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.
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.
I am working on designing a mandelbrot viewer and I am designing hardware for squaring values. My squarer is recursively built where a 4bit squarer relies on 2, 2bit squarers. so for my 16 bit squarer, that has 2 8bits squarers, and each one of those has 2 4bit squarer's.
As you can see the recursivity begins to make the design blow up in complexity. To help speed up my design i would like to use a 4input ROM that emulates a 4bit squarer. So when you enter 3 in the rom, it outputs 9, when you enter 15, it outputs 225.
I know that a normal LUT implemented in a logic cell ay have 3 or 4 input variables and only 1 output, but i need an 8 bit output so I need more of a ROM then a LUT.
Any and all help is appreciated, Im curious how the FPGA will store those ROMs and if storing it in ROM would be faster than computing the 4input Square.
-
Jarvi
To square a 4-bit number explicitly using LUTs, you would need to use 8 4-input LUTs. Each LUT's output would give you one bit of the 8-bit product.
The overall size and fmax performance of your design may be achieved with this approach, using larger block RAM primitives (as ROM), dedicated MAC (multiply-accumulate) units, or by using the normal mulitiplication operator * and relying on your synthesis tool's optimization.
You may also want to review some research papers related to this topic, for example here.
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.
a real junior question with hopefully a junior answer, regarding one of the main assignments of VHDL (concurrent selective assignment) can anyone explain what a VHDL compiler would synthesise the following description into?
LIBRARY IEEE;
USE IEEE.std_logic_1164.ALL;
USE IEEE.numeric_std.ALL;
ENTITY Q2 IS
PORT (a,b,c,d : IN std_logic;
EW_NS : OUT std_logic
);
END ENTITY Q2;
ARCHITECTURE hybrid OF Q2 IS
SIGNAL INPUT : std_logic_vector(3 DOWNTO 0);
SIGNAL EW_NS : std_logic;
BEGIN
INPUT <= (a & b & c & d); -- concatination
WITH (INPUT) SELECT
EW_NS <= '1' WHEN "0001"|"0010"|"0011"|"0110"|"1011",
'0' WHEN OTHERS;
END ARCHITECTURE hybrid;
Why do I ask? well I have previously gone about things the wrong way i.e. describing things on VHDL before making a block diagram of the components needed. I would envisage this been synthed as a group of and gate logic ?
Any help would be really helpful.
Thanks D
You need to look at the user guide for your target FPGA, and understand what is contained within one 'logic element' ('slice' in Xilinx terminology). In general an FPGA does not implement combinatorial logic by connecting up discrete gates like AND, OR, etc. Instead, a logic element will contain one or more 'look-up tables', with typically four (but now 6 in some newer devices) inputs. The inputs to this look up table (LUT) are the inputs to your logic function, and the output is one of the outputs of the function. The LUT is then programmed as a ROM, allowing your input signals to function as an address. There is one ROM entry for every possible combination of inputs, with the result being the intended logic function.
A function with several outputs would simply use several of these LUTs in parallel, with the same inputs, one LUT for each of the function's outputs. A function requiring more inputs than the LUT has (say, 7 inputs, where a LUT has only 4), simply combines two LUTs in parallel, using a multiplexer to choose between the output of the two LUTs. This final multiplexer uses one of the input signals as it's control, and again every possible combination of inputs is accounted for.
This may sound inefficient for creating something simple like an AND gate, but the benefit is that this simple building block (a LUT) can implement absolutely any combinatorial function. It's also worth noting that an FPGA tool chain is extremely good at optimising logic functions in order to simplify them, and to better map them into the FPGA. The LUT provides a highly generic element for these tools to target.
A logic element will also contain some dedicated resources for functions that aren't well suited to the LUT approach. These might include dedicated carry chains for adders, multiplexers for combining the output of several LUTS, registers (most designs are synchronous). LUTs can also sometimes be configured as small shift registers or RAM elements. External to the logic elements, there will be more specific blocks like large multipliers, larger memories, PLLs, etc, none of which can be as efficiently implemented using LUT resource. Again, this will all be explained in the user guide for your target FPGA.
Back in the day, your code would have been implemented as a single 74150 TTL circuit, which is a 16-to-1 mux. you have a 4-bit select (INPUT), and this selects one of 16 inputs to the chip, which is routed to a single output ('EW_NS`). The 74150 is obsolete and I can't find any datasheets, but it's easy to find diagrams of what an 8-to-1 mux looks like (here, for example). The 16->1 is identical, but everything is wider. My old TI databook shows basically exactly the diagram at this link doubled up.
But - wait. Your problem is easier, because you're not routing real inputs to the output - you're just setting fixed data values. On the '150, you do this by wiring 5 of the 16 inputs to 1, and the remaining 11 to 0. This makes the logic much easier.
The 74150 has basiscally exactly the same functionality as a 4-input look-up table (where the fixed look-up data is the same as fixed levels at the '150 inputs), so it's trivial to implement your entire circuit in a single LUT in an FPGA, as per scary_jeff's answer, rather than using a NAND-level implementation. In a proper chip, though, it would be implemented as a sum-of-products, or something similar (exactly what's in the linked diagram). In this case, draw a K-map and find a minimum solution. My 2 minutes on the back of an envelope comes up with three 3-input AND gates, driving a 3-input OR gate. I'll leave it as an exercise to you to check this :)
I'm working on a circuit which performs basic operations such as addition and subtraction using logic gates.
Right now, it takes 3 inputs, two 4 bit numbers, and a 3 bit opcode which indicates what operation to perform.
It seems that a 3-8 decoder would be a good idea here. This is my mockup!
To give a little more context, here is what my adder circuit looks like (+). I designed it to take two 4 bit numbers X & Y:
However, what I am confused about is the fact that I have to feed in 4 inputs or 4 wires to each of the circuit that handles it's respective operations (+, -, =, etc). It appears to only connect one wire to the circuit I need to get to. I need to actually connect 8 wires, as I have to feed in the to 4 bit numbers.
UPDATE: I ended up using a MUX to select the output that I want.
An adder doesn't need an input to tell it to add, because that's all it does.
A 4-bit full adder should have
4 input signals for each operand, total 8
A carry-in input signal if you are also using it for subtraction
5 output signals, the high-order one may be used to generate an overflow flag
Your decoder is a separate component from all the function generators. You could put a tristate buffer on each function generator to connect them to a common data bus, and then the decoder would generate the tristate enable signals. Otherwise, you probably don't need a decoder, but you might look at a multiplexer (mux) instead.