I'm using a Mediatek MT3333 GPS receiver (baudrate: 115200 bpS), but all I'm getting is this:
b'$GNGGA,132002.448,,,,,0,0,,,M,,M,,*5C\r\n'
b'$GPGSA,A,1,,,,,,,,,,,,,,,*1E\r\n'
b'$GLGSA,A,1,,,,,,,,,,,,,,,*02\r\n'
b'$GPGSV,1,1,00*79\r\n'
b'$GLGSV,1,1,00*65\r\n'
b'$GNRMC,132002.448,V,,,,,0.00,0.00,100417,,,N*5A\r\n'
b'$GNVTG,0.00,T,,M,0.00,N,0.00,K,N*2C\r\n'
After some research I found that my receiver doesn't have a fix, any idea how to solve this?
It looks that the received signal strength is low so that your GPS receiver mode doesn't get a GPS FIX. It would be better to place the device outdoor to verify if there is a stable reception.
From the GPS sentences showed above, your Mediatek MT3333 GPS receiver output modified NMEA 0183 Sentence. All the standard sentence should started with $GP as the suffix and with format of $GPaaa, where aaa is alphabetic.
For instance,
b'$GNRMC,132002.448,V,,,,,0.00,0.00,100417,,,N*5A\r\n' should be read as
$GPRMC,132002.448,V,,,,,0.00,0.00,100417,,,N*5A if conforms to NMEA. This sentence tells that at 2017-04-10 12:30:02 (GMT) got no GPS fix with speed at 0 knot and course at 0 degree.
If the output of your GPS receiver conforms NMEA, you can use some free software, such as VisualGPS, to evaluation the GPS signal quality.
If possible, suggest to change the GPS antenna to external one, an active GPS antenna with 2-stage amplifier at around 28dBm gain, to improve the GPS signal reception in order to get a stable fix.
From the datasheet of Mediatek MT3333, it did mention below for improving GPS signal reception:
An external antenna and high gain external LNA connected to the
internal LNA in low-gain mode, which offers high linearity. In this
configuration, external LNA gain ranging from 15 to 20 dB is
recommended. The maximum total external RF front end gain including
active antenna and external LNA can be 43dB.
Hope this help.
The wife asked for a device to make the xmas lights 'rock' with the best of music. I am going to use an Arduino micro-controller to control relays hooked up to the lights, sending down 6 signals from C# winforms to turn them off and on. I want to use NAduio to separate the amplitude and rhythm to send the six signals. For a specific range of hertz like an equalizer with six bars for the six signals, then the timing from the rhythm. I have seen the WPF demo, and the waveform seems like the answer. I want to know how to get those values real time while the song is playing.
I'm thinking ...
1. Create a simple mp3 player and load all my songs.
2. Start the songs playing.
3. Sample the current dynamics of the song and put that into an integer that I can send to which channel on the Arduino micro-controller via usb.
I'm not sure how to capture real time the current sound information and give integer values for that moment. I can read the e.MaxSampleValues[0] values real time while the song is playing, but I want to be able to distinguish what frequency range is active at that moment.
Any help or direction would be appreciated for this interesting project.
Thank you
Sounds like a fun signal processing project.
Using the NAudio.Wave.WasapiLoopbackCapture object you can get the audio data being produced from the sound card on the local computer. This lets you skip the 'create an MP3 player' step, although at the cost of a slight delay between sound and lights. To get better synchronization you can do the MP3 decoding and pre-calculate the beat patterns and output states during playback. This will let you adjust the delay between sending the outputs and playing the audio block those outputs were generated from, getting near perfect synchronization between lights and music.
Once you have the samples, the next step is to use an FFT to find the frequency components. Fortunately NAudio includes a class to help with this: NAudio.Dsp.FastFourierTransform. (Thank you Mark!) Take the output of the FFT() function and sum out the frequency ranges you want for each controlled light.
The next step is Beat Detection. There's an interesting article on this here. The main difference is that instead of doing energy detection on a stream of sample blocks you'll be using the data from your spectral analysis stage to feed the beat detection algorithm. Those ranges you summed become inputs into individual beat detection processors, giving you one output for each frequency range you defined. You might want to add individual scaling/threshold factors for each frequency group, with some sort of on-screen controls to adjust these for best effect.
At the end of the process you will have a stream of sample blocks, each with a set of output flags. Push the flags out to your Arduino and queue the samples to play, with a delay on either of those operations to achieve your synchronization.
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.
I am currently creating an application which signals readiness to other devices using a high frequency sound.
(transmitter): A device will produce a short burst of sound of around 20khz.
(receiver): Another device will be listening for a sound at this frequency at a small distance from the transmitter(10m approx) The device recieves audio data from a microphone
The background noise will be fairly loud, varying from around 0 - 10khz(about human speech range), and would be produced by a small crowd of people.
I need the receiving device to be able to detect the 20khz sound, separated from the noise,
and know the time at which it was received.
Any help with an appropriate algorithm, a library, or even better, code in C or
Objc to detect this high frequency sound would be greatly appreciated.
20 kHz may be pushing it, as (a) most sound cards have low pass (anti aliassing) filters at 18 - 20 kHz and (b) most speakers and microphones tend to have a poor response at 20 kHz. You might want to consider say 15 kHz ?
The actual detection part should be easy - just implement a narrow band pass filter at the tone frequency, rectify the output and low pass filter (e.g. 10 Hz).
You may want to look into FFT (Fast Fourier Transform). This algorithm will allow you to analyse a waveform and convert it to the frequency spectrum for further analysis.
If this is for Mac OS or iOS, I'd start looking into Core Audio's Audio Units.
1 Here's Apple's Core Audio Overview.
2 Some AudioUnits for Mac OS
3 Or for iOS AudioUnit Hosting
A sound with that high frequency will not travel at all with the iphone speaker.
i want to detect heart rate using iphone sdk does someone knows any method for calculating heartbeat rate?
Fast Fourier Transform is a class of algorithms that can quickly turn samples into an analysis that tells you how prominently ceratin frequencies occur in that sample. For more check out:
Wikipedia: FFT
Literate program example: Cooley-Tukey FFT
This is relevant to your problem because: (1) heart rate is itself a frequency, and (2) most of the sound that comes through the body that you can measure will be within a certain frequency range. Dropping frequencies outside this range means dropping all or mostly noise.
Good luck!
Well I've seen various implementations. Some of them use the accelerometer to detect minute movements in your arm/hand when you hold the phone, some of them can use the microphone, you could also do a manual 'tap' interface where you tap the screen while checking your own pulse.