Final Post: Matlab Simulation Results

Hey guys, here is the final post for this semester project.

As I mentioned in my last post, I needed to do some Matlab simulations to wrap this thing up. It took a long time and a lot of help from my professor and a kind graduate student at school but I got some pretty cool results!

I had already decided to utilize the Matlab function "butter" and "filter" to accomplish the task at hand, that being to vary the cutoff frequency of the filter as the signal is passed through the circuit.

Here's how these two functions filter a signal. "butter" receives a cutoff frequency and generates two filter coefficients, named a and b. "filter" takes these coefficients and applies them to a signal. The problem is that "butter" only generates two coefficients for one specific cutoff frequency. No good if I want a changing cutoff frequency. Well I was thinking I would need to implement a loop of some kind to generate a whole heap of coefficients. This is exactly what my professor advise! Yay for having some intuition for this stuff...

So once I implement a loop to generate (and store in a vector, by the way) a whole heap of coefficients, I need to apply all of these to "filter." I needed to know how filter worked. Well Matlab's handy dandy "help" function proved useful as it explained that the function, "filter," is nothing more than a difference equation. That equation being (for a 1st order butterworth filter):

     a(1)*y(n) = b(1)*x(n) + b(2)*x(n-1) - a(2)*y(n-1)

I have no practical understanding of how or why this equation filters a signal but it's what Matlab said and I was going to roll with it. Haha!

So all that was left to do is create another loop that would apply all of those coefficients from "butter" and playback the resulting sound.

So I am not very familiar with Matlab. I've had a few projects at school but I haven't spent extensive time in it. I knew I needed loops but I wasn't sure how to write them! Thankfully a kind friend at school offered to help. Together, we were able to write code that receives an input wave file, applies a filter with a linearly increasing cutoff frequency, and play back the sound. I flipped out when it worked. Who knew you could do this kind of stuff with Matlab? I didn't till now.

Once the ball was rolling I wanted to do some cool stuff like vary the cutoff frequency sinusoidally. Turns out it was not that hard to do! At first, all I did was stick a sine function in front of the cutoff frequency variable inside the loop. However, sine would eventually generate negative frequencies which "butter" did not like. After a minute of thinking I figured it would be easiest to simply get the absolute value of the sine so those negatives would go away. Bam! That worked just fine.

So, my final code varied the cutoff frequency three different ways. My code also generated plots of those variations and created wave files to playback. Below are those results and the code.

Here is the original guitar clip I recorded direct into my Digital Audio Workstation (Reason 6.0!) and exported as a wave file: https://soundcloud.com/visorakmark/ampharos-original-unfiltered

First, I simply "opened up" the filter as time passed.
Here is the resulting sound: https://soundcloud.com/visorakmark/ampharos-linear-increase




Then I applied a sinusoid to the cutoff frequency.
Here is the resulting sound: https://soundcloud.com/visorakmark/ampharos-sinusoidal-variation



Finally, I applied an increasing sinusoid to the cutoff frequency.
Here is the resulting sound: https://soundcloud.com/visorakmark/ampharos-increasing-sinusoid



And here's the code. Note if you try to run this it will take several minutes to execute.

------------------------------------------------------------


% Ampharos_LPF
% This program applies an "opening" low pass filter to a wave file,
% meaning the cutoff frequency of the filter is changed as the filter
% is applied to the signal. This results in a "fading in" of the sound.
% Note that this is not the same as overall signal gain control, but a
% filtering of higher frequencies. As the wave file is played, more higher
% frequencies are allow to pass through and the sound "opens up." The
% cutoff Frequency is also vaired sinusoidaly and varied with an increasing
% sinusoid.
%
% These processes are accomplished through a "brute force" implementation of
% the Matlab function called "filter." "filter" applies two variables to a
% signal. These variables are generated using the Matlab function
% "butterworth." "butterworth" recieves a cutoff frequency and generates two
% variables that can be applied to a signal to filter specific
% frequencies.

% This program requires the cutoff frequency of the filter to
% change with time. "butter" and "filter" cannot accomplish this task
% alone. Considering 'butter" generates two filter variables for a
% specific frequency, a loop is used to generate and store multiple filter
% frequencies. A loop is then used to apply the generated cutoff
% frequencies to the signal. Finally, the signal is played back.

%This process is executed three seperate times, each time applying a
%different type of cutoff frequency variation.


% The wav file is read into Matlab
[x fs]=wavread('MarkGuitar.wav');

%play original signal
sound(x,fs)

%The butterworth loop variables are set up
count=0;
minFc=1;
inc=1;
maxFc=(1+minFc+length(x).*inc);

    %This section of code applies a linearly increasing filter to the signal
    for Fc=minFc:inc:maxFc

        count=count+1

        w = Fc/(2*fs); % Normalized frequency
        [b,a]=butter(1,w,'low'); %1st order butterworth LPF
        PlotFreq(count)=Fc;

        % This loop has been generalized to take any order butterworth filter,
        % hence the length(a) and length(b). coefA and coefB also store the
        % recieve Fc and w for reference.
        coefA(count,1:length(a))=a;
        coefA(count,length(a)+1)=Fc;
        coefA(count,length(a)+2)=w;

        coefB(count,1:length(b))=b;
        coefB(count,length(b)+1)=Fc;
        coefB(count,length(b)+2)=w;

    end
 
% Access filter variables
Avalues=coefA(:,1:2);
Bvalues=coefB(:,1:2);

% Apply filter variables to signal x.
Y=getOut(Avalues,Bvalues,x,fs);

% Plot Fc vs count to vizualize how the cutoff frequency was varied during
% variable generation
figure(1)
plot(PlotFreq)
title('Linearly Increasing Cutoff Frequency Variation');
xlabel('time');
ylabel('cutoff frequency');

%Play filtered signal
sound(Y,fs)






%Reset variables for second filtering implementation
count=0;
minFc=1;
inc=1;
maxFc=(1+minFc+length(x).*inc);

    %For constant sinusoidal filtering%
    for i=minFc:inc:maxFc

        count=count+1

        Fc = 20000;
        Fc = abs(sin(0.0005*i))*Fc;
        PlotFreq(i)=Fc;
        w = Fc/(2*fs); % Normalized frequency  w = (sin(count)*w
        [b,a]=butter(1,w,'low'); %1st order butterworth LPF

        % This loop has been generalized to take any order butterworth filter,
        % hence the length(a) and length(b). coefA and coefB also store the
        % recieve Fc and w for reference.
        coefA(count,1:length(a))=a;
        coefA(count,length(a)+1)=Fc;
        coefA(count,length(a)+2)=w;

        coefB(count,1:length(b))=b;
        coefB(count,length(b)+1)=Fc;
        coefB(count,length(b)+2)=w;

    end

% Access filter variables
Avalues=coefA(:,1:2);
Bvalues=coefB(:,1:2);

% Apply filter variables to signal x.
Y=getOut(Avalues,Bvalues,x,fs);

% Plot Fc vs time to vizualize how the cutoff frequency was varied during
% variable generation
figure(2)
plot(PlotFreq)
title('Sinusoidal Cutoff Frequency Variation');
xlabel('time');
ylabel('cutoff frequency');

%Play filtered signal
sound(Y,fs)






%Reset variables for second filtering implementation
count=0;
minFc=1;
inc=1;
maxFc=(1+minFc+length(x).*inc);

    %This section of code applies an increasing sinusoidal filtering to the
    %signal
    for Fc=minFc:inc:maxFc

        count=count+1
        Fc = abs(sin(0.0005*count))*Fc;
        PlotFreq(count)=Fc;
        w = Fc/(2*fs); % Normalized frequency  w = (sin(count)*w
        [b,a]=butter(1,w,'low'); %1st order butterworth LPF

        % This loop has been generalized to take any order butterworth filter,
        % hence the length(a) and length(b). coefA and coefB also store the
        % recieve Fc and w for reference.
        coefA(count,1:length(a))=a;
        coefA(count,length(a)+1)=Fc;
        coefA(count,length(a)+2)=w;

        coefB(count,1:length(b))=b;
        coefB(count,length(b)+1)=Fc;
        coefB(count,length(b)+2)=w;

    end

% Access filter variables
Avalues=coefA(:,1:2);
Bvalues=coefB(:,1:2);

% Apply filter variables to signal x.
Y=getOut(Avalues,Bvalues,x,fs);

% Plot Fc vs time to vizualize how the cutoff frequency was varied during
% variable generation
figure(3)
plot(PlotFreq)
title('Increasing Sinusoidal Cutoff Frequency Variation');
xlabel('time');
ylabel('cutoff frequency');

%Play filtered signal
sound(Y,fs)

------------------------------------------------------------

And here are those functions I called above:

------------------------------------------------------------
function y=FilterApply(a,b,x,y,sigInd,filtInd)

term1(sigInd)=(b(filtInd,1).*x(sigInd))./a(filtInd,1);
term2(sigInd)=(b(filtInd,2).*x(sigInd-1))./a(filtInd,1);
term3(sigInd)=a(filtInd,2).*y(sigInd-1)./a(filtInd,1);

y(sigInd)= term1(sigInd)+term2(sigInd)-term3(sigInd);

------------------------------------------------------------
function Y=getOut(a,b,X,fs)

% This is essentially the Matlab function "filter" but now able to
% recieved multiple filter variables and apply them. Note this function
% utilizes the mathematical definition for the filter function to apply
% the variables to the singal.

% Given
%     signal X (length n)
%     coef matrix a & b
%     fs = 44100
%     signal y previous
%     signal y at n==1

Y=zeros(1,length(X));
filtInd=2;
for sigInd=2:1:length(X)
    out=FilterApply(a,b,X,Y,sigInd,filtInd);
    Y(sigInd)=out(sigInd);
    filtInd=filtInd+1;
end

------------------------------------------------------------

And that's a wrap! Been a great semester of learning. Hope you all enjoyed following along. Perhaps I will continue this project down the road. Who knows. For now I'm going on break and will see you later.

-Mark

Some Matlab Coding

Hey again readers, spent a couple of hours researching filter simulation in Matlab and got a few things figured out.

Right now my code uploads a wav, applies a 1st order butterworth low pass filter to it, and plays the filtered wav file back. That all is working fine. The challenge I am facing is somehow changing the cutoff frequency while the program is running so the sound "opens up and closes" during playback.

Here's how I'm approaching this right now:

------------

This function samples (at the sample rate, fs) the wav file and stores those samples in the matrix x. x is a 2 by 37000ish matrix that contains a discrete representation of the wav file. I'm currently using a short ringtone for code testing as opposed to a full guitar track.

[x fs]=wavread('Bangarang_Phone.wav');

I then set my cutoff frequency whic for the filter U currently have is 900 hz.

Fc=900;

Then I use a butterworth function to implement a 1st order low pass filter.

w = Fc/(2*fs); % Normalized frequency
[b,a]=butter(1,w,'low'); %1st order butterworth LPF
y=filter(b,a,x); % pass the input signal through the filter

Then play the filtered sound

sound(y,fs)

------------

That's what I have so far. I'm thinking I need to somehow implement a loop and change Fc as the filter is being applied. I just don't know how to do this though. That's for another hour!

More soon,

-Mark

End of the Semester Matlab Simulation

Hey readers, finals are upon us and time is running out for this semester project. Obviously this thing is not ready to be placed in my amp so I will be writing some Matlab code and simulating what I've got so far.

First. I will simulate how I want the circuit to work. I'm going to record a short guitar track, import it into Matlab, and simulate a low pass filter. I'll vary the resistance to get the sweeping wah-wah effect I'v been looking for.

Next I'm going to simulate how the circuit is currently (not) working. Not entirely sure how to do this yet. We'll see what I can cook up over the week.

Expect one more post from me before the 16th of December!
-Mark

Road Un-Blocked, Prototype 2 Simulation and Implementation

Hey folks,

Decided to tank the current component values in my design and just try a basic Low-Pass Filter with component values I used more frequently in Circuit II last Spring. Used 470k ohm resistors and 390 nanofarad capacitor. Not surprisingly the circuit worked! Tested it in the lab and in PSPICE. Cutoff Frequency around 900 hz. Unlike the schematic I provided, the voltage source was set to 0.

Prototype 2. V7 is varied to change overall feedback resistance 

Transient Analysis of Prototype 2. Input (Red) of 870 hz. Output (Green). Output is approx 70% of input at calculated cutoff frequency. Behaves as expected.

Frequency response of Prototype 2. Corner Frequency around 900 hz.

Top: Input of 870 hz. Bottom: Output - As you can see the output is attenuated approximately 70% of the input.  Exactly what is expected by the design.

So! I got a simple active LPF to work. Good news.

As I mentioned earlier, the V7 voltage source is what "sets" the "resistance" of the FET. Applying a voltage opens the pathway between the source and drain. Here's the problem. I've learned that for some reason using small resistances for an active LPF does not work. It just doesn't behave like a filter. Not sure why... The current FET I'm using (TIS74) can only vary between 30 to 800 ohms.  Adding 800 ohms to 470000 is not going to make much of a significant change in cutoff frequency. I doubt I would even see noticeable change if I implemented this.

I already tried simulation with the FET resistance on. Quite unsuccessful. Circuit does some weird stuff.  For voltages between 0 and 2.8, I get the output seen above. However, Once V7 is around 3 volts the entire output is shifted down by about 0.4 volts.  I don't know what model transistor PSPICE uses... But I'm thinking it's acting like a switch (like FET's are usually used for). Once a certain point is reached it completely opens/closes the drain/source path. What I'm looking for is a gradual shift in resistance. Since I don't have a TIS74 PSPICE model (and I'm not equipped to program one!) simulation might be futile for this design.

Here's the output with V7 set to 3 volts:

FET biasing voltage set to 3 volts. Basically drops the output by some constant. No good.

I've got a huge test in Power Distribution Systems on Tuesday so I've got to go study for that. We'll come back to this later. I'll see what happens when I add the FET in the actual circuit.

-Mark

Roadblock

Hey folks,

I have spent the last two hours trying to test an active low pass filter. The idea was similar to the passive LPF design: Test a filter without the FET to ensure proper operation and then add an FET. Well I've used four different oscilloscopes, tested multiple op amps, changed my resistors and capacitors, and I still haven't gotten a simple filter to work. WHAT THE HECK? I've got a major test tomorrow and multiple homework assignments to finish tonight so I'll have to come back to this and try again later. Sorry there is nothing exciting to report. I'll simulate the circuit before I come back in for a practical test, hopefully something will have changed. I've checked my circuit tons of times and everything is as it should be. It's possibly just faulty equipment... who knows...

-Mark

Prototype 2 Construction and Testing

Hey readers,

I was hoping to update you guys on my latest prototype last week but other assignments took priority. I did construct and test the circuit you see below but I could not get the circuit to behave as designed.

Prototype 2

Input and Output for Prototype 2

I haven't a clue to exact reason for this strange result but I went ahead and ordered a couple of new components to use (LM471's to be exact) in case something was faulty. Thankfully they arrived swiftly and I plan to retry this week and see what happens.

Expect another post soon!
-Mark

Active VCF Design (Part 1)

So I have not fallen off the face of the earth. Been getting in the groove of the semester and haven’t set aside time to write till now. I have some updates to share!

My professor and I agreed to make this Arduino-Filter circuit an active filter. An active filter differs from a passive in that it allows gain. Which basically means the output signal can be bigger than the input signal. This is useful for me in that I don’t want the guitar signal to lose too much voltage going through the VCR stage, therefore I can include a “recovery” stage in my design. Cool!

This means my design needs to include some kind of operational amplifier. The LM471 is an op-amp I am familiar with and what I plan to use in prototyping and possibly final implementation.

So, let's dive into active filters! Below is the schematic of a typical Active Low-Pass Filter. (Courtesy of Wikipedia)



Just as my passive filter design over the summer, I need to integrate an FET into this circuit to make it voltage-sensitive. Now any Average-Joe-Engineering-Student knows from AC Circuit Analysis that the cut-off frequency of this circuit is controlled by the feedback impedance (C and R2) while the gain is set by the ration of R2 to R1. Therefore, the FET will need sit in parallel or series with R2 to vary the feedback resistance.

I failed to understand that last point in my first design. The circuit below isn’t a VCF at all!



In this design, the FET simply acted as a voltage-controlled voltage divider followed by a gain stage. Possibly useful but not what I needed.

I haven't drawn it up in PSPICE yet, but my next design corrected this and has the FET in the feedback section of the circuit. Much better! I'll include some schematics of this better design in my next post.

And we have arrived at my current position! Next step is to simulate, analyze, and implement the circuit  I have to see what happens.

As you saw in the August 27th post, the amplifier is complete! It is now just waiting for the VCF to be installed. If I have time, I would like to include a switch in the amp to select between analog EQ controls and my new VCF, that way I don’t lose the good old-school EQ controls forever with this modification.

I’ll be back with a report on that circuit soon. For now, bye friends!

An Announcement, Amp Update, and Voltage Controlled Filter (VCF) Prototype 1

Hey readers, I have a quick announcement before getting to the core of the post. This project will be continuing through the Fall semester! Quite excited. I did not get where I wanted in the summer session and my professor offered me the opportunity to continue this as a Project II class. Great news! More fun ahead.

Well, since the end of the summer session I've been busy finalizing my guitar amp build, just finished last week. My professor and I agreed that building the amp unmodified first would be a good idea, that way I am actually see what kind of voltages I will be dealing with as I design the VCF.

Here's it the completed amp!

It sounds great. I was blown away it worked when I first switched it on. I've heard horror stories of people spending a week tracking down incorrect wiring or bad components on their first builds. Phew!

Now, I do have a prototype VCF, it is not at all designed to be placed inside the amp, it was more of a proof of concept. And it does work as a VCF under certain conditions. One of those conditions being an input signal with a magnitude of around 200mV-500mV, any greater or lesser voltage causes the transistor to enter saturation and the circuit no longer functions as a VCF. It also has a rather small frequency range/sweep.

Have a look:

I don't have a schematic for this yet. I'll scan in my paper draft when I get the chance. I plan to modify this design within the week. I want to see if I can increase the filter frequency range by placing another FET in series with the first. The FET's I am using have a rather small ohm range, from around 70 ohms to 800 ohms. I will be ordering some different FET's to experiment with this week as well. I need to start some PSPICE simulations soon. We'll see. I have not worked with FET's in PSPICE before.

Well that's all for now,
Mark

Voltage-Controlled Filter Design Theory

Heya folks. I'm currently in the lab at school testing out a simple voltage-controlled filter. I'll show you that in another post soon. But first, I want to share my thought process thus far. I sat down the other night and began writing everything I knew about designing a voltage-controlled filter so I knew what blanks needed to be filled. That writing is below.

Simple RC Filter:

This circuit attenuates particular frequencies (in this case, low frequencies) depending on the value of the components. With a variable capacitor or resistor, the specific frequencies can be attenuated or swept through. A simple mechanical device, called a potentiometer is a variable resistor that can be used in this way. The tone knobs on the front of any typical guitar amp utilize this phenomena.

If the Thevenin resistance and total capacitance of the filter are known, then the corner/cutoff frequency of the filter can be mathematically calculated with the following equation:

F_c = 1/(2*pi*R*C)

For low and high pass filters, the cutoff frequency is the frequency at which the output magnitude equals 0.707 times the input magnitude. Or in layman's turn, the frequency where the filter starts working.

So, with this filter theory behind us, let's talk about designing a voltage-controlled filter. We need some device to replace a potentiometer. So it needs to be a three-pronged device where one input controls the flow of current between two of the other inputs. Well this sounds eerily like our good friend the Field-Effect Transistor!

An FET is an electrical device made of semi-conducting material (commonly silicon). The voltage between the gate and source (V_gs) controls the current flow through the source and drain (I_ds), well not entirely. An FET can operate in two different regions. In the saturation region, the drain current is nearly independent of the drain/source voltage. In the ohmic region, the drain current does depend on the drain/source voltage.

My understanding as of now is that small changes in V_gs result in different resistances between the drain and source, however, you've got to be careful to stay operating in the ohmic region, that depends on what V_ds is. Man this is confusing. The circuit I'm testing now seems to be working this way. I just arbitrarily chose some resistors and a capacitor. I need to sit down and derive a circuit with specific characteristics, then test it in the lab. I was just getting familiar with the topology in the lab today.

Here are the documents I have been studying today. Most of what I've discussed has come from them.

http://hyperphysics.phy-astr.gsu.edu/hbase/electric/filcap.html (Basic high pass filter theory)

http://users.ece.gatech.edu/~lanterma/sdiy/datasheets/transistors/vishay_fet_cvr_an.pdf (The circuit I'm testing today is a modification of Figure 7 in this publication)

http://graffiti.virgin.net/ljmayes.mal/comp/vcr.htm (This discusses making an FET operate linearly in the ohmic region)

Well, I've got o quit for today. More again Monday.
-Mark

Back to the Grind

I drilled the chassis with the benevolent guidance of a great friend at Ardent Studios in Memphis. Thanks Chris! Got to run a drill press for the first time. It was a great learning experience, only botched one of the output holes. Chris was able to fix it though. Phew! Here are a few pics of the process.

Drill plan taped to chassis

Running the press

All done!

Next step is to paint the chassis. That will probably happen tomorrow afternoon. Hopefully I can get it built by Tuesday to start testing out.

-Mark

Ax84 Kit Arrival and Resources on Arduino-Controlled Filter

A couple of updates on the project!

The Amp:
When I initially decided on the AX84, I wanted to save little cash by collecting all of the parts separately. After about a week of reading, searching, and phone calls with a couple of professionals, I decided to buy the kit for convenience. Much thanks to Robert Hull of TubeDepot.com for spending half an hour on the phone with me discussing this project. He advised I buy the kit for my first build considering I may not know all of the factors to buy the exact parts the design demands.

The kit arrived today and I couldn't be more excited! Have a look:

I am working on preparing the drill plan for the chassis today. Hopefully, I can get the holes drilled within the next couple of days. I need to get the amp operational by the end of the week. Muse's song "Our Time is Running Out" keeps coming to mind...

The Mod:
Well I've been scratching my head over this one all summer. I'm no electronics designer so getting this Arduino-controlled filter idea off the ground is coming slow. About a week ago I posted a thread on AllAboutCircuits.com to get some advice on implementing this idea. The users their offered some great advice and helped me eliminate a few ideas I had. Here's the link to that thread. http://forum.allaboutcircuits.com/showthread.php?t=87333

My professor emailed me this website as a starting point for this design: http://graffiti.virgin.net/ljmayes.mal/comp/vcr.htm

An FET can operate in one of two regions. Either the Ohmic region (pentode) or the saturation region (triode). While I understand the difference between the regions, I'm not sure why or how pentode/triode correlate with them. At the time of writing I came across another great resource:
http://users.ece.gatech.edu/~lanterma/sdiy/datasheets/transistors/vishay_fet_cvr_an.pdf
I might discus the contents of this publication in future post.

Well, I need to get those drill plans together. More than likely the frequency of posts will increase now that I have the kit.

Cheers!
-Mark

Amp Design Selected, Time to Assemble the Parts!

Well, after a solid month of reading, googling, and talking to people smarter than me, I have selected an amp design.

http://www.ax84.com/home.html

The ax84 is a common beginner's tube amp project and one I am excited to start on. I knew a few weeks ago I wanted a smaller wattage, single-ended, class A amp. These are the most simple designs and I NEED simple. Haha.

I'm tempted to buy the kit, but have decided to at least research the cost of assembling the parts separately. My professor has told me I could get some sheet metal from the machine shop on campus to build a chassis. The tubes and transformers can be bought from local electronics providers. I already own plenty of resistors and capacitors. I plan to put together a parts list tomorrow and determine what all I have.

I plan to build the amp as designed to ensure I can build an operational piece of equipment. But I need to integrate an arduino somehow. Tone control seems the simplest and most rewarding mod.

Here are a few thoughts I have going forward with this idea:

I need filters that are voltage controlled as the arduino outputs voltages. The current filters are controlled with variable resistors, I'll probably need to pull that entire stage out and replace with my arduino circuit. This may be very challenging as impedance matching and a host of other commonsense circuit design necessities become apparent. I imagine PSPICE simulation of this circuit will be helpful. SPICE and I have a rocky past, so this very well may be the most frustrating part of this project.

Well, here is the circuit I'll be soldering together within a couple of weeks!

Notice similarity this schematic has with my black box sketch from last week. Glad I learned something this summer!

I'm out for today,
-Mark

Black Box Model and Power Supply Simulation

So it's been close to a month since the beginning of this project. Part of me feels like I have gotten no where in the last three weeks. But I have to remember I have had to familiarize myself with amplifier models and jargon. I have learned about guitar amplifier classes, common guitar amp vacuum tubes, rectifiers, smoothing capacitors, theory of distortion, theory of filter design, and much more. My professor recommended I begin with power supply design, but as I began my research I realized I had to go through guitar amp boot camp before even begin that.

So! The purpose of this post is to discuss a black box model of what goes on inside a guitar amp. The following scan is a quick sketch I made of the essentials of an amp. By no means is it exhaustive, but I feel it is a good assessment of the basics.

I plan to build a Class A, single ended 10 watts (ish) amp. Probably will use a couple of 12Ax7's and an EL84, or at least cheaper copies of these vacuum tubes. I've read about Class AB Push-Pull designs, but feel it is a bit complex for my first build. An acquaintance and fellow U of M student also encouraged me to use a simple power supply design. No tubes, just a rectifier bridge and a few filter stages. This will save on money and time.

I decided to test a simple rectifying circuit/power supply in PSPICE. Took a while to get the output I wanted, but it seems to be doing what I need it to. I plan to try and implement this circuit sometime this week to see if I can get it to work in the lab.

I have also been studying the schematic for my own tube amp in my current guitar rig. The Epiphone Valve Junior seems to be the kind of amp I would like to build. Only a few stages and a power supply, nothing fancy. I still plan to integrate an arduino into my design, just not sure how yet.

That's all for now. More to come soon.
-Mark

Tube Selection and Power Supplies

I had my first meeting with my professor this Monday. It seems the best place to start in amplifier design is power supply and vacuum tube selection. Considering vacuum tubes have special needs (high voltages, max inputs before overheating, other stuff I don't know about yet, etc.), deciding which tubes I will use is essential in design.

But honestly, I've been wandering a bit. Even with a starting point, I still find many of the resources I have collected overwhelming. There is a lot of jargon in tube circuits and I'm still catching up.

I've read the first three chapters of the ancient "Vacuum Tubes" by Karl Spangenberg from 1948, which was helpful, but very theoretical. (http://www.tubebooks.org/Books/Spangenberg_vacuum_tubes.pdf) I'm attempting to find the happy medium between theory and practical application. As my professor said, engineers aren't necessarily good at getting stuff to work, we are more interested in the scientific method. Prediction, implementation, measurement, analysis. When do I apply this? After I've actually got something to predict!

Power supplies seem less mysterious than tubes. I've studied half wave and full wave rectifiers in EECE 3211 (Electronics 1) and transformers in EECE 3201 (Circuit Analysis). I get the basics of those and the power supplies I've been looking at make sense.

The basic function of a power supply seems to be the following:

120 V 60 Hz "wall" input signal ==> transformer to step down/up voltage ==> rectifier to change AC signal to DC signal with lots of ripple ==> some kind of filtering capacitor to smooth out ripple ==> DC signal to power circuit ==> Face-melting guitar solo

I guess that ripple reduction stuff my prof was talking about during filter design in EECE 3201 last semester was important... Not that I didn't pay attention or anything...

I'm thinking my best option is going to be choosing a common, simple tube amp design that can get me comfortable with the basics. Dave Funk's Tube Amp Workbook, which I've read most of this week, mentioned the Vox AC15 being the simplest of simple designs:

What do you guys think?

I'm thinking: Get Behind Me Satan! If this is simple I've got a LOOOOONG way to go.

About lunch time. I'll see you folk later!
-Mark

Project Outcomes

Clear outcomes are a necessity for any project. While I'm sure some of the following will be adapted over the summer, I feel this list defines the scope of this project.

1. Build a working guitar amplifier circuit with some or all of the following effects:

     Low pass filter

     Mid frequency notch filter

     High pass filter

     Distortion (these circuits seem fairly simple, an op amp and a couple of resistors right?)

2. Control sections of the guitar amplifier circuit with an arduino. (http://www.arduino.cc/) Below are some ideas I have for the arduino:

     Tremolo effect (program the arduino to vary the output magnitude of the circuit)
     Phasor effect (program the arduino to vary the filter stages)

These all are theoretically simple, the biggest hurdle now is to actually get a physical circuit to do these things.

3. Write a formal report on my research and outcomes. I just took Technical Writing, gotta keep my skills up!

It should be noted this project is not completely "practical." Given my limited electronics knowledge, I doubt this circuit is going to sound that great. The ultimate purpose of this is to understand amplifier design and modify what I can (i.e. using the arduino).

Cheers for now!
-Mark

Welcome

Heya folks, 

Welcome to The Ampharos Project. As described in the project info module, I plan to build a guitar amplifier by the end of summer. Let me share a little background on this project:

I study electrical engineering at the University of Memphis. I currently am a junior and will be taking my EECE (Electrical Engineering and Computer Engineering) electives within the next three semesters. EECE students can fulfill these credits through typical semester classes or research projects with professors. My advisor and I share a love of guitars so we decided building a guitar amplifier would be a great summer project. And here we are!

I named the project after one of my favorite childhood characters, the (fittingly) electric-type Pokemon, Ampharos. He's basically a bipedal-electric-Pharoh-lighthouse-sheep.

Epic.

I will be posting my thoughts, analysis, diagrams, pictures, and circuits to document my first major electronics project. I have little to no experience in electronics so I imagine the road will be bumpy. That being said, if you have input or suggestions, please comment and share your thoughts.

Thanks for reading!
-Mark

Picture credit goes to Ho-Roudon from Seribii.net:
http://www.serebiiforums.com/showthread.php?452046-My-Pok%E9mon-Drawings