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Is it possible to automatically correct the phase using REW arithmetics?
OP
AudioBoy
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Are you talking about this part?
If not, I want to ask if anyone knows why the line is not straight? I've never noticed this before on an unmodified graph. Usually, before the impulse begins, this line is strictly straight.
I've have made multi-point measurement, but I don't know if it suitable or not. Anyway I'll try to use it.
OP
AudioBoy
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yes to all your questions.
There's no filter because I used ASIO and disabled OS sound features.
The device is RME Babyface Pro FS
Could the subwoofer affect this?
In my case, the signal goes along the route PC->DAC->subwoofer->amplifier->main speaker
Last edited:
palm
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I‘m not familiar with windows but I meant a FIR from previous experiments as sometimes we forget to disable it.
The sweep measurement method is quite robust, much more than when you had to blow balloons !
When we were using CoolEdit and Aurora it was a bit easier to what’s going on, like the distorsion that you find back on the left of the main impulse after deconvolution. I doubt this is the case here though.
Other problems would manifest when different clocks and sampling rates were involved in the chain, but there are now timing chirps to correct for this.
The sweep measurement method is quite robust, much more than when you had to blow balloons !
When we were using CoolEdit and Aurora it was a bit easier to what’s going on, like the distorsion that you find back on the left of the main impulse after deconvolution. I doubt this is the case here though.
Other problems would manifest when different clocks and sampling rates were involved in the chain, but there are now timing chirps to correct for this.
It will only be straight with perfect minimum phase behaviour. There's negligable non-minimum phase action (pre-ringing) going on in the speaker prior to the excitation.
OP
AudioBoy
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I understand. But I'm used to turning off everything that isn't related to measurements, so I'm pretty sure that no filters can be accidentally left on. Even if this happened, ASIO (which bypasses all the system audio additions that the APO Equalizer core uses) would simply bypass them and route all the flow from the microphone to the DAC.
OP
AudioBoy
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yes I use acoustic time reference
This is weird because I never notice this thing before. I have some old measurements and measurements of the same speaker without subwoofer and there are no such pre-ringing thing. So I started to think it's subwoofer DSP (which I don't use) or some else.
Speaker driver itself acts as a minimum phase device but as soon as you add a crossover and other drivers it isn't anymore. Also cabinet diffraction aren't minimum phase. But there's nothing remotely audible there so I wouldn't worry.
I couldn't agree more.
OP
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Yep, all you say makes complete sense to me.
I get the "room correction" motivation loud and clear.
And is why i keep an inquisitive eye on it....
I can assure you, this will not be in your top 20 problems during filter creation.I just think could this be a problem when I try to create FIR filter from such measurements. Escpecially when I want REW to show me simulation of actual step response that I'll get using the filter I've created.
What is the start frequency on the REW sweep?
If it's not well below the start of the sub's response, I sometimes see waves in impulse before the rise. Math gack i think.
I'm fairly convinced when the Fourier math has to deal with mag and phase traces that don't match up ala min phase & excess phase, step response goes wonky on left side of the impulse peak.
Cool. And you're rock dead certain there's no high pass filter in play on the sub, (or main speaker) ?
If so, seems to me the sub / main speaker interplay is showing some phase and mag relationship the math can't match up.
Can you kill the low freq that overlaps sub from the main speaker (disconnect the woofer maybe)...and then see what you get?
OP
AudioBoy
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Sub is in manual operation (non DSP) mode and main speakers are passive. But of course sub has HP filter with 80Hz XO.
I think you're absolutely right. i never ran measurements in that range before. And just as I've wrote earlier without a sub main speaker measures pretty straight in that region.
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I will try to explain the maths (much as I can understand) behind the ripples seen with measurements that are not full range Nyquist (ie 0-24,000Hz for 48,000Hz sampling rate):
REW's uses a "refined" version of the measurement method called ESS (exponentially swept sine). While the exponentially swept sine signal formulated (in discrete terms) below is being played,
x(k)=sin(w1 x K / ln(w2/w1) x exp(k/K x ln(w2/w1)); k=0,....,K (w1,w2 angular frequencies)
the recorded signal is convolved with the time reversed and -6dB/octave damped (amplitude rolled off) version of that excitation signal. Then, linear impulse response is separated and most of the harmonic distortion is passed on to the non-causal part.
The sweep needs to begin and end in zero phase to minimize any ripples due to the boundary effects at the start and end points. To achieve this, there are two adjustments made:
1. Fix the start and end frequencies and end the sweep in exactly the Nyquist frequency (w2=PI),
2. Let the sweep span a "fixed" number of octaves (w1=PI/2^p) where p is the number of octaves
With these changes, the start and end phase becomes:
phase(0)=(PI x K) / (2^p x ln(2^p)), phase(K)=(PI x K) / (ln (2^p)
Choosing K = M x 2^(p+1) x ln (2^p) , where M is an integer ensures that both endpoints will have the correct phase. However, since K is discrete and integer, two different values are needed for it; one ideal according to the previous criterion, and one rounded to the nearest integer. Having two different K values causes a frquency stretch by a factor of Kideal / K which should also be compensated in the time inversed x(k). The error is typically in the order of 10^(-6):
fs 48,000
Kmax 480,000
p 10
Kideal 468,456.591
K 468,457
Kideal/K 0.999999126
REW's uses a "refined" version of the measurement method called ESS (exponentially swept sine). While the exponentially swept sine signal formulated (in discrete terms) below is being played,
x(k)=sin(w1 x K / ln(w2/w1) x exp(k/K x ln(w2/w1)); k=0,....,K (w1,w2 angular frequencies)
the recorded signal is convolved with the time reversed and -6dB/octave damped (amplitude rolled off) version of that excitation signal. Then, linear impulse response is separated and most of the harmonic distortion is passed on to the non-causal part.
The sweep needs to begin and end in zero phase to minimize any ripples due to the boundary effects at the start and end points. To achieve this, there are two adjustments made:
1. Fix the start and end frequencies and end the sweep in exactly the Nyquist frequency (w2=PI),
2. Let the sweep span a "fixed" number of octaves (w1=PI/2^p) where p is the number of octaves
With these changes, the start and end phase becomes:
phase(0)=(PI x K) / (2^p x ln(2^p)), phase(K)=(PI x K) / (ln (2^p)
Choosing K = M x 2^(p+1) x ln (2^p) , where M is an integer ensures that both endpoints will have the correct phase. However, since K is discrete and integer, two different values are needed for it; one ideal according to the previous criterion, and one rounded to the nearest integer. Having two different K values causes a frquency stretch by a factor of Kideal / K which should also be compensated in the time inversed x(k). The error is typically in the order of 10^(-6):
fs 48,000
Kmax 480,000
p 10
Kideal 468,456.591
K 468,457
Kideal/K 0.999999126
