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Minimum Phase vs Linear Phase

maty

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Hardware improve. I would say I now have the bass that class D amps usually provide. I know, it is incredible but it is true. And with a second-hand AVR that cost me less than € 120 put at home! Much more bass and fast. It looks like it has other loudspeakers, much bigger.

The key: CLEAN power, again.
 

KeithPhantom

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Hardware improve. I would say I now have the bass that class D amps usually provide. I know, it is incredible but it is true. And with a second-hand AVR that cost me less than € 120 put at home! Much more bass and fast. It looks like it has other loudspeakers, much bigger.

The key: CLEAN power, again.
And how an amp can provide "better" bass? Doesn't that mean that it could be adding something such as distortion or tone controls?

In Spanish: ¿Cómo un amplificador puede añadir frecuencias al menos que no sea por distorsión o control de tono? Es raro que opines que un amplificador puede cambiar tanto las características del sonido.
 

Music1969

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Hi @mansr

I read your Troll Audio post on this topic - very imformative, thanks!

Using the same filters (and associated parameters) used in your post, could you show min vs lin phase step response plots?

I've read somewhere that our hearing is logarithmic. How would these same step response look in log scale?

If I find the source of where I read this I'll post here.
 

Julf

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Ouch! Don't wake up the troll!
 

mansr

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I read your Troll Audio post on this topic - very imformative, thanks!

Using the same filters (and associated parameters) used in your post, could you show min vs lin phase step response plots?
Linear phase:
1616156688566.png


Minimum phase:
1616156719225.png


I've read somewhere that our hearing is logarithmic. How would these same step response look in log scale?
It is true that we perceive sound intensity logarithmically. An increase in SPL of 10 dB subjectively sounds about twice as loud. Plotting the time-domain waveform with a logarithmic scale doesn't make sense, though. Simply put, the logarithmic part of hearing takes place after the sound has been separated into frequencies. That's why we usually use logarithmic scales for spectrum plots.
 

Music1969

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Linear phase:
View attachment 119088

Minimum phase:
View attachment 119089


It is true that we perceive sound intensity logarithmically. An increase in SPL of 10 dB subjectively sounds about twice as loud. Plotting the time-domain waveform with a logarithmic scale doesn't make sense, though. Simply put, the logarithmic part of hearing takes place after the sound has been separated into frequencies. That's why we usually use logarithmic scales for spectrum plots.

Thanks!

Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.

If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?
 

Mnyb

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Thanks!

Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.

If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?

That's the central missunderstanding , there is no pre echo signals coming out of the DAC with music . They are there with the test signals .
 

Music1969

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That's the central missunderstanding , there is no pre echo signals coming out of the DAC with music . They are there with the test signals .

Aren't these impulse / step responses being shown in this thread coming from analogue output of DAC?

Isn't music just a large number of these test signals?
 

Julf

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Aren't these impulse / step responses being shown in this thread coming from analogue output of DAC?

Isn't music just a large number of these test signals?

Those test signals are basically synthetic, illegal inputs. You will never find them in the recording.

Music is a large number of perfectly well-behaving sine waves.
 

Mnyb

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Those test signals are basically synthetic, illegal inputs. You will never find them in the recording.

Music is a large number of perfectly well-behaving sine waves.

For testing purpose you bypass the required bandwidth limiting ? that's what i understood about it . One can argue if similar signals appear in post process (ADC at inputs will bandwidth limit ) well written DAW plugins to , but finally the sample rate conversion to your CD or download file ?
 

Julf

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For testing purpose you bypass the required bandwidth limiting ? that's what i understood about it . One can argue if similar signals appear in post process (ADC at inputs will bandwidth limits ) weel written DAW plugins to , but finnally the sample rate conversion to your CD or download file ?

It uses/assumes an artificial signal with infinite bandwidth, without any limiting. You will not get that kind of signals from post processing or DAW plugins.
 

Mnyb

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It uses/assumes an artificial signal with infinite bandwidth, without any limiting. You will not get that kind of signals from post processing or DAW plugins.

Suppose its well understood since decades :) in the pro world
 

Julf

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charleski

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Thanks!

Another thing. I've often heard the pro-min phase people say that speakers and headphones are minimum phase devices, so it doesn't make sense to use linear phase filters.

If we look at headphones (easier because it's a single driver) and linear phase filters, is the driver attempting to play these pre-echo signals coming out of the DAC?
If you’re doing EQ in the audible range then you can hear a difference between linear and minimum phase. It’s easy to demonstrate this, here’s an example.

From what I’ve read, when EQing speakers you use min-phase filters as a way of inverting the minimum phase errors in the speakers’ frequency response, thus cancelling out both FR and phase error.

None of this has any relevance to over-sampling filters used in DACs, though.
 

Rottmannash

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Abuot raedibaltiy: I awlyas touhgt tihs was fnuny:

It deosn't mttaer in waht oredr the ltteers in a wrod are, the olny iprmoetnt tihng is taht the frist and lsat ltteer be at the rghit pclae. The rset can be a toatl mses and you can sitll raed it wouthit a porbelm. Tihs is bcuseae the huamn mnid deos not raed ervey lteter by istlef, but the wrod as a wlohe.
That just freaked me out- was shocked I was able to read that so easily.
 

charleski

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It's kind of funny, people that do RIAA via software using Audacity never say there is any problem with linear phase equalization.
RIAA equalisation should be minimum phase. But you need a pretty steep filter slope to produce enough ringing to be audible, which is probably why people can get away with linear phase. Also, well, we’re talking about vinyl here ...

[Edit] Oops, replied before I noticed your name. I’m sure you know a lot more about RIAA than I do.
 
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