Ever happened to you that you unplug the cable from the guitar? It must have! Either you stepped on the cable or you moved around too much. Here is a cheap and quick way how to take care of this problem.
Leave a loop of 10-20 cm between the clip and the jack. Adjust this length until you are comfortable and its not bothering you. When you are finished unplug the jack first then remove the clip. Keep cables in their own space and its best to hang them in long loops. Don’t wind them around your hand.
Do this every time. Don’t get lazy and leave the cable in just because you will play again tomorrow!
I’ve been studying the past couple of months some ways to improve the field coil performance and in a way to make my tools better, to sharpen them. My goal is to get the most flux density in gap for a given current in field coil.
I tried many magnetic circuits meant to take advantage of the shape of a field coil because being a solenoid it will work better having a length per diameter ratio above unity. Works better means a more uniform field in the core.
I managed to get a simulated result of 1.5T in a 1mm wide and 12mm high gap using standard 1010 steel and a field coil consuming 7.231 Watts!
The field coil is made with 1 mm EC wire so it will be able to handle a lot more current. You will get more flux density this way but the curve will get more and more bell-shaped.
As you can see for 1 Amp you get a very nice increase in flux density. Its not as linear but the difference is small.
There is a balance, hard to achieve, between linearity and maximizing flux density. And this has to do with the magnetic circuit and saturating the pole plates. I usually want to saturate them right at the gap. This way it’s less prone to modulation.
In placing the saturation area, the central pole piece and the top plate geometries have a large role. Always remember that the thin parts of steel saturate faster and zones with transitions from one dimension to another or sharp angles are also prone to saturation.
Below you can see a motor with a linear flux density curve over a 20 mm travel. The central pole piece is of the same diameter as before. You can see that making the top plate thicker killed the saturation around the gap. To bring it back, we must use a larger diameter for a voice coil. But more on that in a future article.
Do you remember BOSS Slow Gear pedal? If your a guitarist you most likely do or at least you’ve heard of it. It was a great pedal sold from 1979 to 1982 and it was made in Japan. The pedal would cut the attack of your notes giving a swelling sound. It god famous for making the guitar sound kinda like a violin.
I always liked that effect and i even made a clone a few years back. It is based on a 2SK30 JFET and it was a pain getting these transistors. It was a lot of fun though and i though i should make a Project Ryu swell effect pedal and so LAGGER was born!
Recently i worked on a few projects with LM13600/LM13700, one of them is a nice noise gate / compressor unit which i will present at a later date, and i really like the VCAs that can be built with these chips.
To cut the attack of a note and then swell the volume basically we need a triggered fade in effect. This means that we need to control our VCA with a rising voltage using what i call a ramp generator.
Below you can see the block diagram of the Lagger:
The input is fed into an ADC channel to be rectified and averaged in order to detect when a note is played. Once it is detected, the ramp generator is triggered and provides the control voltage for the first VCA.
Since LM13600/LM13700 is a dual amplifier the second one is configured as a VCA with manually set control voltage. In the picture below you can see how the circuit works. The top signal is the input signal, the middle signal is the output of the ramp generator and the bottom signal is the trigger.
There is a problem with using the ramp generator circuit this way. The capacitor is discharged too quickly when the trigger is interrupted and this causes an audible thump noise when trigger goes off. Looking below at the schematic we can see the discharge current goes through CE junction of Q1.
We can lower this current by inserting a resistor between ground and Q1’s emitter but in our specific application that will cause an offset and the output will not be totally silent in absence of input signal.
Another way to solve the problem is by paralleling a capacitor with R3 (Q2’s emitter resistor) This will cause a fade out effect and eliminate the thump noise.
Below you can find the schematic for the Lagger:
U5 shows as TL071 but you need an opamp with higher output current sink capability. Something like HA17358 with 50mA capability is good:
Trigger for the ramp generated is created when the microcontroller detects a signal from guitar. In my last article i have explained a way to rectify and average an analog signal using ADC and software. If the input level is higher than a set threshold level then ramp generator is triggered.
In the first units the middle pot was used to set a sustain period but that was changed to sensitivity control as it proved to be much more helpful.
J1 is a push-button which will generate an interrupt for the microcontroller and provide a true bypass via the SPDT relay.
You will notice some unusual supply voltages. For example the microcontroller’s Vdd is set to GND and Vss to -5V. This is done in order to provide correct trigger levels and avoid using other active components to shift the level.
Below you can see the PCB for the unit:
Here are some pictures with Project Ryu Lagger:
Here is a short video with the unit in action:
I will be supplying the hex file for the PIC18f1320 microcontroller in my next newsletters so if you want to built the unit and your not a subscriber yet please use the top right form to subscribe.
Also in my newsletter you will find offer for kits and complete units for those who don’t do well with electronics.
Often in my projects i need to detect the signal level and then feed it to an A/D Converter. Since Digital circuitry operate on a single rail power supply usually from 1.8V to 5V you cannot feed the analog signal to the ADC’s input referenced to digital ground because the negative polarity will be ignored or worse, could damage the ADC.
One way is to use an analog rectifier and convert this output to digital. This requires some extra components though and depending on maximum frequency of the signal can be expensive.
In audio domain I usually rectify it in software and I provide a DC offset on the ADC input in order to keep both positive and negative side of the signal within the ADC’s input range.
To have a symmetric voltage swing the offset is VDD/2. This of course gives a first limitation of peak input voltage of VDD/2 but it can be easily resolved by attenuating the signal first and keeping the attenuation level in mind in software side. This way we can scale the signal to fit our input requirements.
In the picture above (a) shows you the signal with a VDD/2 offset presented at the input of our A/D converter. We will sample this signal in time as shown on the x axis. and the black lines represent the result after sampling
(b) shows how to reconstruct the analog signal. You can see that for values above the VDD/2 line the green part exists in our sample value from (a) and for samples below VDD/2 line we need to add the green part.
Looking at our analog input signal and samples in (a) and looking at the green parts in (b) we can say that the signal value v(t) for each sample s[n] is:
v(t1) = s-VDD/2
v(t2) = s-VDD/2
v(t3) = s-VDD/2
v(t4) = s-VDD/2
v(t5) = s-VDD/2
What the above list describes are actually the green parts in (b). The problem that remains is that v(t3) and v(t4) (corresponding for the negative voltage swing in the input signal) are negative because sample 3 and 4 are lower in value than VDD/2.
What we did so far is eliminate the input offset and now in order to rectify it we need to make all the negative values positive. This is simply done this way:
v(t1) = s-VDD/2
v(t2) = s-VDD/2
v(t3) = VDD/2-s
v(t4) = VDD/2-s
v(t5) = s-VDD/2
Now the signal is rectified. v(t3) and v(t4) are positive and the list above describes now figure (c), the rectified output.
There are a couple of drawbacks in using this technique. First is the stability of the offset VDD/2. This voltage will need to be extremely stable for accurate result. If offset fluctuates then sample values will fluctuate too. If you can eliminate drift caused by temperature, load fluctuations etc then you can have a very accurate and fast rectifier.
There is a way to make this problem disappear by using a second A/D converter channel. And this is the second drawback if your number of channels is very limited. When using a microcontroller this isnt really a problem as usually there are a sufficient number of channels available.
I usually use the second solution and i always get the VDD/2 value from the second ADC channel and then get the input signal value. Lets look below at at example on how to do this;
/* we will use channel 0 for VDD/2 input and channel 1 for signal input */
int READ_ADC(int channel); //function prototype to read ADC value on channel
int INPUT_OFFSET, SAMPLE_VALUE;
INPUT_OFFSET = READ_ADC(0);
SAMPLE_VALUE = READ_ADC(1);
if(SAMPLE_VALUE >= INPUT_OFFSET)
SAMPLE_VALUE = SAMPLE_VALUE – INPUT_OFFSET;
SAMPLE_VALUE = INPUT_OFFSET – SAMPLE_VALUE;
Output of RECTIFY function can then be stored into a vector and perform various calculations to get your desired result. I usually average the values:
#define SUM_LENGTH 512 //how many samples you want to average
double SUM=0; // depending on the ADC resolution choose the data type
int RESULT, i;
RESULT = RECTIFY();
SUM = SUM + RESULT;
RESULT = SUM/SUM_LENGTH;
To choose the SUM_LENGTH you should consider the speed of the A/D conversion and the lowest frequency of the input signal so that you can capture an entire period. This, of course is application dependant but you can play around with the values and check the result.
Holding the RECTIFY value in a vector you can then get max value which is peak value or RMS, envelope and so many more things.
I hope this will be useful and if there is anything you want to ask go ahead and leave a comment.
Some years ago, i got my hands on an old and obscure mono mixing console. It was labeled just as PM5200. It wasnt of much use to me so i tore it down and salvaged parts for my own projects.
The other day i was looking through my things and found the mic preamp modules from that console and i thought this would be a great weekend project. I remember i used one of the modules for a guitar recording rig. Battery powered and with a transformer input it worked very well with Shure SM57.
The circuit uses only solid state discrete components and uses capacitor coupling. It allows for 3 gain settings and a nice line in/mic feature.
I do like the shield. 1mm thick steel.
You can see the original module with the busted gain setting. First i had to do is to reverse engineer and draw a schematic. It wasn’t very difficult, old PCB single sided, resistors all 500mW clear marked. You can see the schematic below:
With +/-15V the circuit draws about 10mA so its running pretty hot.
T101 along with D101, D102, R109, R110 for a constant current source. T102 and T103 form a long tail pair. You can attack it with balanced/unbalanced signal. You can use an input for more feedback or if you use the inputs together you can get a nice line in input with about 10x gain. T104 and T105 forms the main voltage amplification stage.
Next i took down all the components. Nothing was worth keeping.
I changed the NPN transistors to 2SC2240 from Toshiba. I really like these transistors in audio applications. PNP transistors i used BC559C, low noise and i have alot of these.
Also i replaced R119 with a multiturn pot of 5k in value. I hope i can get rid of the last coupling capacitor. We will see. One thing that surprised me was the lack of any power rail decoupling on board. I added 2 100n MKT caps.
After that i recorded frequency response which you can see below. It is pretty linear with just 1dB dropping at high frequency.
In next part i will explore some improvements and just put it in a box, add controls and make it ready to be used.
There were many discussions about what is the most important component in the music reproduction audio chain. I won’t get into that, we know everything is important but the better question i think is when starting an audio system what components should come first? Where should the investment start?
In my opinion, loudspeakers bring the greatest change in a system and therefore it seems the best choice of a first investment. The sonic differences are clear and the quality of reproduction can have a drastic increase when upgrading to real speakers.
So you want to build a better audio system and you set aside a specific budget for it. You are thinking perhaps to split it to get a good source (record player, cd player, reel to reel tape deck), a good amplifier and a good pair of loudspeakers. This would be a great way to do it if you have a really big budget. Most often than not tho that is not the case. With a limited amount of money at hand i would say to first invest it in a pair of loudspeakers. Especially if you still have an old amplifier and source from previous systems. You can use the new speakers in the old system for a while and still benefit greatly from the increased quality of reproduce sound. Then put aside money again and get the best amplifier to drive the speakers and then save more money and get a good source.
Then reason behind this way is that sources are more abundant. With the availability of high resolution formats you can live with PC-based sound for a while until you save for your favorite source. You can still enjoy better reproduction than your old system as opposed to getting a great record player and using less than good speakers.
With great speakers you immediately feel the increase in performance and this will give you a good feeling that you made the right investment. Money don’t come easy so you will look for an excuse in spending the money. Hearing the difference in sound will make up for that I’m sure.
Starting from the other end of the audio chain might not render such a big difference and you might loose confidence in spending money to upgrade your system.
When choosing components don’t hurry. Excitement can get you to take rush decisions, believe me, i know. Study, do your homework, go to audio expos listen and try out as many products as you can. Take notes, keep a record of what you hear, what you like, what you don’t, what surprised you, features and specs of anything you hear. Don’t rely on your memory. You will soon find out you like a certain type of sound and you will have a list of products that meet your requirements.
In a home recording environment equipment often doesn’t come in a large variety because of either limited budget or limited space… or both. I am presenting here an balanced attenuator which comes in between fixed gain preamp and recording device/soundcard.
The purpose of this device is to adjust the level and monitor it to prevent saturation of the next stage in the recording chain. It uses a L-pad followed by a balanced buffer stage. The attenuation steps are 0dB, -3dB, -6dB, -9dB and -12dB.
The meter section uses a microcontroller with a 10 bit ADC. It monitors both polarities of the signal and detects the peak within a frame of 1000 samples. ADCs samples the signal every 12us.
In the schematics above J7 will be used later on future revisions to indicate symmetry in the balanced signal.
J3 will connect the attenuation selector.
There are certain modifications i did the initial schematic tho. R7, R8 must be of greater value. At least 100k. With 100k you will get a -0.8dB signal at 0 dB setting.
The circuit is pretty straightforward, you can use any quad opamp chip for U3 as long at it operates from a 9V single supply. I do recommend a FET input opamp if R7 and R8 >= 100k as they tend to have lower noise than bipolar with high input impedance.
U2 i used a LM324 as it allows operation at 5V single supply. U2:C is used to bias the ADC inputs at 2.5V.
I will explain how to rectify the signal with a PIC in a future project but you will find the hex file for this one at the end of this article.
Here are the PCB drawings:
pcb top layer
pcb bottom layer
pcb top silk
Warning: NONE of the PCB images are mirrored!
Here is how the PCBs turned out:
I used a rotary switch to select the attenuation levels. For 0dB you can just omit R15. Use POT2 to calibrate 0dB on your meter. I usually set it to 0dBV.
After building the first unit i made some measurements. I used 5% tolerance resistors so i wanted to see if the attenuation levels are correct. Here are the results:
-6dB and -12dB settings are about 1dB off and it seems -3dB setting is also a bit off. For -12dB i soldered a 47k resistor in parallel with the 6k8 one and for -6dB i soldered one 100k resistor in parallel with the 22k. For -3dB i soldered a 470k resistor in parallel with the 47k one. Results were much better:
I will be making kits available for this device with PCBs and programmed microcontroller as well as fully built units. Please subscribe to receive more details about this offer in the next newsletter.