn on (in configure) when building the library? Thanks. Regards, Andy Ngo ----- Original Message ---- From: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> To: ndno72-speex@yahoo.com Cc: jtrantow@ieee.org; speex-dev@xiph.org Sent: Tuesday, December 5, 2006 2:52:02 PM Subject:

is there a dowload for mac users? __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

>There's plenty of areas for improvements that don't require incompatible changes like this one. can u please tell me what do I do to make it more exact waveform coder for music rather than speech. I understand that its meant for speech, but I was just using it for music... I am interested in getting the residual as small as possible using speex. Can you please tell me the areas to

Hi all, We all know that mismatch between clocks of ADCs of far-end voice and near-end voice is not allowed in a time-domain or frequency-domain LMS based AEC system. It means that capture and render audio streams must be synchronized to a same sample rate. However, I found that this restriction is removed in microsoft AEC from Windows XP SP1. Anyone knows how microsoft AEC do it? This technology

Hello all, I am attempting to integrate the AEC algorithm from Speex into an RTP-based voice-over-IP application. The application is supposed to operate on a TI C6412-based board, using Code Composer Studio for development. Everything seems to be working, but the problem is that there is no discernible difference in the output, whether the echo cancel function is called or not. Let me first