similar to: AEC and different sound cards

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

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,

Try using SPEEX_ECHO_SET_SAMPLING_RATE to specify your sampling rate. Also, don't forget that the tail need to be longer (proportional to the sampling rate). Last thing, if you use resampling, make sure you use a decent resampler (the Speex one is fine) because otherwise, any aliasing left will not be cancelled. Jean-Marc Mihai Balea a ?crit : > Hi all, > > I am attempting to test

Hi all, I am attempting to test AEC behavior at various sample rates. I ran a little experiment: I recorded a 10 seconds voice clip and the resampled at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000. I have a small applications that plays a wave file, records whatever comes in from the microphone and applies the Speex AEC and preprocessor on the input. It then saves the raw

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

Hi, is it somehow possible to use the internal data of the echo canceller to get some kind of a double talk detector? This would allow to perform echo suppression in scenarios where the cancellation fails, like with webcams, where AEC doesn't work at all. Best regards Andreas

> Today's Topics: > > 1. AEC with WB mode (Jean-Christophe.Berge@etu.enseeiht.fr) > 2. Multiple frame encode and decode (Reza Fatahillah) > 3. cant link speex_echo.h (jesus) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Sep 2006 08:46:03 +0200 (CEST) > From:

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

> 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 ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the

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

I'm developing an application that have a video conference component. For that I need echo cancellation, and is looking around for algorithms/implementations of that, and the one in speex is an alternative. In the documentation for speex I find the following sentence however. "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may

Hello, We are experiencing a few problems with Speex AEC. We are using it to process audio data on a real time stream over IP with Speex codec (frame size = 320). We initialize the echo state like this : SpeexEchoState * echo_state; echo_state = speex_echo_state_init(320,512); int sr = 16000; speex_echo_ctl(echo_state, SPEEX_ECHO_SET_SAMPLING_RATE,&sr); We are not really sure about

Hello, The denoiser was good with a car noise or a raod noise before the r11739. But for some situation, the actual version is perfect. That's why I think that if we can adjust the agressivity of the denoiser, we can respond to all of the situations. Maybe two parameters should be enough : high or low. For the AEC, I have to test with older versions because it's another person that

Hi, I have a question regarding the viability of using Speex on the Windows Phone 8. The constraints of my project have led me to use the Windows Audio Session API (WASAPI) for the capturing and rendering of audio. It works pretty dang well, except for the fact that there is no AEC unit. Hence my use of your library. I've been able to get some pretty good echo cancellation, but there are

Ouss a ?crit : > Hello, > > Is it possible to adjust the level of the denoiser ? Not for now. > In an old beta, before you change the aggressiveness of the denoiser, it > works very well with the noise of a car, a road etc... but not now. When did that change? > The AEC too works very well in a old beta, but now, I think there is a > problem... When did that change?

> 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

Hello, I'm using AEC for a VoIP application on mobile handsets. I am doing experiments to learn how to work with it, and I have a problem: As long as I play through the device's normal speaker and record using the mic, I have absolutely no clock drift (according to echo_diagnostic.m). The echo is being cancelled and all is fine. Once I route to the earpiece (and still use the mic, which

Ok, let me be more clear on this. AFAIK, Windows OS doesn't expose speaker input as other OS ( Linux, Mac OS...). That puts you in bad spot in using Speex AEC with windows. Only way to work is to use Soundcard with ASIO functionality which does give you speaker input. But then you would have to impose that requirement on all your users. I heard Speex AEC works great from developer who

Well, I am not the person to be asked about how Speex AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at