similar to: Sound card problem in acoustic echo

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

Hi All, In order to deal with acoustic echo cancellation problems of most PCs which sound cards have different capture and play frequencies. I made a trial. At first, a 1000Hz sine wave is played for a long time via a speaker and its acoustic echo is recoreded. Seconds, get the frequency of the echo by a FFT analyser. So the difference between capture and play frequencies is obtained. Thirdly,

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

Hi Underwood, Thank you for your help. I agree with your opinion. But it is almost impossible to further reduce the frequent difference between play and capture. 1. I used a 2^18 step FFT to analyse the echo frequency. So the freq resolution is 8000HZ/(2^17)=0.0625Hz. The analyser need at least 2^18/8000=32 seconds acoustic echo record signal from the microphone. Better freq resolution relies

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

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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

Hi all, We have discussed so many about sampling rate asynchronous (or offset) between rendering (D/A converter) and capturing (A/D converter) of most PC soundcards. It seems all acoustic echo cancellers, include AEC in speex, can not deal with this trouble, because it causes a drift of echo path and also buffer overflow and underflow which jumps the delay of echo path seriously. Unfortunately,

Simply to say, in a quiet room, you can play a impulse signal and then find it's impulse response signal from the microphone. For example, if the delay between the impulse signal and its response signal range from 500 to 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the filter length to 4000. It is also called to align far-end signal and near-end signal. BTW: Speex

>>>> Hi, >>> >>> I have a scenario in a mobile VoIP app that requires echo cancellation but >>> is somewhat different from what's described in the docs. >>> >>> Audio is received from and sent to the network at 8000Hz. Each packet >>> contains 160 samples worth a playback of 20ms. >>> >>> But the hardware

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

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

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

Ubuntu 10.4, Wine 1.1.42, Native Instruments Kontakt 4, Akai MPK88 I've installed Kontakt (sampled musical instruments). Sound is working OK with the OSS driver, but the program doesn't respond to the USB keyboard. :- ( The keyboard shows up in `winecfg` under "Audio", OSS, MIDI In/Out devices. I checked the MIDI options for the Kontakt program and it DOES show the Akai

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

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

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

In winecfg is alsadrv selected but i cant see any asio audio device in a asio compatible aplication and no ports in jack too if i do: env WINEDEBUG=+asio wine '/home/xxx/.wine/drive_c/Program Files/FXpansion/Geist/Geist.exe' Code: p11-kit: couldn't load module: /usr/lib/i386-linux-gnu/pkcs11/gnome-keyring-pkcs11.so: /usr/lib/i386-linux-gnu/pkcs11/gnome-keyring-pkcs11.so: no se