Passive Summing Mixer


Hi Martin, hope all is well. Haven’t heard from you in a while. I sent you an email a couple of weeks ago. Still trying to get my head around the electronics thing. Came across this the other day what do you think?
I’m working on a summing mixer right now. Its going to be passive, but I think it still needs an output amplifier to bring back up the lost voltage later going through resistors. Right? I was thinking of putting 2 different ones in there with different sonic flavors so i could have a choice. Still doing the research though. What are your thoughts
I’ve added a few books that might build on the knowledge for this post – SP


  1. Hi Paul
    I see what you mean now, and why you want a little six channel mixer, because it matches the converters on your soundcard. You’ve got several line level sub-mixes in the pc, what used to be called busses on a mixing console, and they are stereo in your case. I used to do the same as we had a recording track shortage (as it was tape), so that we’d get all the drums down into two tracks, panned for stereo. Doing it like this meant that the mix became part of the song composition as once something was done, it was set in time and the only way to change it was to re-record the tune from scratch!
    When you get it off the ground, pay attention to your panning as you can get weird nulling effects. The good panning hardware is expensive, a cheap pot might give you unpredictable results that you WILL hear as you are running very high quality stuff, which will be a waste of your gear. Check out what Douglas Self has to say. He is not very complimentary about valves or transformers!
    My gut feeling is to keep your “stems” totally separate for your sound processing (which is what you are trying to do), and then do a final stereo mixdown using your normal method in the pc. You may have to mix in a weird, non-intuitive way to do this, but you know, horses for courses. And you might come across a new “sound” or hook signature with the experimenting.

    The thing is, when you mix on a big proper (analogue) console, everything is mono really. It’s only at the final stages that you set the stereo image. As you know, lots of the physical desk space is taken up with the various routeing switches and buss send controls (both pre and post fader). That’s what I was saying ages ago – the electronics is very well devoped and can take care of most things. The trick is in deciding how much desk space you are prepared to use and how much you are willing to spend on knobs, switches, pots etc. It very rapidly mounts up. Sometimes, you need as many busses as there are input channels! You are already there – you’re using your six audio inputs as six audio buss sends… Think what it’d be like on a 48 channel mixer. That’s what all the coloured sticky tape is for otherwise you just get lost!

    As for changing your bitrates etc. I’d keep everything at the max so you don’t lose anything. Watch out for high frequencies. A recent SOS article explained all this. What it means is to use a steep low pass filter or else you get reflections back down the audio spectrum that are unpleasant. It all comes from the original maths behind the technology.

    As for colouring the sound. I remember reading somewhere that Mr Neve actually put a little bit of a midrange peak in his frequency response, so that they weren’t really flat, unlike the Soundcraft consoles. If that’s true, can you not do the same? I think it was a couple of capacitors, that’s all.

    Enough of these idle musings, I’m off! Keep well.


  2. From my sound card i have 6 outs and 6 ins(all analogue)
    certain driver amps have certain color/tonal qualities(is NEVE or API A12=API 312
    N72=NEVE 1072)
    from my sound card they are line level, thats why i’m not interested in the pre amp side for now

    Stereo stem: say, inside your computer you have you drums; kick, snare, hats, toms. now say all together you have 6 tracks of these, you set the level and pan for each one, then you route them all to a strereo stem. This way you could put a 1 compressor or limiter across them all to help them gel or just using the one fader raise or lower the overall level or run out of computer to use an analogue comp or limiter or whatever processing

    hope this helps>

  3. Hi Paul

    How many DA/AD converters have you got for all these channels?

    Remember, you’re only going to get stuff to work if analogue is going through the mixer. To do six channels needs at least that in converters.
    When you say “different coloured op amps” I suppose you meant pre-amps or driver amps? And is “colour” is it just an expression for the tonal quality?
    I don’t know what you mean by the germanium transistor…But to wire one up is the same as wiring anything up! You can still buy them!

    What sort of stuff would you be putting through the mixer? Just the pre-amped mics, guitars etc? Or is it the degraded output from the PC workstation alone like you described? the key is “degraded”. As a rule when you drop bits or speed, you’ll lose quality

    What’s a stereo stem? Is it the basic stereo track that’s close to the final print?

    Must go to bed…Rees

  4. Cool, thanks for that. I still don’t completely understand all things yet, but i do feel that i am getting slightly closer each time.

    The idea is to build a 6 input summing mixer, that can be used as a mono 1-6 channel input or a 1-3 stereo channel input. I would like the output to be able to be switched through 2 different coloured op amps(ie neve 1084 and API 312) also maybe germanium transistor(not sure if this is possible?). As i often compose in one program and then mix/produce in another program, i use a process called rewire(virtual patchcords) form one program to the other. I’m sometimes going from 44.1 to 96. So rather than doing it internally, i was thinking of running certain sounds through the summer to give a sonic colouring. Also at the end of mix down, instead of dithering form 24bit 96khz down to 16bit 44.1khz i am looking at having 3 x stereo stems inside computer then running them through summer and back in at 16bit 44.1khz. our thoughts on if this is possible? Thanks

  5. Paul

    classa_amp.jpgThe neat bit of op-amp info is at the bottom of this scan of an ETI Magazine (Summer Circuits) issue that I used to refer to all the time. (the neat little headphone amp circuit is at the top)

  6. Hi Paul

    You’ve got two things going on here. The first was the tape overload sound copier thingy and the second is the mixer.

    ONE: Tape “Sound” copier

    Well you could always get a tape recorder! Failing that, I can’t see how the diode chain works. This link from St Andrews University shows a nice graphic of the operation of a diode. What it means is that a diode has to have a certain voltage across it before it starts to conduct.

    The 1N34 is a germanium thing from the fifties, which means this voltage is about 0.1V and the 1N914 is silicon with 0.6V. Funnily enough, the first one is called a “point-contact diode” cos it’s got a little pointy wire pressing on a piece of germanium. This was the original “cat’s whisker” that people used to have as the heart of their early radios…

    As you can see from the circuit, the thing shorts the signal to earth via the back-to-back diode pairs, but only when they conduct. Looking at just a positive swing, say, the positive signal voltage has to exceed ((3 x 0.1) + 0.6)Volts which is about 0.9V. So at small signal levels (about line level which is 0.775V) nothing happens. Above that, i.e. a signal level of > 1.8V peak to peak, then the signal is shorted to earth, which means it clips, making the nice sine waves go square. That’s not really compression. It’s a fuzz box…

    This isn’t what happens with tape, plus, the article mentions the use of pre-emphasis. This is used on cassettes a lot, a bit like the RIAA curve on vinyl pickups. However, on studio tapes the tape whizzes along so fast that the use of a recording/playback eq was negligible as it wasn’t needed. The best thing if you can get access to a tape deck, to try is to stick a sine wave into a tape and (being careful not to damage your speakers!) turn up the signal gradually and listen to the distortion kick in. It’s easy with a three head machine. If you haven’t a signal generator, use a flute or (none reedy)church organ sound. Try it with different notes. It’s more like a squelching and then a bit of crackle kicks in as it can’t take any more.

    TWO: Passive Summing Mixer

    Make sure that you use log (not linear) pots for your faders. Also, make sure that the bottom of the pot goes to earth. If you use inline faders then you’ll still have a bit of bleed-thru and you’ll never be able to completely cancel a channel.

    What you are talking of is what you should be doing – use a virtual earth summing amp. This is quite easy to do, but quite hard to do well. If you are only doing a few tracks you’ll be okay. Use a good qualtity op-amp for the active component. The resistors chosen will be the impedance seen by each channel. See this article by Douglas Self ( – it’s that man again!). You need the sub heading “6. Summing Technology” although the whole article is excellent.

    The thing about the virtual earth is that the signal coming down one channel “dissappears” into the virtual earth so it can’t interfere with the other channels. This is brilliant! Use it!

    You’ll need two op-amps, one to do the mix and another as an inverting buffer to fix the phase, (the virtual earth is at the inverting input of the op-amp, so you need to invert the phase so that everthing you actively use in the studio is IN PHASE). It’s usual to stick a bit of gain and/or EQ into this circuit element as well as it’s sitting there doing nothing! You can see a buffer amp as the output stage on the circuit diagram link you sent me for the tape imitator. If the signal went to the inverting (-) input, that would be an inverting buffer amp… Easy innit?!!!

    The mixer resistors should come after the channel fader (which shorts to earth if necessary, remember). The value should be somewhere between 4k7 and 100k. The top range will be enough so that a electric guitar plugged straight in will sustain for weeks. If the values are too low, they will load the feed, if they are too high, then it will be noisy(it’s that Boltzmann thing again). Usually, you’d use preamps so that everything was about the same level by the time it gets to the summing resistors, and then set these at the low end. I think they were 10k in the one I made.

    Hope this helps a bit. Check out the op-amp circuits I sent before. They are used everywhere. Mr Self uses two nice clean transistors as his front-end amp stage followed by the op-amp (that’s Fig 5 on that page). Some modern op-amps have these included in the package which means less clutter and soldering. I think they were included in some of the circuits you sent me earlier to look at for making your own high quality pre-amp. Speaking of which, Mr Self was one of the designers of the Soundcraft mixing desks. He says this in the page I linked to…


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