similar to: problem with echo cancellation

Hi Julia, Version 1.2-beta1 has a bug in the echo canceller. Try either svn or 1.1.12 -- or apply this patch to 1.2beta1: https://trac.xiph.org/changeset/11882 Jean-Marc julia rg a ?crit : > Hello Jean-Marc, > > > > I solved the variable delay problem, but I still have trouble with > speex_echo_cancel(). When i try testecho.c with clean speech for far-end input > and

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

Hello Jean-Marc and all, I?ve got a problem with speex_echo_cancel usage. I only managed to get an attenuation of 5-8 db using testecho.c and it depends heavily on the playback/capture delay. I used clear speech as far-end input and same speech with a bit of reverb, attenuation and delay as mic nput. I also tried using played and recorded speech for mic input. I tried different constant delays

I'm pretty much sure of it. When I test inverting the inputs, my output is pretty much the same as my speaker signal. Whereas the way that I normally test the output is my mic signal with very little attenuation. If you are interested I can send my test files; they are about 94KB each. -Jason --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Are you sure you're

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

Had a try. The reason why a simple delay is not that good is mainly due to the initialization of the filter parameter that still takes a few seconds (if they are perfectly in sync, you sort of get lucky). Otherwise, you real recording seems to have something odd in it. Are you sampling from a different card then the one that's playing the sound? or maybe the mic (or something else) in the room

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 Andras, others, Andras Kadinger <bandit <at> surfnonstop.com> writes: > > > Daniel, > I recommend you to start from a simple case and gradually progress > towards your goal. > Can you make things work with the "Speex in a Disco" (Example 6) > testcase at http://ns.surfnonstop.com/~bandit/speex/echocard1/ ? > These files

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

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, I am using speex-1.2 beta1 version. When I am using sample files from http://www.surfnonstop.com/~bandit/speex/echocard1/ ,example 6, as input to mdf through testecho the output is fine However when I am recording the signals in echocancel.cxx in OpalEchoCanceler::ReceivedPacket taking speaker signal from echo_buf and mike signal from ref_buf, echo cancellation is not affective. Can

Hello Bernard, Friday, December 13, 2002, 7:22:54 AM, you wrote: Bernard> I've one small request - an option on speexenc that allows you to Bernard> specify a speex file to append to, allowing you to concatenate Bernard> streams without losing quality by decoding & encoding. Ideally, it Bernard> would: But you can `cat speex1.ogg speex2.ogg> unionspeex.og` and still have

Are you sure you're not just inverting the two inputs? Jean-Marc On Wed, 2005-11-09 at 22:16 -0800, Jason Harper wrote: > I ran some further tests on mdf and here are the > results: > 1. reduced tail length to 100ms, aligned mic and > speaker signals to within 10ms - almost no echo > attenuation > 2. aligned mic and speaker signals to within 5 samples > - still almost

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.

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

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

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.

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

The echo suppression in SVN is shaping up quite nicely! I have also observed some - occasional, apparently data-dependent - misbehaviour in AGC. http://www.surfnonstop.com/~bandit/speex/echocard1/