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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

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

If you want to test the AEC, I recommend using the svn version of Speex. Jean-Marc Aymeric Moizard a ?crit : > > I finally solved my problem: > > My **static** speex library is compiled with: > -D_LIB > > While I should compile it with: > -D_USRDLL > > I'm not able to understand completely why: anyway, it > now can be compiled without breaking anything.

On Wed, 6 Dec 2006, lianghu xu wrote: > Hi Jean-Marc, > > I remember you said that the SDP tells the mode information. Right. But a decoder like speex can always be configured from information contained in the incoming stream. That makes your application more interroperable. Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 -

In a "normal" frame that doesn't have in-band signalling (which nobody seems to use anyway), the frame starts with the narrowband information. If the frame is in wideband, then this will be followed by a bit of wideband information that starts with a bit set to 1 (narrowband starts with a bit set to 0). Have a look at what the Speex decoder does. You may actually want to have similar

On Mon, 6 Nov 2006, Jean-Marc Valin wrote: > That's quite a strange problem. Considering that Speex doesn't have any > "init code" (stuff that executes when it's loaded), there's no way this > could be a bug in Speex. True. > I see two likely possibilities (plus lots of > unlikely): > 1) A bug in your code. Linking with Speex pushes your code to a

Hi, I'm experiencing weird problems while testing speex (I want to test the echo canceller first) on PDA with WM5 version pocket PC 2005. Here is the strange scenario: 1-> App running without being linked with speex -> no problem 2-> App running while being linked with speex -> crash at always the same place. NO ACCESS TO

On Sat, Nov 1, 2008 at 2:40 PM, Aymeric Moizard <jack at atosc.org> wrote: > The vcproj file provided for 2005 in 1.2rc1 does not include > _USE_SSE when compiling the project for "Release_SSE". This > define is only used when compiling under "Release_Dynamic_SSE". > > Is this just because the vcproj file is wrong or is it because > this #define must not

Hi, Speex-dev: ? I have some problems about complie speex on pxa-270. I use libspeex.so.1.4.0 at my platform after complie speex and all function is ok, but the cpu loading is 99%. How can i do to reduce the cpu loading? ? note: 1. My Platform are os: linux 2.6.19 , cc: gcc-2.95.3 , cpu: pxa-270, ogg: libogg-1.1.3, speex: speex-1.2beta3 2. I configure speex by the follow command. ./configure

Sorry, got the subject wrong in the last mail. My e-mail client is playing games with me...

I finally solved my problem: My **static** speex library is compiled with: -D_LIB While I should compile it with: -D_USRDLL I'm not able to understand completely why: anyway, it now can be compiled without breaking anything. I'll now test wether the aec have an impact or not :) tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 -

Dear Sir/Miss: I'm very sorry to bother you,but I've found a bug in the Speex,Version 1.2beta1,and also I've have some questions. I'm deeply impressed by the work you have done in the coding and decoding of speech signal.But when using speex,I encoutered a bug,that is when in 32K mode,the window size is still 200 samples,but not 400 samples,the bug will bring the problem of

Hi Jean-Marc, I used the following code after init of the encoder to get the frame size of my RTP packets. char pcm_data[3200]; char speex_data[3200]; SpeexBits bits; memset(pcm_data, '\0', 3200); memset(speex_data, '\0', 3200); speex_bits_init(&bits); speex_bits_reset(&bits); speex_encode_int(codec_info->enc,

Aymeric Moizard a ?crit : > > I'm working on the echo canceller with the SVN, with Windows Mobile 5 > PocketPC. > > Obviously, I can't make it run correctly. What happens/doesn't happen? > My question: If I use the SPEEX_PREPROCESS_SET_ECHO_STATE, I have > changes in the sound: noise seems to be removed. Noise should be removed even without

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

Hi, We are starting a new project, and are trying to decide the best way to proceed. We want to setup a LAMP configuration using Centos, something we have been doing in the past with great success. The question is load balancing. We antisipate the potential for the system to receive 500,000 requests/ day with in the next year. We want to plan for that extra load now as we start the project. What would you suggest for setups for multiple servers for redundancy and load balancing? I have setup MySQL replication and that works fine bu...

> In those cases, when you get let's say 1000 packets of 20ms from the mic > you may have only 990 packets of 20ms from RTP incoming stream. > > Thus, before sending outgoing mic/RTP stream, you would wait for 1000 > incoming packets: where last packet in fact arrive 10*20ms = 200ms > after it was supposed to. I have from my experience already seen 4s > of clock deviation

Isn't this the same problem described starting at the bottom of this page? http://www.embeddedstar.com/articles/2003/7/article20030720-11.html Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > As it says in 5.4.1 of the good book "Using a different soundcard to do > > the capture and playback will *not* work, regardless of what you may > > think.

On Thu, 16 Oct 2008, Jean-Marc Valin wrote: > Aymeric Moizard a ?crit : >>> None of that is defined yet, though I'm open to suggestions on how to do >>> the mapping. >> >> CELT/44100 and CELT/48000 >> a=fmtp:105 stereo=on >> >> probably a latency value? > > It would definitely need a frame_size value > >> "CELT"

This paper is very interesting. Just to add my own opinion on this: My main problem concerning those elements is to synchronize different soundcard on different PC! The aec is working on both outgoing (mic) and incomnig (RTP) stream. The problem most show up with USB wich appear to have very different clock than "classic" soundcard hardware. In those cases, when you get let's say