similar to: AEC - Echo is cancelled however.....

I usually use the cross-correlation between the speaker signal and the mic signal. It works pretty well. The problem is the you need enough data in order to get a good S/N. I compute the cross-correlation between 1.5 second data of each signals. In this way the application detection of any anomalies is quite slow. I am looking for something faster, but I fear it will be something very application

2010/5/11 Elston Sa <jose at rebaca.com>: > Is there an API in speex or have you implemented your own? I don't know if there is an API in speex for this, probably not. There is a test for AEC which does something very similar. I implemented it starting from a FFT library. It is quite easy to do. > > I am currently trying to find one. But if you could point me to one it 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.

I tried cross-correlation. It does locate the sync point and it's almost perfect. I guess now I have to make it more perfect. Thanks for the pointers. On the side note: I have a test environment. Where I created the stream such a way that I have one "captured stream" purely with echo (i.e. no one else is speaking) and one stream with echo + someone speaking. I confirmed that I

Is there an API in speex or have you implemented your own? I am currently trying to find one. But if you could point me to one it will reduce my search time. :) Many Thanks, Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Tuesday, May 11, 2010 5:55 PM To: Elston Sa Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] AEC - Echo is cancelled

Marco, Thank you for your response. Is there a way I can determine if the synchronization is perfect or not? If not by how much it is off. - Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Monday, May 10, 2010 3:46 PM To: Elston Sa Subject: Re: [Speex-dev] AEC - Echo is cancelled however..... Hi Helton, it is now several months that I am using

Well, AFAIK AEC engine needs time to adapt. So probably that's why at the beginning of the conversation you hear the consequences of this adaptation. Btw, what sound system do you have? I mean laptop or arbitrary acoustic system? I'd like to know, cause I couldn't get enough echo cancellation with speex in my environment with standalone speakers AT ALL. -----Original Message-----

Hello, I am using the speex AEC in a real time application. I have experienced that when the mic and the speakers tracks are on sync or with a small delay the AEC works very well. I understood that when they are out of sync the AEC cannot works, so what the "user" should do is to focus in order to have the tracks on sync. Since I am working in an environment where it is not rare to

The problem is that now I have two files very similar (one for mic, the other for speaker). If I play them together I can hardly say that they are two different file for most of the file length. When I try to cancel the echo, I obtain as output almost the same input. The update_foreground variable is always 0. I don't get the conditions which must be fulfilled in order to have

If more than one speaker receives the *same* signal, it doesn't matter the number of speakers. It only gets tricky when the speakers are playing slightly different signals (e.g. from a stereo song). Jean-Marc Quoting Greger Burman <greger at mobile-robotics.com>: > One thing I can think of is if you are using two or more speakers. If the > speakers are not at the exact same

Hello. I have the following situation. AEC is used in network chat software over DirectSound API. Echo and reference signals are almost aligned (delay is no more than 30ms). When echo is emulated in notebook (built-in speakers + mic) everything goes fine and echo is cancelled. But when configuration includes stand-alone speakers and mic no echo is removed. Audio is in 22050 hz at 16 bit

Hi, I would like to have some more information about the parameter of speex_echo_state_init. I have a clear idea of the importance of the filter_length in the echo cancellation, what it is more obscure for me is the role of frame_size and how its value affect the algorithm. Let use, for example, frame_size = 400, would I obtain the same result of using frame_size = 200 but with half method call?

I have build celt with FIXED_POINT option (latest 0.6.1 as well as from the git repo) on windows. However I am not getting a valid output (all samples are saturated) when I try to decode with this version. The input file was encoded with the same fixed point version. Does fixed point version work at all? Following are the command line settings; Samplerate; 48000 Channels: 2 Framesize: 256

On 2010-03-16 14:22, Josh Gargus wrote: >> If more than one speaker receives the *same* signal, it doesn't >> matter the number of speakers. It only gets tricky when the >> speakers are playing slightly different signals (e.g. from a stereo >> song). > > Does "tricky" mean that the Speex AEC won't handle such situations > well? Or just that you

Hi Anton Thanks for the info... What if I use this function do I still need to sync the playback and capture? Assuming that the captured voice (*rec) already contains the echo which I need to cancel out... -------------------------------- void speex_echo_capture (SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out) Parameters: st Echo canceller state rec signal

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

I have tried both linear and non-linear (residual echo) parts of SPEEX AEC on MS Windows platform. Great job. Most echo can be cancelled. However, the voice volume becomes too small. Does any one have suggestions that makes voice volume louder but echo still can be cancelled? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Is it possible to use only the speex echo cancellation module w/o using the speex codec? Here's the scenario: 1. I have my voice recorded in PCM audio file format 2. I want to cleanup the recorded voice by removing any echo included in the audio file/buffer 3. can I just use the ff APIs? - SpeexEchoState* speex_echo_state_init() - int speex_echo_ctl() - void speex_echo_capture()

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, I couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal