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.
Related
I'm trying to write a code in which every 1 ms a number plused one , should be replaced the old number . (something like a chronometer ! ) .
the problem is whenever the cpu usage increases because of some other programs running on the pc, this 1 milliseconds is also increased and timing in my program changes !
is there any way to prevent cpu load changes affecting timing in my program ?
It sounds as though you are trying to generate an analogue output waveform with a digital-to-analogue converter card using software timing, where your software is responsible for determining what value should be output at any given time and updating the output accordingly.
This is OK for stationary or low-speed signals but you are trying to do it at 1 ms intervals, in other words to output 1000 samples per second or 1 ks/s. You cannot do this reliably on a desktop operating system - there are too many other processes going on which can use CPU time and block your program from running for many milliseconds (or even seconds, e.g. for network access).
Here are a few ways you could solve this:
Use buffered, hardware-clocked output if your analogue output device supports it. Instead of writing one sample at a time, you send the device a waveform or array of samples and it outputs them at regular intervals using a timing signal generated in hardware. Unfortunately, low-end DAQ devices often don't support hardware-clocked output.
Instead of expecting the loop that writes your samples to the AO to run every millisecond, read LabVIEW's Tick Count (ms) value in the loop and use that as an index to your array of samples: rather than trying to output every sample, your code will now say 'what time is it now, and therefore what should the output be?' That won't give you a perfect signal out but at least now it should keep the correct frequency rather than be 'slowed down' - instead you will see glitches imposed on the signal whenever the loop can't keep up. This is easy to test and maybe it will be adequate for your needs.
Use a real-time operating system instead of a desktop OS. In the case of LabVIEW this would mean using the Real-Time software module and either a National Instruments hardware device that supports RT, such as the CompactRIO series, or installing the RT OS on a dedicated PC if the hardware is compatible. This is not a cheap option, obviously (unless it's strictly for personal, home use). In any case you would need to have an RT-compatible driver for your output device.
Use your computer's sound output as the output device. LabVIEW has functions for buffered sound output and you should be able to get reliable results. You'll need to upsample your signal to one of the sound output's available sample rates, probably 44.1 ks/s. The drawbacks are that the output level is limited in range and is not calibrated, and will probably be AC-coupled so you can't output a DC or very low-frequency signal. However if the level is OK for what you want to connect it to, or you can add suitable signal conditioning, this could be a neat solution. If you need the output level to be calibrated you could simultaneously measure it with your DAQ card and scale the sound waveform you're outputting to keep it correct.
The answer to your question is "not on a desktop computer." This is why products like LabVIEW Real-Time and dedicated deterministic hardware exist: you need a computer built around dedication to a particular process in order to consistently serve that process. Every application in a regular Windows/Mac/Linux desktop system has the problem you are seeing of potentially being interrupted by other system processes, particularly in its UI layer.
There is no way to prevent cpu load changes from affecting timing in your program unless the computer has a realtime clock.
If it doesn't have a realtime clock, there is no reason to expect it to behave deterministically. Do you need for your program to run at that pace?
I am having a problem porting a GNU Radio setup from PC (windows 10, USB3) to Raspberry Pi 2 (USB2). USB bandwidth and CPU should not be a problem I think (only around 30% utilization while running). Essentially it looks like the RPi is 'pausing' during transmission, while the PC is not. The receiver is running on PC in both cases. I am including a pic of what I see after the FSK demod when running transmitter on PC vs Pi (circled 'pause' area), as well as a picture of my (admittedly sloppy) schematic. Any help/tips is greatly appreciated.gnuradio schemreceived signals
Edit: It appears it may actually be processing limitations. Switching from 9400 baud to 2400 baud makes the issue go away. If anyone has experience with GNURadio...am I doing anything overly inefficient or should I just drop comm rate?
The first thing I would do would be to lower your sample rates.
You don't need 1.5Ms/s if you are going to keep only the lowest 32k in your low pass filter.
Then you could do the same for your second stage after the quadrature demod if it's not enough (by the way, the sample rate of your second low pass filter does not seem to match the actual sample rate of the stage which is still 1.5Ms/s if I'm not mistaken).
Anyway, Gnuradio uses a lot of processing power so try not to use a sampling rate way above what you actually need ;)
In your case, you could cut the incoming sample rate down to 64k (say 80 for safety). 18 times less samples to process might do the trick :)
I am trying to set up 5 USRP1 and some daughterboards on 2.4 and 5 GHz.
Some of them are out of order and some work properly, but I don't know which is which. I am trying to send a symbol (QAM modulation) sequence then I am trying to pass it with a file source to both a USRP sink and an FFT sink.
I am trying to find articles and tutorials, how sample rates are connected and how to set up them but I can't understand what am I missing. Could somebody please help with the schemes?
128 MS/s is not a rate that is possible with the USRP1. The console will contain a UHD warning that a differen, possible rate was chosen (most likely, 8MS/s).
Now, you contradict that rate by having a "Throttle" block in your flow graph - that block's job is only (and nothing more) to slow down the average rate at which samples are being let through – and that is something your "USRP Sink" already does. In fact, modern versions of GRC will warn you that using a throttle block in the same flow graph as a hardware sink or source is a bad idea.
Now, you'll say "ok, if the USRP sink actually needs to consume but 8MS/s, and my interpolator makes 128 MS/s out of my nominally 1M/s flow (really, signals within GNU Radio don't have a sampling rate), then that's gotta be fast enough to satisfy the 8MS/s demand!".
But the fact is that a 128-interpolator is really a CPU-intense thing, and the resulting rate might not be that high, made even worse by the choppy nature of how Throttle works.
In fact, your interpolator is totally unnecessary. The USRP internally has proper interpolators for integer fractions of its master clock rate 64MS/s, which means that you can set the USRP Sink to a sampling rate of 1MS/s and just directly connect the file source to it.
I have a very low speed data connection over serial (RS485):
9600 baud
actual data transmission rate is about 25% of that.
The serial line is going through an area of extremely high EMR. Peak fluctuations can reach 3000 KV.
I am not in the position (yet) to force a change in the physical medium, but could easily offer to put in a simple robust forward error correction scheme. The scheme needs to be easy to implement on a PIC18 series micro.
Ideas?
This site claims to implement Reed-Solomon on the PIC18. I've never used it myself, but perhaps it could be a helpful reference?
Search for CRC algorithm used in MODBUS ASCII protocol.
I develop with PIC18 devices and currently use the MCC18 and PICC18 compilers. I noticed a few weeks ago that the peripheral headers for PICC18 incorrectly map the Busy2USART() macro to the TRMT bit instead of the TRMT2 bit. This caused me major headaches for short time before I discovered the problem. Example, a simple transmission:
putc2USART(*p_value++);
while Busy2USART();
putc2USART(*p_value);
When the Busy2USART() macro was incorrectly mapped to the TRMT bit, I was never waiting for bytes to leave the shift register because I was monitoring the wrong bit. Before I realized the inaccurate header file, the only way I was able to successfully transmit a byte over 485 was to wait 1 ms between bytes. My baud rate was 91912 and the delays between bytes killed my throughput.
I also suggest implementing a means of collision detection and checksums. Checksums are cheap, even on a PIC18. If you are able to listen to your own transmissions, do so, it will allow you to be aware of collisions that may result from duplicate addresses on the same loop and incorrect timings.
I have an embedded device (Technologic TS-7800) that advertises real-time capabilities, but says nothing about 'hard' or 'soft'. While I wait for a response from the manufacturer, I figured it wouldn't hurt to test the system myself.
What are some established procedures to determine the 'hardness' of a particular device with respect to real time/deterministic behavior (latency and jitter)?
Being at college, I have access to some pretty neat hardware (good oscilloscopes and signal generators), so I don't think I'll run into any issues in terms of testing equipment, just expertise.
With that kind of equipment, it ought to be fairly easy to sync the o-scope to a steady clock, produce a spike each time the real-time system produces an output, an see how much that spike varies from center. The less the variation, the greater the hardness.
To clarify Bob's answer maybe:
Use the signal generator to generate a pulse at some varying frequency.
Random distribution across some range would be best.
use the signal generator (trigger signal) to start the scope.
the RTOS has to respond, do it thing and send an output pulse.
feed the RTOS output into input 2 of the scope.
get the scope to persist/collect mode.
get the scope to start on A , stop on B. if you can.
in an ideal workd, get it to measure the distribution for you. A LeCroy would.
Start with a much slower trace than you would expect. You need to be able to see slow outliers.
You'll be able to see the distribution.
Assuming a normal distribution the SD of the response time variation is the SOFTNESS.
(This won't really happen in practice, but if you don't get outliers it is reasonably useful. )
If there are outliers of large latency, then the RTOS is NOT very hard. Does not meet deadlines well. Unsuitable then it is for hard real time work.
Many RTOS-like things have a good left edge to the curve, sloping down like a 1/f curve.
Thats indicitive of combined jitters. The thing to look out for is spikes of slow response on the right end of the scope. Keep repeating the experiment with faster traces if there are no outliers to get a good image of the slope. Should be good for some speculative conclusion in your paper.
If for your application, say a delta of 1uS is okay, and you measure 0.5us, it's all cool.
Anyway, you can publish the results ( and probably in the publish sense, but certainly on the web.)
Link from this Question to the paper when you've written it.
Hard real-time has more to do with how your software works than the hardware on its own. When asking if something is hard real-time it must be applied to the complete system (Hardware, RTOS and application). This means hard or soft real-time is system design issues.
Under loading exceeding the specification even a hard real-time system will fail (hopefully with proper failure indication) while a soft real-time system with low loading would give hard real-time results. How much processing must happen in time and how much pre/post processing can be performed is the real key to hard/soft real-time.
In some real-time applications some data loss is not a failure it should just be below a certain level, again a system criteria.
You can generate inputs to the board and have a small application count them and check at what level data is going to be lost. But that gives you a rating specific to that system running that application. As soon as you start doing more processing your computational load increases and you now have a different hard real-time limit.
This board will running a bare bones scheduler will give great predictable hard real-time performance for most tasks.
Running a full RTOS with heavy computational load you probably only get soft real-time.
Edit after comment
The most efficient and easiest way I have used to measure my software's performance (assuming you use a schedular) is by using a free running hardware timer on the board and to time stamp my start and end of my cycle. Or if you run a full RTOS time stamp you acquisition and transition. Save your Max time and run a average on the values over a second. If your average is around 50% and you max is within 20% of your average you are OK. If not it is time to refactor your application. As your application grows the cycle time will grow. You can monitor the effect of all your software changes on your cycle time.
Another way is to use a hardware timer generate a cyclical interrupt. If you are in time reset the interrupt. If you miss the deadline you have interrupt handler signal a failure. This however will only give you a warning once your application is taking to long but it rely on hardware and interrupts so you can't miss.
These solutions also eliminate the requirement to hook up a scope to monitor the output since the time information can be displayed in any kind of terminal by a background task. If it is easy to monitor you will monitor it regularly avoiding solving the timing problems at the end but as soon as they are introduced.
Hope this helps
I have the same board here at work. It's a slightly-modified 2.6 Kernel, I believe... not the real-time version.
I don't know that I've read anything in the docs yet that indicates that it is meant for strict RTOS work.
I think that this is not a hard real-time device, since it runs no RTOS.
I understand being geek, but using oscilloscope to test a computer with ethernet/usb/other digital ports and HUGE internal state (RAM) is both ineffective and unreliable.
Instead of watching wave forms, you can connect any PC to the output port and run proper statistical analysis.
The established procedure (if the input signal is analog by nature) is to test system against several characteristic inputs - traditionally spikes, step functions and sine waves of different frequencies - and measure phase shift and variance for each input type. Worst case is then used in specifications of the system.
Again, if you are using standard ports, you can easily generate those on PC. If the input is truly analog, a separate DAC or simply a good sound card would be needed.
Now, that won't say anything about OS being real-time - it could be running vanilla Linux or even Win CE and still produce good and stable results in those tests if hardware is fast enough.
So, you need to simulate heavy and varying loads on processor, memory and all ports, let it heat and eat memory for a few hours, and then repeat tests. If latency stays constant, it's hard real-time. If it doesn't, under any load and input signal type, increase above acceptable limit, it's soft. Otherwise, it's advertisement.
P.S.: Implication is that even for critical systems you don't actually need hard real-time if you have hardware.