The Great [Volume] Leveler (AVR Audio Compressor)

Like many people, I have fairly diverse taste in music.  My media library holds tracks from Bach, Beethoven, Billy Joel, Bonobo, Brent Lamb, Brahms, Brian Hughes, and the Bee Gees (to name just the “B” section).  I love variety.  The trouble is, all of these different genres tend to require slightly different volume settings.  Worse still, in the case of some classical music, you can get quite a wide range of volumes within a single piece (e.g. O Fortuna).  So if I hook up my BlackBerry and set it to shuffle, I find myself having to continually adjust the volume knob – either because I can hardly hear the current track, or because my neighbors are about to come banging on my door.

Well, I’m not the first one to have this problem.  Nor am I the only one to attempt to solve it.  In fact, it’s already been solved.  As you may know, there are plenty of software solutions out there for so-called volume leveling.  But before the advent of the BlackBerry, or the personal computer, there was the analog compressor.

The Buzzaudio SOC-1.1 Compressor

The sole purpose of such a device is to compress the dynamic range of an incoming audio signal: amplifying the soft parts and reducing the volume of the loud parts (thus decreasing, or compressing, the range of the track’s volume).  It’s quite a simple thing, really.  And you can buy one of these units for between $100 and $200.  But why buy something when you can build it from spare parts? 🙂

Now for all of you comp-sci majors out there who are planning on commenting on any one of a million different audio leveling programs out there, don’t worry, I know they exist.  The trouble is, even if I leveled every single MP3 I possess, what about internet radio?  Specifically, the Pandora app for my BlackBerry?  Alright, maybe there’s a software-only solution for that too (although I haven’t found it), but you know what?  Just for the fun of it, I’m going with hardware this time.  Well, mostly.  Actually, it’s going to be a mix of analog hardware and an 8-bit Atmel AVR running some embedded C code.  Sound good?  Alright, then let’s get started!

First Prototype: Powered by LabVIEW

As frequent readers of my blog will know, I’m a big fan of the LabVIEW programming language.  I can use it to whip up fairly complex data acquisition and processing applications in a matter of minutes (examples).  Much faster than writing code for an MCU.  So I decided to use it, along with my handy Mobile Studio IOBoard and Bus Pirate, for my first compressor prototype.  Before we get into the brains though, let’s talk about the guts (and by that I mean let’s talk about the analog circuitry which will be common between the LabVIEW and the MCU prototypes).

First off, we need some way of detecting the volume of our incoming audio signal.  To do this, you could simply connect your left and right audio channels directly to your ADC inputs.  Two problems with this though.  First, unless you’ve really cranked up your source’s volume, you may only see peak signals of around 100mV.  Such small voltages will mean poor resolution.  In other words, without some type of amplification, we won’t be able to detect fine changes in the input’s volume.

To counter this, I have added two operational amplifiers (op-amps, U1) which connect, via high-pass filters, to our incoming left and right audio signals.  Each op-amp increases the amplitude of its respective signal by about 50x (click to enlarge):

Audio Compressor: Analog Stage Schematic (click to enlarge)

The purpose of those two AC-coupling high-pass filters is to remove any DC offset from the input signals.  Without these filters, we would also be amplifying that offset, such that even a 0.1V bias would saturate this first set of op-amps (which are supplied by +5V and -5V).

Once amplified, there is one more thing we need to consider.  Since audio waveforms change rapidly, and swing between positive and negative voltages, our input signal may appear to change wildly between ADC samples.  What we need then is a circuit that can smoothly follow the peak amplitude of our input signal.  Kindof like an old VU meter.

Peak Hold Circuit Results (Blue = Input Audio, Green = Output)

The amplified input audio signal is shown above in blue, while the output from our “peak hold” circuit is shown in green (this plot displays 1V/div vertically, 50ms/div horizontally).  Just how does this work, you ask?  Well, scroll back up to my earlier schematic.  The key is the two sets of diodes (D1 and D2) and 10uF capacitors (C3 and C4) you see just past the first amplifier stage.  Diodes only allow current to flow in one direction (from left to right, in this case).  Thus, when the left channel amplifier’s output is greater than the voltage across C4 plus 0.35V (due to the voltage drop across the diode), C4 will be charged to a higher voltage.  Once the op-amp’s output drops, the diode D2 prevents C4 from discharging.  Thus, it “holds” the op-amp’s peak voltage.  However, because we don’t want C4 to hold this voltage indefinitely, I’ve added resistor R15 (R14 on the right channel), which gives C4 a path through which it may slowly (kindof) discharge.  The result is the scope plot above.

Now at this point, I could’ve stopped and wired both peak voltage outputs directly to my ADCs.  However, I would then have needed to read and average two voltages within my leveling program.  Since I’m on a hardware kick, I decided to simply sum the two using an inverting summing amplifier configuration (R7, R8, R9, U2).  Because this circuit’s output is inverted (negated), I then passed the signal through a unity gain (1x), inverting amplifier (R10, R11, U2).  Now we’re ready for measurements!  By the way, in case you have as much trouble as I do remembering what all of these op-amp circuits look like, you should download or bookmark this nifty guide from TI.  I use it all the time.

Audio Compressor First Prototype (Click to enlarge)

Before I move onto the brains of the device, I should explain how I intend to digitally control volume.  Recently, I acquired a Bus Pirate and a few SPI digital potentiometers.  My plan here is to use these potentiometers (P/N MCP42010), as 256-position voltage dividers.  They will be wired in-line with each audio channel.  You can see one of them (one IC containing two pots) shown as a green box in my earlier schematic.  The output from each pot will be fed into a buffer op-amp in order to keep the outputs strong (capable of driving headphones or line inputs).  For my initial LabVIEW prototype, I will be communicating with the potentiometers via an RS232 connection to the bus pirate.

LabVIEW Audio Compressor VI

Alright, so onto the brains!  [Insert zombie joke here]  The image above is a screenshot of my VI (virtual instrument, also known as a LabVIEW program).  As you might expect, it is quite simple.  I have a control for the potentiometers’ wiper positions, as well as indicators for the instantaneous and filtered ADC inputs.  The analog input is being provided by my old friend, the RPI IOBoard.  Its interface consists of just a few blocks on the diagram (for those of you unfamiliar with LabVIEW, what you see here is equivalent to written code):

LabVIEW Audio Compressor Block Diagram (Click to enlarge)

I have divided the program into two while loops (the large grey-border rectangles), each running at a 50ms rate.  The top loop handles communication with the bus pirate (initial bus pirate configuration I perform once, by hand, through the terminal).  This communication is quite simple, and consists of just two bytes.  The first byte is a command which tells the potentiometers to write the second byte to their wiper position registers.

The bottom loop is responsible for determining the appropriate wiper position, based on the incoming voltage measurements.  My logic is quite simple.  Perhaps you can determine it from the diagram alone?  Basically, I compute a difference between the current (instantaneous) voltage measurement and our last filtered value.  Then, if the current measurement is greater than the filtered measurement, I divide the difference by 40 and add it to the filtered measurement.  If the current measurement is less than the filtered measurement, I divide the difference (which is negative) by 240 and add it to the filtered measurement.  The result of this is that for loud sounds, the filtered measurement increases fairly quickly (within a second or two).  However, if the input volume drops, the filter will take perhaps five times longer to respond.

Once the filtered measurement has been computed, the appropriate wiper position is determined via a look-up table.  That table, when plotted, looks something like this:

Look-Up Table Graph

These values were determined empirically.  I simply listened to the circuit’s output for various inputs and adjusted the wiper position manually to achieve the best sound.  However, as you can see, it implements a log(x) function, as you might expect with an audio signal.  But rather than attempting to determine a precise formula for this function, as I am designing this for use on an 8-bit MCU, I chose the look-up table approach.

With my VI wired and ready to go, I proceeded to tune my filtering algorithm to provide leveling over the whole musical spectrum.  The results sounded pretty good.  But before I demonstrate anything, let’s talk about the final implementation using the ATMega328.

Second Prototype: Powered by Atmel

For my embedded prototype, I again chose an AVR microcontroller.  I’ve long been a fan of these chips.  They’re just so easy to work with, and quite powerful.  Implementing the SPI interface required just a few simple commands, since the protocol is natively supported by the ATMega’s hardware.  This particular chip also has a built-in, six-channel ADC, so that was no big deal either.  And since I’d already developed the algorithm and settings in LabVIEW, the most time-consuming part of this whole process was wiring up this lovely 10-segment LED bar graph.  Now I need more resistors…

AVR-Powered Audio Compressor (Limiter)

Although I would have liked two bar graph displays, I had only enough IO to fully support one of them.  So rather than just hard-coding the chip to display one value, I added a small button (you’ll notice it just to the left of the MCU, the largest IC) which allows you to select between displaying the instantaneous voltage input (VU meter mode) and the current wiper position.  This proved very helpful in debugging.

For simplicity, I’ve chosen to use the ATMega’s internally 8Mhz RC oscillator.  Timing isn’t very critical for this application, so I coded up a simple delay routine which works based on repeating an operation for a calibrated number of iterations.  I probably could’ve done a better (more generically useful) job on the look-up table too, but with only a few datapoints, I didn’t feel like spending much time on it.  Lazy me.  Anybody out there have some good code written for interpolating look-up tables? 🙂 Well, what I do have, you may download via the links at the bottom of this post.


In order to demonstrate the performance of my device, I created an MP3 containing about 90 seconds of audio (music) at different volumes.  This file was played back on my laptop, whose volume setting was adjusted until the loudest sound produced just barely hit the maximum  input of the ADC.  The Audacity screens you see below show the original waveforms on the top and the “leveled” waveforms on the bottom (recorded via my laptop’s microphone input jack).  Enjoy:

As you can see and, hopefully, hear (sorry about my lousy sound card – the signal is much clearer than it sounds here) in the video above, the soft parts in the original track were amplified somewhat, while the loud parts were reduced in volume.  If not aurally obvious, I suspect the results are pretty easy to see in this screenshot from Audacity:

Audacity Demo Waveforms

Final Thoughts

Overall I think this project turned out alright.  I particularly liked the method of first prototyping with LabVIEW, the bus pirate, and my IOBoard.  This definitely sped up development of the final application and made debugging easier.  The only thing I wish is that I could use my VI to generate code for the AVR.  Such a thing is available, but only for 32-bit processors.  Oh well, maybe it’s time for me to upgrade by a few bits.

In the future, I may try amplifying the audio outputs.  Obviously, potentiometers by themselves can only reduce signals.  And that’s fine, as long as my input source is sufficiently loud on even the soft songs (because we can always cut the volume of the loud pieces).  But for even greater range, amplification would be nice.

You may wonder why I didn’t include any other adjustments, like an additional volume control, or knobs to adjust the filtering speed (these would be the attack and release knobs, which, by the way, are awesome names).  Well frankly, I wanted to keep things simple.  The point of this project was to reduce the amount of knob-turning I do.  If I left things adjustable, I’d probably be tweaking stuff all the time.  It’s just a compulsion.

Anyway, I hope you’ve enjoyed this project as much as I have.  As always, if you have questions, comments, or want help building something like this, please leave a comment!

List of Parts Used, with Prices  (Excluding Passives)

  • Bus Pirate ($29.95)
  • IOBoard (not readily for sale; you might be able to buy one off an RPI/Rose-Hulman student who doesn’t understand/care about the value of what the have)
  • MCP42010 Digital Potentiometer ($2.40)
  • TLC2272, 2.25Mhz, Rail-to-Rail, Dual Op-Amps ($1.83)
  • ATMega328 ($3.93)
  • LED Bar Graph, SSA-LXB10IW-GF/LP ($3.33)
  • LT1054, for generating the -5VDC rail ($1.62)


15 thoughts on “The Great [Volume] Leveler (AVR Audio Compressor)”

  1. Good job on building this. I like how you documented it. I have some suggestions in simplifying the schematic:
    Since audio is a.c. with no d.c. component, there is no importance if overall your circuit is inverting or not, so you can give up the last inverter. Also, since you are not reading the two channels separately, you can simply connect D1s cathode to D2s cathode and only keep C4+R15 as peak measurement for both channels, which will go directly to the microcontroller.. This way you can read the greatest peak of both channels. Summing the peaks migh make your circuit less sensitive to an amp saturating peak on one channel when the other has a low level. It is not likely that this happens, but there is a chance.

    1. Thanks Bogdan, I appreciate the comments! Good point about the peak hold – I could definitely have joined the two circuits at the diodes and then would’ve eliminated the need for both the summing and inverting amplifiers. As I was building this I wasn’t sure whether I was going to read in two voltages or just one, so I went with two separate peak-holds. But you’re right, I probably want the max peak anyways. Going with the summing method I have here though does require the inverter as the final stage, since the summing amplifier inverts (negates) the output. Since this is going into my AVR’s ADC, which can’t read negative voltages, I had to add the inverter to flip it positive once again.

  2. You could use logarithmic digital pots (designed for audio), so no look up table would be required in MCU.

    Set a higher gain for the audio line, so it could normalize the level, not just compress it.

    1. Yes, both good points. As is usually the case for me, I used parts I had on hand, hence the linear vs logarithmic choice. I do agree there should probably be some gain in the circuit to really maximize the range. Thanks!

  3. Hi,

    Nice project and very well documented.

    Regarding the two graph bars, why don’t you multiplex them ?
    It’s often used, and you have enough pins to do so.


  4. Hello Mike,

    This is an awesome project and thanks for posting it!
    I have a stupid question. I’m looking to do something similar to this except to use the MCP42010 (or MPC42050?) purely for remote volume control from RCA cables. Since I won’t be doing any signal analysis with ADC’s, is there need for any op-amps of capacitors in such a circuit?
    Thanks for any help!

    1. Hi Nemik,

      Sorry it’s taken me so long to reply… for some reason I didn’t get an email about your post and only just noticed it.

      So that’s a good question! If you’re just looking for volume control in a line-level circuit, no, you probably don’t need anything more than a digital potentiometer. If you were trying to drive something like a pair of headphones, you would need a simple buffer op-amp between the digipot and the headphones to provide the necessary current. But if you’re just connecting the output of an MP3 player to a stereo or something, there’s probably no need to buffer the signal. The stereo’s input likely has a fairly high impedance and thus won’t load down your potentiometer.

      Although, by now maybe you’re already tried it out? If so, how’d it go?

  5. There is no DC offset, according to digital pot’s data sheet input should never be less than -0.6 V, have you considered this?

  6. Hi there,

    Nice job! Congrats!

    I was wondering if you have made some modification to this furing this year!

    And another question…. what is the best leveler software…. for leveling the volume of a huge base of mp3 files?


    1. You want to use something like ReplayGain to analyze your files and a music player that supports reading in the data (foobar, AtomicPlayer). Or use a layer between your music player and speakers such as Breakaway Audio Enhancer.

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