I am running long simulations using project_flow and SUMO. but i don't need the GUI to refresh every step of the simulation. is there any way to
de-couple the simulation step and the gui refresh rate?
I do not know how this works in the context of flow but in general you can simply use the sumo executable instead of sumo-gui to have the same simulation without a GUI. Modifying the refresh rate is not possible, as far as I know, but minimizing the window should at least avoid all draw requests from the system.
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've set up an experiment in the Builder to obtain rapid reaction times to audio stimuli, and I've subsequently been playing with the code to get the experiment to do exactly what I want. In particular, I'd like very accurate reaction times, so the program would ideally hog the CPU from the onset of each stimulus until a fixed point afterwards, and record keypresses of "w" and "e" during this time.
In an attempt to achieve this, I've been resetting the clock at the start of the audio stimuli, then hogging the CPU for for 2secs, as follows:
event.clearEvents(eventType='keyboard')
response.clock.reset()
core.wait(2,2)
if response.status == STARTED:
theseKeys = event.getKeys(keyList=['w', 'e'])
This seems to work fine. However, I have one concern: the documentation for the core.wait command says:
If you want to obtain key-presses during the wait, be sure to use pyglet.
How would I know if I'm using pyglet? Is it automatic, or do I need to alter the script in some way to ensure that I'm using it?
This refers to the type of window (pyglet or pygame) that you are using to display your stimuli. PsychoPy will generally use pyglet, but to be sure, you can explicitly set the window type when you create it. See the window API at http://www.psychopy.org/api/visual/window.html:
winType : None, ‘pyglet’, ‘pygame’
If None then PsychoPy will revert
to user/site preferences
More importantly, make sure you are using the pyo audio library rather than the default pygame. Set this in the PsychoPy Preferences -> General -> Audio Library dialog box field. Pygame definitely has sound latency problems: you should assume that there is a substantial lag between telling a sound to play and sound actually being produced. Pyo does better apparently, but I think you should validate this independently in some way to ensure that your reaction times to auditory stimuli are meaningful.
We're undertaking a large project that is focused on delivering automated testing of the software that we produce.
We have a lot of "events" that trigger certain behavior at specific times. Ideally, we would be able to exercise these tests in an automated fashion without the need to move the system clock in intervals to specific points in time.
To that end, I'm wondering if there is a way (with VMWare, or any other virtualization software) to increase the speed of the system clock of the guest operating system. I'm not interested in measuring performance in these tests, only functionality.
Is there anything out there that would allow for this behavior?
It works for VirtualBox:
VBoxManage setextradata "VM name" "VBoxInternal/TM/WarpDrivePercentage" x
where x is the percentage you want (for instance, 200 is doubling, 50 is halving)
You can also more information here, on the section "Accelerate or slow down the guest clock". Regards.
I was able to work around this using the Win32 API SetSystemTimeAdjustment()
This allows you to increase the amount of time added to the system clock for each OS tick interval. It's meant generally for addressing clock skew, but can be used outside of that particular context.
I don't see what the benefits are of testing this in a fast-forwarding VM instead of unit testing the event trigger using a mock implementation of the date/time dependency.
The only thing you "gain" by testing this in a fast-forwarding VM is that you test both the system's and the programming language's date/time implementation, which I think you are save to trust because it is used, developed and tested by so many for such a long time.
The BOINC client (does distributed processing jobs like SETI#home does) is able to turn processing on or off based on whether other processes are using a certain percentage of CPU time. That is, if the user starts to do some work and their processes start using 60% CPU, BOINC can pause to avoid interfering with the user's work.
I would like to do the same thing (monitor CPU usage by other processes). The difficulty as I see it is not monitoring CPU usage, but rather making sure that the information isn't skewed by my own usage. For example, if my process is using a ton of CPU time it may prevent another process from using enough to trigger the pause.
Can someone point me in the right direction? Even a suggestion for what to search for would be useful. I'm not really sure what this feature would be called.
You can use NSTask to set the 'nice' value of the process when your process starts.
Also [[NSThread mainThread] setThreadPriority:0.0]
where priority value is between 0.0 and 1.0 is a Cocoa API which may save you frakking about with sudo
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.