I am using the Skype4Com COM object to intercept audio from the microphone. I know that it is 16 bit mono, but how do I find out what its sample rate is?
I know that it's possible to manually set input through a wav-file. The only rate that is allowed, is 16,000 Hz. I think that the rate that is used to transfer, also is 16,000 Hz.
It would seem that there is no facility in Skype4Com to determine the microphone sample rate at the moment. My experiments indicate that it is 16kHz.
Related
I'm trying to improve the sample rate of the MLX90393 sensor with ARDUINO. The safe highest sample rate is about 500Hz according to the datasheet.
However, I use Bluetooth to connect to the sensor with SPI protocol and read the sensor using Python language, the highest sample rate I obtain is about 100Hz when I set the FILTER to 2 and OSR to 0.
Also, if I use wire to read the sensor, the sample rate will reach about 300Hz in Serial Monitor, however, if I use the PySerial package in Python to read the sensor and the sample rate decreases to 70~80Hz. I think the reason may be the read speed of PySerial is slow?
Hence, I'm wondering how to improve the sample rate of MLX90393. I want to use the Burst Mode provided by the MLX90393 code, but I can't find the proper Arduino code, anyone can help me?
Thanks a lot!
I am trying to get the sample code for the 'FM_receiver' sample code working with my limeSDR and it seems not to pickup anything. Here's some screenshots of my gnuradio and its output:
I have it tuned to 98.5MHz which I can hear clearly with a typical FM receiver but can only hear static with this. What am I missing?
You may need to set the Device Name in your Audio Sink block. See https://wiki.gnuradio.org/index.php/Audio_Sink You didn't say what OS you have.
If you continue to have problems, I suggest that you work through the https://wiki.gnuradio.org/index.php/Guided_Tutorial_Hardware_Considerations
I observe that while the RF Frequency is set to 98.5 MHz in your LimeSuite Source block the frequency range that is detected by your FFT Sink is centered at 100.1 MHz and going from 99.1 MHz to 101.1 MHz, so we don't really know if the device is receiving any signal at 98.5 MHz.
You clarified that you can clearly hear the FM station in your FM receiver. Considering that the flow diagram is set up correctly , I think that you are probably hearing the 98.5 MHz station from a digital radio receiver. That explains why you cannot hear anything with your GNU Receiver.
Another way to test your design is by setting the FR Frequency parameter of the LimeSuite Source block to 100.1 MHz, where we can clearly see that you receive an FM signal.
Im currently creating a satellite ground station which will be used to control our cubesat in coming months. The modulation scheme used is GFSK and the baud rate is 9600. I have tried to run some tests by using a USRP board before I could try to communicate with the satellite by directly connecting the tx and rx blocks in the flowgraph.I was able to send and receive a png file using this flowgraph.
However, when I connect the tx and rx output to my USRP B210 TX/RX(transmission sink) and RX2(reception source) as shown below, I receive no data even though the two the source and sinks have been connected to each other carefully by RF cables with attenuators.
Below are the assumptions I took into account when I was making the second flowgraph. Please tell me if im on the right path.
Transmitter side : The packet decoder and GFSK mod blocks use 20 samples per symbol. Baud rate is 9600 and sample rate is 20*baud rate = 192K. Since the expected symbol rate by the satellite is baud_rate = 9600, I included a rational resampler and set UHD symbol rate to baud_rate. Is my logic correct?
GFSK mod and demod : For both of these blocks, I calculated sensitivity as S = Pi * Modulation_index/Samples_Per_Symbol. The default BT value of 0.5 is used. Are my calculations sound? Is there a link for to find documentation for GFSK blocks? My derivations are based on the GFSK python source code which is a poor substitute for documentation.
Packet Encoder/Decoder : The output of packet decoder is null even though the GFSK demod block give some kind of output which is rather meaningless. Is this normal? What is the meaning of the threshold variable and why its value is -1?
I'm a newbie in GNU Radio as well as GFSK in general. So please drop me any further references.
Thanks in advance.
Moses.
I was finally able to solve the problem. All I did was re-implementing the GFSK demod in GRC. If you go into source of gfsk.py, you will find out that the blocks used are Quadrature Demod --> M&M Clock recovery --> Binary slicer which can easily be connected in GRC directly. As Marcus suggested in my other thread, GFSK demodulation with Xlating filter in GNU Radio , I replaced the M&M Clock recovery block with PFB block. My flowgraph is shown below.
Even if I can not answer all of your questions, I provide below some thoughts:
When using hardware devices the Throttle MUST be removed from the flow-graph. The hardware device is now responsible for the rate limiting. Mixing hardware device and the Throttle block may break the real-time boundary of your flow-graph required by the device. Underflows or Overflows messages should be produced by the UHD driver in such a case.
Are you sure that the USRP can support the requested sampling rate? You may need also to change the master_clock_rate of the device, if the requested sampling rate is not an integer decimation of the clock. If this is not possible consider some kind of re-sampling.
EDIT: The B200 can not provide 192e3 sampling rate with the default clock. You can set the master_clock_rate at 19.2e6. The hardware will apply then the proper decimation. The master_clock_rate can be changed either by the device specific arguments or the Clock Rate field of the UHD Sink/Source blocks that presents in the latests GNU Radio versions.
I'm using ublox NEO-M8N-0-01 GNSS module.
This module supports up to 5Hz GPS+GLONASS and 10 Hz GPS only.
However, when I try to change the sampling rate (via UBX-CFG-RATE in the messages view) I can only increase it to 5 Hz (Measurement period = 200ms). Any value below 200ms is impossible (changes the box to pink).
It happens even if I only produce NMEA message GxGGA.
The way I made it only GPS was via UBX-CFG-GNSS
Has anyone encountered this issue?
Thanks in advance
Roi Yozevitch
You don't say how you are setting the rate however going by your description I'm assuming you are using the ublox u-center software.
There is a simple explanation for this issue and a simple solution: Their software has a bug in (or wasn't updated to match the final specification of the part).
The solution is to not use u-center, it's the PC software that's complaining not the receiver. The receiver itself doesn't care what the spec sheet says, it will try it's best to run at whatever rate you request.
Sending commands directly I've managed to get a fairly reliable 10Hz GPS+Glonass. There is the occasional missing point but most of the time it keeps up.
Running GPS only you can get faster than 10Hz. If you play with the settings and restrict it to 8 channels 18-19Hz is fairly reliable. Unfortunately 20Hz is pushing it too far, you end up getting positions at 10Hz.
Obviously when running at these update rates make sure that your baud rate is high enough to cope with the requested messages and rate.
Hi Im using the following RF module
http://www.apogeekits.com/rf_receiver_module_rx433.htm
on an embedded board with the PIC16F628A. Sadly, I realized that the signal strength was in analog form and couldn't get any ideas to get the RSSI reading off the pin because well my PIC is digital DUH!.
My basic idea was
To get the RSSI value from my Receiver
Send it to the PIC
Link the PIC to a PC via RS232
Plot a graph of time vs RSSI of the receiver (so I can make out how close my TX is to my RX)
I thought it was bloody brilliant at first but ive hit a dead end here. Any ideas on getting the RSSI data to my PC from this receiver would be nice.
Thanks in Advance
You can get a PIC that has an integrated ADC for sampling the analog signal. Or, you can use an external ADC chip to do the conversion. You would connect that to your PIC using SPI or I2C.
The simplest thing to do is obviously to use a more appropriate microcontroller - one with an ADC! There are many (most), including PICs (though that wouldn't be my first choice).
Attaching an external SPI or I2C ADC might be a bit tedious since having no SPI or I2C on your part, you'd have to bit-bash it. If you do that, use an SPI part - its simpler. Your sample rate will suffer and may end-up being a bit jittery if you are not careful.
Another solution is to use a voltage controlled PWM, then use the timer input capture to time the pulse width. That will give you good regularity and potentially good resolution. You can get a chip (example) to do that, or grow your own. That last option requires a triangle wave input as well as the measured (control) voltage, but on the same site...
In a similar vein, you could use a low frequency VCO (example) and use the output to clock one of the timers, then using a second timer periodically sampling the first and reset it. The count will relate to the voltage, though not necessarily a linear relationship, linearisation could be none on the PIC or at the receiving PC - I'd go for the latter - your micro will suck at arithmetic (performance wise) - even integer arithmetic, especially if it involves division.