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Imagine this experiment:
Two independent audio signal generators, "A" and "B", present both their signals to the input of a single amplifier.
"A" produces a sine wave of frequency 70Hz
"B" produces a sine wave of frequency 1100Hz
The amp comprises two gain stages so phase is inverted twice.
The two frequency components in the signal are voltage summed and maintain registration with respect to each other - no intermodulation distortion.
Then, a single speaker direct radiator is connected. The paper moves forward and backward at 70Hz while vibrating at 1100Hz, so the 1100Hz is acoustically Doppler shifted sharp when the paper is moving forward into the room air, and shifted flat when receding back from the room air... the speaker exhibits intermodulation distortion.
Now imagine this version of the experiment:
Two independent audio signal generators, "A" and "B", each present their signal to an independent amp, each amp is connected to its own independent speaker... so two separate systems "A" and "B", each comprised of a generated signal, amp, and speaker.
"A" produces a sine wave of frequency 70Hz
"B" produces a sine wave of frequency 1100Hz
Each amp comprises two gain stages so phase is inverted twice.
This time a single microphone is used to record the sound from both speakers. That recording is later played back through a single amp to a direct radiator. The wiring to the recording microphone and the wiring to the later playback speaker are such that the over-pressure that made the microphone diaphragm displace inward will make the playback speaker's paper displace outward and present an over-pressure to the air in the room, and vice versa...
The diaphragm of the microphone moved forward and backward at 70Hz which made the diaphragm encounter the 1100Hz while in motion moving back and forth, generating the signal frequency of the 1100Hz sharp when moving toward the sound source and flat when moving away from it. Doppler shift was recorded from the diaphragm's motion so that shift will be present in the signal of the later playback amplifier and speaker.
Now during playback, the single speaker direct radiator paper moves forward and backward at 70Hz while vibrating at 1100Hz, but since the Doppler shift is in the signal, the flattened Doppler shifted part of the signal is when the paper is moving forward into the room air, and the sharpened part of the shift is when receding back from the room air... the speaker motion corrects the shifts recorded in the signal
because of the same mechanics by which it was captured by the microphone. The sound does not exhibit intermodulation distortion.
If you're still with me, consider one more version (but it's simple):
Same as the last on, but the playback speaker's wires are reversed!
Now during playback, the single speaker direct radiator paper moves forward and backward at 70Hz while vibrating at 1100Hz, but since the Doppler shift is in the signal, the sharpened Doppler shifted part of the signal is when the paper is moving forward into the room air, and the flattened part of the shift is when receding back
from the room air... the speaker motion doubles the shifts recorded in the signal and the sound exhibits twice the intermodulation distortion.
Questions:
- Have I thought this through correctly, are my assumptions and understanding right?
- Is this something that recording studios routinely take into account when setting up mics, managing tracks, monitoring playback, and generally engineering recording of music? Or do they not think about it?
- It seems to me that even with complicated modern recording methods with multiple mics and tracks and such, there could be best practice standards that would result in a world where absolute polarity of speaker connection would ensure best sound. But is it the case that none of this was ever worked through so that now it's generally all mixed up and random? Individual songs on the same album may sound better with polarity one way or the other, an incidental artifact or accident of chance in the studio?
Follow Ups:
I think no, and I think that doppler shift is a red herring. The two different frequencies combine to make a time domain waveform that is their combination and encompasses their phase and amplitude relationship. That is what the speaker reproduces, or what a microphone would capture. If that time domain waveform is asymmetrical then it will be captured/reproduced differently if the system polarity is changed. In your experiment I suspect the waveform is symmetrical, though musical transients can be asymmetrical.
Intermodulation distortion is generated by the same mechanism that produces harmonic distortion when you pass more than one signal through the distorting mechanism. As well as each tone generating its harmonics, new non-harmonically related tones appear dependent on the difference frequency. Intermodulation would not be produced by a speaker cone producing 70Hz and 1100Hz simultaneously unless one, or both tones, is large enough to make the cone move non-linearly.
Regards,
13DoW
'I think that doppler shift is a red herring'
it doesn't have to be, things need to move! he's not using the right speakers!
[just kidding don't mind me]
geoffkait confirmed my suspicion; no polarity studio recording standard.
However, now another question arises (still wondering if there is ever a real reason to swap speaker leads and reverse polarity)...
You have all certainly seen a picture of the scope display of a sine wave exhibiting 2nd harmonic distortion from a single ended triode output - asymmetric, the upper half curved to be more circular and the lower half extended and curved more sharply. I have never seen it displayed inverted, but this choice of orientation would apply to all signals at the speaker.
Anyone using SET amps know anything about differences in sound of music depending on whether the rounded or extended half of the wave is on the room-side of the driven element?
What does inverted polarity sound like? Diffuse sound/unfocused, lack of deep bass, compressed, boring. Of course, a lot of other things can cause those things to happen. Nobody said it would be easy. :-)
Edits: 11/20/21
Thanks John,
I'm not trying to establish or determine polarity of my system.
I was just thinking the other day about the concept of absolute polarity, and concluded that it would be something rather important for us listeners if the recording studios included strict attention to preserving it throughout their recording and engineering process, possibly only if they did so.
But I was curious what others thought; wondering if my conclusion is correct, and if so, wondering if others know to what degree the studios have actually known about it and controlled this.
There are no standards for absolute polarity. The out of phase track on XLO Test CD or similar will show if your audio system is in correct polarity or reverse polarity.
Edits: 11/20/21
If you're trying to determine whether your recording system inverts polarity, all you need to do is record a pulse. If the pulse is reproduced correctly, it will point upward just like the one that was recorded. On the other hand, if the reproduced pulse points down, then your recorder inverts polarity.
I have a Denon test record with a rectangular waveform with a 3:7 duty cycle. When I play the record and look at the output from my phono cartridge, if the narrow part of rectangular waveform points upward, the cartridge is not inverting polarity. However, if the narrow part of the waveform points downward, the cartridge inverts absolute polarity. Then, I can reverse the leads from the phono cartridge to get correct absolute polarity.
Good luck!
John Elison
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