I was able to recorded an audio signal through the MIC and i applied FFT and Hanning window function to it. Now I want to know how do I identify CUTOFF frequency from it to apply the High Pass Filter.
my sample rate = 8000 Hz, FFT block size = 256
Can anyone help me to identify this ? I would really appreciate it ! :)
Thanks!
Related
I have a simple flowgraph for QPSK transmitter with USRP.
After execution, there is lage sidelobes, that pulsate.
During the periods of large sidelobes, there is a drop in amplutude of main lobe.
There is no such pulsations if I make similar transmitter with Matlab.
I suscpect discontinues in sorce.
Comments and advice are appreciated.
Your pool of random data is far too short; you'll see data periodicity in spectrum very quickly; it might be that this is exactly what happens. So, try with num_samples 2**20 instead.
You can observe your transmit spectrum yourself before even transmitting it: use the Qt GUI frequency sink or waterfall sink with an FFT length that corresponds to the FFT length you use in gqrx.
Your sample rate is at the least end of all possible sampling rates. Here, the roll-off of the interpolation filters inside the USRP will definitely show. Don't do that to yourself. Use sps = 16, samp_rate = 1e6 instead.
Make sure you're not getting any underruns in your tranmitter, nor overruns in your receiver. If that happens at these incredibly low sampling rates, something is wrong with your computer setup
Changes make no difference. The following is # 2**20 number of samples, 1 MHz sample rate and 20 samples per symbol. There is no underrun.
# 5 Mhz sample rate I start receiving underrun.
I found the problem and a solution.
The problem is that the level of the signal after modulator is too strong for the USRP input. After modulator the abs value of the signal reach 9. I don't know the maximum level of the signal that USRP expects. I presume something like 1 peak to peak
The solution is to restrict the level by multiplication with a constant. With constant=0.5, there is still distortions. Value of 0.2 is ok.
Here is the new flowgraph:
I want to create some simple heart rate monitor in LabVIEW.
I have sensor which gives me heart workflow (upper graph): Waveform
On second graph (lower graph) is amount of hills (0 - valley, 1 - hill) and that hills are heart beats (that is voltage waveform). From this I want to get amount of those hills, then multiply this number by 6 and I'll get heart rate per minute.
Measuring card I use: NI USB-6009.
Any idea how to do that?
I can sent a VI file if anyone will be able to help me.
You could use Threshold Peak Detector VI
This VI does not identify the locations or the amplitudes of peaks
with great accuracy, but the VI does give an idea of where and how
often a signal crosses above a certain threshold value.
You could also use Waveform Peak Detection VI
The Waveform Peak Detection VI operates like the array-based Peak
Detector VI. The difference is that this VI's input is a waveform data
type, and the VI has error cluster input and output terminals.
Locations displays the output array of the peaks or valleys, which is
still in terms of the indices of the input waveform. For example, if
one element of Locations is 100, that means that there is a peak or
valley located at index 100 in the data array of the input waveform.
Figure 6 shows you a method for determining the times at which peaks
or valleys occur.
NI have a great tutorial that should answer all your questions, it can be found here:
I had some fun recreating some of your exercise here. I simulated a squarewave. In my sample of the square wave, I know how many samples I have and the sampling frequency. As a result, I calculate how much time my data sample represents. I then count the number of positive edges in the sample. I do some division to calculate beats/second and multiplication for beats/minute. The sampling frequency, Fs, and number of samples, N or #s are required to calculate your beats per minute metric. Their uses are shown below.
The contrived VI
Does that lead you to a solution for your application?
I'm using RTL-SDR generic dongle for receiving frames of Z-Wave protocol. I use real Z-Wave devices. I'm using scapy-radio and I've also downloaded EZ-Wave. However, none of them implements blocks for all Z-Wave data rates, modulations and codings. I've received some frames using original solution of EZ-Wave, however I assume I can't receive frames at all data rates, codings and modulations. Now I'm trying to implement solution according to their blocks to implement all of them.
Z-Wave procotol uses these modulations, data rates and coding:
9.6 kbps - FSK - Manchester
40 kbps - FSK - NRZ
100 kbps - GFSK - NRZ
These are my actual blocks (not able receving anything at all right now):
For example, I will explain my view on blocks for receiving at
9.6 kbps - FSK - Manchester
RTL-SDR Source
variable center_freq = 869500000
variable r1_freq_offset = 800e3
Ch0: Frequency: center_freq_3-r1_freq_offset, so I've got 868.7 Mhz on RTL-SDR Source block.
Frequency Xlating FIR Filter
Center frequency = - 800Khz to get frequency 868.95 Mhz (Europe). To be honest, I'm not sure why I do this and I need an explanation. I'm trying to implement those blocks according to EZ-Wave implementation of blocks for 40 kbps-FSK-NRZ (as I assume). They use sample rate 2M and different configurations, which I did not understand.
Taps = firdes.low_pass(1,samp_rate_1,samp_rate_1/2,5e3,firdes.WIN_HAMMING). I don't understand, what should be transition bw (5e3 in my case)
Sample rate = 19.2e3, because data rate/baud is 9.6 Kbps and according to Nyquist–Shannon sampling theorem, sampling rate should be at least double to data rate, so 2*9.6=19.2. So I'm trying to resample default 2M from source to 19.2 Kbps.
Simple squelch
I use default value (-40) and I'm not sure, if I should change this or not.
Quadrature Demod
should do the FSK demodulation and I use default value of gain. I'm not sure if this is a right way to do FSK demodulation.
Gain = 2(samp_rate_1)/(2*math.pi*20e3/8.0)*
Low Pass Filter
Sample rate = 19.2k to use the same new sample rate
Cuttoff Freq = 9.6k, I assume this according to https://nccgroup.github.io/RFTM/fsk_receiver.html
Transition width = 4.8 which is also sample_rate/2
Clock Recovery MM
Most of the parameters are default.
Omega = 2, because samp_rate/baud
Binary Slicer
is for getting binary code of signal
Zwave PacketSink 9.6
should the the Manchester decoding.
I would like to ask, what should I change on my blocks to achieve proper receiving of Z-Wave frames at all data rates, modulation and coding. When I start receiving, I'm able to see messages from my devices at FFT sink and Waterfall sink. The message debug doesn't print packets (like from original EZ-Wave solution) but only
Looking for sync : 575555aa
Looking for sync : 565555aa
Looking for sync : aa5555aa
what should be value in frame_shift_register, according to C code for Manchester decoding (ZWave PacketSink 9.6). I've seen similar post, however this is a bit different and to be honest, I'm stuck here.
I will be grateful for any help.
Let's look at the GFSK case. First of all, the sampling rate of the RTL source, 2M Baud is pretty high. For the maximum data rate, 100 kbps - GFSK, a sample rate of say 400 ~ 500kbaud will do just fine. There is also the power squelch block. This block prevents signals below a certain threshold to pass. This is not good because it filters low power signals that may contain information. There is also the sample rate issue between the lowpass filter and the MM clock recovery block. The output of the symbol recovery block should be 100kbaud (because for GFSK, sample rate = symbol rate). Using the omega value of 2 and working backward, the input to the MM block should be 200kbaud. But, the lowpass filter produces samples at 2Mbaud, 10 times than expected. You have to do proper decimation.
I implemented a GFSK receiver once for our CubeSat. Timing recovery was done by the PFB block, which is more reliable than the MM one. You can find the paper here:
https://www.researchgate.net/publication/309149646_Software-defined_radio_transceiver_for_QB50_CubeSat_telemetry_and_telecommand?_sg=HvZBpQBp8nIFh6mIqm4yksaAwTpx1V6QvJY0EfvyPMIz_IEXuLv2pODOnMToUAXMYDmInec76zviSg.ukZBHrLrmEbJlO6nZbF4X0eyhFjxFqVW2Q50cSbr0OHLt5vRUCTpaHi9CR7UBNMkwc_KJc1PO_TiGkdigaSXZA&_sgd%5Bnc%5D=1&_sgd%5Bncwor%5D=0
Some more details on the receiver could also be found here:
GFSK modulation/demodulation with GNU Radio and USRP
M.
I appreciate your answer, I've changed my sample rates. Now I'm still working on 9.6Kbps, FSK demodulation and Manchester decoding. Currently, output from my M&M clock recovery looks like this:
I would like to ask you what do think about this signal. As I said, it should be FSK demodulation and then I should use Manchester decoding. Do I still need usage of PCB block? Primary, I have to do 9.6kbps, FSK and Manchester, so I will look at 100Kbps GFSK NRZ if there will be some time left.
Sample rate is 1M because of RTL-SDR dongle limitations (225001 to 300000 and 900001 to 3200000).
Current blocks:
I don't understand :
Taps of Frequency Xlating FIR Filter firdes.low_pass(1,samp_rate_1,40e3,20e3,firdes.WIN_HAMMING)
Cuttoff Freq and Transition Width of Low Pass filter
Clock Recovery M&M aswell, so consider its values "random".
ClockRecovery Output:
I was trying to use PCB block according to your work at ResearchGate. However, I was unsuccessful because I still don't understand all that science behind the clock recovery.
Doing Low-pass filtering twice is because original Z-Wave blocks from scapy-radio for 40Kbps, FSK and NRZ coding are made like this (and it works):
So I thought I will be just about changing few parameters and decoder (Zwave PacketSink9.6).
I also uploaded my current blocks here.
Moses Browne Mwakyanjala, I'm also trying to implement that thing according to your work.
Maybe there is a problem with a clock recovery and Manchester decoding. Manchester decoding use transitions 0->1 and 1->0 to encode 0s and 1s. How can I properly configure clock recovery to achieve correct sample rate and transitions for Manchester decoding? Manchester decoder (Z-Wave PacketSink 9.6) is able to find the preamble and ends only with looking for sync.
I would like to also ask you, where can I find my modulation index "h" mentioned in your work?
Thank you
I am a labview new starter.
Here is my problem,I am working on a data processing LabVIEW program,now I use Threshold Detector VI to get the signals which have crossing the threshold,and I have the indices of these signals.but the indices are related to the amplitude of the signals,what I want is the time crossings of these signals.
What can I do?
Thank you very much!
You can use Get Waveform Time Array to get an array of timestamps, then use your threshold indices to select from the timestamp array.
I need to calculate the FFT of audiodata in an Android Project and I use jTransforms to achieve this.
The samples of the audiodata are a few seconds long and are recorded with a samplerate of 11025 Hertz.
I am not sure how to set the length of the FFT in jTransforms.
I do not really need high frequency resolution, so a size of 1024 would be enough.
But from what I have understood learning about the FFT, if I decrease the FFT size F and use a sample with a lenght of N > F, only the first F values of the original sample are transformed.
Is that true or did I understand something wrong?
If it is true, is there an efficient way to tranform the whole signal and decreasing the FFT-Size afterwards?
I need this to classify different signals using Support Vector Machines, and FFT-Sizes > 1024 would give me too much features as output, so I would have to reduce the result of the FFT to a more compact vector.
If you only want the FFT magnitude results, then use the FFT repeatedly on successive 1024 chunk lengths of data, and vector sum all the successive magnitude results to get an estimate for the entire much longer signal.
See Welch's Method on estimating spectral density for an explanation of why this might be a useful technique.
Im not familiar with the jTransform library, but do you really set the size of the transform before calculating it? Amplitude values of the time-domain signal and the sampling frequency (11.025 kHz) is enough to calculate the FFT (note that the FFT assumes constant sampling rate)
The resolution in frequency domain will be determined by Nyquist's theorem; the maximum resolvable frequency in your signal will be equal to half your sampling rate. In other words, sampling your signal with 11.025 kHz, you can expect your frequency graph to contain frequency values (and corresponding amplitudes) between 0 Hz - 5.5125 kHz.
UPDATE:
The resolution of the FFT (the narrowness of the frequency bins) will increase/improve if your input signal is longer, thus 1024 samples might not be a long sequence enough if you need to distinguish between very small changes in frequency. If thats not a problem for you application, and the nature of your data is not variying quickly, and you have the processing time, then taking an average of 3-4 FFT estimates will greatly reduce noise and improve estimates.