Can Frame Channel Conversion Library Download
Robert Trimble
When dealing with CAN, most new developer struggle with the Frame and Channel (or Signal) conversion. Developers are trying to get away with using the cheapest CAN hardware, and as a result are opting to not use XNet hardware but instead use things like the USB 8473, or using non-NI hardware like the ValueCAN by Intrepid, or using Vector hardware, or CAN to serial adapaters. The problem with all of these is you generally just read and write frames which is the raw form of CAN. When using these cheap devices you can't ask the CAN bus what the signal Bus_Voltage is, instead you need to perform a frame read, find the frame associated with the signal, then pull out the bits for that signal, then scale the signal based on the signal selected.
All of this is doable but a pain and can be very custom. That's one reason NI came out with the Frame Channel Conversion Library. This library converts from Frames to Channels using an industry standard CAN database file. The problem with this library is it hasn't been updated in 5 years, and has known issues which likely will never be fixed.
So I wrote a wrapper around the XNet conversion library to handle going from signals to frames, or from frames to signals. Anyone looking to use a DBC file, on hardware that only supports frame API should use this conversion library, or at least use the XNet conversion sessions.
Oh wow I did not know of this limitation. So just to clairify. If you have multiple frames in a database. And not all frames are of the same payload, when the conversion takes place it will try to use each frame. But when converting a frame of say 4 bytes and you provide 8 an error occurs. So even if the frame is not associated with the conversion you are performing, it will throw an error.
Convert from OpenCV img to PIL img will lost transparent channel. While convert PIL img to OpenCV img will able to keep transparent channel, although cv2.imshow not display it but save as png will gave result normally.
Using the struct module, you can take the wave frames (which are in 2's complementary binary between -32768 and 32767 (i.e. 0x8000 and 0x7FFF). This reads a MONO, 16-BIT, WAVE file. I found this webpage quite useful in formulating this:
I needed to read a 1-channel 24-bit WAV file. The post above by Nak was very useful. However, as mentioned above by basj 24-bit is not straightforward. I finally got it working using the following snippet:
A wave file can carry more than one audio channel. It could be mono, stereo, surround or other multichannel-configurations. Mono would be the simple case, but in multichannel wavs the samples are typically interleaved. That means one frame will carry samples from all channels in alternating fashion. Assuming Stereo, that might be:
Your project is going to need to have some way of controlling many channels of lights. You can use the 6 analog out channels the Arduino has built-in, or you can use various PWM expansion chips/shields available.
You can customize the name and color of your channels. Just right click on a channel name and select Channel properties. This will help you track what the channels do. For example, see how I setup the channels for a recent project:
There are two common types of operations that impact the frame and sample rates of a signal: Frame rebuffering and direct rate conversion. Frame rebuffering, which is used to alter the frame size of a signal in order to improve simulation throughput, usually also changes either the sample rate or the frame rate of the signal. Direct rate conversions such as upsampling and downsampling can be implemented by altering either the frame rate or the frame size of a signal. For more details on the direct rate conversion technique, see Convert Sample and Frame Rates in Simulink Using Rate Conversion Blocks.
Sometimes you might need to rebuffer a signal to a new frame size at some point in a model. For example, your data acquisition hardware may internally buffer the sampled signal to a frame size that is not optimal for the signal processing algorithm in the model. In this case, you can rebuffer the signal to a frame size more appropriate for the intended operations without introducing any change to the data or sample rate.
Buffering operations provide another mechanism for rate changes in signal processing models. The purpose of many buffering operations is to adjust the frame size of the signal M without altering the sample rate of the signal Ts. This operation usually results in a change to the frame rate of the signal Tf according to the following equation:
However, this equation is true only if no samples are added to or deleted from the original signal. Therefore, this equation does not apply to buffering operations that generate overlapping frames, that only partially unbuffer frames, or that alter the data sequence by adding or deleting samples.
Some forms of buffering alter the signal data or sample period in addition to adjusting the frame size. This type of buffering is desirable when you want to create sliding windows by overlapping consecutive frames of a signal, or when you want to select a subset of samples from each input frame for processing.
A signal with a sample period of 0.125 seconds is rebuffered from a frame size of 8 to a frame size of 16. This rebuffering process doubles the frame period from 1 to 2 seconds, but does not change the sample period of the signal, . The signal is then unbuffered into a sequence of sample outputs using the Unbuffer block. The frame period then changes to 0.125 seconds, which is equal to the value of the sample period of the signal.
The Signal From Workspace block has the Sample time parameter set to 0.125, and the Samples per frame parameter is set to 8. Each frame in the generated signal contains 8 samples and has a sample period of 0.125 seconds.
The Buffer block has the Output buffer size (per channel) parameter set to 16, and the Buffer overlap parameter is set to 0. The Buffer block rebuffers the signal from a frame size of 8 to a frame size of 16.
To view the effect on the frame period of the signal, enable color coding, annotations, and timing legend by selecting Information Overlays > Colors, Text, Timing Legend. In the Timing Legend, you can view the value of the frame period for each signal in the model, the color associated with the frame period, and the corresponding annotation.
As you can see, the input frame period of the signal (denoted by D2 in the model), is given by or and equals 1 second. The Buffer block doubles the frame period from 1 to 2 seconds. The Unbuffer block that follows unbuffers the signal into a sequence of scalar outputs. The frame period of the unbuffered sequence equals 0.125 seconds, which matches the sample period of the signal.
Some forms of buffering alter the signal data or sample period in addition to adjusting the frame size. In the following example, a signal with a sample period of 0.125 seconds is rebuffered from a frame size of 8 to a frame size of 16 with a buffer overlap of 4 samples.
The Buffer block has the Output buffer size (per channel) parameter set to 16, and the Buffer overlap parameter is set to 4. The Buffer block rebuffers the signal from a frame size of 8 to a frame size of 16. After the initial output, the first four samples of each output frame are made up of the last four samples from the previous output frame.
where is the output frame size and equals 16, is the overlap and equals 4, and is the input sample period and equals 0.125 seconds. Substituting these values, the output frame period of the Buffer block becomes or seconds. The corresponding sample period of the signal equals or seconds. When you unbuffer the signal into a sequence of sample outputs, the frame period of the signal (shown as D2 in the model) matches the sample period value of 0.0938 seconds. Thus, both the data and the sample period of the signal have been altered by the buffering operation.
For seekable output streams, the wave header will automatically be updatedto reflect the number of frames actually written. For unseekable streams, thenframes value must be accurate when the first frame data is written. Anaccurate nframes value can be achieved either by callingsetnframes() or setparams() with the numberof frames that will be written before close() is called andthen using writeframesraw() to write the frame data, or bycalling writeframes() with all of the frame data to bewritten. In the latter case writeframes() will calculatethe number of frames in the data and set nframes accordingly before writingthe frame data.
Make sure nframes is correct, and close the file if it was opened bywave. This method is called upon object collection. It will raisean exception if the output stream is not seekable and nframes does notmatch the number of frames actually written.
Write audio frames and make sure nframes is correct. It will raise anerror if the output stream is not seekable and the total number of framesthat have been written after data has been written does not match thepreviously set value for nframes.
A config/init pattern is used throughout the entire library. The idea is that you set up a configobject and pass that into the initialization routine. The advantage to this system is that theconfig object can be initialized with logical defaults and new properties added to it withoutbreaking the API. The config object can be allocated on the stack and does not need to bemaintained after initialization of the corresponding object.
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