similar to: Performance and Optimization

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

Quoting Greger Burman <greger at mobile-robotics.com>: > The reasons I have posted these questions are: 1) To find out if Speex can > take advantage of SIMD extensions. If you define _USE_SSE, CELT is already able to use SSE instructions. You must be careful on Windows though because it doesn't support C99 var-arrays and alloca() isn't properly aligned for SSE, so you have to

Why haven't you tried using release build with compiler optimizations? It's quite possible that the performance picture could be substantially different. You might end up wasting a lot of time if you do much performance analysis or optimization on a debug build. Debug build not only has no optimization - it also has extra checks that may have a significant performance impact

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

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

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

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

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

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

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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...

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

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

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

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 >

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

On Mar 15, 2010, at 8:46 AM, Jean-Marc Valin 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 had to be

Hi, I have a setup that rsyncs (using the rsync protocol without rsh/ssh logins) from one windows server (A) to a NAS (B) via a second windows server (C) (A is on a separate location from B & C, and the NAS doesn't support rsync directly), where C stores data on an SMB share on B. My problem is that I often get "No such host or network path (136)" from rsync (robocopy or

Hi, I've used rsync successfully for several years, syncing between two Windows 2000 servers using daemon mode, but today I stumbled accross something peculiar. I'm using cygwin with rsync 2.6.0 at both ends (the latest available at this date) and I have a file that rsync considers up to date even though both the md5 and a normal diff show differences. I've tried calling rsync