Interesting question.The DPA mic site notes that condenser mic output relates to the sound pressure, while dynamic mic output relates to sound velocity. At top pressure, sound molecules are at zero speed. (I think I get this.) That means a condenser is at max output while a dynamic would be at zero. Ergo, according to the site, the waveforms between two dynamic and condenser mics that have theoretically frequency response would be 90 degrees out of phase, even if they have same polarity.Is this right? I checked my copy of John Eargle's Microphone Book 2nd edition. I've no doubt the answer is in there but 10 min of skimming didn't fully answer. He does note the result of an impulse test done by Boré in 1989 which seems to provide a contradictory result to the DPA conclusion above. The difference between these signals is not that they're 90 degrees out of phase, but mainly that the dynamic mic (on top) exhibits a slower excitement and dampening response while also showing additional oscillations compared to the capacitor up top. These anomalies would affect the sound if you were to combine the two mics together, but not due to a phase difference. (The OP's question about mono compatibility introduces the question of separating the two signals in the stereo field and then recombining them, which isn't what I'm getting at.)Something that I assume but isn't totally clear from the graph above is whether or not these are taken at the exact same time. If they aren't, but were later aligned for comparison purpose, I suppose some 90 degree difference could exist between them? Maybe? I'm a little confused.Long post. Going to continue with another one where I test it out on my own.= Justin

Justin Foley wrote:This may be a little tough to read, but it tells us enough. The top is one cycle of the DAW's 500hz tone, the middle is the dynamic recording and the bottom the condenser. The big time lag between the first one and second two is recording latency. But since the bottom two share the same latency, we can compare between them.A bit of a nitpick, but PT should be correcting for that after the pass. That the time delay is caused by the physical distance between the voice coil and your microphones. In a practical scenario, this would also be demonstrated if you were combining a raw bass DI signal with its amplified counterpart, regardless of recording medium.

Justin Foley wrote:I need to give a little more thought on the impact on a sound that's more complex than a sine wave.Seems like a square wave would be an ideal candidate for such a task. Program material would probably be too difficult to decipher with all of the variables.

djensen wrote:Justin Foley wrote:I need to give a little more thought on the impact on a sound that's more complex than a sine wave.Seems like a square wave would be an ideal candidate for such a task. Program material would probably be too difficult to decipher with all of the variables.A square wave will help us verify that the effect exists. But because the difference between the two microphones depends on signal frequency and not time, changing the relative position of the mics or sliding the waveform in a DAW won't align the signal for any waveform more complex than a single frequency (or its higher fundamental, I guess). You couldn't shift the bottom waveform of the bottom image in my last post to exactly line up with the one above it; what would work for one frequency would make another worse.Got a couple of ideas for some further tests on this; will post later when I get the chance.= Justin

... Square waves don't exist in air... And therefore maybe aren't the best thing to expect a clinical result from when judging microphones.

œTrue square waves don t exist anywhere, since theoretically they represent instantaneous changes in voltage and current.

Perfect square waves don't exist in DAWs either, they are frequency limited by the sample rate.So technically speaking it's not really square, it's the combined sine waves of all the odd harmonics from the fundamental up to maybe 22k or so, and it will be a continuous curve with wiggles in it, never a ruler-straight square shape.That's not just true in the air, it's also true of the original digital square wave you start with.

Whenever I have to combine two mic signals from the same source I not only time align them in my DAW but also put a phase rotator on one of them and set it to the degree of rotation that sounds the most flattering (even if that puts them technically out of phase).I basically treat it as a form of EQ.

Indeed. EVERYTHING is frequency limited. It's never straight up and down; we're living in a wiggly world.

A guitar split in two amps and a mic on each speaker is pretty regular here. When i have the guitar player plug/unplug his guitar to get a short signal i can see on the DAW which mic needs to go closer or further to it's speaker. I often noticed that just having both capsules the exact same distance from its speaker does not guarantee they will give a signal that is arriving at the same fraction of a second. I always used to wonder why this or that mic would need to be closer or further to get signals that are pretty good in phase. Of course i always check by ear and flip polarity to make sure it all combines well when these two signals get together again- depending on amps stages the polaritiy in some amps is flipped anyway. Having dismantled many mics i know with a good number of models where the capsule is sitting.