The following image is from the datasheet of SRAM IS64WV51216BLL(page 15).
It is a SRAM's write timing diagram.I don't know the meaning of double transitions for WE signal.I have circled it in red.
SRAM write timing diagram
Timing Diagram Basics
Understanding Timing diagrams of digital systems
How to Read Timing Diagrams: A Maker’s Guide
Acturally I have searched many material,but nothing I wanted.Please help me!Thank you very much.
That is not multiple transitions, rather it is a timing window where the transition can happen which is bounded by two timing constraints: tSA and tPWE.
tSA is the Address Setup time which is the earliest that W̄ can be asserted. tPWE is the latest W̄ can be asserted and not violate the Pulse Width.
Related
I'm working on a guitar effects "pedal" using the NEXSYS A7 Board.
For this purpose, I've purchased the I2S2 PMOD and successfully got it up and running using the example code provided by Digilent.
Currently, the design is a "pass-through", meaning that audio comes into the FPGA and immediately out.
I'm wondering what would be the correct way to store the data, make some DSP on this data to create the effects, and then transmit the modified data back to the I2S2 PMOD.
Maybe it's unnecessary to store the data?
maybe I can pass it through an RTL block that's responsible for applying the effect and then simply transmit the modified data out?
Collated from comments and extended.
For a live performance pedal you don't want to store much data; usually 10s of ms or less. Start with something simple : store 50 or 100ms of data in a ring (read old data, store new data, inc address modulo memory size). Output = Newdata = ( incoming sample * 0.n + olddata * (1 - 0.n)) for variable n. Very crude reverb or echo.
Yes, ring = ring buffer FIFO. And you'll see my description is a very crude implementation of a ring buffer FIFO.
Now extend it to separate read and write pointers. Now read and write at different, harmonically related rates ... you have a pitch changer. With glitches when the pointers cross.
Think of ways to hide the glitches, and soon you'll be able to make the crappy noises Autotune adds to most all modern music from that bloody Cher song onwards. (This takes serious DSP : something called interpolating filters is probably the simplest way. Live with the glitches for now)
btw if I'm interested in a distortion effect, can it be accomplished by simply multiplying the incoming data by a constant?
Multiplying by a constant is ... gain.
Multiplying a signal by itself is squaring it ... aka second harmonic distortion or 2HD (which produces components on the octave of each tone in the input).
Multiplying a signal by the 2HD is cubing it ... aka 3HD, producing components a perfect fifth above the octave.
Multiplying the 2HD by the 2HD is the fourth power ... aka 4HD, producing components 2 octaves higher, or a perfect fourth above that fifth.
Multiply the 4HD by the signal to produce 5HD ... and so on to probably the 7th. Also note that these components will decrease dramatically in level; you probably want to add gain beyond 2HD, multiply by 4 (= shift left 2 bits) as a starting point, and increase or decrease as desired.
Now multiply each of these by a variable gain and mix them (mixing is simple addition) to add as many distortion components you want as loud as you want ... don't forget to add in the original signal!
There are other approaches to adding distortion. Try simply saturating all signals above 0.25 to 0.25, and all signals below -0.25 to -0.25, aka clipping. Sounds nasty but mix a bit of this into the above, for a buzz.
Learn how to make white noise (pseudo-random number, usually from a LFSR).
Multiply this by the input signal, and mix or match with the above, for some fuzz.
Learn digital filtering (low pass, high pass, band pass for EQ), and how to control filters with noise or the input signal, the world of sound is open to you.
This is an architectural question regarding gnuradio(-companion) and since I am not sure how to tackle this problem in the first place I first describe what I want to achieve and then how I think I would to it.
Problem
I implement a special form of an RFID reader with an Ettus X310 SDR: The transmitter sends an OOK/AM modulated (PIE encoded) request, followed by a pure Sine wave. The RFID tag backscatters its response onto this sine wave using OOK/AM modulation in FM0 or "Miller subcarrier" coding (a form of a differential Manchester coding). I want to receive its response, translate it into bits (and form a PDU), buffer different responses in a FIFO and send them for further processing. The properties of the tag response are:
It is asynchronuous. I do not know when the response is coming and if it does, when the proper sampling times are: I cannot simply filter, sample, decimate the signal and use a simple slicer because I do not know what the sample points are.
The response comes into very small "bursts" (say, 100 bits). Hence I cannot afford performing timing recovery on bits and waste them (except I buffer the entire signal somehow which I do not think is the way to do it).
The signal starts with a small preamble (UHF RFID Gen2 preamble) which is 6 bits (~8 bit transitions). This may not be enough for for time recovery but can be used to identify the start of a response somehow.
It uses mentioned FM0 encoding, so I have a guaranteed transition every bit. For that reason, I do not have to sample them but could detect the transitions and convert them into bits. I would not need conventional clock recovery (e.g. M&M) either.
My Thoughts
"Ordinary" gnuradio preprocessing brings me to the received oversampled bits: Downconversion, filtering; possibly a slicer which uses a lowpass filter to subtract the mean value and a comparator (note that even this may be challenging because the lowpass filter may have a large settling time of few bits until it obtains the right mean value).
In order to detect the actual transmission, I do not think I have much choice other than a simple squelch that detects a higher signal level than the noise floor (is this true or is there a way to detect the transmission using the preamble only?)
Once the squelch block detects a transmission, I could use a differentiator (or similar) to get the edges. But my understanding of the transition between this "baseband land" and "bits/PDUs" ends: I would need a block that triggers asynchronously (rather than samples at fixed intervals). In an actual system, the edges from the described detector could act as clock input of a flip flop. However, I do not see which standard gnuradio block would allow me to do this.
Once in "bits land", the bits (or PDUs) would be processed at a much lower rate. However, two clock domains are crossed: the normal baseband sampling rate, an irregular rate by which the transitions are detected and the rate at which the bits are read. For that reason, I would be looking for a FIFO or shift register, in which the detected bits are shifted in at whichever edge transition rate they come in and read out at the regular bit rate on the other side.
Question
What is the correct architecture/approach to implement this in gnuradio?
I could imagine to implement this with my own blocks. But as much as possible I would like to use standard block, gnuradio-companion. I would like to resort to own blocks (in particular C++) only as last resort if either not possible otherwise or if it would really not be the right way to so it otherwise.
I need to use keyboard as input for musical notes, and digilent speaker as output.
I plan to use only one octave.
My most intriguing questions are:
How do I represent the musical notes in VHDL code.
How do I (or do I need to) implement a DAC module that uses Spartan 3E Starter's built-in DAC? I have read on other forums that it can't be implemented. I need to use it in order to transmit the note to the speaker. The teacher who supervises my and my colleagues' projects suggested me to look into PWM for that(but all I've found is explained in electronic manner, no accompanying code, or explanation on implementation).
Besides keyboard controller, a processing module(for returning the note corresponding to the pressed key from the notes vector) and DAC, that I have figured out so far that I need, what else do I need.
There is a DAC (see comments)
There is no DAC on the Spartan-3E Starter Kit. Using a low-pass PWM signal is a common way to generate analog signal level from digital output.
You need to define a precision for your PWM, let's say 8 bits or 256 levels. For each audio sample you want to output, you need to count from 0 to 255. When the counter is less than the desired sample level, output 1, otherwise output 0. When the counter reach 255, reset it and go to the next sample.
Thus, if you want 8 bits precision (256 levels) and 8KHz signal, the counter will have to run at 256*8000 = 2.048MHz.
For your other questions, there is no easy answer. It's your job, as designer, to figure that out.
I am doing a VLSI Project and I am implementing a Barrel Shifter using a tool called DSCH.The schematic for the same is realized using Transmission Gates.
What the circuit does is, it ROTATES the 8 bit word(8-bit shifter) with as many rotations chosen from a decoder in one clock cycle.
But I want to know the use of a rotator and why is it still called a shifter even though it's rotating.
Also please help me with some applications regarding Rotator which could be added to the present circuit to show it's use?
Rotation is just shifting with the bit exiting from one end fed back into the input at the other end, possibly by way of the carry flag bit. At the level of a simple implementation, it would make sense to have one circuit for both operations, with some additional control lines to select the source at the input side between the output of the other side, 0, or 1. Sign extension during right shift of 2's complement numbers would be another selectable option often built in.
The stackexchange sites aren't really suited to "list" questions including about applications, but a couple come to mind:
If you want a vector to test every bit of another value in turn, and to do so repeatedly, you could just keep rotating an initial one-bit-active value over an over, never having to re-initialize it.
You could swap a two-part (typically double-byte) value to imitate the opposite endieness of encoding by rotating it half way. Or putting it another way, it can be a single-operation swap of the values of two pairable but also independently accessible registers (think AL and AH together making up AX in real mode x86). But this will not work two endian swap a four-part value, such as a 32-bit value on a byte-addressable machine.
Various coding, checksum, and hashing schemes may wish to transform a value
While doing clock domain conversions (rate matched) we usually double flop the data to avoid meta-stable states. Double flopping just reduces the probability of meta-stability. Triple flopping will reduce it further.
How to calculate the probability/relationship between meta-stability and number of clock domain flops used?
The canonical answer to metastability queries always involves referring to articles written by the late, great, Peter Alfke. In particular, the XAPP094 appnote - don't worry about the age of it, the theory is still the same.
There are also numbers for some more recent families available - although I can't see anything for the 6 and 7 series as yet.