It still seems strange that the soundcard, which is doing the 
upsampling/downsampling in each case, would use a different rate for each.  I'm 
not familiar with the architecture of PC audio hardware, but it seems like it 
would be harder to make it asynchronous then if playback and record used the 
same clock.  Is the 48kHz the exact same 48kHz for both playback and record? 
Thanks for your info!

-- 
Jon Harris
SPAM blocker in place:
Remove 99 (but leave 7) to reply

"Yehuda Mittelman" <liayehud@netvision.net.il> wrote in message 
news:dd22e7$ob$1@news2.netvision.net.il...
> The sound card can play/record only at a certain sample rate, usually it is 
> 48khz. When you want to play a wav file at 8khz it has to internally upsample 
> your 8khz data to 48khz in it's dsp and play it at 48khz.
> When you want to record at 8khz the sound card samples your microphone at 
> 48khz and downsamples to 8khz.
> Some sound cards don't do it accuratly and that is why you have a mismatch.
>
>
>
> "Jon Harris" <jon99_harris7@hotmail.com> wrote in message 
> news:oQZIe.1406$2j.784@trnddc07...
>> "Yehuda Mittelman" <liayehud@netvision.net.il> wrote in message 
>> news:dd05t8$cml$1@news2.netvision.net.il...
>>> You will have two problems:
>>>
>>> 2) some sound cards can only play the audio at 48khz. when such a soundcard 
>>> has to
>>>    play at 8000hz and record at 8000hz it's dsp uses a 
>>> downsampling/upsampling
>>>    algorithm.
>>>    therefor you will see that you actually play the audio at 8000hz and 
>>> record it at 8005hz or 8010hz or.... who knows...
>>>    so you will have to detect that and convert your near_end or far_end 
>>> samples
>>>    accordingly.
>>
>> I've heard this before, but never really understood why this is the case. Can 
>> you explain?  Is the problem that the OS isn't locked to the sample rate on 
>> the audio card?
>>
>
> 

Well, if you have to do sampling rate conversion anyway, then why not 
simply doing the sampling rate conversion between 8<->48kHz by yourself 
and running your sound card at 48kHz.... (while all your processing are 
still at 8kHz)  then you completely eliminate the second  problem!

Isn't that much simpler and makes more sens Mr. Mittelman ?


Yehuda Mittelman wrote:
> The sound card can play/record only at a certain sample rate, usually it is 
> 48khz. When you want to play a wav file at 8khz it has to internally 
> upsample your 8khz data to 48khz in it's dsp and play it at 48khz.
> When you want to record at 8khz the sound card samples your microphone at 
> 48khz and downsamples to 8khz.
> Some sound cards don't do it accuratly and that is why you have a mismatch.
> 
> 
> 
> "Jon Harris" <jon99_harris7@hotmail.com> wrote in message 
> news:oQZIe.1406$2j.784@trnddc07...
> 
>>"Yehuda Mittelman" <liayehud@netvision.net.il> wrote in message 
>>news:dd05t8$cml$1@news2.netvision.net.il...
>>
>>>You will have two problems:
>>>
>>>2) some sound cards can only play the audio at 48khz. when such a 
>>>soundcard has to
>>>   play at 8000hz and record at 8000hz it's dsp uses a 
>>>downsampling/upsampling
>>>   algorithm.
>>>   therefor you will see that you actually play the audio at 8000hz and 
>>>record it at 8005hz or 8010hz or.... who knows...
>>>   so you will have to detect that and convert your near_end or far_end 
>>>samples
>>>   accordingly.
>>
>>I've heard this before, but never really understood why this is the case. 
>>Can you explain?  Is the problem that the OS isn't locked to the sample 
>>rate on the audio card?
>>
> 
> 
> 

"qfu72" <qiangfu72@gmail.com> schrieb im Newsbeitrag
news:wbydnZ2dnZ1aPCjVnZ2dnXPWbt-dnZ2dRVn-y52dnZ0@giganews.com...
> we want to implement an acoustic echo canceller on a windows PC platform
> for chatting tools like MSN/skype.
>
> A serious problem we found in this kind of platform is the time delay
> (sound card play+speaker+air+mic.+ sound card record) between Ref. signal
> and Echo signal is not consist in every session  1)how could we esimate
> the time dealy? ...

Don't know if it's very advanced, but why not generate a little burst and do
a correlation with incoming signal. So you would know the delay time and you
may get useful IRs for the future processing.

The sound card can play/record only at a certain sample rate, usually it is 
48khz. When you want to play a wav file at 8khz it has to internally 
upsample your 8khz data to 48khz in it's dsp and play it at 48khz.
When you want to record at 8khz the sound card samples your microphone at 
48khz and downsamples to 8khz.
Some sound cards don't do it accuratly and that is why you have a mismatch.



"Jon Harris" <jon99_harris7@hotmail.com> wrote in message 
news:oQZIe.1406$2j.784@trnddc07...
> "Yehuda Mittelman" <liayehud@netvision.net.il> wrote in message 
> news:dd05t8$cml$1@news2.netvision.net.il...
>> You will have two problems:
>>
>> 2) some sound cards can only play the audio at 48khz. when such a 
>> soundcard has to
>>    play at 8000hz and record at 8000hz it's dsp uses a 
>> downsampling/upsampling
>>    algorithm.
>>    therefor you will see that you actually play the audio at 8000hz and 
>> record it at 8005hz or 8010hz or.... who knows...
>>    so you will have to detect that and convert your near_end or far_end 
>> samples
>>    accordingly.
>
> I've heard this before, but never really understood why this is the case. 
> Can you explain?  Is the problem that the OS isn't locked to the sample 
> rate on the audio card?
> 

"Yehuda Mittelman" <liayehud@netvision.net.il> wrote in message 
news:dd05t8$cml$1@news2.netvision.net.il...
> You will have two problems:
>
> 2) some sound cards can only play the audio at 48khz. when such a soundcard 
> has to
>    play at 8000hz and record at 8000hz it's dsp uses a 
> downsampling/upsampling
>    algorithm.
>    therefor you will see that you actually play the audio at 8000hz and record 
> it at 8005hz or 8010hz or.... who knows...
>    so you will have to detect that and convert your near_end or far_end 
> samples
>    accordingly.

I've heard this before, but never really understood why this is the case.  Can 
you explain?  Is the problem that the OS isn't locked to the sample rate on the 
audio card? 

I used a calibration procedure since the project was not final, but it is 
possible to do it online without calibration.

Here is what I did.
For problem 1:

When the program started I played a recorded beep sound from a wave file and 
recorded the near_end and far_end signals.
then I calculated the long cross correlation between the recordings to 
estimate the delay ( I did that in the frequency domain to be more 
efficient).
If you want to do it online you will have to calculate the long cross 
correlation of the near_end and far_end all the time.
another solutionis to use multiple lms filters, each one with a different 
delay. this will be more efficient if you work with subbands (using dft 
filterbank) since you will be able to run these multiple lms only in one 
subband.
you can also combine these methods to get a fine delay estimation.


For problem 2:
I used a calibration when the program was installed. I played a wav file of 
music and recorded the near_end and far_end. then I started checking for the 
right sample rate conversion by grid searching.
the sample rate conversion worked like that:
suppose your far_end played the sound at 8001hz and your near end recorded 
it at 8000hz. this means that you will drop one sample from your far_end 
signal every 8000 samples. to be more accurate you can drop half a sample 
every 4000 samples or quarter every 2000samples or... you can go on with 
that.

after guessing the sample rate conversion I calculated the coherence (matlab 
function: COHERE) of the near_end and the corrected far_end.
I started by guessing that the far end sample rate is 8010hz, converted the 
sample rate and calculated the coherence. I did the same procedure for 
8020hz etc.

so I got a table:
frequency   | coherence
8010          |   0.5
8020          |   1
8030          |   0.5
...
in this case the right sample rate is 8020hz.

The correct sample rate was the one that gave the maximal coherence. and 
this sample rate does not change in time. it will be the same all the time 
so only have to do it once.

you can search with a resolution of 50hz and when you find the right 
frequency conversion you can switch to a fine resolution of 1hz.
There might still be a drift, but it will be very slow and since you 
calibrate the delay online it will compensate for the drift.

I hope this helps

Yehuda







"NS" <NS@NoSpam.com> wrote in message news:42F3FCEA.5070105@NoSpam.com...
> Yehuda Mittelman wrote:
>> You will have two problems:
>>
>> 1) The delay is not constant and it changes sometime. when you reboot the 
>> pc there will
>>     be a different delay.
>
> That's simple one!
>
>> 2) some sound cards can only play the audio at 48khz. when such a 
>> soundcard has to
>>     play at 8000hz and record at 8000hz it's dsp uses a 
>> downsampling/upsampling
>>     algorithm.
>>     therefor you will see that you actually play the audio at 8000hz and 
>> record it at 8005hz or 8010hz or.... who knows...
>>     so you will have to detect that and convert your near_end or far_end 
>> samples
>>     accordingly.
>
> Can't it be 8017.35125874 Hz...? and then your sampling rate conversion is 
> likely to not be 100% accurate, hence there will probably be drift between 
> your signals, and then the AEC filter will drift, and even if you track 
> drift, the performance still degrades as a result of the (perhaps small 
> but still) inaccuracy/mismatch in the sampling rates!
>
> Good AEC needs more professional solution to this problem!
>
>>
>> You should try to do all that online using long buffers of your near_end 
>> and far_end.
>> It is not simple to make an echo canceller for a pc, but it is possible 
>> once your figure out all the problems.
>
> We already figured out the problems..., it may be possible once you figure 
> out the SOLUTIONs to the problems.... 

A little more patience and persistence works. I found this:
http://cnx.rice.edu/content/m10481/latest/
atleast to clear the doubt what an adaptive filter is. But any more
references will be more helpful.

thanks and regards
--Himanshu

Hello!

> As to your second question, an adaptive FIR filter with a fixed filter length
> needs all of the "echo" to "fit" within the filter's length.  If there is a

Sorry if it sounds lame, but what are adaptive FIR filters? Where can I
learn more about them?

thanks and regards
--Himanshu

Yehuda Mittelman wrote:
> You will have two problems:
> 
> 1) The delay is not constant and it changes sometime. when you reboot the pc 
> there will
>     be a different delay.

That's simple one!

> 2) some sound cards can only play the audio at 48khz. when such a soundcard 
> has to
>     play at 8000hz and record at 8000hz it's dsp uses a 
> downsampling/upsampling
>     algorithm.
>     therefor you will see that you actually play the audio at 8000hz and 
> record it at 8005hz or 8010hz or.... who knows...
>     so you will have to detect that and convert your near_end or far_end 
> samples
>     accordingly.

Can't it be 8017.35125874 Hz...? and then your sampling rate conversion 
is likely to not be 100% accurate, hence there will probably be drift 
between your signals, and then the AEC filter will drift, and even if 
you track drift, the performance still degrades as a result of the 
(perhaps small but still) inaccuracy/mismatch in the sampling rates!

Good AEC needs more professional solution to this problem!

> 
> You should try to do all that online using long buffers of your near_end and 
> far_end.
> It is not simple to make an echo canceller for a pc, but it is possible once 
> your figure out all the problems.

We already figured out the problems..., it may be possible once you 
figure out the SOLUTIONs to the problems....

The problem may be even more complicated when you randomly use different 
audio devices, for example Camcorder or other device connected to PC via 
USB or Firewire etc., and then speaker connected via soundcard...

qfu72 wrote:
> we want to implement an acoustic echo canceller on a windows PC platform
> for chatting tools like MSN/skype. 
> 
> A serious problem we found in this kind of platform is the time delay
> (sound card play+speaker+air+mic.+ sound card record) between Ref. signal
> and Echo signal is not consist in every session  1)how could we esimate
> the time dealy? 2) why the AEC is so sensitive to time delay setting?
> 
> 
> 
> 
> 		
> This message was sent using the Comp.DSP web interface on
> www.DSPRelated.com