similar to: AEC troubleshooting

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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

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 am a new user of speex.I am currently working on speex frames and I have some questions. I am using narrowband and long tail length, and it works very well with speex test DEMO. But it is very difficult to have speaker input in perfect sync with mic input. Speex does not work at all. Any suggestion? Regards? -------------- next part -------------- An HTML attachment was scrubbed...

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

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

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

The reasons I have posted these questions are: 1) To find out if Speex can take advantage of SIMD extensions. 2) To maybe learn from someone with previous experience in optimizing Speex for moderns x86 architectures before I set off trying all kinds of things on my own. See answers inline: 2009/6/15 Tom Grandgent <tom at grandgent.com> > Why haven't you tried using release build

Sorry for not being accurate in the original question. My testing environment is set to 20ms frames (PCM 16 8 1 to be precise). I have verified that all the frames that I feed to the AEC have the fixed size of 320 bytes. So this is why I am presuming that the the dumped files should be of equal size. The output signal is good and not distorted. It has echo, though. In a try to diagnose that I

I have a question about the overall performance of Speex and what I can do to improve it. I'm running Speex Windows x86, Visual C++ EE compiler. I will say right away that I've only compiled debug so far and used no compiler optimizations at all. I use the uwb-mode, preprocessing, denoising and echo cancellation. I've noticed that speex consumes a lot of cpu resources. When I run this

I believe there is a documentation error regarding speex_preprocess_ctl(preprocess_state, request, ptr); The manual states: SPEEX_PREPROCESS_SET_DENOISE Turns denoising on(1) or off(2) SPEEX_PREPROCESS_SET_AGC Turns automatic gain control (AGC) on(1) or off(2) I noticed that setting a value of 0 turns the setting off. Setting a value of 1 or 2 turns the setting on. While I do think 1 and 0 is more

Hi Guys, I am trying to incorporate the speex AEC into my RTP stack. Trying to diagnose why the echo is not being cancelled I enabled DUMP_ECHO_CANCEL_DATA as per the docs. The problem is that the dump files the AEC generates are of different size. As I understand this and looking at echo_diagnostic.m, the files are supposed to be of same size. Can someone give me a hint as to why this could

Hi everyone, I think this should be interesting to all of those with echo cancellation problems. I finally check in some code to make these problems easier to debug. You'll need to have svn version for this. First, you need to manually define DUMP_ECHO_CANCEL_DATA in the compilation (sorry configure switch for now). With that, the AEC will automatically save the near-end, far-end and output

> Thanks for the great work. It indeed helps to diagnose of echo > canceller and narrow down the > problem area. Good, so it worked for you? > I am not an expert in the signaling processing but I > wonder if it is possible > to add the linearity check in the echo_diagnostic function? Theoretically, yes. In practice, I'm not sure how accurate it would be. Do you have any

On 2010-03-16 06:35, Greger Burman wrote: > Ok. Thanks J-M for clearing that up. > What if you mix up the polarity on one speaker (180 degree phase > change), would that matter? Not at all. It's still all linear. You can even apply a different equalizer to each speaker and it'll still be linear. Jean-Marc > cheers > Greger > > 2010/3/15 Jean-Marc Valin

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

Hi I've been debugging and troubleshooting echo cancellation myself recently and I have made some observations. First of all playback and recording must be synchronized. There cannot be any clock drift between the microphone signal and the speaker (echo) signal. This has been said many times in the mailing list, but I will repeat it anyway. You have to first make sure that this is not your

I'm trying to understand a bit more about the inner workings of the speex resampler. I want to do resampling on a voip stream. High quality and low latency is important. 1) What does speex_resampler_magic() do? 2) I find that resampling has quite an impact on performance and I'm trying to figure out where the cpu cycles are going and see what can be done to reduce the cost. I ran a profile

Ok. Thanks J-M for clearing that up. What if you mix up the polarity on one speaker (180 degree phase change), would that matter? cheers Greger 2010/3/15 Jean-Marc Valin <Jean-Marc.Valin at usherbrooke.ca> > 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 >