Room correction options that wont ruin the sound of tube amp?

[marked sound]

To break it down very simply, it doesn't matter which amplifier you use, provided the power is sufficient even with DSP.
A test file is played, which is recorded at the end, i.e. after the amplifier and loudspeaker. This recording is compared with the original, evaluated and a filter/mask is designed that ensures that the test file and playback no longer differ (ideally).
Of course, this only works within a certain framework that the DSP can map and within the technical specifications of the amplifier, loudspeaker, etc.

Purely fictitious as an example!!! if a device can technically only play down to 30 Hz and up to a maximum of 17 kHz, then these technical limits cannot be shifted with a DSP.

So if I am to understand correctly, the chain goes like this...

#1. Source (Streamer/DAC, cd player etc),
#2. Room correction device with mic.
#3. Amplifier.
#4. Speakers.

Then if I am to further understand correctly, if I do use a PC with room correction software, or use come other external device that does room correction (miniDSP, McIntosh MEN220, etc), what happens is this...

#1. The room correction device has the proper measurements of what is ideal stored in its memory.
#2. A tone is played by the room correction device through the amplifier and speakers.
#3. That tone is then picked up by the microphone, that is hooked up to the room correction device.
#4. The room correction device then compares the measurements it received to the ideal measurements.
#5. The room correction device adjusts what info is being fed to the amp accordingly, so that the sound being outputted through the speakers matches ideal measurements.

Do I get the gist of what is going on?

Thanks

 

[JustJones]

So if I am to understand correctly, the chain goes like this...

#1. Source (Streamer/DAC, cd player etc),
#2. Room correction device with mic.
#3. Amplifier.
#4. Speakers.

Then if I am to further understand correctly, if I do use a PC with room correction software, or use come other external device that does room correction (miniDSP, McIntosh MEN220, etc), what happens is this...

#1. The room correction device has the proper measurements of what is ideal stored in its memory.
#2. A tone is played by the room correction device through the amplifier and speakers.
#3. That tone is then picked up by the microphone, that is hooked up to the room correction device.
#4. The room correction device then compares the measurements it received to the ideal measurements.
#5. The room correction device adjusts what info is being fed to the amp accordingly, so that the sound being outputted through the speakers matches ideal measurements.

Do I get the gist of what is going on?

Thanks

No

 

[MaxwellsEq]

So if I am to understand correctly, the chain goes like this...

#1. Source (Streamer/DAC, cd player etc),
#2. Room correction device with mic.
#3. Amplifier.
#4. Speakers.

Then if I am to further understand correctly, if I do use a PC with room correction software, or use come other external device that does room correction (miniDSP, McIntosh MEN220, etc), what happens is this...

#1. The room correction device has the proper measurements of what is ideal stored in its memory.
#2. A tone is played by the room correction device through the amplifier and speakers.
#3. That tone is then picked up by the microphone, that is hooked up to the room correction device.
#4. The room correction device then compares the measurements it received to the ideal measurements.
#5. The room correction device adjusts what info is being fed to the amp accordingly, so that the sound being outputted through the speakers matches ideal measurements.

Do I get the gist of what is going on?

Thanks

Mostly wrong, I'm afraid. Your key mistake is to think that the measurement is done in real-time. Assuming you get the results right, you do the measurements once only and reap the benefit from then on.

Let's use the simplest example. A laptop with Foobar2000 and EAPO installed. You get the results of your scan - let's say you have a room mode at 150Hz - so you put a filter into EAPO reducing gain at 150Hz by 3dB. That's it! Now you play your music on Foobar2000 through EAPO into a DAC and then into a power amplifier. From now on all your music has less energy at 150 Hz and the bass sounds less boomy.

 

[Galliardist]

So if I am to understand correctly, the chain goes like this...

#1. Source (Streamer/DAC, cd player etc),
#2. Room correction device with mic.
#3. Amplifier.
#4. Speakers.

Then if I am to further understand correctly, if I do use a PC with room correction software, or use come other external device that does room correction (miniDSP, McIntosh MEN220, etc), what happens is this...

#1. The room correction device has the proper measurements of what is ideal stored in its memory.
#2. A tone is played by the room correction device through the amplifier and speakers.
#3. That tone is then picked up by the microphone, that is hooked up to the room correction device.
#4. The room correction device then compares the measurements it received to the ideal measurements.
#5. The room correction device adjusts what info is being fed to the amp accordingly, so that the sound being outputted through the speakers matches ideal measurements.

Do I get the gist of what is going on?

Thanks

Better:
The chain is conceptually as you point out, but we should include 5 - the room, and 6- the listener/s.

The room correction system can never reach an "ideal". It can at best ameliorate faults in the in room response. It's really a good idea to do everything else that you can first, before using DSP. giving the best chance for it to improve things further. Ideally you want a really good setup and room first, before you apply correction.

Measurements are performed with a series of different signals, and systems that take a range of different measurements with the microphone in different positions will usually give a better result.

The measurements are used to produce a profile or setting, a collection of filters that is used to modify all music when you play it. You only need to take measurements again if something significant in the room or system changes, or if you get a poor result. You can also modify those filters to sound closer to your preferred sound, and use broadband tone controls to modify for individual recordings.

DSP is a tool, not a panacea. My system is not ideally set up: no amount of EQ is going to make it sound as good as if it were properly set up in a dedicated room. Ever.

 

[marked sound]

Better:
The chain is conceptually as you point out, but we should include 5 - the room, and 6- the listener/s.

The room correction system can never reach an "ideal". It can at best ameliorate faults in the in room response. It's really a good idea to do everything else that you can first, before using DSP. giving the best chance for it to improve things further. Ideally you want a really good setup and room first, before you apply correction.

Measurements are performed with a series of different signals, and systems that take a range of different measurements with the microphone in different positions will usually give a better result.

The measurements are used to produce a profile or setting, a collection of filters that is used to modify all music when you play it. You only need to take measurements again if something significant in the room or system changes, or if you get a poor result. You can also modify those filters to sound closer to your preferred sound, and use broadband tone controls to modify for individual recordings.

DSP is a tool, not a panacea. My system is not ideally set up: no amount of EQ is going to make it sound as good as if it were properly set up in a dedicated room. Ever.

I have had amps with room correction before, and I remember having to position the mic in a few different places for the amp to perform a test. However, I have never added external room correction to an amp before and that is what sort of confuses me.

Am I at least now correct in thinking that... #1. the external room correction device (with the use of the mic), develops a profile based on the tone it passes through the amp and speakers and how that tone reacts with the room? And from then on... #2. That created profile edits the sound before it reaches the amp, then speakers, then room, then listener?

 

[MaxwellsEq]

Am I at least now correct in thinking that... #1. the external room correction device (with the use of the mic), develops a profile based on the tone it passes through the amp and speakers and how that tone reacts with the room? And from then on... #2. That created profile edits the sound before it reaches the amp, then speakers, then room, then listener?

Yes that's pretty close. You run a bunch of measurements using sweeps, tones and white/pink noise to get a sense of your loudspeakers and their room interaction. You then STORE an INVERSE of the measured peaks and troughs in a DSP device (such as DAC with DSP, a laptop with DSP, a DSP-only device, an amplifier with DSP). When you play music, the DSP, in real time, "edits" the music to reduce certain frequencies in the music which would be incorrectly loud with your speakers in your room.

 

[marked sound]

Yes that's pretty close. You run a bunch of measurements using sweeps, tones and white/pink noise to get a sense of your loudspeakers and their room interaction. You then STORE an INVERSE of the measured peaks and troughs in a DSP device (such as DAC with DSP, a laptop with DSP, a DSP-only device, an amplifier with DSP). When you play music, the DSP, in real time, "edits" the music to reduce certain frequencies in the music which would be incorrectly loud with your speakers in your room.

I DAC with I am getting has built in EQ and allows you to store profiles and use created with REW.

 

[marked sound]

Better:
The chain is conceptually as you point out, but we should include 5 - the room, and 6- the listener/s.

The room correction system can never reach an "ideal". It can at best ameliorate faults in the in room response. It's really a good idea to do everything else that you can first, before using DSP. giving the best chance for it to improve things further. Ideally you want a really good setup and room first, before you apply correction.

Measurements are performed with a series of different signals, and systems that take a range of different measurements with the microphone in different positions will usually give a better result.

The measurements are used to produce a profile or setting, a collection of filters that is used to modify all music when you play it. You only need to take measurements again if something significant in the room or system changes, or if you get a poor result. You can also modify those filters to sound closer to your preferred sound, and use broadband tone controls to modify for individual recordings.

DSP is a tool, not a panacea. My system is not ideally set up: no amount of EQ is going to make it sound as good as if it were properly set up in a dedicated room. Ever.

The DAC I want to get has EQ settings and allows you to upload, store, and use, profiles created with REW.

 

[Roland68]

I have had amps with room correction before, and I remember having to position the mic in a few different places for the amp to perform a test. However, I have never added external room correction to an amp before and that is what sort of confuses me.

Am I at least now correct in thinking that... #1. the external room correction device (with the use of the mic), develops a profile based on the tone it passes through the amp and speakers and how that tone reacts with the room? And from then on... #2. That created profile edits the sound before it reaches the amp, then speakers, then room, then listener?

You're making it far too difficult for yourself, it's actually always the same thing that happens. It doesn't matter whether the DSP area is integrated in the amplifier, or in a streamer like Wiim, or on a PC.
An AVR today consists of, for example:
Input/streamer -> DSP -> DAC -> preamplifier/volume control -> power amplifiers with speaker connections.
An AVR is just a device that is made up of several parts. Why should anything change if all the parts are individual devices?
With an amplifier with DSP you have: DSP -> DAC -> preamplifier/volume control -> power amplifiers with speaker connections
With a pure DSP device: DSP -> DAC -> (preamplifier/volume control)

You can imagine the basic processing like your eyes and an optician. You see blurry (uncorrected music signal), the optician measures your eyes and tests how sharp different lenses are (calibration process), the optician makes your glasses (filter), you put the glasses on and see clearly (filter is applied in real time = corrected music signal)

 

[marked sound]

You're making it far too difficult for yourself, it's actually always the same thing that happens. It doesn't matter whether the DSP area is integrated in the amplifier, or in a streamer like Wiim, or on a PC.
An AVR today consists of, for example:
Input/streamer -> DSP -> DAC -> preamplifier/volume control -> power amplifiers with speaker connections.
An AVR is just a device that is made up of several parts. Why should anything change if all the parts are individual devices?
With an amplifier with DSP you have: DSP -> DAC -> preamplifier/volume control -> power amplifiers with speaker connections
With a pure DSP device: DSP -> DAC -> (preamplifier/volume control)

You can imagine the basic processing like your eyes and an optician. You see blurry (uncorrected music signal), the optician measures your eyes and tests how sharp different lenses are (calibration process), the optician makes your glasses (filter), you put the glasses on and see clearly (filter is applied in real time = corrected music signal)

I get it now. I actually watched a video from about the Streamer/DAC I want to buy and how easy it is to implement room correction via computer and REW.

 

[Roland68]

I get it now. I actually watched a video from about the Streamer/DAC I want to buy and how easy it is to implement room correction via computer and REW.

Then I would advise you to take a look at Acourate, in my opinion one of the most powerful solutions, extremely customizable and scalable. REW not necessary.
You need the software, a convolver, a small powerful PC, a calibrated measurement microphone and a DAC or sound card for the output. Expansion such as subwoofers, partially or fully active speakers are no problem.
Just take your time and get to grips with it.
Acourate
Acourate FAQ
I was visiting the Pure Acourate Sound Project in Germany last month
THE PURE ACOURATE SOUND PROJECT. The project is supported by Klippel, among others.

 

[marked sound]

Then I would advise you to take a look at Acourate, in my opinion one of the most powerful solutions, extremely customizable and scalable. REW not necessary.
You need the software, a convolver, a small powerful PC, a calibrated measurement microphone and a DAC or sound card for the output. Expansion such as subwoofers, partially or fully active speakers are no problem.
Just take your time and get to grips with it.
Acourate
Acourate FAQ
I was visiting the Pure Acourate Sound Project in Germany last month
THE PURE ACOURATE SOUND PROJECT. The project is supported by Klippel, among others.

Rew works really good with the DAC I want to buy.

 

[marked sound]

For me there would be a very simple but very serious reason not to buy it.
I don't want to have anything from BOSE at home

Now that Bose bought McIntosh, I wouldn't be surprised if you see McIntosh as discount electronic stores and places like Best Buy.

 

[ivanj]

Now that Bose bought McIntosh, I wouldn't be surprised if you see McIntosh as discount electronic stores and places like Best Buy.

It's already in Bug Buy/Magnolia.