Re: [Discuss-gnuradio] Audio Sample rate problem

2009-07-11 Thread Oswald Berthold

for future searches and in case it's linux, i did this when i bumped
into the same problem a while ago:

$ cat ~/.asoundrc 
pcm.rate_convert {
type plug
slave {
pcm hw:0,0
rate 32000
}
}


bst, opt

Bob McGwier rwmcgw...@gmail.com writes:

 In gnuradio-examples/python/audio there is a completely worked out
 example block to do EXACTLY what was requested, go to and from 32000
 and 48000.  It is good practice to look in the examples directory
 because with very high probability, the developers who check stuff in
 have faced the kinds of problems you want to tackle.  Ask after you
 have looked there.

 In this case it is test_resampler.py

 It does a PERFECT 3/2 polyphase resampling ( and back 2/3) since that
 is a rational fraction, not an approximation.  If you use the
 low_pass_2 filters, you gain better control over the number of taps
 required.  It uses the brilliant study by fred harris of finding the
 minimum number of taps that will just exceed the number required to
 get in band ripple, transition bandwidth, and stop band rejection you
 are targeting.

 Bob


 davek wrote:
 On Tue, Jul 7, 2009 at 11:20 PM, Abdalla Sokarabdallaso...@yahoo.com wrote:


 thanx very much this way really worx and wonderfully too

 --- On Wed, 7/8/09, davek davidki...@gmail.com wrote:

 From: davek davidki...@gmail.com
 Subject: Re: [Discuss-gnuradio] Audio Sample rate problem
 To: Abdalla Sokar abdallaso...@yahoo.com
 Date: Wednesday, July 8, 2009, 5:14 AM

 i dont know much about this so it might be wrong but
 if you cant figure it out you could put a fractional interpolation
 filter before your audio block set to 2/3


 On Tue, Jul 7, 2009 at 8:58 PM, Abdalla Sokarabdallaso...@yahoo.com wrote:
 my audio sampling rate is 48000 not 32000 so all the the gnu examples that
 output an audio dont work and due to this 48000 i cant configure the
 decimation and interpolation to be even integers as required

 so is there away to change sampling rate of my audio card to 32000

 i am using Dell Inspiron 1525 with ubuntu 9.04 with the latest version of
 GNU
 and my USRP is rev 4.2

 Please i need help ASAP
 Thanx



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[Discuss-gnuradio] question regarding usrp_fft.py

2009-07-11 Thread adib_sairi

Good day to all,

   i have some question regarding the y-axis in usrp_fft.py.. it is labeled
in unit dB. so how does it convert from ADC value to dB? the output from
after c2mag block we got the magnitude value of the power isn't it? so how
does from the ADC value it convert to the power? what is the formula? 

second, what is this formula mean? 
  log = gr.nlog10_ff(10, self.fft_size,
-20*math.log10(self.fft_size)-10*math.log10(power/self.fft_size)). i think
this is to convert from power in magnitude domain to dB.. is it? what is the
actual formula on this? thank you very much in advance

Adib
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Re: [Discuss-gnuradio] question regarding usrp_fft.py

2009-07-11 Thread adib_sairi

one more question, if the y-axis is in dB so this picture show the reading of
around -42dB.. this reading is taken without any transmission in 2.5G
channel which mean it is the reading of noise level. so isn't -42dB is very
high for noise? thanks

http://www.nabble.com/file/p24440598/noise%2Bpower%2Bmeasurement%2Bin%2B2.5GHz.png
noise+power+measurement+in+2.5GHz.png 
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Re: [Discuss-gnuradio] question regarding usrp_fft.py

2009-07-11 Thread Firas Abbas

Hi,


--- On Sat, 7/11/09, adib_sairi adib_sa...@yahoo.com wrote:
 
    i have some question regarding the y-axis in usrp_fft.py.. it is 
 labeled in unit dB. so how does it convert from ADC value to dB?

Using the formula below.


 
 second, what is this formula mean? 
   log = gr.nlog10_ff(10, self.fft_size,
 -20*math.log10(self.fft_size)-10*math.log10(power/self.fft_size)).

 
 Adib

This formula is used to convert the FFT output value to dB. This is relative 
conversion and the values does not mean actual power because the actuall power 
value  depends on other factors such as RF gain used. What is meant by 
relative, is that if the input signal power is (for example) -20dBm and we 
measured it in this formula as (say) 50dB, then when we change the signal power 
to -30dBm, we get 40dB from the output of this formula. So, lowering signal 
power level by 10dBm result a 10dB value reduction from the formula output.

So, what is the factors that effect dB conversion?. The first value is the FFT 
size, so that when we change the FFT size, the formula should compensate for 
this. Second we should compensate for the taps of the window used for FFT 
(Hanning, Hamming,...etc).  



Best Regards,

Firas


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Re: [Discuss-gnuradio] question regarding usrp_fft.py

2009-07-11 Thread adib_sairi

thank you firas for the explanation..


Firas A. wrote:
 
 
 So, what is the factors that effect dB conversion?. The first value is the
 FFT size, so that when we change the FFT size, the formula should
 compensate for this. Second we should compensate for the taps of the
 window used for FFT (Hanning, Hamming,...etc).  
 
 

is the fft_size effect the value of db? for plotting the GUI (the display of
the spectrum analyzer) in fftsink_nongl.py, inside the set_data function, we
can see that each of the bin of the fft is treated as individual i.e
evt.data[0], evt.data[1], evt.data[2], and so on. this data is use to plot
the graph inside the GUI of the spectrum analyzer. so if the bin of fft is
treated as individual, what is the relationship between fft_size and the db
calculation? what is the usage of fft_size in the calculation?

if i choose to monitor the 2.5G center frequency, so the evt.data[0] is the
place where the 2.5G is right? if the fusb is 4Ms/s and our fft bin is 256,
so our bandwidth of monitoring is only 2MHz right? and evt.data[1] to
evt.data[128] is the positive side of my signal while evt.data[129] to
evt.data[256] is my negative side.. am i right?

Adib

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Re: [Discuss-gnuradio] FFT Plot and Benchmark

2009-07-11 Thread adib_sairi


Ali Siddiqi wrote:
 
 Hi all,
 I looked at fftsink_gl.py. It couldnt be executed because of import error
 of
 fft_window. Also, if in case I manage
 to run it, still I am finding it difficult to incoporate it in
 benchmark_tx.py. Can someone elaborate a bit more on how to add FFT sink
 in
 benchmark_tx?
 
 Regards,
 Ali
 
 

Good day,
  is there any progress in this Mr Ali? i also very interested to try this..
if you have any progress please post it again..:-) thank you very much in
advance

Adib
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Re: [Discuss-gnuradio] Frequency dependant behaviour of wired transmission using USRP

2009-07-11 Thread Matt Ettus

Harry Jones wrote:

Hello,

I am currently doing a project at university on channel sounding. I have 
been using pseudo-random noise and complex chirps as the base signals in 
order to do this task. In order to get used to using the USRP, I have 
been using a coaxial cable of around 3m connecting the two USRPs (on two 
different computers). I am currently using the DC-30/40MHz 
daughterboards for the initial test stage. I am using a very basic 
transmission and receive structure. I write a file in MATLAB, generally 
pseudo-random noise, read this in Python and then use it as a vector 
source into the USRP. The transmit works fine and has been tested using 
oscilloscopes etc. The received side takes the data into a complex file 
sink. The receive end also takes in data well. I am running a NetBSD 
system which has proven itself to work well.


When I look at the frequency content of the received data however I see 
a strange tappering on the low/high frequencies sides of the signal, 
almost as if the coaxial cable is applying a bandpass filter to the data 
(I apply an FFT to the data on MATLAB after receiving it. For example I 
take a 1000 chunk of the data, if the transmitted data was 1000 long, I 
then take a complex fft.). The complex chirp and PR noise are both 
spectrally flat and should not be changed drastically by the wired 
channel. When I do a time domain cross correlation, I get nice 
delta-esque peaks, which implies that the impulse response of the wired 
channel is a delta (what I expect). Thus the frequency domain and time 
domain descriptions are conflicting.


I have:
- Checked this phenomenom at DC, still occurs
- Occurs at 10MHz and 20MHz (the coaxial should surely work up into the 
GHz range)

- Changed USRP, still occurs




I believe you are seeing the rolloff of the CIC interpolator.  If you 
use a lower interpolation ratio in the FPGA and do some interpolation in 
software, you should see it much flatter.


In any case, you can't expect to use 100% of the nyquist bandwidth in 
any real DSP system due to practical filter rolloff issues.


Matt


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Re: [Discuss-gnuradio] question regarding usrp_fft.py

2009-07-11 Thread Firas Abbas

Hi,


--- On Sat, 7/11/09, adib_sairi adib_sa...@yahoo.com wrote:
 so if the bin of fft is treated as individual, what is the relationship  
 between fft_size and the db calculation? what is the usage of fft_size 
 in the calculation?


Ok let me explain. Let x(n) = cos(2*n*PI*f/fs)  , n=0,1,2,..,N

Now take the FFT 3 times with N = 256, N = 1024 and N = 2048.

You will see that the output of the FFT X(n) will be different in magnitude for 
the same input sequence because of different FFT size. The spectrum shape is 
similar in the 3 cases, but the the amplitude of each spectrum bin is different.



 
 if i choose to monitor the 2.5G center frequency, so the evt.data[0] is
  the place where the 2.5G is right? if the fusb is 4Ms/s and our
 fft bin is 256, so our bandwidth of monitoring is only 2MHz right? and
 evt.data[1] to evt.data[128] is the positive side of my signal while
 evt.data[129] to evt.data[256] is my negative side.. am i right?
 

No some numbers are wrong. Assuming complex samples, when Fusb is 4MB/s the 
bandwidth is 1 MHz (2 bytes for I and 2 for Q). For FFT 256, evt.dat[0] is the 
center frequency (2.5G in your case) evt.data[1] to evt.data[127] is the 
positive side and evt.data[128] to evt.data[255] is the negative side.



BR

Firas


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[Discuss-gnuradio] Having a higher samples per symbol (greater than 7) with lower bitrate

2009-07-11 Thread Mir M. Ali
Hi,
the maximum samples per symbol value that benchmark_tx has is 7. What
happens if I change it to a higher value and reduce the bitrate such that
the usrp constraints are still met. I read somewhere that having a sample
per symbol value greater than 7 is impractical. Why?
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