similar to: AEC Intermittently Fails

> So 9316 works and 9320 doesn't? How about latest SVN. I just ran > everything in valgrind and saw no error at all. Can you give more info > on how to reproduce (with speexenc)? > > Jean-Marc > I went to check my code and it turned out to be a fault in the speex_encoder_destroy being used to destroy a decoder state. It seems that the new revision thriggered this error.

I have been working on a project importing Speex to a embedded platform for a while. When I tried to shut down the macro TWO_PATH in Speex AEC to save space, an error occurred. Dbf is undefined. The 1.2rc1 code is like this Dbf = 0; See = 0; #ifdef TWO_PATH /* Difference in response, this is used to estimate the variance of our residual power estimate */ for (chan = 0; chan < C;

Hi, Can anybody show me how to enable stereo AEC, I tried with the following code and the result is bad, degraded output. Init: ec_state = speex_echo_state_init_mc(frame_size, aec_tail, 2, 2); speex_echo_ctl(ec_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sample_rate); preprocess_state_left = speex_preprocess_state_init(frame_size, sample_rate); preprocess_state_right =

I've recently made changes to the AEC. Please try the code in SVN and see if it works better. Jean-Marc Le jeudi 03 novembre 2005 ? 22:36 -0800, Jason Harper a ?crit : > I've tried some further debugging to see what mdf is > actually doing. Instead of sending: > tmp_out = (float)ref[i] - st->y[i+st->frame_size] > to the output, I just sent >

Sorry, I've forgot to mention init params: Frame_size is 441 samples and filter_length is 4096. ..don't think it makes any sense though.. Regards, Anton A. Shpakovsky From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On Behalf Of Anton A. Shpakovsky Sent: Tuesday, June 15, 2010 4:15 PM To: speex-dev at xiph.org Subject: [Speex-dev] AEC init crashes

This kind of behaviour is odd. One of the reason could be the fact that you're using a really long impulse response. Try syncing your signals and making the tail length more in the order of 100 ms to 300 ms. Jean-Marc Le dimanche 06 novembre 2005 ? 21:25 -0800, Jason Harper a ?crit : > Thanks for alerting me to the new changes. I just > tried the latest code from SVN, but

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.

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

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'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

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

Hi Guys, I'm attempting to use the libspeexdsp component with a Coldfire M5329. I've downloaded the speex-1.2beta3 and configured using the following: ./configure --host=m68k-uclinux --prefix=/home/uclinux/uClinux-dist --enable-fixed-point LDFLAGS='-Wl,-elf2flt' The library builds fine (or appears to) an is placed in the subdir 'lib' of the prefix shown above; my routine

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

If the left and right channels are processed separately as you do, the parameter Mic-Number in speex_echo_state_init_mc should be 1, not 2 On Wed, Feb 10, 2010 at 4:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello, I've just caught a strange crash in speex_echo_state_init routine. It happened only on one WinXP machine, while on others using XP, Vista and 7 everything is fine. Crash occures in mdf.c line 434: st->spec_average = DIV32_16(SHL32(EXTEND32(st->frame_size), 15), st->sampling_rate); Got any ideas or should I provide more information of the OS? Thanks a lot!

I've tried some further debugging to see what mdf is actually doing. Instead of sending: tmp_out = (float)ref[i] - st->y[i+st->frame_size] to the output, I just sent st->y[i+st->frame_size] to see what was being subtracted from the microphone input. When I open this in Audacity, I see a very small signal at about -40dBm. The actual echo in my sample has a power closer to -20dBm.

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

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 output have very fine tolerances on most standard audio