Absolute phase - which one is it, the one of the instrument or the one of the microphone?

To get a good frequency accuracy with an FFT (or more generally a DFT), you need to have a small bin width. The bin width is inversely proportional to the time interval over which you look, so you need a long time interval. When you calculate a moving DFT over long time intervals, you don't know accurately when the tone started or ended.

The Hilbert-Huang transform is meant to solve that issue. It's an algorithm to split a signal into a bunch of amplitude- and frequency-modulated sines.
 
According to https://en.m.wikipedia.org/wiki/Auto-Tune Auto-Tune doesn't use anything like the Hilbert-Huang transform, but a different method based on autocorrelations.

Maybe the 2008 plug-in that you linked to, the one that identifies notes in polyphonic music, uses it, maybe not.

In any case, I haven't seen any claims of uniqueness in the book so far, so it could be that someone else invented something similar.

This discussion is very off topic. Do we stop it or ask the moderating team to put it in a separate thread?
 
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We were talking about phase, maybe of LF transients (say, for example, a kick drum, maybe with a plastic beater to add a little 'click' to the sound), and possibly even measurements thereof.

Typical audio FFTs are steady-state and phase-free (magnitude only). Can't measure time-domain transients that way.

Why don't we have a better way to measure time-domain transient performance?
If we did it might also help measure dynamic imaging differences between a stereo amp and a dual mono-block setup?
Can we make effective us of more modern signal analysis techniques, maybe something like Hilbert-Huang?
 
When talking about polarity we have to face that all positive waves are gone through npn transistors and all negative waves through pnp types. Thats a fact in nearly every amplifier and in most preamps even in D/A converters. So perhaps this is a further reason we can hear these differences when changing polarity.


It would be interesting to compare the polarity with a sound file reorded with a class A microphone direct to an A/D converter and then played through a single ended class A amp.
 
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An example of absolute phase might be if you mic'ed a kick drum at the front head. When struck by the drummer at the rear side, the front head moves out towards the audience. So, maybe a reproduction speaker should do the same?
 
This has been a fascinating discussion. However, since most speakers don't pay attention to phase coherence or time alignment in their design, much of it may not apply to home listening. Not to mention that we are completely at the mercy of decisions made in the recording studio.

I have preferred listening to my Vandersteen 2C speakers for over 20 years. It is hard to find others that do as well at preserving the impact of percussion instruments. The Theil designs are similar in their dedication to preserving both phase coherence and time alignment. Big planar magnetic speakers and electrostatic speakers also do a good job, but it can be hard to find ones that extend well into lower bass frequencies.

The Pass designed Slot-loaded Open-baffle (SLOB) speakers that a few of us got to build last month have an advantage in their use of a single full range driver that covers most of the audio band. The big 15" drivers that cover the lower bass frequencies. They just need amplifiers with the same phase for bi-amping (or taking the phase into account with the speaker connections). Mine are still going through the run-in and level adjustment process. They sound very good so far.
 
This has been a fascinating discussion. However, since most speakers don't pay attention to phase coherence or time alignment in their design, much of it may not apply to home listening. Not to mention that we are completely at the mercy of decisions made in the recording studio.
My speakers are time correct, but the latter is indeed a major issue leaving much of the sw with mixed up phase.

dave
 
I know all-pass filters are one of the many tricks used to make radio transmitters sound louder - not better, just louder.


True, but when you use a minimum-phase equalizer to equalize out some abberation caused by a minimum-phase thing, the equalizer's phase response actually corrects for the phase shift of the thing. Typical example: the usual minimum-phase RIAA correction networks correct for the phase shift of the RIAA filters used during recording.

I think PRR's second sentence also means that everything he knows of that causes tilted frequency responses in microphones is minimum phase, therefore the phase response is automatically corrected when you flatten the magnitude response with some minimum-phase means.
A minimum phase EQ curve only fully restores phase response if the applied curve is an exact mirror of the minimal phase curve that was applied first. This is not a situation that occurs in the recording studio.
 
IBP works over a span of freq but it is limited. If you have access to one hardware ( or plug in) IBP you can compare between allpass and time aligning the signals on takes needing compensations , there will be difference in outcome.
Is it a deal breaker? Not in my view as IBP can be a great help and is fast to implement ( time aligning can be time consuming and during a session you most always fight the clock).

About mixing analog and digital: nothing stop anyone to track in analog and then digitalize at later stage... or the other way around! I've done both to great effect.

You don't know in advance how a track will sound in a mix BUT you can know if the multi mic induce some 'phasing': for this we have tools to visualize ( lissajous figure) and the most lmportant button on monitoring section called 'mono'. When locating mic seting up amps i monitor in mono and compare mics to the 'main one' ( usually a sm57) modifying location until it sound good and 'full'. Doing so will limit the amount of work at later stage ( anyway and beside what people usually think you can't do many things in 'post production' stage if it doesn't sound good at takes and better spend time wisely, 1hour to set up save 3 to 4 of editing...).


That said I most always now what i want - or what musician want- (kind of sound) and which mic ( or blend) to use to reach it ( tracking cab is for me almost always: sm57/e609/md421/ ribbon ( Coles, Aea, Beyer M160.../ large capsule capacitor -u87/47 fet/67/c414/...).
This is my typical set up. Most of the time max 3 mic open, more probably 2. Large condenser located away from cab to catch early reflection, the 4 drivers at once and room/air ( used most only on soli the 47fet being the exception - works well for rock/blues in 'close' micing) ( typical cab 4x12").
But this is a way to do it, there is many more and possibly better ways to do it. In no way i claim to have THE answer or THE workflow.
I like to do unconventional things when I have the time during a recording session, just to hear how it sounds. That's how I discovered that the Neumann Gefell M582 with an M94 capsule is a great close mic for cabs. It just sounds right. It is my standard now (often with a close ribbon and a distant LDC).
Might be worth trying out!
 
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The effect of the polarity on the sound is quite subtle; there has been at least one study showing that polarity changes are audible in music under double blind conditions, but they only found an effect with acoustically very dry recordings played back in mono in an acoustically dead space if I remember well.
If there is a difference i agree that it wil be extremly subtle. Maybe you have a link to that paper?