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time aligning with 'REW'

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matty w View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote matty w Quote  Post ReplyReply Direct Link To This Post Topic: time aligning with 'REW'
    Posted: 09 September 2017 at 2:07am
Greetings all

So I recently got a laptop , steinberg interface dbx rta condenser Mic and installed the 'Room Eq Wizard ' program .

Iv been getting to know the software over about 5 weeks now and everything is making a lot more sense .

Theres a few things I'm unsure about ,like for starters' time alignment'
There's so much information online on how to go about this but it would seem a little contradicting or Atleast multiple paths to acheive a similar result !

So far ......

I run a loop back on the unused interface channel for timing reference .. to get me in the ball park , rite ?.

Then ... I have options ,

I can view the impulse response and align on the basis the impulses are aligning ( if anyone has a link to a good video on this that would be great as I'm not 100% confident in my ability to read it correctly ) .

Or ...

Some people say to concentrate on getting the phase graph aligned


Question..
For example , if the time alignment is wrong , surely it's possible to still align the phase graph ? Or is that why you unwrap the phase ,to see what the phase is really doing ?


Any incite or chat would be great :p

On a side note ... What smoothing do people usually apply before using the eq generator ?


Also ... :) is it possible create a 'house curve' manualy by either entering in parametric eq or dragging the line on a graph ?

Theres actually heaps of material online for this program and I feel like iv read it all lol

If anyone cares to explain it to this dummy I'd be forever gratefull

Ta

Black to black , red to red , blue to bits ....

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rubjel View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote rubjel Quote  Post ReplyReply Direct Link To This Post Posted: 11 September 2017 at 10:46am
Very interesting questions, I have been curious about this as well! 
I think the phase approach makes a lot of sense, too bad the phase cannot be viewed real time in the RTA.
I will be following this threadThumbs Up
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odc04r View Drop Down
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Post Options Post Options   Thanks (1) Thanks(1)   Quote odc04r Quote  Post ReplyReply Direct Link To This Post Posted: 11 September 2017 at 11:31am
Time aligning via impulse response is as you say - measure the delay from t=0 to when the impulse is recorded. Do that for all of your bands and you have a good first approximation of delays between cabinets. Shouldn't be affected by room modes and echo much either, as the first sound to reach the mic should be via the most direct route.

Set the delays relative to the cabinet that has the longest delay time. If you sub has 0.5 second delay and your upper bass has 0.2s, then mids 0.1s, delay the upper bass by 0.3s and the mids by 0.4s. Repeat the measurements and the impulse responses should now be quite close together in delay.

After that is the critical part, phase integration. This will be affected by your crossover alignment and slope choices. In fact you can choose them to assist the process. Measure the amplitude and phase of each cabinet at least an octave aside from their crossovers (might as well measure the lot tbh) and overlay the responses of adjacent bands. Unwrap the phase and try to figure out what constant phase delay works best to align the responses on top of each other as best as possible. If it is possible to subtract one phase from the other then even better, fit a straight line to the data and make the adjustment. Keep doing this for pairs of cabinets and fixing one at a time as you move up or down in frequency. Keep measuring and checking. Bear in mind that this approach is going to be very affected by room modes etc and the results are only really valid for the one mic measurement location. But once you have your initial settings dialled in, there is nothing to stop you moving the mic around and taking more readings .

Your ultimate goal is to take a transfer function measurement of all cabs acting together via your LMS such that they integrate and appear as one unit with one smooth phase and SPL response. Any SPL drop outs or abrupt phase shifts around a crossover point are an indication that there is still work to do. It is very much a iterative process. Judicious use of your LMS band disable options are highly useful when measuring cabinets solo and together.
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ceharden View Drop Down
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Post Options Post Options   Thanks (1) Thanks(1)   Quote ceharden Quote  Post ReplyReply Direct Link To This Post Posted: 11 September 2017 at 9:17pm
Personally I prefer to use Holm Impulse (http://www.holmacoustics.com/holmimpulse.php) for doing time alignment jobs.  REW is good but better for frequency domain analysis in my opinion.
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odc04r View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote odc04r Quote  Post ReplyReply Direct Link To This Post Posted: 12 September 2017 at 10:20am
A few people say the same thing. HolmImpulse for time alignment, REW for phase. Clue's in the name possibly... I still have yet to give HI a whirl.
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bob4 View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote bob4 Quote  Post ReplyReply Direct Link To This Post Posted: 12 September 2017 at 12:24pm
APL TDA is nice for time aligning too….
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Post Options Post Options   Thanks (0) Thanks(0)   Quote ceharden Quote  Post ReplyReply Direct Link To This Post Posted: 12 September 2017 at 2:44pm
Originally posted by odc04r odc04r wrote:

A few people say the same thing. HolmImpulse for time alignment, REW for phase. Clue's in the name possibly... I still have yet to give HI a whirl.

It's very powerful but doesn't have a 'non-expert' mode!  There are a few tricks you need to know such as locking the auto peak-detection when doing time alignment of multiple measurements.
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Post Options Post Options   Thanks (0) Thanks(0)   Quote Timo Beckman Quote  Post ReplyReply Direct Link To This Post Posted: 21 November 2017 at 10:00am
Hi Matty

I do not work with REW or any analyzers mentioned in the above comments. I work with just a couple of other analyzers (Smaart/SIM3/SAT-live).
So getting drivers working together with Impuls response measurements would not be the way to go for me.
I would synchronize on the high driver/Mid-high's (no sub)  and after that start phase aligning all at the selected x-overs.

The reason for me to look at the phase response and not at the IR is simple:
The IR (lin so not LOG or ETC) wil give me a time on where the analyzer has the most data so that would be the Mid-high's.

If i implement a High-pass filter on a high driver there's a lot of data left for the analyzer to work with but as soon as i implement a low-pass filter on a low driver the data will be diminished substantially (from a linear point of view) and by implementing a low-pass filter the timing of frequencies will spread over time...(more group delay so the phase trace of the low driver will move down under a steeper angle).
In the screen capture you can see the difference between the IR (lin view) on the High-pass and Low-pass filters implemented (x-vr @1K2).

From about 2 min in to the screen capture i mess with different High and low pass filters and you can clearly see what those filters do to the IR. Timing doesn't change when implementing different filter orders when implementing High-pass filter but the IR moves to a different time when implementing a Low-pass filter. Also the IR get's time stretched and in linear view you see the level dropping.

I do not know how REW goes about FFT sizes but with Smaart you have a bunch of choices on FFT size.
If you set Smaart to a FFT size of 128 the frequency resolution of Smaart will be 375Hz steps. The analyzer will not see data points below 375Hz and will only see multiples of 375Hz (so next points would be 750Hz/1125Hz/1500Hz and so on).

You can calculate these steps by deviding sample rate/FFT size=48000/128=375Hz
If i set Smaart to a 32K FFT size you'll get 48000/32768=1.46Hz
So at a 32K FFT size Smaart will have a frequency resolution in 1.46Hz steps from 1.46Hz up to 24kHz.
(it's a lot of data ;-) )
Since a FFT analyzer is a linear device presenting data in a Logarithmic fashion you could say that all of those data point sum up to the value of 1 in a (live) IR measurement.

As soon as i implement a low-pass filter at 12kHz the data will be half of the data without the low-pass filter. So 1/2 in linear means -6dB in logarithmic (20log(½)=-6dB

As soon as i implement a low-pass filter at 6kHz the data will be half of the data without the low-pass filter. So 1/4 in linear means -12dB in logarithmic (20log(1/4)=-12dB

As soon as i implement a low-pass filter at 3kHz the data will be half of the data without the low-pass filter. So 1/8 in linear means -18dB in logarithmic (20log(1/8)=-18dB

As soon as i implement a low-pass filter at 1k5 the data will be half of the data without the low-pass filter. So 1/16 in linear means -24dB in logarithmic (20log(1/16)=-24dB

In short by implementing a Low-pass filter you'll reduce the number of data points for the analyzer to work with and the data will be time stretched (group delay on the driver + group delay from the low-pass filter). Still the analyzer is able to give you a time but it's in the area where it has the most data points: the upper range where the x-over is implemented and in that area phase will go horizontal.

Since you want to sum both drivers together to get a full range system you would need to synchronize your analyzer on 1 of the 2 drivers (preferably the high driver = more data for the analyzer to work with) and keep the time set at that point. After that get the phase angles on both drivers to match in there x-over range: i.e. Phase align them both in the x-vr.

Phase aligning both can be done by using delay or my preferred way All-pass filters. Another way would be to modify the IR trough a FIR filter if you do not mind more latency.....

Small note: response on the system in the screen cap is sort of flat. If this system with these settings would be used in a live situation you would probably need to temper the mid-highs a bit if pushed to a 100dBa or more.....(something to do with our ears perceiving this as kind of harsh sounding MH's) ;-)



Edited by Timo Beckman - 21 November 2017 at 11:09am
The sound will be as good as the band play's
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toastyghost View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote toastyghost Quote  Post ReplyReply Direct Link To This Post Posted: 21 November 2017 at 10:52am
Originally posted by bob4 bob4 wrote:

APL TDA is nice for time aligning too….


Can you explain why? Every rig I've heard aligned with it was well, shit.
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