I want to modify the frequency or the voltage in Gem5 to save power.
How I can change them in Gem5?
Related
I am quite confused about the relationship between the declared MCU Clock frequency ( which we set from edit project window) and the oscillator frequency (which we set by codding). So for an example let's say I need 31 kHz frequency. Do I also need to set the MCU Clock frequency from edit project window? If yes, in case of I want to change the frequency during the process, how am I supposed to change the clock frequency during run time?( Since I will not be able to change declared clock frequency.)
If you are using MPLAB X, the Harmony Plug-In will create code from your settings in UI. So it's the same and initial frequency on startup. If you want to change the frequency during runtime, you have to set the registers accordingly - but this sounds somehow odd.
I'm trying to learn how the SPI protocol works, and I'm working on a basic project using the STM32F407G-Discovery board.
This board has a built-in accelerometer (LIS3DSH), and it uses the SPI protocol. In the user manual, it states the following:
The LIS3DSH has ±2g/±4g/±6g/±8g/±16g dynamically selectable full-scale
and it is capable of measuring acceleration with an output data rate
of 3.125 Hz to 1.6 kHz.
This accelerometer is using SPI1, which is connected to APB2. I'm using STM32CubeMX to generate the initialization code (including the clock configuration), and it looks like the APB2 peripheral clock has a default value of 84 Mhz.
Does this mean that I need to configure the APB2 peripheral clock to have such a value that it falls between the range of 3.125 Hz and 1.6 kHz? I can't imagine this is true because I can't get the value low enough
in STM32CubeMX since it throws an error if I go too low.
I'm also accounting for the baud rate control SPI register, which allows you to go as low as f-PCLK/256.
In other words, I'm a bit stuck on which clock frequency to use and which baud rate control to use.
I'm still learning embedded programming, and so my terminology might be incorrect.
the two are not related. the max SPI clock rate is 10Mhz (page 14). The out rate of 3.125Hz to 1.6Khz is how fast the chip does an acceleration conversion. At 3.125Hz, a new conversion result is ready every 320ms, and at 1.6Khz, they are available every 625us. There is a trade off between conversion rates, power consumption and accuracy. The data sheet leaves a lot of holes, I would suggest reading the MMA7660 data sheet to get a better understanding of how these types of chips work and then revert back to your datasheet for implementation details.
You could use the SPI clock frequency with up to 10MHz to get data from this chip.
(So a prescaler of 16 and the full rate (84MHz) APB2 clock would be ok)
The SPI clock determines how fast the data is transferred from the chip to the controller not how fast the chip generated new results.
To always get the newest data you could use the IRQ lines from the chip or use an timer to trigger the transmission corresponding to sampling rate.
I have a SCB 68A connector from National Instruments and I want to read out the open voltage from it. So I used the example code provided by National Instruments (https://decibel.ni.com/content/docs/DOC-28502):
I got 5 mV which is a reasonable value (I measured the noise signal with an oscilloscope). Now I want to read out the noise signal from few channels. So I sightly changed the VI (according to the documentation I need to create an array of channels and flatten them):
But now I read out approximately 200 mV on both channels (and one of them is the same as in the first VI). It doesn't make any sense.
What am I doing wrong?
I want the user to be able to choose the channels, so I can't just write "Dev1/ai0:4".
Edit: I'm using the DAQ 14.0.0.
Edit 2: 1) There is nothing connected to the deivce - I just want to read out the noise signal.
2) I'm using the connector in the MIO with the disabled temperature sensor mode (the default configuration).
You are observing charge injection from the DAQ device's multiplexer. Connect each aiN terminal to aignd and you will be able to measure the noise of the DAQ device.
Charge Injection
Most NI DAQ boards have a single analog to digital converter (ADC) and provide multiple input channels by using a multiplexer (MUX) to switch the input of the ADC to the different analog input terminals ai0, ai1, etc:
As NI explains, when the DAQ device's multiplexer moves from one channel to the next, it can introduce a small charge on each channel. Since the open channel does not have a path for this charge to dissipate, the voltage of the channel will increase. This can also cause the channel to rail, slowly floating up to the maximum input voltage (usually 10 V).
Characterizing Noise
You can determine the noise of each component in your system by:
Measuring the noise of the DAQ device
Measuring the noise of the DAQ device and terminal block
Subtracting the DAQ device noise (step 1) from the system noise (step 2)
When you're finished, the value from step 1 is the noise of the DAQ device, and the value from step 3 is the noise of the SCB-68.
To measure the noise of an electric path, there must be a complete circuit for the ADC to sample. For step 1, connect each aiN terminal to aignd and run your VI. For step 2, connect the terminal block to the DAQ device, disconnect the sensor, and connect the terminal block's channel terminals to its ground terminal and run your VI.
Minimizing Noise
In addition to charge injection, noise can be introduced to a DAQ system from several sources, including the environment. Open terminals act like small antennas and receive radiated energy from other electronics, lights, and the AC mains.
The link also outlines how to find and minimize noise, but the gist is:
Systematically identify the sources of the noise.
Remove sources of noise that aren't necessary for your measurements.
Depending on the nature and source of the remaining noise, use appropriate shielding, cabling, and terminal configuration.
Over-sample and average the signal.
Please have a look on the links below:
http://forums.ni.com/t5/Multifunction-DAQ/How-to-use-DAQmx-Read-to-measure-multiple-analog-channels/td-p/2620949
http://digital.ni.com/public.nsf/allkb/A3A05920BF915F1486256D210069BE49
There is the complete solution to your question.
I don't have much knowledge about microcontrollers. In my project, I need to shift the sine wave. Here, I want to know, if I feed pure sine at port A pin 2. Then, will i get the shifted version of pure sine wave at port B pin 2 . will the following instruction work?
Inialise port A as input and port B as output
call delay
portb=porta
we can generate sine wave using DAC in microcontroller. But, as it is not perfect, it wont meet required conditions.
First of all the input needs to be to an ADC, and the output needs to be from a DAC (or a PWM with appropriate output filtering). It is not clear from your question that the pins you have chosen are appropriate for that.
If you are generating the sine from the DAC, why would you apply it to an input only to output it again? If you need two sine waves shifted in phase, why not simply generate calculated outputs from two DAC or PWM? Either way you need two analogue outputs, but that way you do not need any input. A PWM will need greater analogue filtering than a DAC and is likely to support lower bandwidth, but most microcontroller have more PWMs than DACs.
You cannot simply call a delay than copy port a to port b, that would be simply be a copy of a to b after a delay. You need to take samples from A and place then in a FIFO buffer, then apply the output of the FIFO to B. The length of the FIFO determines the delay.
A microcontroller is not an analogue device, you cannot put in an analogue signal on just any old pin and and transfer that signal to another pin. Most pins are digital GPIO, they except just two states representing 0 or 1. No matter what voltage you apply, it will be interpreted as either high or low.
Rather you will have to use an ADC input, sample at sufficiently high frequency, delay the samples through a FIFO, then apply the delayed samples to a DAC. Reconstruction of a "pure" sine wave from the quantized DAC output requires analogue filtering circuitry. With a filter cut-off lower than half the sampling rate you will recover a reasonably good representation of the original signal (which can be any signal with components below half the sampling frequency - it need not be a sine wave). If you do use a more complex signal, you will need to analogue filter the input to remove components above half the sampling rate to avoid aliasing.
It might be possible to do all that on one chip using a Cypress PSoC, since these are hybrid chips with reconfigurable analogue elements as well as a microcontroller.
How do I feed the waveform pulses into DAQ Assistant to cause a DAQ 6259 board to generate desired voltage pulses?
Using the Simulate Signal express VI I have created a square pulse waveform.
My goal is to allow a LabView user to configure the Frequency and Pulse width using knobs from the GUI as needed in order to generate a desired pulse train. This pulse train should be sent to the DAQ 6259 to generate a voltage pulse train. The voltage pulse train would be captured by an oscilloscope in order to verify its correctness (i.e. the captured pulse train looks exactly like the waveform displayed in the labview GUI).
What is the simplest way this can be achieved? Are there any tutorials that explain how this can be done?
Have you checked out the example finder (Help>Find Examples...)
Hardware Input and Output > DAQmx> Analog Output / Digital Output
There are a bunch of examples in there that will get you 90% of the way