similar to: speex1.2-beta2 and noise suppression problem

Hi, I'm trying to compile speex1.2-rc1 for Symbian 3th edition. I get some errors: ***Invoking bldmake command perl.exe -S bldmake.pl bldfiles winscw \Users\Maurizio\University\ProgettoInterdisciplinare\speex-1.2rc1\speex-1.2rc1\symbian\BLD.INF FATAL ERROR(S): \Users\Maurizio\University\ProgettoInterdisciplinare\speex-1.2rc1\speex-1.2rc1\symbian\BLD.INF(37) : Exported source file

Hello R users, I am excited to introduce a new package for R which is now available on CRAN called ?backbone?. The package can be found here: https://cran.r-project.org/web/packages/backbone/index.html The backbone package provides methods for extracting from a weighted graph a binary or signed backbone that retains only the significant edges. The user may input a weighted graph, or a bipartite

Hello R users, I am excited to introduce a new package for R which is now available on CRAN called ?backbone?. The package can be found here: https://cran.r-project.org/web/packages/backbone/index.html The backbone package provides methods for extracting from a weighted graph a binary or signed backbone that retains only the significant edges. The user may input a weighted graph, or a bipartite

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: =====================================

hy, here is my speex encoder/decoder .. the sampleRate i use is 16000 and quality,complexity are at 5. can someone take a look in to the code and see if there is something that is making that robot voice here is a link to colored and numbered code, same as below http://barvanjekode.gama.us/temp/1257361243.html thanx. code ----------------------- #include "codec.h"

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Anton A. Shpakovsky <saa <at> tomsksoft.com> writes: > > As for me - speex_echo_cancellation is a better choise. Try using it in > capture thread instead > of those speex_echo_capture and speex_echo_playback functions. > > And please, describe your problem in details. Cause the fact that you > "didn get echo cancellation" > doesn't mean you are

hy, recording and playback without speex is working perfectly Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us | |--------------------------| | tel: 00386 31 457 627 | +--------------------------+ 2009/9/2 Tim Rijavec <rico at gama.us> > hy, > > here is my speex encoder/decoder .. the sampleRate i use is 16000 and >

On 24 Jun 2015, at 15:33, Martin Jangowski wrote: > I regularly broadcast with 48kBit streaming rate (mp3, mono, 44kHz). > My fallback-file is a 244kBit, stereo, 44kHz, mp3 file. Works like a > charme.... That depends on the player, if it works you are lucky. It might not work for other formats or with some players.

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

Jesus, Unfortunately, FFT and magic algorithms don't work (yet?). You might want to try this if you're not satisfied with Speex VAD: http://lists.xiph.org/pipermail/speex-dev/2008-August/006860.html It won't perform any miracles, but I think it works pretty well and is easy to tweak. Tom >---- Original Message ---- >From: jmorion at toomeeting.com >To: speex-dev at

constant prediction see this page http://flac.sourceforge.net/format.html oversampling.. i maean digitally change the wave file rate form 44khz to 440 khz it make next sample easyer predictable if i'll found my old work i'll tell you how much compression i could achieve

> The cost of down-sampling, if done efficiently, is probably less then > the cost difference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz

Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }

On Fri, 16 Nov 2001, Jerome Alet wrote: > one thing that would be nice in DarkIce would be to allow the user to pass > specific reencoding options for each server, e.g. DarkIce could acquire > the audio in stereo and send it to a server in mono and in stereo to > another server, which is AFAICT impossible today. I agree! Also, something I've been looking for is a way to pull

I know speex is not supposed to do a great job compressing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down

The attached patch eliminates some warnings while compiling for ARM4 targets. It also simplifies the asm constraints a bit. Now we can use the ARM4 optimisations when compiling for PortalPlayer targets in Rockbox. Cheers, Dan -------------- next part -------------- A non-text attachment was scrubbed... Name: speex_arm4.patch Type: text/x-diff Size: 1550 bytes Desc: not available Url :

I have found that filter_mem2 fixed point does not match the inlined assembly version for arm4. Looking closer there appears to be an off by one error. It occurs when setting the value of mem at the end of the inner loop. In the C fixed point version this is done with a subtract. In the arm4 version instead of multiplying den by yi, it is multipled by nyi and then an add is done. This