Akankshu,
I have taken a look at the code, but I'm not really able to say what is
significantly different between the two other than the fact that
SoundLocalizer is designed to sample the microphone as fast as possible,
so there are probably some optimizations that Hu Siquan (the author of
the code, his email is inside the code if you need it) made to make it
perform better. You would have to talk to him if you have questions
about it as I have no hands-on experience with any of the mts300
functionality.
Your problem may be that the MicP implementation is that it has more
overhead and samples the microphone fewer times per second. I have no
idea of the speeds, but if for example, the MicP implementation takes 10
seconds to collect 1000 samples, you may be missing all of the peaks of
the sound events (clapping, shouting, or whatnot) when you are sampling
the ADC, versus SoundLocalizer which is designed to sample the ADC as
fast as possible. Without access to the hardware, and performing some
profiling of each implementation, i can't say for sure if this is the case.
Sorry that i couldn't help you more.
-Paul
Akankshu Dhawan wrote:
Hi Paul
Can you tell me what might be the different between the microphone
implementation in MicrophoneC inside SoundLocalizer Vs the MicP inside
mts300 ?
Thanks a lot
Akankshu
On Thu, Nov 12, 2009 at 4:44 PM, Paul Johnson <oewyn...@gmail.com
<mailto:oewyn...@gmail.com>> wrote:
CVS under /tinyos-2.x-contrib/tinyos-programming/
-Paul
Martin Osterloh wrote:
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Hi all,
sorry for interfering this thread. But, where can I find the
sourcecode to the soundlocalizer demo?
Best,
Martin
Akankshu Dhawan wrote:
Hi Michael and Zainul
Thanks a lot for your response. I am grateful to be hearing from you.
I will look into the corrected mean solution before I suggest some
hardware changes to my team.
Although, my bigger question was why direct sampling by calling
getData->dataReady on a free running ADC (this is what is used in
the SoundLocalizer example in Tinyos-2.x described in the book also)
responds better ? if I try to do threshold detection of every sample
using the same interface and direct sampling I get values which are
very responsive and high rate of sampling. I know I am loosing on
some precision by sampling at a rate other than the default
ATM128_ADC_PRESCALE but since I am mainly using it for thresholding
I think my application can handle it.
Can you tell me some comparison between the microphone setup used by
the MicP and MicStream compared to the SoundLocalizer direct
sampling setup ?
Thanks a lot once again
Sincerely
Akankshu Dhawan
On Thu, Nov 12, 2009 at 1:30 PM, Michael Schippling
<sc...@santafe.edu <mailto:sc...@santafe.edu> <mailto:sc...@santafe.edu>>
wrote:
I suspect that the straight averaging thing is not what you want.
I haven't looked at the mic data, but it's sampling an AC
signal (sound wave pressure alternates between compression
and rarefaction at the frequency you perceive) and is probably
biased so a no-pressure signal is pretty much in the middle of
the ADC range. Averaging that AC signal over a number of high-low
pressure waves will just give you the middle again.
If that is the case, what you want to do is rectify the signal --
half-wave would just chop off the samples below the mid-value,
full-wave would invert them around the mid-value -- before
trying to average to get a "loudness". If you are trying
to get the "Sound Pressure Level" -- how loud the sound is at
any particular time -- it's called "Envelope Following" in the
good old days of signal processing. If you know any electronics,
what you want to build is a diode or diode bridge, to rectify
the signal, and a capacitor to filter, or average, it to a slowly
varying DC value.
MS
Akankshu Dhawan wrote:
Hi All
I am using two mechanisms for high sampling.
1. Using MicStreamC and changing the prescalar value inside
MicP to ATM128_ADC_PRESCALE_32 and the gain value is set to
64. I create a buffer of 1000 samples and every time the
buffer gets full I take the average and print it out. The
problem is that the microphone does not seem to be
responding or is not showing me sufficiently precise values.
The average when I dont make a noise is around 500 ADC.. and
even if I am clapping shouting (for long durations) it still
shows me slightly */lower values /like 497 etc. So I am not
sure why this is so ?*
2. When I created the low level interfaces on my own using
the sample SoundLocalizer example in Tinyos programming
book, I am giving the microphone ADC a free run at the same
prescale setup (there is a difference of some parameter
being passed through getData as FALSE in that example which
is suppose to cater to LEFT_ADJUSTMENT ) but when I compare
the values that I am getting inside dataReady they respond
well to sounds. This would work for me but I am just curious
why the MicStream does not work. This way I would not have
to recreate all the buffer code.
If anyone can please tell me what setting inside MicSetting
or MicP have I done wrong that it is not responding.
Thanks a lot.
SIncerely
Akankshu Dhawan
--
First they ignore you, then they laugh at you, then they
fight you, then you win.
- Mahatma Gandhi
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