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My first REW measurements(KEF R3 META)

staticV3

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But then again, I started thinking that if I want to integrate my subs I will need to to use a software, which will again undermine the bit perfect playback. Is this true? If I want to use wasapi exclusive mode I will need to forget any kind of intervention?
You can use a miniDSP 2x4 HD/DDRC-24/Flex to get bit-perfect audio transmission and still apply room correction and sub integration.
 

Keith_W

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I wouldn't worry about bit perfect playback if I were you. What the room does to your audio is far more important and it is what needs to be corrected.

The number 1 cause of unsatisfactory DSP correction is improper corrections made by the user. Subwoofers need bass management. In fact all systems need some form of bass management. Bass management is more than cutting peaks, you also need to align phase, time align, and much more.

At this point I encourage you to persist with DSP, but you will need to choose a DSP program. Some of them are "one button DSP" where a software algorithm decides on a correction for you. Other software have minimal automation, or no automation at all. This gives you finer control, but with a steep learning curve (and also the potential to make many mistakes along the way). I am skeptical of how accurate software automation can be compared to the manual approach. I have seen some good results, and some bad results. REW + RePhase are examples of software with no automation. If you go beyond its normal measurement functions and start considering using it for correction, you will quickly hit a brick wall and you will need a lot of guidance and reading. But trust me, it's worth it. You learn so much. You need to decide whether you want to go the easy route or take a more manual approach.

To get you started, here is a list of common software packages used by many members of ASR with a very brief description. Remember: more automation = easier, but also more dependent on software algorithm and less flexible. Less automation = steeper learning curve, but more user control.

- Dirac: a lot of automation, minimal user controls. Expensive.
- Focus Fidelity: new software, less automation than Dirac, user can over-ride algorithm. Inexpensive.
- Audiolense: more complete than Focus Fidelity, more advanced functions. Has some automation. User can over-ride algorithm. Inexpensive.
- Acourate: very minimal automation, more manual approach, very powerful. Extremely flexible. Inexpensive.
- REW + Rephase: zero automation, very manual, surprisingly powerful. Extremely flexible. Free.
- REW + MSO: some automation (software does the analysis for you) but the rest is manual. For bass only. MSO has a learning curve. Free.
- Audioweaver, Matlab, Octave: forget it unless you are an engineer or a maths nerd.

You could also use Room Shaper. This is a VST (meaning it is installed in your signal path) and it is almost plug and play. Corrects bass up to mid frequencies only.

Do some research, download a trial, and have a play.
 
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kolestonin

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You can use a miniDSP 2x4 HD/DDRC-24/Flex to get bit-perfect audio transmission and still apply room correction and sub integration.
I gave a look and I would say that DDRC24 and Flex are my best options given they come with a Dirac module.

And let's suppose that Flex would be my choice.
My current chain is pc—> DAC —> preamp —>power amp

Do I have the options to?

a) keep all my current gear and insert the Flex in the chain? If yes, where in the chain?
b) replace my current DAC with Flex and use it in DAC mode with my current preamp?
c) completely remove my current DAC and preamp, and use Flex as DAC and preamp(volume controller)?

I need to say, that this last scenario is the most interesting.
Not only cause I am a big fan of simplicity, but also cause I will be able to sell my current equipment and cover the cost of the Flex. So I can give the money I was intended to spend for buing Flex in order to buy a Dirac license.
You need to know I use my Topping pre90 purely as a volume controller and not as a source selector cause I only have one source. But my understanding is that a big benefit of using a preamp as volume controller is it's analogue volume control. Cause a device with digital volume control require full volume in order to get the best out of it. And below maximum volume it's ability to expose detail is reduced cause SINAD and dynamic range will be reduced. If that's the case with c scenario, I will need to stick with a or b cause I am not willing to sacrifice absolutely nothing in terms of sound of quality no matter the simplicity or money gain.
I wouldn't worry about bit perfect playback if I were you. What the room does to your audio is far more important and it is what needs to be corrected.

The number 1 cause of unsatisfactory DSP correction is improper corrections made by the user. Subwoofers need bass management. In fact all systems need some form of bass management. Bass management is more than cutting peaks, you also need to align phase, time align, and much more.

At this point I encourage you to persist with DSP, but you will need to choose a DSP program. Some of them are "one button DSP" where a software algorithm decides on a correction for you. Other software have minimal automation, or no automation at all. This gives you finer control, but with a steep learning curve (and also the potential to make many mistakes along the way). I am skeptical of how accurate software automation can be compared to the manual approach. I have seen some good results, and some bad results. REW + RePhase are examples of software with no automation. If you go beyond its normal measurement functions and start considering using it for correction, you will quickly hit a brick wall and you will need a lot of guidance and reading. But trust me, it's worth it. You learn so much. You need to decide whether you want to go the easy route or take a more manual approach.

To get you started, here is a list of common software packages used by many members of ASR with a very brief description. Remember: more automation = easier, but also more dependent on software algorithm and less flexible. Less automation = steeper learning curve, but more user control.

- Dirac: a lot of automation, minimal user controls. Expensive.
- Focus Fidelity: new software, less automation than Dirac, user can over-ride algorithm. Inexpensive.
- Audiolense: more complete than Focus Fidelity, more advanced functions. Has some automation. User can over-ride algorithm. Inexpensive.
- Acourate: very minimal automation, more manual approach, very powerful. Extremely flexible. Inexpensive.
- REW + Rephase: zero automation, very manual, surprisingly powerful. Extremely flexible. Free.
- REW + MSO: some automation (software does the analysis for you) but the rest is manual. For bass only. MSO has a learning curve. Free.
- Audioweaver, Matlab, Octave: forget it unless you are an engineer or a maths nerd.

You could also use Room Shaper. This is a VST (meaning it is installed in your signal path) and it is almost plug and play. Corrects bass up to mid frequencies only.

Do some research, download a trial, and have a play.
Can't thank you enough for that!
Gathering all possible eq solutions with a brief explanation on each one of them is almost sticky material.
Very helpful.
 

staticV3

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keep all my current gear and insert the Flex in the chain? If yes, where in the chain?
PC->Flex Digital->DAC->Preamp->Power Amp

But really, PC->Flex->Power Amp would make a whole lot more sense.

Use the Flex as DSP, DAC, and Preamp.

But my understanding is that a big benefit of using a preamp as volume controller is it's analogue volume control. Cause a device with digital volume control require full volume in order to get the best out of it. And below maximum volume it's ability to expose detail is reduced cause SINAD and dynamic range will be reduced. If that's the case with c scenario, I will need to stick with a or b cause I am not willing to sacrifice absolutely nothing in terms of sound of quality no matter the simplicity or money gain.
That's a flawed understanding of digital vs analog volume control.

Unless you can actually hear the DAC's noise floor through your speakers at your MLP, which is highly unlikely with modern electronics, nothing is lost compared to an analog preamp in terms of detail retrieval and dynamic range.
 

dogmamann

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Finally found the time to take some REW measurements.

I made 3 measurements in the main listening position.
Red=Left speaker only
Green=Right speaker
Blue=Both speakers

Sharing only the screenshots.
mdat files are available and can be uploaded if you think is necessary.

I just wasn't to be sure my REW settings are correct before moving forward, taking more measurements, sharing them on line etc.

That's why I also added 2 screenshots with my soundcard and measure settings.

Any comment, help, suggestion will be highly appreciated.
That’s a wobbly response right there, which makes everyone wonder why the speakers costs so much, and why would anyone prefer this over r11.
 

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kolestonin

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That’s a wobbly response right there, which makes everyone wonder why the speakers costs so much, and why would anyone prefer this over r11.
I think it's my (difficult)room to blame and not the speakers.

But your clearly negative comment on my curves is very interesting to me.
As I understand you see something that is undoubtedly bad and I would be interested to know what is this and where I should focus my efforts to improve it.
This the purpose of this thread after all as I am not capable to read and interpret my measurements.
Like a patient showing his blood tests to a doctor and asking for the best cure:)
 

Keith_W

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I think it's my (difficult)room to blame and not the speakers.

But your clearly negative comment on my curves is very interesting to me.
As I understand you see something that is undoubtedly bad and I would be interested to know what is this and where I should focus my efforts to improve it.
This the purpose of this thread after all as I am not capable to read and interpret my measurements.
Like a patient showing his blood tests to a doctor and asking for the best cure:)

Ignore him. You are right, it is your room that is causing the issue and not the speakers. That wobbly curve is normal. And you might want to read what Floyd Toole said about wobbly curves - link. In particular, pay attention to this: "Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain."

You can easily prove this to yourself. Place your mic where your left ear would be and do a sweep. Then move it to where the other ear would be and repeat. Apply 1/24 smoothing and overlay the two curves. What you will see: bass remains almost the same. But the wobbles are vastly different at high freqs. Fortunately this is not what you hear, it has to do with Equivalent Rectangular Bandwidth (don't bother learning about it in detail, just be aware that it exists). A sweep from an omnidirectional microphone in a single point in space does not reflect reality. You have to be aware of its limitations and what information you can derive from it. You can derive phase and timing information from a single sweep and information about bass. But it should not be used to correct treble freqs at high resolution.

What I would suggest is to use 1/3 smoothing and look at the high freqs. 1/3 smoothing better reflects what we hear. If you see broad wobbles in the FR, you might want to fix it because it will affect the tone - or as Toole says, "broadband, low Q, tone controls".
 

dogmamann

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Ignore him. You are right, it is your room that is causing the issue and not the speakers. That wobbly curve is normal. And you might want to read what Floyd Toole said about wobbly curves - link. In particular, pay attention to this: "Don't worry about little ripples. When I see exceptionally smooth high-resolution room curves I strongly suspect that something wrong has been done. The measurement microphone is no substitute for two ears and a human brain."

You can easily prove this to yourself. Place your mic where your left ear would be and do a sweep. Then move it to where the other ear would be and repeat. Apply 1/24 smoothing and overlay the two curves. What you will see: bass remains almost the same. But the wobbles are vastly different at high freqs. Fortunately this is not what you hear, it has to do with Equivalent Rectangular Bandwidth (don't bother learning about it in detail, just be aware that it exists). A sweep from an omnidirectional microphone in a single point in space does not reflect reality. You have to be aware of its limitations and what information you can derive from it. You can derive phase and timing information from a single sweep and information about bass. But it should not be used to correct treble freqs at high resolution.

What I would suggest is to use 1/3 smoothing and look at the high freqs. 1/3 smoothing better reflects what we hear. If you see broad wobbles in the FR, you might want to fix it because it will affect the tone - or as Toole says, "broadband, low Q, tone controls".
What’s the point in then buying the reference 3 instead of r11 if both are going to have ripples in a real room?
 

staticV3

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What’s the point in then buying the reference 3 instead of r11 if both are going to have ripples in a real room?
The R11 Meta has deeper bass extension and less THD than the R3 Meta.

The R3 Meta costs less and is more compact.
 

dogmamann

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Ah my bad.
Compared to the R11 Meta, the Reference 3 Meta has better controlled vertical directivity and less THD.
Measurements says r11 has lower distortion. Please check it. Can you post the vertical directivity comparison here?
 
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