[Speex-dev] Acoustic echo cancellation

[Yue Meng Chen]

Assumption of a quiet room is inapplicable in reality. Plus, the impulse
response might change in the middle of a call … so, you basically need to
find a good initial alignment point, then, track it along the way. Make it
reliable is very much the key.

 

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From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On
Behalf Of Li Maoquan
Sent: Wednesday, April 20, 2011 5:36 PM
To: speex-dev at xiph.org
Subject: Re: [Speex-dev] Acoustic echo cancellation

 

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 AEC is sensiive to mismatch between sample rates of capturing and
rendering. But most low-cost
computer soundcards have this problem.

 




At 2011-04-21 03:00:01，speex-dev-request at xiph.org wrote:
>> >>>
>> >>> 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 requires aggregation for both playback and capture.
So
>> for
>> >>> playback, I coalesce 4 packets in a buffer and queue them as a larger
>> buffer
>> >>> for playback.
>> >>> On the send side, I read a large buffer (worth 4 packets) and send
them
>> out
>> >>> over time 20ms apart.
>> >>>
>> >>> I tried using speex_echo_playback just when a 160-sample packet
arrives
>> from
>> >>> the network, before coalescing and speex_echo_capture just before a
>> packet
>> >>> is sent out to the network but that doesn't seem to work properly
>> (doesn't
>> >>> cancel any echo).
>> >>
>> >> The most likely reason is that you didn't align the far-end and
near-end
>> samples.
>> >> So the filter can not converge.
>> >
>> >Thanks for your response. Can you please explain what you mean by
>> >align samples from near-end and far-end? And how is that usually
>> >accomplished?
>> 
>> You need to know the total delay caused by DAC buffer before speaker, ADC
>> buffer
>> after microphone and acoustic path between speaker and microphone. Simply
>> to say,
>> if you play an impluse signal and its first echo appears after N sample
>> cycles,
>> you can call N as the delay between y (echo in near-end signal) and x
>> (far-end
>> signal). Then you can buffer far-end signal for N-M cycles before sending
>> to AEC.
>> M is a little number (such as 100) in order to avoid filter failure when
>> echo
>> path drifts.
>> 
>> 
>Thanks again. I am trying to model the delay between the near and far end
>signals using a circular queue of length n. Every time a frame is received
>and queued for playback, it is also entered into the queue. Each frame
being
>read from the mic is echo-cancelled ( speex_echo_cancellation ) using the
>oldest frame in the queue if the queue is filled up, thus I am cancelling
>the recorded frame using a playback frame that is N-frames old.
> 
>I have played with different values of N from 2 to 50 (320 samples to 8000
>samples), attempting to align the input and output but the cancellation
>doesn't seem to work. The echo is steady as ever.
> 
>Is this model correct and expected to converge with a right value of "N"?
Or
>do I need some other adaptation to account for drifts here. Right now, it's
>a black box for me. I am not sure how to get some feedback from this system
>to tune the AEC (and the delay parameters) correctly.
> 
>Also, I did not follow the use of "M" in your description above and how it
>helps with drifts. My queue stores frames (160 samples each). So a number
of
>100 samples seems too small.
> 
>Btw, I am assuming that speex AEC API can be used even though I am not
using
>the speex encoder/decoder.
> 
> 
> 
>> >>
>> >>>> So, in this scenario above, please recommend a good place to insert
>> >>> speex_echo_playback and speex_echo_capture. Should I be just before
the
>> read
>> >>> and write to hardware? In that case, should I use a larger "frame
size"
>> of
>> >>> 160 samples x 4?
>> >>
>> >> Of course you can set frame size to 160*4. Otherwise you can feed
>> samples 4 times
>> >> to the AEC if you don't want to modify the frame size.
>> >>
>> >>>
>> >> Thanks in advance,
>> >> Daniel.
 






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