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Daniel,<br>
<br>
I recommend you to start from a simple case and gradually progress
towards your goal.<br>
<br>
Can you make things work with the "Speex in a Disco" (Example 6)
testcase at <a class="moz-txt-link-freetext" href="http://ns.surfnonstop.com/~bandit/speex/echocard1/">http://ns.surfnonstop.com/~bandit/speex/echocard1/</a> ?<br>
<br>
These files were captured via a stereo soundcard, so the channel
sample clocks are exactly synchronized, making for a simple baseline
testcase for AEC.<br>
<br>
Download the uncompressed far end and near end unaltered files
there, pretend in your code they were your inputs and outputs, and
make sure your results using them is about good as on the example.
If not, you probably have a problem in your code that you need to
fix first, before worrying about unsynced sample clocks.<br>
<br>
Andras<br>
<br>
2011.04.21. 10:20 keltezéssel, Daniel K írta:
<blockquote
cite="mid:BANLkTi=4YaL=7hYCfXp-3wwZuR=-P7kFkg@mail.gmail.com"
type="cite">
<div class="gmail_quote"><br>
</div>
<div class="gmail_quote">2011/4/20 Li Maoquan <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:limaoquan2000@126.com">limaoquan2000@126.com</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div class="im">Simply to say, in a quiet room, you can play a
impulse signal and then find it's impulse response signal
from the <br>
microphone. For example, if the delay between the impulse
signal and its response signal range from 500 to<br>
3000 cycles, you can buffer the far-end signal to 0-300
cycles and set the filter length to 4000. It is also called<br>
to align far-end signal and near-end signal.<br>
<br>
BTW: Speex AEC is sensiive to mismatch between sample rates
of capturing and rendering. But most low-cost<br>
computer soundcards have this problem.<br>
</div>
</blockquote>
<div><br>
</div>
<div>Based on your explanation above, I have attempted to
measure the delay between the near-end and far-end signal. I
send in a sharp impulse sound, have it play on the speaker and
see when it comes out of the mic. I am looking at peak
amplitudes in each frame (160 samples) to identify the impulse
sound.</div>
<div><br>
</div>
<div>The time from when the sound enters the stack (before it is
queued with the speaker) to the time when it comes out of the
mic ready to go out, varies between 0.44 to 0.57 seconds. So I
queue the far-end samples for 0.4 seconds, and then send them
to speex_echo_playback. Also, I am using a filter length of
0.2 seconds (1600 samples). Each outgoing (near-end) sample
coming from the mic is passed through speex_echo_capture to
cancel the echo and the output is sent out of the device.</div>
<div><br>
</div>
<div>Despite this, the cancellation does not seem to work. I
have also tried using speex_echo_cancellation. That doesn't
help either.</div>
<div><br>
</div>
<div>What could I be doing wrong? Any suggestions?</div>
<div>Also, I am not using the speex encoder/decoder, just the
echo cancellation. I hope that's not an issue. Please correct
me if it is.</div>
<div><br>
</div>
<div>Thanks for helping me with this!</div>
<div><br>
</div>
<div>
<br>
</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<pre>
At 2011-04-21 03:00:01，<a moz-do-not-send="true" href="mailto:speex-dev-request@xiph.org" target="_blank">speex-dev-request@xiph.org</a> 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.
</pre>
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