Speex dev - Apr 2011 - Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams?

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 is
much helpful for us to implement AEC in common PC. We know that most low-cost
soundcards have different sample rates in capturing and rendering which prevents
LMS based AEC from being used in most computer.
http://msdn.microsoft.com/en-us/library/ff536174(VS.85).aspx
In Windows XP, the clock rate must be matched between the capture and render
streams. The AEC system filter implements no mechanism for matching sample rates
across devices. ............. In Windows XP SP1, Windows Server 2003, and later,
this limitation does not exist. The AEC system filter correctly handles
mismatches between the clocks for the capture and render streams, and separate
devices can be used for capture and rendering.
Maoquan
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I would imagine that it is handle through basic asynchronous sample rate
conversion.  There is a lot of literature out there on the different techniques
to do this.  A common method is sinc interpolation.  This is how I have handle
these types of things in the past.

Vasant Shridhar

From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org] On
Behalf Of LiMaoquan2000
Sent: Tuesday, April 12, 2011 12:36 AM
To: speex-dev
Subject: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches
between clocks of capture and render streams?


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 is
much helpful for us to implement AEC in common PC. We know that most low-cost
soundcards have different sample rates in capturing and rendering which prevents
LMS based AEC from being used in most computer.

http://msdn.microsoft.com/en-us/library/ff536174(VS.85).aspx
In Windows XP, the clock rate must be matched between the capture and render
streams. The AEC system filter implements no mechanism for matching sample rates
across devices. ............. In Windows XP SP1, Windows Server 2003, and later,
this limitation does not exist. The AEC system filter correctly handles
mismatches between the clocks for the capture and render streams, and separate
devices can be used for capture and rendering.

Maoquan
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Hi Shridhar,

Sample rate conversion is not enough to solve this problem. I have tried this
method several months
ago. The first step is to measure the difference between sample rate of
capturing and rendering. Then
resampling (by what you said "sinc interpolation") one signal to
eliminate the difference. The frequency
step in my experiment is less than 0.1Hz. I have tried speex AEC after
resampling, much more echo is
cancelled than the one without resampling. But there is still echo can be
heared.
After all, frequency step of sample rate conversion is limited, mismatch is
still exist after resampling.
Someone told me that capture and render codec have different clock generator
which shift independently.
And LMS algorithm is very sensitive to the difference between sample rates.

Sincerely
Maoquan


At 2011-04-12 21:46:26?"Shridhar, Vasant" <vasant.shridhar at
harman.com> wrote:


I would imagine that it is handle through basic asynchronous sample rate
conversion.  There is a lot of literature out there on the different techniques
to do this.  A common method is sinc interpolation.  This is how I have handle
these types of things in the past.

 

Vasant Shridhar

 

From:speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org]On
Behalf OfLiMaoquan2000
Sent: Tuesday, April 12, 2011 12:36 AM
To: speex-dev
Subject: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches
between clocks of capture and render streams?

 

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 is
much helpful for us to implement AEC in common PC. We know that most low-cost
soundcards have different sample rates in capturing and rendering which prevents
LMS based AEC from being used in most computer.

http://msdn.microsoft.com/en-us/library/ff536174(VS.85).aspx
In Windows XP, the clock rate must be matched between the capture and render
streams. The AEC system filter implements no mechanism for matching sample rates
across devices. ............. In Windows XP SP1, Windows Server 2003, and later,
this limitation does not exist. The AEC system filter correctly handles
mismatches between the clocks for the capture and render streams, and separate
devices can be used for capture and rendering.

Maoquan
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I am doing this right now with no problem.  I am not using speex for this at the
moment though.  Group delay is the biggest problem.  I implemented a version
where the input and output sample rates are known up front.  The routine than
interpolates between the jitter.  This should solve the problem.  The crystals
used to clock the input and output have very fine tolerances on most standard
audio cards.

Vas
________________________________________
From: Li Maoquan [limaoquan2000 at 126.com]
Sent: Tuesday, April 12, 2011 2:48 PM
To: Shridhar, Vasant
Cc: speex-dev
Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with
mismatches     between clocks of capture and render streams?

Hi Shridhar,

Sample rate conversion is not enough to solve this problem. I have tried this
method several months
ago. The first step is to measure the difference between sample rate of
capturing and rendering. Then
resampling (by what you said "sinc interpolation") one signal to
eliminate the difference. The frequency
step in my experiment is less than 0.1Hz. I have tried speex AEC after
resampling, much more echo is
cancelled than the one without resampling. But there is still echo can be
heared.
After all, frequency step of sample rate conversion is limited, mismatch is
still exist after resampling.
Someone told me that capture and render codec have different clock generator
which shift independently.
And LMS algorithm is very sensitive to the difference between sample rates.

Sincerely
Maoquan

At 2011-04-12 21:46:26?"Shridhar, Vasant" <vasant.shridhar at
harman.com> wrote:
I would imagine that it is handle through basic asynchronous sample rate
conversion.  There is a lot of literature out there on the different techniques
to do this.  A common method is sinc interpolation.  This is how I have handle
these types of things in the past.

Vasant Shridhar

From: speex-dev-bounces at xiph.org<mailto:speex-dev-bounces at xiph.org>
[mailto:speex-dev-bounces at xiph.org<mailto:speex-dev-bounces at
xiph.org>] On Behalf Of LiMaoquan2000
Sent: Tuesday, April 12, 2011 12:36 AM
To: speex-dev
Subject: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches
between clocks of capture and render streams?


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 is
much helpful for us to implement AEC in common PC. We know that most low-cost
soundcards have different sample rates in capturing and rendering which prevents
LMS based AEC from being used in most computer.

http://msdn.microsoft.com/en-us/library/ff536174(VS.85).aspx<http://msdn.microsoft.com/en-us/library/ff536174%28VS.85%29.aspx>
In Windows XP, the clock rate must be matched between the capture and render
streams. The AEC system filter implements no mechanism for matching sample rates
across devices. ............. In Windows XP SP1, Windows Server 2003, and later,
this limitation does not exist. The AEC system filter correctly handles
mismatches between the clocks for the capture and render streams, and separate
devices can be used for capture and rendering.

Maoquan

Hi  Vasant,
>I am doing this right now with no problem.  I am not using speex for this at
the moment though.  Group delay is the biggest problem.  I implemented a version
where the input and output sample rates are known up front.  The routine than
interpolates between the jitter.  This should solve the problem.  The crystals
used to clock the input and output have very fine tolerances on most standard
audio cards.
Are you sure?
1. What's the core algorithm of your AEC algorithm?
2. Have you tried common low-cost AC97 soundcards? I have tested more than 20
low-cost soundcards. Only 2 of them have exactly the sample rate of capturing
and rendering. Differences of others range from 0.5Hz to 80Hz. The exact reason
in detail is still unknown.
3. Group delay is not a problem compared to different sample rate. Because the
latter one will cause delay shift and overflow and downflow (sudden change of
delay) of buffers, which is a big obstacle to acoustic echo cancellation.
Who knows any AEC which is not sensitive to different sample rate? After all,
microsoft said its new AEC can do it.
>________________________________________
>From: Li Maoquan [limaoquan2000 at 126.com]
>Sent: Tuesday, April 12, 2011 2:48 PM
>To: Shridhar, Vasant
>Cc: speex-dev
>Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with
mismatches     between clocks of capture and render streams?
>
>Hi Shridhar,
>
>Sample rate conversion is not enough to solve this problem. I have tried
this method several months
>ago. The first step is to measure the difference between sample rate of
capturing and rendering. Then
>resampling (by what you said "sinc interpolation") one signal to
eliminate the difference. The frequency
>step in my experiment is less than 0.1Hz. I have tried speex AEC after
resampling, much more echo is
>cancelled than the one without resampling. But there is still echo can be
heared.
>After all, frequency step of sample rate conversion is limited, mismatch is
still exist after resampling.
>Someone told me that capture and render codec have different clock generator
which shift independently.
>And LMS algorithm is very sensitive to the difference between sample rates.
>
>Sincerely
>Maoquan
>
>At 2011-04-12 21:46:26?"Shridhar, Vasant" <vasant.shridhar at
harman.com> wrote:
>I would imagine that it is handle through basic asynchronous sample rate
conversion.  There is a lot of literature out there on the different techniques
to do this.  A common method is sinc interpolation.  This is how I have handle
these types of things in the past.
>
>Vasant Shridhar
>
>From: speex-dev-bounces at xiph.org<mailto:speex-dev-bounces at
xiph.org> [mailto:speex-dev-bounces at xiph.org<mailto:speex-dev-bounces
at xiph.org>] On Behalf Of LiMaoquan2000
>Sent: Tuesday, April 12, 2011 12:36 AM
>To: speex-dev
>Subject: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches
between clocks of capture and render streams?
>
>
>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
is much helpful for us to implement AEC in common PC. We know that most low-cost
soundcards have different sample rates in capturing and rendering which prevents
LMS based AEC from being used in most computer.
>
>http://msdn.microsoft.com/en-us/library/ff536174(VS.85).aspx<http://msdn.microsoft.com/en-us/library/ff536174%28VS.85%29.aspx>
>In Windows XP, the clock rate must be matched between the capture and render
streams. The AEC system filter implements no mechanism for matching sample rates
across devices. ............. In Windows XP SP1, Windows Server 2003, and later,
this limitation does not exist. The AEC system filter correctly handles
mismatches between the clocks for the capture and render streams, and separate
devices can be used for capture and rendering.
>
>Maoquan
>
>

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