similar to: AEC with different soundcards

AFAIK, that's a common point for all AECs. But some of them solve the problem by resampling on of the end to keep it in sync with the other. On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: > Thank you John. > > On 07/06/2009 11:03 PM, John Ridges wrote: > > ly synchronized, and therefore the clock drift adds a non-linear > factor to the audio path. The AEC

Hi ? I used this "sample counting " method to?resample and put my audio signals in synch. It worked perfectly?in XP machines using a SoundMax?audio card, but it failed in other XPs using Realtek cards. As seen on http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html?my application?continously checked my AEC level to slighly modify resample frequency, but convergence was

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

I've been testing the AEC and have experienced it working on one sound card, and not working when using different cards for capture/playback so I know it's a real problem. And yes I know the documentation says it won't work "regardless of what you may think." I'm the inquisitive type; I'm looking to understand just what exactly is happening here. One discussion in

Hi Shridhar, Sample rate conversion is not enough to solve this problem. I have tried this method several months ago. The first step is to measure the difference between sample rate of capturing and rendering. Then resampling (by what you said "sinc interpolation") one signal to eliminate the difference. The frequency step in my experiment is less than 0.1Hz. I have tried speex AEC

I understand that the capture/playback signals need to be sync'd for an AEC to adapt. I'm a little bit confused on the requirements of synchronous sampling between the near end (mic/speaker) and the far end (phone line). I have an embedded DSP system with mic and speaker getting 1msec packets containing 8 samples. We can watch the DSP and ISDN clock frames drift and every few minutes we

On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a version where the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's

Le vendredi 11 novembre 2005 ? 01:21 -0800, Duane Storey a ?crit : > This is a very real problem though.. I've encountered many sound cards that > use different clocks for input and output (even on the same card!) Also, if > you open up a sound device on windows at 8kHz, the microphone is often > around 8100Hz, while the output is 8000Hz.. I'm not sure if there's a bug >

Hi all, We have discussed so many about sampling rate asynchronous (or offset) between rendering (D/A converter) and capturing (A/D converter) of most PC soundcards. It seems all acoustic echo cancellers, include AEC in speex, can not deal with this trouble, because it causes a drift of echo path and also buffer overflow and underflow which jumps the delay of echo path seriously. Unfortunately,

May I still try to sync them using some kind of DPLL (digital phase locked loop) mechanism? Something like matching clocks and interpolating samples to try to sync them together? Do the interpolated samples get to be eligible for an aec in your opinion? Thanks, Dario > To everyone on the list: do *NOT* attempt to do echo cancellation with > signals sampled using different clocks. This will

Yes. I guessed that too, however I am not sure why it keeps repeating every time the user stops / pauses and starts speaking again in a single session. I am using a laptop with standalone speakers. For echo cancellation to work one has to make sure that the ref and echo buffers are synchronized. I guess this is the most common problem. -Elston -----Original Message----- From: Anton A.

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > Hi All, > Many Thanks to Underwood for her excellent review of our big trouble > which prevent LMS-based AEC algorithms to be used in most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > low-cost soundcards (In my experiments over more than 20 soundcards,

To everyone on the list: do *NOT* attempt to do echo cancellation with signals sampled using different clocks. This will *NOT* work. Just a 0.1% difference between the two sampling rates (it's sometimes worse than that) means that the impulse response drifts by 8 samples every second. There's just no way to efficiently track this. Or at least no way that doesn't involve something 100x

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 >

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

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

Hello? I am a new user of speex.I am currently working on speex frames and I have some questions. I am using narrowband and long tail length, and it works very well with speex test DEMO. But it is very difficult to have speaker input in perfect sync with mic input. Speex does not work at all. Any suggestion? Regards? -------------- next part -------------- An HTML attachment was scrubbed...