similar to: SpeexJitter and TCP/IP

Probably the level of your signal is too low and/or you're just not letting it time to adapt. Jean-Marc Le mercredi 19 juillet 2006 ? 19:00 -0400, ac2491@columbia.edu a ?crit : > On closer looks and debugging I always end up in > > speex_echo_cancel function with comment > /* Temporary adaption rate if filter is not adapted correctly */ > > > Does this give any clue

Hi, If I pass the same ref and the echo data to the echo cancellation API, I am expecting silence as output. I get back the original audio data. Is this correct? Thanks -Anurag Quoting ac2491@columbia.edu: > Hi Jean, > > I got the earlier problem fied with correct NN and tail values. > But > I dont see any echo being cancelled. To the echo cancel API I am > giving, audio

Mandi! Rowland Penny via samba In chel di` si favelave... > Not sure what you are proposing is going to work, AD expects every user > to be a member of Domain Users, even though there is nothing in AD to > show membership. Ah. > Do you require this user to visible on all domain machines ? [...] > It might help if you could explain how you are going to use your new > user

hi, this patch should fix the symbian build. /alfred -------------- next part -------------- A non-text attachment was scrubbed... Name: speex-symbian2.patch Type: text/x-patch Size: 2669 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20061115/5ac4915e/speex-symbian2.bin

Hi, I have a few questions about the speex_jitter_buffer when used with DTX here goes: I want to use the speex_jitter_buffer with muliple streams (multi client) and with DTX so that clients will only transmit data when they are actually saying anything, but Im unsure of how the ticking is handled. I have 1 x SpeexJitter struct per client stream Im using portaudio and Im planning on

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

Jean-Marc Valin wrote: > I suggest you have a look at the new API I introduced in svn (see > testecho.c). It's easier to use, harder to screw up, and it means I can > improve it in the future without breaking things. > > Jean-Marc Thanks for your response. Yes, I followed The World Institute for Mental Health recommendation. :) But I think the problem might still present

? hi all, i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" in my VOIP application. Is it that the frame length to be chosen should always be 20ms or can i have flexibility in chosing the frame lenght? on what parameters does the frame length choise depends? thank you all, Shri. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All We are working on an ARM-based speaker phone application where the speaker and microphone is placed roughly 8 cm from each other (similar to this one: http://www.voipsupply.com/images/CHAT50SPLASH.jpg). We are using Speex AEC and preprocessor to handle the acoustic echo. The AEC typically contribute with ERLE of 15-20 dB which should be as expected from the algorithm. Additionally we need

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

hi Jean-Marc, I just updated from svn and it looks like there are some errors in the speex.mmp file. the filters.c is duplicated, and the fft/kiss.c files are still missing.. /alfred Jean-Marc Valin wrote: > Applied in svn. Thanks. > > Jean-Marc > > Alfred E. Heggestad a ?crit : >> hi, >> >> this patch should fix the symbian build. >> >> >>

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call

Dear vegan team, Vegan development happens now completely in github. R-Forge repository is no more in sync with github. I tried to commit all github changes to R-Forge, but a week ago I got a conflict in file and I haven't had time to resolve that conflict. You can follow vegan development and vegan discussion also without signing to github. The system seems to be completely open and does not

I have a video conference like application that I've been working on for a while now, and a recent change is causing some odd problems, and I was wondering if anyone else had seen problems like this. The issue I'm seeing is that when using the sound card for capture, the audio will eventually get about 1-2 seconds out of synch (delayed), from the video. However, if I use USB devices

To the support team, I am getting confused while studying the manual of speex1.2 Beta 2. If audio frame capture and playback are used asynchronously then speex_echo_cancel() function is preferred that is simpler then speex_echo_cancellation() function. However when I go through the API manual, In section 5.4.4.1 It is mentioned that this function is deprecated. Please tell me the set of

Hello everybody. I have the same problem like you. I use ARM microcontroller too. Here is the list of all functions called by speex_echo_cancel until goes into infinite loop: speex_echo_cancel() | spx_fft | kiss_fftr | kiss_fft | kiss_fft_stride | kf_work Function kf_work looks like: static void kf_work(kiss_fft_cpx * Fout, const kiss_fft_cpx * f, const size_t fstride, int

As you know, the ability to use a bundled version of pjproject was introduced with Asterisk 13.8.0. More info on the Asterisk Wiki <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled> and in this email thread <http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>. Since then I've fixed a few

Hello, I would like to test the echo cancel algorithm available in unstable version 1.1.4. This echo canceller can be used with other codecs like G711? Somebody could send me some documentation or sample, or explain the next functions: SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); void speex_echo_state_destroy(SpeexEchoState *st); void

hi all, i want to use the echo canceller feature of Speex 1.1.10. For that i hav added speex_echo.h in my code.and have called the function. till compilation everything is fine but on linking it is giving me uresolved symbol error ... speexenc.obj : error LNK2001: unresolved external symbol _speex_echo_state_init Debug/speexenc.exe : fatal error LNK1120: 1 unresolved externals Error executing

dear all I got this error when I try to use libspeex in visual studio. when I include speex/speex_echo.h, it complains about unresolved external symbol _speex_echo_state_init. And when I include mdf.c into the source, it generates whole bunch of errors. Any idea? thanks so much --------------------Configuration: audio - Win32 Debug-------------------- Compiling... mdf.c C:\Documents and