Whenever I input a = 1 and b = 1 I still get 0 and my inner pin of aAndNotb shows 1, however if I delete the Not gate I get a normally functioning Nand gate, what's the deal?
/**
* And gate:
* out = 1 if (a == 1 and b == 1)
* 0 otherwise
*/
CHIP And {
IN a, b;
OUT out;
PARTS:
// Put your code here:
Nand(a=a, b=b, out=aAndNotb);
Not(in=aAndNotb, out=out);
}
Given the symptoms you describe, the problem appears to be in your Not gate. Perhaps that component is not correctly implemented?
You can test this by implementing the Not gate directly using a Nand gate, which if you've implemented a Not gate, you already know how to do.
If that works, then the issue is the Not gate implementation. If it doesn't, then the issue is the wiring of the gates for some reason (but they look OK to me).
Related
I am trying to implement a system so that it retrieves sound and extracts the mfcc of it. I'd like to implement my own mfcc function because librosa library wasn't implemented in C and other implementations of mfcc extractions doesn't yield the same outputs as librosa library does.
So I wrote a code, however, when I would like create hanning window, program doesn't take a step further and always stays the same statement while debugging. The statement is below:
float *mul = malloc(sizeof(float)*fftsize);
The whole code is as follows:
float* hanning(int fftsize){
float *mul = malloc(sizeof(float)*fftsize);
for (int i = 0; i<fftsize; i++){
mul[i] = 0.5 * (1 - cos(2*PI*i/(fftsize-1)));
}
return mul;
}
I put an LCD code to all error handler functions in stm32f7xx_it.c file to determine which fault I'm facing, and I see that it is hard_fault.
So what's the problem? I hope the issue is explained clearly. Due to the privacy, I couldn't put here whole code. Sorry for that. Thx in advance for your response.
Edit: I am chaning malloc to normal array with a variable length array. But still it takes me to HardFault_Handler function. SCB->SHCSR returns sometimes 65535 and sometimes 1.
I am trying to make a 10-bit adder/subtractor. Right now, the logic works as intended. However, I am trying to set all bits to 0 iff there is overflow. To do this, I need to pass the output (tempOut) through a 10-bit Mux, but in doing so, am getting an error.
Here is the chip:
/**
* Adds or Subtracts two 10-bit values.
* Both inputs a and b are in SIGNED 2s complement format
* when sub == 0, the chip performs add i.e. out=a+b
* when sub == 1, the chip performs subtract i.e. out=a-b
* carry reflects the overflow calculated for 10-bit add/subtract in 2s complement
*/
CHIP AddSub10 {
IN a[10], b[10], sub;
OUT out[10],carry;
PARTS:
// If sub == 1, subtraction, else addition
// First RCA4
Not4(in=b[0..3], out=notB03);
Mux4(a=b[0..3], b=notB03, sel=sub, out=MuxOneOut);
RCA4(a=a[0..3], b=MuxOneOut, cin=sub, sum=tempOut[0..3], cout=cout03);
// Second RCA4
Not4(in=b[4..7], out=notB47);
Mux4(a=b[4..7], b=notB47, sel=sub, out=MuxTwoOut);
RCA4(a=a[4..7], b=MuxTwoOut, cin=cout03, sum=tempOut[4..7], cout=cout47);
// Third RCA4
Not4(in[0..1]=b[8..9], out=notB89);
Mux4(a[0..1]=b[8..9], b=notB89, sel=sub, out=MuxThreeOut);
RCA4(a[0..1]=a[8..9], b=MuxThreeOut, cin=cout47, sum[0..1]=tempOut[8..9], sum[0]=tempA, sum[1]=tempB, sum[2]=carry);
// FIXME, intended to solve overflow/underflow
Xor(a=tempA, b=tempB, out=overflow);
Mux10(a=tempOut, b=false, sel=overflow, out=out);
}
Instead of x[a..b]=tempOut[c..d] you need to use the form x[a..b]=tempVariableAtoB (creating a new internal bus) and combine these buses in your Mux10:
Mux10(a[0..3]=temp0to3, a[4..7]=temp4to7, ... );
Without knowing what line the compiler is complaining about, it is difficult to diagnose the problem. However, my best guess is that you can't use an arbitrary internal bus like tempOut because the compiler doesn't know how big it is when it first runs into it.
The compiler knows the size of the IN and OUT elements, and it knows the size of the inputs and outputs of a component. But it can't tell how big tempOut would be without parsing everything, and that's probably outside the scope of the compiler design.
I would suggest you refactor so that each RCA4 has a discrete output bus (ie: sum1, sum2, sum3). You can then use them and their individual bits as needed in the Xor and Mux10.
I'm tryng to build this chip:
// This file is part of www.nand2tetris.org
// and the book "The Elements of Computing Systems"
// by Nisan and Schocken, MIT Press.
// File name: projects/01/Mux4Way16.hdl
/**
* 4-way 16-bit multiplexor:
* out = a if sel == 00
* b if sel == 01
* c if sel == 10
* d if sel == 11
*/
CHIP Mux4Way16 {
IN a[16], b[16], c[16], d[16], sel[2];
OUT out[16];
PARTS:
// Put your code here:
And that what i wrote ontil now -
PARTS:
// Put your code here:
Xor(a=sel[0], b=sel[1], out=finalSel);
Not(in=finalSel, out=notFinalSel);
Mux16(a=a, b=b, sel=finalSel, out=aAndB);
Mux16(a=c, b=d, sel=notFinalSel, out=cAndd);
Mux16(a=aAndB, b=cAndd, sel=sel[0], out=out);
}
And for some reason it's not working..
Screenshot from the HardwareSimulator
someone know why?
Your logic for selecting what input to pass through appears to be incorrect. You should test it by creating a truth table for finalSel, notFinalSel, aAndB, cAndd and out for each of the 4 control conditions.
In general, when doing these kinds of problems, the KISS principle holds; Keep It Simple and Stupid. You don't need any fancy logical manipulation of your sel[] bits, you can just use them directly. So once you get your version fixed (and understand where you went wrong), try doing a version that just consists of 3 Mux16's and nothing else. Once you have both versions working, you'll then understand the error that caused you to go down the wrong path in your first attempt, and that will be a valuable lesson going forward.
Have fun!
Suppose our program has just started after Reset, so all FIODIRs are 0 and corresponding pins are inputs, and therefore they are in the third state (right?).
And suppose I want instaneously switch some pin to output with 1. Then probably I'll write this:
FIOxDIR_bit.Px_xx = 1;
FI0xSET_bit.Px_xx = 1;
But what will be state of the pin between these two commands? Presumably output with 0.
Another option to perform such switching is:
FI0xSET_bit.Px_xx = 1;
FIOxDIR_bit.Px_xx = 1;
But the datascheet claims:
If any pin is configured as an input or a secondary function, writing 1 to the corresponding bit in the IOSET has no effect.
Probably the same is true about FIOSET as well.
And that's exactly what we do in our latter code snippet. Though in my expierence this approach works well, but I'd like to have some theoretical basis for this.
I've been doing research on trying to understand the way sounds and sine waves work, particularly with chords. So far, my understanding is as follows:
b(t) = sin(Api(t)) is the base note of the chord at frequency A.
T(t) = sin(5/4piA(t)) is the major third of the base b(t).
D(t) = sin(3/2piA(t)) is the dominant (fifth) of the base b(t).
A(t) = sin(2Api(t)) is the octave.
Each one alone is a separate frequency which is easy for a computer generator to sound. However, the major chord of the note with frequency A is as follows:
Major Chord = b+T+D+A
I was wondering if anyone has a way to make a computer synthesizer play this function so I can hear the result; most programs I have found only take Hz as an input, an while this function has a wavelength, it's different from the simple sine wave with the same wavelength.
Note: will post this in the physics and computer sections as well - just wondering if you musicians know something about this.
It's a bit unclear to me what you're trying to do, so I'm explaining a few things, and giving you a few options to investigate further, depending on what your purpose is.
The de facto means of synthesizing music on computers uses MIDI (Musical Instrument Digital Interface). Because musicians rarely think directly in terms of frequencies or wavelengths (they count steps or half-steps along a scale), MIDI represents each pitch with an integer, that represents a number of half-steps. By default, MIDI assumes these pitches are tuned using a standard called "12-Tone Equal Temperament" (12TET), and, unfortunately, it isn't particularly easy to make it use other tunings.
What does this mean? And why is it a problem? (I'm not sure how much of this you know already, so I apologize if I'm repeating what you know)
In theory what you say about tuning being based on frequency ratios is 100% absolutely correct -- this is a system called Just Tuning. The major third is a 5/4 ratio and the perfect fifth is a 3/2 ratio. However instruments with fixed-pitches (keyboards, fretted instruments, etc...) are rarely tuned that way in practice. Instead, they compromise, by rounding each note in a chromatic scale to the nearest 12th of an octave. Since an adding octave is equivalent to multiplying the initial frequency by 2, adding a 12th of an octave is the equivalent of multiplying the initial frequency by 2^(1/12). This is how all the half steps on a piano are usually tuned.
So instead of the pure ratios, you would actually have:
sin(A pi t)
sin(2^(4/12) A pi t)
sin(2^(7/12) A pi t)
sin(2^(12/12) A pi t)
Note: Compare 2^(4/12) ~ 1.26, with 5/4 = 1.25. Also compare 2^(7/12) ~ 1.498, with 3/2 = 1.5.
These are exactly the frequencies that any MIDI synthesizer will play, given the MIDI notes numbered n, n+4, n+7, and n+12. So, if you are only looking to play a chord, and don't care about the frequency ratios being pure (just), you can just use MIDI.
However, if you are looking for something that will play the actual just ratios, it will be a bit trickier. You might start with looking at some of the things here: https://music.stackexchange.com/questions/351/software-that-allows-playing-in-different-temperaments
If you just want to see what they sound like, you can check out this youtube video:
https://www.youtube.com/watch?v=6NlI4No3s0M
If you can write software, you might try writing your own, but I don't want to go into how to do that if that's not going to be helpful.
I'm not sure what kinds of programs you're describing that "only takes Hz as input". Is this a software library (like an API call?) or something different? There are (obviously) API calls that can send more complex data to the soundcard than a single-frequency wave.
EDIT: I've not used it, but it looks like perhaps this software is capable of doing what you want: https://www.youtube.com/watch?v=z0NZQMiDdNU
I think you are going at the problem from a wrong direction. You are using sinoid signals as the basis of your "chords" and pure intervals.
The output of that is strictly periodic, with a period that is the least common multiple of the individual periods. So basically you have not as much a "chord" but rather a "tone".
Organs use that technique: you can combine an 8" pipe with a 5⅓" pipe in order to arrive at a tone sounding like it came from some funny 16" pipe. That's not a "chord", that's registration. Classical composition theory does not allow quint parallels to avoid that effect: quints must only occur transitorily and more often than not moving in a different direction or out of synch with the base note.
"chords" play with aural ambiguities between tone colors and voices: they excite the same region in your inner ear. However, real "chords" and choral effects also have beatings and interferences from non-ideal frequency ratios, and the tones they are made of have harmonics of their own, making it possible to discern them as independent entities.
The experienced music listener perceives all that as an independent phenomenon. However, if you start with pure sinoids or highly frequency-locked comparatively pure sources like snare-less organ pipes, this becomes hard to do.
So I'm not sure you are doing yourself a favor by looking at sinoids. It's like trying to understand a painting based on primary color components. It's more of a reproduction toolkit than anything tied to higher-level perception.
A very low-barrier way to play is to use wavepot
The code to do what you ask in your question is
var A = 440
export function dsp(t) {
var b = Math.sin(t * Math.PI * 2 * A);
var T = Math.sin(t * Math.PI * 2 * 5/4 * A);
var D = Math.sin(t * Math.PI * 2 * 3/2 * A);
var O = Math.sin(t * Math.PI * 2 * 2 * A);
return (b + T + D+ O) * 1/4
}
which is at this link.
Note that this might not sound much like a chord to you due to the fact that sine waves have no harmonics. Here is the same example, but using a saw-tooth waveform, which has many harmonics.
var A = 440
//function f(x){return Math.sin(x) }
function f(x){return 2 * ( (x/(2*Math.PI) % 1) -.5) } //sawtooth
export function dsp(t) {
var b = f(t * Math.PI * 2 * A);
var T = f(t * Math.PI * 2 * 5/4 * A);
var D = f(t * Math.PI * 2 * 3/2 * A);
var O = f(t * Math.PI * 2 * 2 * A);
return (b + T + D+ O) * 1/4
}