similar to: AEC algorithm explained

Hi: I have been trying to test the speex AEC by playing a voice sample (pre-recorded) as a echo and capturing a new voice sample and playing the pre-recorded one, for reference. [Sampling rate 8 KHz, frame size 64] For some reason, I dont see any removal of echo. Does anybody have a idea of * Testing the AEC in some other mechanism or * Getting some sample files for testing AEC Thanks Su ---

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,

Hi: Can anyone give some insight on: 1) As per the Speex manual, there is a mention of the relationship between input frame and echo frame (Pg 19, Para 2). I was wondering of what would be the ideal delay between these two frames, so that the echo is already fed to the canceller and the echo cancellation filter is not inefficient. 2) Corresponding to the above mentioned delay, what filter

On Mon, Aug 11, 2008 at 12:34 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > OK, here's what happens. There is indeed a small difference between > beta3 and rc1, but the fundamental problem isn't there. I've attached > plots of the speaker signal (blue) alongside the mic signal (green). You > can see the delay is in the order of 1000 samples.

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 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: By 'no success', I mean result gets worse in terms that, there comes in a shrill noise with high gain and also *no* attenuation of the echo. I am trying to validate the DC offset part. But can you please feed some light on how does the DC offset matter for AEC. As I have already mentioned, I tried adding silence (from 2 ms to 10 ms) in the start of the reference signal. This is for

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

Hi everyone, I've just made the first public release of some new *experimental* codec work I've been doing (part of the vague Ghost project) with help from Monty and Timothy. This is mainly intended with developers with DSP knowledge, not for doing anything useful with it (but it does encode and decode already). Also, the main idea is *not* to replace either Speex or Vorbis, but to code

When I ran test 4 as originally described there is substantial echo cancellation (but not as good as when the files are perfectly aligned). When I invert the inputs, there is no noticeable cancellation. I'm using testecho with the preprocess line commented out. Preprocess seems to work very well at cleaning up the residual echo when mdf does its job, so I'm just focusing on testing mdf.

On Sat, Aug 9, 2008 at 12:59 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > Hi Benny, > > Can you send me your pair of testecho input files that work well with > beta3 and not with rc1? I'll have a look. > > Thanks for the help. The files are on their way now, the upload will take few more minutes to complete. In the mean time let me explain more

I wasn't implying that anyone do anything about it, just that's it a real problem. Unfortunately, most of the crappy sound cards are the ones that ship with your typical PC, so it's just something that people should be aware of. The solution is pretty straightforward -- just resample the audio data in real time using a reference clock. -----Original Message----- From: Jean-Marc

Thanks for taking a look. There was no motion; however you are right about sampling from a different card. The speaker is connected to the Sound Blaster card, while the microphone is part of a USB webcam. I don't think that this is likely to be too unusual a configuration among users. I can retry the test using a sound card microphone to see if there is a difference. If it turns out that

Thanks Jean, I missed that detail. I have a few more compile errors if you could help me with it => In function speex_echo_state_init in mdf.c I am getting error: "undefined reference to exp". I don't see a fixed point implementation of "exp". Can you point me to the correct files. I am using the following files from speex 1.2 beta1 => mdf.c,

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

Hello, Is it possible to adjust the level of the denoiser ? 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. The AEC too works very well in a old beta, but now, I think there is a problem... I have tested it with the same sound card (WB 16000). In any case, your project is the best VOIP library in this

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

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 somewhere in some of the OS resampling algorithms, but I've seen that on many machines.