similar to: Work with speex

I have been tyring to use the preprocessor standalone by feeding it pcm data from a headerless pcm file (mono, 22050hz) and then write the result to another file. Unfortunately, all I get back is noise (first frame is set to 0, rest is noise). Frame size is set to 440. I just modified testdenoise.c slightly to read/write from a file (see below) but I can't get useful results. What am I

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

Hi Richard, Richard McGravie a ?crit : > We are looking for a consultant to assistance in the following areas. > 1. Shim in the pre filter from speex to all trunk interfaces in > asterisk. (E1, ISDN and FXO) > > 2. Optimize and improve the pre-filter for airplane noise floor. > Is this something you would be willing to assist us with or do you have > any contacts that could

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

I'm getting confused, what's the problem exactly? The AGC doesn't remove the DC or it adds one? Jean-Marc Aymeric Moizard a ?crit : > Hi Jean-Marc & all, > > I'm currently working on a wav file with very low volume. > This wav also contains a DC component. > > Using testdenoise with modification (AGC turned on with > 20000.0 value) I'm able to

These are my results with the Speex AGC and denoiser: Over the past couple of months, I've spent probably 10-20 hours of conversation with Speex (wideband) doing AGC + denoiser on both ends of the call. I think it does an excellent job of removing most noise while keeping speech intact. Yes, it's possible to make some constant noise (like humming) and have that get removed, but

Hi, I am considering to use Speex in a particular environment and I want to ask if you thought it realistic. I am going to use Wireless SIP (G.711) handsets in an environment that is typcially noisey (background), connecting to an Asterisk PBX. I am considering converting the G.711 to Speex so I can "massage" out the background noise. Is this realistic? Any input would be great.

Hello! I tried quite a few things before I could reach out to the community. The problem I am facing is that I am unable to run SPEEX in fixed point at WB 15ms or 20ms frameSize for some files. I tried using rc1.1/1.2/1.3 releases and all exhibit the same issue. I tried enabling the debug option during configuration step: ./configure –enable-fixed-point-debug; but this gives bit exact

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

The absolute values may be different for a variety of reasons; you can hardwire them all to 0 to see if your noise goes away. Make sure the values on the decompressed (PCM) side are little-endian (low byte first in memory), assuming you're working in Windows on x86. -- john RAD Game Tools -----Original Message----- From: Carine Liang [mailto:carineliang@yahoo.com.sg] Sent: Sunday,

Can you post the input and output file on the web and specify what options you're using (use speexenc/speexdec, *not* your own code). Jean-Marc Yanxin Cui a ?crit : > Would it help if I scale the input wav data before encoding to spx? > One interesting thing is if I encode the same wav file to spx(or > decode the same spx file to wav ) on Windows platform, then there is > no

Would it help if I scale the input wav data before encoding to spx? One interesting thing is if I encode the same wav file to spx(or decode the same spx file to wav ) on Windows platform, then there is no such big clicking noise, samething is true if I use floating_point speex library to encode/decode on WM2003. Problem happens only on WM2003 using FIXED_POINT library. Thanks Yanxin -----

When built with --with-fft=gpl-fftw3, libspeexdsp is not linked with libfftw3, while it uses symbols from it. This means every object linked with libspeexdsp should also be linked with libfftw3, but it is not something reflected in the pkgconfig file. Instead, link libspeexdsp with libfftw3 when --with-fft=gpl-fftw3, and remove explicit link with libfftw3 in objects build by the speex Makefile.

I am running speex 1.2beta1 on WM2003 using FIXED_POINT option, I got big clicking noise during encoding or decoding randomly, I got more noise if I spoke louder or made the MIC sensitivity higher, for example, when I encode the wav file to spx file, the original saved wav file does not have such noise, the noise is added only after it's converted to spx format. The floating point version of

Sorry everyone, but I figured it out; it's working now. The problem was in my monoToStereo and stereoToMono functions (the Linux OSS driver on the ARM side only supports stereo so I had to convert it to mono before feeding it to Speex); also I had an alignment issue with my buffers I was using on the DSP side. Thanks Jim/Jean-Marc for your help! Regards, Andy ----- Original Message ----

Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles

Hi John, I noticed the values after decode is very much different from the values of the original PCM samples. For cases when the PCM sample values are constant (because no one is speaking), the decoded values fluctuate too. These fluatuations are causing the background noise. Is my observation expected? Thanks, Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To:

Hi Jean-Marc, Thanks. I've got the code for sampleenc and sampledec. But I didn't know how to verify whether the decoded file is correct. So I tried to analyse the decoded output and compile with my input. Test case: PCM samples, 8 bit, 8000 samples per sec. When I pulled all my input to 0s, after encode and decode, the output fluctuates around -0.12 to 0.2 (in float), so when finally

I'm not sure if this is due to the same problem, but I get the same clipping behavior when trying to use VAD+denoiser on a particular mic+speaker setup that has significant background noise (enough to really foul up the codec should the denoiser be turned off.) Leaving VAD off and transmitting continuously seems to be the only way to avoid this clipping. Another person I regularly

Hi, I've been trying to get Speex to compile and run on the DSP of TI's new DaVinci System-On-Chip processor, which has both an ARM (ARM926) and a DSP (C64+, based on the C6400). I used the latest code (1.2beta) and followed the example in the speex- 1.2beta1/ti/speex_C64_test trunk to build the Speex library for the DSP. Basically I have a loopback application on the ARM that samples