HydrogenAudio

MathAudio Room EQ for Foobar2000
(http://mathaudio.com/images/mathaudio-room-eq.png)

• Corrects deficiencies of room acoustics (multipoint compensation).
• Corrects acoustic imperfections of speakers.
• Optimized for maximum accuracy of the correction.
• Corrects both amplitude and phase components of frequency response.
• Quells resonance peaks of frequency response while leaving the deep notches to prevent large excursions of the speaker diaphragm. Avoids the muddy sound of conventional linearizing equalizers.
• Manually adjustable level of compensation allows one to reach the maximum transparency of the sound.
• Supports high sample rates.
• Applies 64-bit signal path throughout.
• Works with USB measurement microphones (e.g. MiniDSP UMIK-1) or standard measurement microphones (e.g. NADY CM100 or Dayton Audio EMM-6).
• Supports microphone calibration files.
• Applies a patented method of frequency response correction.

Download: mathaudio.com/download.htm (http://mathaudio.com/download.htm)

This looks magnificent.  I looked into room EQ some years ago, but the whole process was so complicated, I never got round to it.  I did buy a Radioshack Sound Level Meter for this purpose though.  Any chance I could use it with this plugin (or a future version)?

Why is the plug-in an .exe? No thanks.

The plug-in is packed into an 'exe' installer because it includes a Help manual. The  package also includes a standard Windows uninstaller, so you can easily remove all installed files from your PC.

This looks magnificent.  I looked into room EQ some years ago, but the whole process was so complicated, I never got round to it.  I did buy a Radioshack Sound Level Meter for this purpose though.  Any chance I could use it with this plugin (or a future version)?

Yes, you can use it as a measurement microphone with Room EQ. Try to find a calibration (correction) file for your model and copy it to the Documents -> MathAudio Room EQ folder. It will improve the accuracy of your measurement.

Why is the plug-in an .exe? No thanks.

You can extract it with 7-Zip (or 7-Zip Portable (http://portableapps.com/apps/utilities/7-zip_portable)) and use just the DLL.

Avast AV detected uninstall.exe as malware.

From the icon, it appears to be a bog standard NSIS uninstaller. From what I've found from basic searching, to sign the uninstaller, you need to compile your own uninstaller and sign it, then bundle that with the installer instead of scripting it to generate one on the fly.

https://www.virustotal.com/en/file/d34a0b13...sis/1385536852/ (https://www.virustotal.com/en/file/d34a0b13f2f6be6e7645eecf7eb67bac9f083467cb8e6107fea1001f367d3860/analysis/1385536852/)

I'm calling false positive due to unsigned code.

Also, just extracting the component will not work, because it appears to be nothing more than a VST wrapper, as the installer also places the equalizer VST in a separate directory, alongside the documentation and the uninstaller.

Avast AV detected uninstall.exe as malware.

uninstall.exe doesn't contain any malware. This is a false positive error of AVAST.

I got the plugin installed and working.

I did some measurements and got a reasonable looking frequency response.

Unfortunately, I do not hear any difference between the Bypass and Room EQ positions (both with music and sinusoidal tones) 


 

You can extract it with 7-Zip (or 7-Zip Portable (http://portableapps.com/apps/utilities/7-zip_portable)) and use just the DLL.

This method will not work because foo_room_eq.dll depends on other Room EQ files that are located in another directory. It was not practical to install all Room EQ files into the "components" directory of Foobar2000 because it could create some delay on startup of Foobar2000 (Foobar2000 checks all files in its "components" directory one-by-one every time when you start it up).

Room EQ applies well known and very reliable NSIS installer/uninstaller which is used in thousands of software products around the world. Just use it and don't worry: Room EQ does not contain any malware.

I got the plugin installed and working.

I did some measurements and got a reasonable looking frequency response.

Unfortunately, I do not hear any difference between the Bypass and Room EQ positions (both with music and sinusoidal tones)

It is impossible. Play music and move the vertical slider up and down in the Room EQ mode. You will hear noticeable changes in timbre and volume of the sound.

Quote from: oruam57 on 2013-11-27 18:35:58

I got the plugin installed and working.  I did some measurements and got a reasonable looking frequency response.  Unfortunately, I do not hear any difference between the Bypass and Room EQ positions (both with music and sinusoidal tones)

It is impossible. Play music and move the vertical slider up and down in the Room EQ mode. You will hear noticeable changes in timbre and volume of the sound.

 

Sorry, I misunderstood the function of the vertical slider. Now I see that the frequency response is corrected for the part above the white line only.

Sorry, I misunderstood the function of the vertical slider. Now I see that the frequency response is corrected for the part above the white line only.

Yes, Room EQ removes everything above the dashed line while leaving the notches below the dashed line. This is one of the most important features of this plug-in. Conventional room correction systems often try to “equalize everything”. As a result, they create large excursions of the speaker diaphragm at certain frequencies. The large excursions of the speaker diaphragm create non-linear distortion and the sound becomes “dirty”. The vertical slider of this plug-in allows you to choose between “full” and “partial” types of correction.  Consequently, you can get the best possible sound. 

Please note that the sound of your speakers becomes more silent when you remove the room resonances. Don't forget to move left the “Bypass signal volume” slider to make the volume of the bypass signal equal to the volume of the processed signal. After that you will be able to correctly compare the quality of the processed sound with the quality of the original sound.

News
A new version of the plug-in has just been released. The new version includes a custom target curve feature. This new feature can be used for drawing your own reference frequency response.

News
A new version of the plug-in has just been released. The new version includes a custom target curve feature. This new feature can be used for drawing your own reference frequency response.

Hi k2k, fantastic work!

To top it off, maybe you could offer correction curves for a few cheap stock microphones (Best Buy, etc.). Would be more than willing to pay $1 a piece.
For example, here's a quite popular one: http://www.amazon.com/Logitech-980240-0403.../dp/B00022TNHM/ (http://www.amazon.com/Logitech-980240-0403-Desktop-Microphone/dp/B00022TNHM/)

Hello, I have a Behringer ECM8000 microphone. I let it calibrate and got a correction file like these (http://lasip.hifi-selbstbau.de/downloads/MicCalDemo.sen).
Can the software read ASCII (text) microphone correction files with file extensions like .sen/.mic/.cal etc. ?

Hi k2k, fantastic work!

To top it off, maybe you could offer correction curves for a few cheap stock microphones (Best Buy, etc.). Would be more than willing to pay $1 a piece.
For example, here's a quite popular one: http://www.amazon.com/Logitech-980240-0403.../dp/B00022TNHM/ (http://www.amazon.com/Logitech-980240-0403-Desktop-Microphone/dp/B00022TNHM/)

Your example mic is not so cheap. Just compare its price ($35) with the price of a professional calibrated measurement microphone like EMM-6 (http://www.parts-express.com/pe/showdetl.cfm?partnumber=390-801) ($48) or UMIK-1 (http://www.minidsp.com/products/acoustic-measurement/umik-1) ($75).

Hello, I have a Behringer ECM8000 microphone. I let it calibrate and got a correction file like these (http://lasip.hifi-selbstbau.de/downloads/MicCalDemo.sen).
Can the software read ASCII (text) microphone correction files with file extensions like .sen/.mic/.cal etc. ?

Yes, Room EQ can read ASCII (text) microphone calibration files if you save them with the extension .CAL or .FRD.

Use "Save As" in your browser to save the text of the calibration file as a text file (e.g. calibration.txt). After that change the file type from “txt” to “cal”: open the text file in Notepad, click File – Save as -  Save as type – All Files, then type the new name (e.g. calibration.cal) in the “File name” edit window and click the “Save” button.

Copy the calibration file to the Documents -> MathAudio Room EQ folder. Then exit Foobar2000 and start it once again. The name of the calibration file will be shown at the top right corner of the Room EQ graphical area.

Hm. Anyone know whether the IMM-6 ("I", not "E") is good for anything? Small and practical to bring around ... and when a few friends have borrowed it I can just buy another one.
http://www.amazon.com/Dayton-Audio-iMM-6-C...t/dp/B00ADR2B84 (http://www.amazon.com/Dayton-Audio-iMM-6-Calibrated-Measurement/dp/B00ADR2B84)

HELLO,
Room correction mode, foobar when playing a song, I hear a beep every 5 to 10 seconds.
Es it possible
Hi

Hm. Anyone know whether the IMM-6 ("I", not "E") is good for anything? Small and practical to bring around ... and when a few friends have borrowed it I can just buy another one.
http://www.amazon.com/Dayton-Audio-iMM-6-C...t/dp/B00ADR2B84 (http://www.amazon.com/Dayton-Audio-iMM-6-Calibrated-Measurement/dp/B00ADR2B84)

This mic is designed to be used on iPhones and Android phones, not on a PC. If you want to use it with your PC, you will need to use an adapter cable to go from TRRS (4 wires) to TRS (3 wires). Besides, you will need to use the Mic Input of your PC, which can be less accurate than the Line Input. You may want to look at Dayton Audio UMM-6 (http://www.parts-express.com/dayton-audio-umm-6-usb-measurement-microphone--390-808) or MiniDSP UMIK-1 (http://www.minidsp.com/products/acoustic-measurement/umik-1) instead.

HELLO,
Room correction mode, foobar when playing a song, I hear a beep every 5 to 10 seconds.
Es it possible
Hi

There are two versions of the Room EQ plug-in: Room EQ for Foobar2000 and Room EQ VST. Room EQ for Foobar2000 is free and works without any beeps. Room EQ VST is designed to be used with professional audio workstations (DAW) and it is not free (demo beeps). You can use Room EQ VST with Foobar2000, but you don't need to do it because Room EQ for Foobar2000 includes all features of Room EQ VST. Just install Room EQ for Foobar2000 and you will have no beeps.

Quote from: eahm on 2013-11-27 05:20:25

You can extract it with 7-Zip (or 7-Zip Portable (http://portableapps.com/apps/utilities/7-zip_portable)) and use just the DLL.

This method will not work because foo_room_eq.dll depends on other Room EQ files that are located in another directory. It was not practical to install all Room EQ files into the "components" directory of Foobar2000 because it could create some delay on startup of Foobar2000 (Foobar2000 checks all files in its "components" directory one-by-one every time when you start it up).

Room EQ applies well known and very reliable NSIS installer/uninstaller which is used in thousands of software products around the world. Just use it and don't worry: Room EQ does not contain any malware.

It's not because of worries, I am sure I can trust you. I don't install anything (unless I really really need to), always make or use portable versions. This way I fully use ONLY one drive and keep the OS drive (SSD) clean and at the maximum of speed and efficiency and it's better for backup purposes, the folder/drive is just one.

It's not because of worries, I am sure I can trust you. I don't install anything (unless I really really need to), always make or use portable versions. This way I fully use ONLY one drive and keep the OS drive (SSD) clean and at the maximum of speed and efficiency and it's better for backup purposes, the folder/drive is just one.

MathAudio Room EQ is not able to slow down your system. However, if you want to keep everything under your control, you can use the following manual method of installation:
1. Download mathaudio-room-eq-for-foobar2000.exe from http://mathaudio.com/download.htm (http://mathaudio.com/download.htm)
2. Open it with 7-Zip (http://www.7-zip.org/).
3. Copy foo_room_eq.dll to the "Program FIles (x86)\foobar2000\components" folder (Windows x64) or "Program FIles\foobar2000\components" folder (Windows x32/x86).
4. Open the "$SHELL[17]" folder and copy the "MathAudio" folder with all its content to the "Program FIles (x86)"  directory (Windows x64) or "Program FIles" directory (Windows x32/x86).
Please note that you will need to manually delete all these files if you will want to uninstall the plug-in.

thank you for this plugin!

MathAudio Room EQ is not able to slow down your system.

Of course not, Windows is

It's really a choice more than the slowing down, with modern PCs takes much more than that to "slow down" a system. I ONLY use portable apps for 100% or control/management over the data, apps, backup.

Hi, this thing really improves the sound

Just a suggestion however, would it be possible to add an output option "DSP list", so that the test tone outputs through foobar and through any proceeding DSPs in the list? this would be cool for comparing.
Thanks!

Hi, this thing really improves the sound

Just a suggestion however, would it be possible to add an output option "DSP list", so that the test tone outputs through foobar and through any proceeding DSPs in the list? this would be cool for comparing.
Thanks!

The Room EQ measurement program sends the sweep sound directly to the sound card. If we would process the sweep sound with any DSP effects, we could get incorrect measurement results. However, you can use Room EQ with any other DSP effects during the playback of audio files. Just make sure that Room EQ is the last effect in your "Active DSPs" list.

Hi, thanks for response,

Well, I ask because in order for maximally flat response, I need to apply linkwitz transfrom to all signals getting sent to the subwoofer, including sweeps.

A sweep without linkwitz transfrom active, the sub does not portray flat response when being tested.

I could apply the transfrom to sound card eq, so that the sweep processes through that... but was curious if a 'dsp list' path could be added... 


Is it also possible to export the curve to use with winamp in some way?



Again thanks for this - REW really is complex to set up.

Hi, thanks for response,

Well, I ask because in order for maximally flat response, I need to apply linkwitz transfrom to all signals getting sent to the subwoofer, including sweeps.

A sweep without linkwitz transfrom active, the sub does not portray flat response when being tested.

I could apply the transfrom to sound card eq, so that the sweep processes through that... but was curious if a 'dsp list' path could be added... 


Is it also possible to export the curve to use with winamp in some way?



Again thanks for this - REW really is complex to set up.

There is no easy way to export the curve to Winamp.
Do you want to implement a digital Linkwitz transform?

hi and Yes,

hi and Yes,

The VST version of this plugin can help you to process the sweep sound if you need to. The demo version is enough to make the measurement and to save the plot as a preset. Then you can load the preset to the Foobar2000 version of the plugin. You can use the VST version of the Room EQ plugin with the demo version of Reaper or any other DAW. Reaper allows you to include any effects into the sweep sound pass.

easy, thanks!

Salut k2k

Thanks for working and sharing.

I have a similar requirement to giro1991 - I use a DSP crossover plugin in Foobar so the measurement would need to proceed this, of course. There are no crossover components in my speakers and the crossover plugin is not VST.

Can you think of a solution, please ?

Thanks and regards

Vernon

vtaylor,
I think adding "DSP Direct" option in the output dropdown list is a good idea, this would help you...

Salut k2k

Thanks for working and sharing.

I have a similar requirement to giro1991 - I use a DSP crossover plugin in Foobar so the measurement would need to proceed this, of course. There are no crossover components in my speakers and the crossover plugin is not VST.

Can you think of a solution, please ?

Thanks and regards

Vernon

There is no easy solution to this problem. Foobar2000 components cannot activate each other...

Would it be possible to port this to OS X, or at least export the current settings as a stereo impulse WAV file? While I realize that may allow people to dodge your licensing controls, my only reason for needing this is so I can attempt to use it as a system wide correction in AU Lab on OS X.

Also, just FYI, attempting to load this plug-in's configuration when running foobar2000 under Wine causes the whole program to crash. I could give you a crash report and minidump if you think you can make use of them, if it even matters.

Would it be possible to port this to OS X, or at least export the current settings as a stereo impulse WAV file? While I realize that may allow people to dodge your licensing controls, my only reason for needing this is so I can attempt to use it as a system wide correction in AU Lab on OS X.

Also, just FYI, attempting to load this plug-in's configuration when running foobar2000 under Wine causes the whole program to crash. I could give you a crash report and minidump if you think you can make use of them, if it even matters.

The VST version of Room EQ is already ported to OS X, however, AFAIK AU Lab doesn't support VST plug-ins. It supports only AU plug-ins. It is possible that Room EQ will be ported to the AU format in the future, but I don't promise.

Room EQ applies a FIR-free technology which doesn't add the convolution-specific artifacts to the sound. This is one of the main advantages of this product. It is theoretically possible to export a time-limited piece of the impulse response as a WAV file for using it with external convolvers. However, convolvers add specific artifacts to the sound, so you will not get the same quality of the sound. The convolver-specific distortion is usually not large, however, it is better to avoid using convolvers in room correction systems.

Room EQ doesn't work with current version of Wine.

Hi! Thank you for this plugin. Аre you going to add mode processing DSD?

Hi! Thank you for this plugin. ?re you going to add mode processing DSD?

No. DSD needs to be converted to PCM in order to run through a DSP engine.

Interesting, what about a VST multichannels version?

Thx

Interesting, what about a VST multichannels version?

You can use multiple instances of Room EQ VST in your DAW.

Quote from: mfast on 2014-12-21 23:08:09

Interesting, what about a VST multichannels version?

You can use multiple instances of Room EQ VST in your DAW.

Thanks for your answer, this sounds good.

However for to measure the center channel ( in a 5.1 or 7.1 speaker configuration ), it's mandatary consider it like a mono channel: Is possible to do correctly this?

Regards
                     

Thanks for your answer, this sounds good.

However for to measure the center channel ( in a 5.1 or 7.1 speaker configuration ), it's mandatary consider it like a mono channel: Is possible to do correctly this?

Room EQ VST can work as a mono plug-in if you include it into a mono track of your DAW.

Room EQ VST can work as a mono plug-in if you include it into a mono track of your DAW.

Ok, I will try it, thanks.

I installed MathAudio Ro+om EQ for Foobar2000, but cannot get any sound when Room EQ is loaded in by File -Preferences-select RoomEQ+ Configure. As soon as RoomEQ is moved from  left to right column under Preferences I get sound. I have Windows 8, and realtek Soundard in my Lenovo Flex 15D PC
Can any one help please?
Best regards
Sleiven

I installed MathAudio Ro+om EQ for Foobar2000, but cannot get any sound when Room EQ is loaded in by File -Preferences-select RoomEQ+ Configure. As soon as RoomEQ is moved from  left to right column under Preferences I get sound. I have Windows 8, and realtek Soundard in my Lenovo Flex 15D PC
Can any one help please?
Best regards
Sleiven

The sound is absent because you didn't measure the frequency response of your room.
You need to measure the frequency response in accordance with the instruction: mathaudio.com/room-eq.htm (http://mathaudio.com/room-eq.htm)
After that click the "Room EQ" radio button and move down the vertical slider to cut out the resonances of your room.

Quote from: Sleiven on 2015-01-04 10:05:10

I installed MathAudio Ro+om EQ for Foobar2000, but cannot get any sound when Room EQ is loaded in by File -Preferences-select RoomEQ+ Configure. As soon as RoomEQ is moved from  left to right column under Preferences I get sound. I have Windows 8, and realtek Soundard in my Lenovo Flex 15D PC
Can any one help please?
Best regards
Sleiven

The sound is absent because you didn't measure the frequency response of your room.
You need to measure the frequency response in accordance with the instruction: mathaudio.com/room-eq.htm (http://mathaudio.com/room-eq.htm)
After that click the "Room EQ" radio button and move down the vertical slider to cut out the resonances of your room.

Hi,
I need help and have a question about calibrated microphone I have. It came with my Velodyne DD-10 subwoofer, looks almost the same as one pictured in manual and it is calibrated but I don't know how to use it correctly with MathAudio Room EQ for Foobar2000 because I don't have calibration file and don't know what it is.
On line manual says:
"Conventional measurement microphones work perfectly with MathAudio Room EQ without calibration. However, if you have a calibrated measurement microphone, you can use it too. Just rename its calibration file to calibration.frd or calibration.cal and copy it to the Documents -> MathAudio Room EQ folder. Then exit your Foobar2000 and start it once again."
Can you explain what is calibration file and how to obtain or make it? Should it come with all calibrated microphones...? My microphone kit didn't come with any CD and Velodyne doesn't offer calibration file. Please help me.

Can you explain what is calibration file and how to obtain or make it? Should it come with all calibrated microphones...? My microphone kit didn't come with any CD and Velodyne doesn't offer calibration file. Please help me.

Measurement microphones usually have a very flat frequency response (in comparison with conventional microphones). Nevertheless, their frequency response is not ideally flat. That is why manufacturers often measure the frequency response of every measurement microphone and record the measurement results to individual calibration files. If you use the calibration file, the results of your room measurement are as accurate as if you would use an ideal measurement microphone. If your measurement microphone is supplied without a calibration file, you can send it to Cross-Spectrum Acoustics (http://www.csacoustics.com/products/) and they will make an individual calibration file for your microphone.

Idea: Android  support would be nice, just load the correction file and listen to the music.

First, thanks for your work on this component.  Simple to use and it has made a great improvement in what I'm hearing from my little PC sound system.

Two things:

I didn't find any release notes or change log on your website.  I've updated three times now, would be interested to know what gets fixed or changed with the new versions.

Also, I'm wondering if there is a simple way you could suggest to make a "before and after" measurement to more objectively evaluate the effectiveness of the filtering.  The results sound good to me, I'm curious what the measured response actually looks like on a graph.

Thanks again for one of the very few add-on components that actually makes Foobar sound better.

First, thanks for your work on this component.  Simple to use and it has made a great improvement in what I'm hearing from my little PC sound system.

Two things:

I didn't find any release notes or change log on your website.  I've updated three times now, would be interested to know what gets fixed or changed with the new versions.

Also, I'm wondering if there is a simple way you could suggest to make a "before and after" measurement to more objectively evaluate the effectiveness of the filtering.  The results sound good to me, I'm curious what the measured response actually looks like on a graph.

Thanks again for one of the very few add-on components that actually makes Foobar sound better.

All the latest updates can be qualified as “minor polishing”. No features added. For example, VST v2.4.8 includes a more convenient installer.

There is a simple method to evaluate the accuracy. You can connect the input of your sound card with its output by means of a RC circuit with cut-off frequency of about 1..2 kHz.  Then you can measure the frequency response of the RC circuit using Room EQ and compare it with the calculated frequency response. You can take the necessary formulas from  http://en.wikipedia.org/wiki/RC_circuit (http://en.wikipedia.org/wiki/RC_circuit)
The measured frequency response must coincide with the calculated frequency response. After that you can correct the frequency response using Room EQ and check the linearity of the resultant frequency response using sinusoidal signals of different frequencies.

There are a few other parameters that can be verified. Two of them have crucial importance: absence of pre-echo and absence of resampling. It is relatively easy to automatically equalize a room, but it is much more difficult to do it without pre-echo and resampling. The corrected sound becomes unnatural if it includes even a small amount of pre-echo. The absence of pre-echo in MathAudio Room EQ can be easily checked using any DAW and wave editor. The absence of resampling can be checked by means of FFT analysis.

Hello

I'm a MAthaudio french user

It does really great job

I take the measures with my Anthem MRX microphone (with calibration file)

Is this a suitable microphone ?

thanks

Hello

I'm a MAthaudio french user

It does really great job

I take the measures with my Anthem MRX microphone (with calibration file)

Is this a suitable microphone ?

thanks

The Anthem MRX microphone is a conventional measurement microphone, so it must work with Room EQ.

Quote from: fcouma on 2015-03-29 11:00:04

Hello

I'm a MAthaudio french user

It does really great job

I take the measures with my Anthem MRX microphone (with calibration file)

Is this a suitable microphone ?

thanks

The Anthem MRX microphone is a conventional measurement microphone, so it must work with Room EQ.

Yes, it works perfectly but my question was about the microphone Anthem's quality compared to UMIK (for example)

The Mathaudio curves are really closed to Anthem ARC curves

Yes, it works perfectly but my question was about the microphone Anthem's quality compared to UMIK (for example)

The Mathaudio curves are really closed to Anthem ARC curves

I don't have a personal experience with the Anthem MRX microphones, but I think that the presence of the individual calibration file is a good sign. IMHO, there should be no much difference between UMIK-1 and your microphone.

Quote from: fcouma on 2015-03-29 15:40:55

Yes, it works perfectly but my question was about the microphone Anthem's quality compared to UMIK (for example)

The Mathaudio curves are really closed to Anthem ARC curves

I don't have a personal experience with the Anthem MRX microphones, but I think that the presence of the individual calibration file is a good sign. IMHO, there should be no much difference between UMIK-1 and your microphone.

Thanks !

Curves with Mathaudio 2.8 and :

DAC Matrix i-mini  (coax)
Vincent SP331
Infinity Kappa 8.2i


(http://img11.hostingpics.net/pics/754404Sanstitre1.png)

News
A new version of the plug-in has just been released. The new version includes a point-to-point drawing feature. Simply use the right button of your mouse to draw straight lines.

News
New version released (v2.5.2).
The new version is compatible with Windows 10. It also includes the latest version of Foobar2000 SDK.

Thanks

The 2.5.1 version works fine with windows 10

Thanks

The 2.5.1 version works fine with windows 10

The 2.5.1 version doesn't normally show the "In/Out" window when you click the "In/Out" button (only on Windows 10). This problem is solved in the 2.5.2 version.

News
New version released (v2.5.3).
The sampling frequency range is extended up to 384 kHz.

News
New version released (v2.5.4).
The new version can read calibration files of iSEMcon measurement microphones (http://www.isemcon.net/en/emx-7150.htm) (they use decimal comma instead of decimal point).

News
New version released (v2.5.4).
The new version can read calibration files of iSEMcon measurement microphones (http://www.isemcon.net/en/emx-7150.htm) (they use decimal comma instead of decimal point).

Hi!

I tried to install the new version today but the installer blocked my computer. Win10.

Quote from: k2k on 2015-11-17 07:44:37

News
New version released (v2.5.4).
The new version can read calibration files of iSEMcon measurement microphones (http://www.isemcon.net/en/emx-7150.htm) (they use decimal comma instead of decimal point).

Hi!

I tried to install the new version today but the installer blocked my computer. Win10.

After two restart now all working. I don't know what happens.

After two restart now all working. I don't know what happens.

I've just tested the installer with both x32 and x64 versions of Windows 10. Everything works normally. This new version of the Room EQ plug-in uses exactly the same installer as it was used in the previous version. You may want to scan your computer with Avira Free Antivirus (https://www.avira.com/en/avira-free-antivirus) for safety.

Quote from: DiasDePlaya on 2015-11-22 01:49:58

After two restart now all working. I don't know what happens.

I've just tested the installer with both x32 and x64 versions of Windows 10. Everything works normally. This new version of the Room EQ plug-in uses exactly the same installer as it was used in the previous version. You may want to scan your computer with Avira Free Antivirus (https://www.avira.com/en/avira-free-antivirus) for safety.

I did everything, no viruses, no registry problems, all OK now. But I needed two restarts to recover my PC.

Great plugin! My home setup is not high end and the biggest problem is that on some frequencies you can hear resonance from one low frequency which is quite hard to locate and adjust using most of common equalizers.

Two tips from my side.

Go to Preferences->Playback->Output->Buffer length and change it to quite low. I used 6s for normal playback but because of that any change in MathAudio needed to pass 6s buffer to apply. Note that reducing it to low value will also change Fading fields because they need buffer to be applied.

Second tip is that I heard no difference until I've moved conversion to surround after MathAudio. Otherwise MathAudio was applying only to stereo I guess.

I just installed from fresh the FooBar2000 and the MathAudio (I had very old versions) and did a new calibration, and the result was way better than before! Thanks for the new version!!!

Wonderfull plugin !
I spent most of the night playing with it and it definitely brings significant sound improvement to my hifi setup. Congratulations.

News
New version released (v2.5.6).
EXE installer is replaced with a ZIP file.

News
New version released (v2.5.7).
Minor algorithm improvements.

Hello I'm from Brazil an excellent system pity that I can not use because my dac IFI only works of foobar using the super audio CD decoder plugin. Any new version will enable it?

Hello I'm from Brazil an excellent system pity that I can not use because my dac IFI only works of foobar using the super audio CD decoder plugin. Any new version will enable it?

It looks like your IFI dac supports PCM: http://ifi-audio.com/portfolio-view/nano-idsd/
I don't have any personal experience with that dac, however, if it really supports PCM, it should work with Foobar2000 without the decoder plug-in. It is possible that you need to select the correct output device in the Foobar2000 settings.

Please update binary on official site, http://mathaudio.com/foo_room_eq.zip old ver 2.5.5. Where can I get ver, 2.5.7?

Please update binary on official site, http://mathaudio.com/foo_room_eq.zip old ver 2.5.5. Where can I get ver, 2.5.7?

I've just checked the binary. Its version is correct (2.5.7). Simply remove v.2.5.5 in your Control Panel, then install v.2.5.7 in accordance with the instruction which can be found in the ZIP file.

Please update binary on official site, http://mathaudio.com/foo_room_eq.zip old ver 2.5.5. Where can I get ver, 2.5.7?

Found two instance of file in two folders "components" and user-components"... :) Thank you.

Very interesting software! I have just experimented with this in Foobar. It has certainly cleared up the low end in my room and brought the mids and highs into greater focus. I'm not sure if I "prefer" the sound yet, it will take a while to get used to. Is it possible to re-run the test with the corrections activated to see a new set of results with the MathAudio amendments in place?

Maybe someone can clarify this.

In the MathAudio guide for the Foobar room EQ plugin, it only mentions interfacing the speakers and microphone
via a computer soundcard. I haven't got one of them.

My system is: Computer - USB DAC/Preamplfier - Amplifier - Speakers. If I had one of the recomended USB microphones
is that enough for this to work or would I still need a soundcard to act as an interface for some reason?

If I am good to go with my present system does anyone else use a similar setup - got any advice/tips? 

Thanks

Very interesting software! I have just experimented with this in Foobar. It has certainly cleared up the low end in my room and brought the mids and highs into greater focus. I'm not sure if I "prefer" the sound yet, it will take a while to get used to. Is it possible to re-run the test with the corrections activated to see a new set of results with the MathAudio amendments in place?

The component doesn't include such a feature: the measurement subsystem is implemented as an independent block which doesn't include the correction algorithm. The component corrects the 'average' response of the room which is calculated as a result of the multipoint measurement.  You may safely trust the averaged response plots (taking into consideration that they are smoothed for better readability).

My system is: Computer - USB DAC/Preamplfier - Amplifier - Speakers. If I had one of the recomended USB microphones
is that enough for this to work or would I still need a soundcard to act as an interface for some reason?

You may connect your USB microphone directly to your PC. After that click the "In/Out" button in the Room EQ component window and select the USB input (this input will be visible after connecting the microphone to your PC).

Hey!

Really nice plug-in. I have a little experience in DRC programs and this seems like a very good and cheap(est) alternative to high cost analogues.

I have two questions:

- After applying correction, the overall volume level significantly dropped. Is there a way to add some gain lost due to correction? And foobar plugin you would recommend to compensate for the lost volume?

- Although, a very nice addition to a home-audio system, do you think this plug-in will work as nicely in the car environment? I am currently contemplating a CarPC install with PC being the central DSP and am choosing a DRC solution.

Hey!

Really nice plug-in. I have a little experience in DRC programs and this seems like a very good and cheap(est) alternative to high cost analogues.

I have two questions:

- After applying correction, the overall volume level significantly dropped. Is there a way to add some gain lost due to correction? And foobar plugin you would recommend to compensate for the lost volume?

- Although, a very nice addition to a home-audio system, do you think this plug-in will work as nicely in the car environment? I am currently contemplating a CarPC install with PC being the central DSP and am choosing a DRC solution.

- This plug-in doesn't change the maximum amplitude of the audio signal. Adding some gain can cause clipping in your DAC. It is recommended to use the volume control of your audio amplifier to compensate for the lost average volume.
- Yes, this plug-in will work in the car environment. Use no less than 10 measurement points.

Using this on my PC with Foobar2000 and a Dayton Audio UMM-6 calibrated mic did a very nice job of correcting the two channel system in room response at my listening position.

Thanks for the great capability!

News
New version released ( v2.5.8 ).
The new version supports microphone calibration files with the .txt file extension.

This software needs the volume adjustment feature as the Math Audio processing results in digital clipping.

News
New version released ( v2.5.9 ).
The new version includes an additional gain control slider.

This software needs the volume adjustment feature as the Math Audio processing results in digital clipping.

Room EQ doesn't clip the sound, however, its output signal can exceed 0 dB. Old versions of Room EQ relied on the host's volume control to avoid clipping. The latest version includes an amplitude indicator which helps to set the "Room EQ gain" control to the position corresponding to 0 dB output (see Fig.8 at http://mathaudio.com/room-eq.htm (http://mathaudio.com/room-eq.htm) ).

News
New version released ( v2.6.0 ).
The new version includes an additional amplitude indicator.

The sound is absent because you didn't measure the frequency response of your room.
You need to measure the frequency response in accordance with the instruction: mathaudio.com/room-eq.htm (http://mathaudio.com/room-eq.htm)
After that click the "Room EQ" radio button and move down the vertical slider to cut out the resonances of your room.

So I can't use this as a simple click-and-drag parametric equalizer?

So I can't use this as a simple click-and-drag parametric equalizer?

You are right.

I'm requesting a button to simulate a perfectly flat frequency response white line so I can notch it as I see fit.  What say you?

Then I'd say what you're requesting is not a room correction filter, as room correction is not something you can do by merely tweaking parameters until it sounds good.

This component already has the hard part down.  The only thing missing is to add a simple button that provides a perfect response to play with.

Here comes the semantics: "room correction" is arbitrary and ambiguous in the first place.  My hand-made correct curve is not objectively any worse than something spit out and adjusted from a "good" microphone at a "reasonable" listening position.

foobar has a built-in equalizer if you want to do it yourself.

a simple click-and-drag parametric equalizer

The closest you'll get is foo_dsp_xgeq, which is already a better equalizer than the one that comes with the player, simply by virtue of the available open source solution, SuperEQ, being a piece of crap if you don't work around its shortcomings manually. I say manually, because for most/all frequency bands, it has a sheer dropoff on both sides, and the only way to have a smooth slope of frequency response between wide points is to manually configure SuperEQ with more ranges and interpolate the response levels.

xgeq is already capable of fine grained and also smooth response slopes, if that's what you're after. Maybe you can ask xnor to make a parametric equalizer that plots a logarithmic frequency line, and lets you redraw the line almost like a paint program. It won't be an automated room correction filter like this plug-in, but it will be what you want.

Basically, this plug-in is probably never going to be a parametric equalizer, because that's not what it was designed to be. It plays reference sounds, it collects the impulse response using a reference microphone, and it performs deconvolution to produce an impulse simulating the conditions of your room. Then it calculates an inverse to that impulse response, assuming that to compensate for relative frequency losses caused by your speakers or room, it needs to pump up the frequency ranges inverse to the effects of that filter.

Maybe it could be a parametric equalizer, but it doesn't even need to know the numeric frequencies that it's correcting. It merely needs to take a frequency domain plot of your room/speakers' impulse response, normalize it, then inverse the values of each band, so accentuated frequencies are attenuated, and attenuated frequencies are accentuated.

If I were feeling adventurous, I could try to make an overlapped FIR-based parametric equalizer, using the same FFT library as foobar's visualization system, KissFFT. But I have no idea how the different bands of a frequency domain conversion relate to the base or the nyquist frequencies, or what their relative frequency steps would be. I would have to play with it a bit and measure the results.

I understand how this component is intended to be used.  The thing is, it already has clicky-draggy support, so it seems like it would be easy to make it generate a perfectly flat response for manual frequency twiddling because manual frequency twiddling is already supported.  I appreciate that only attenuation is allowed, because that protects the natural excursion of the audio components, as the author describes.

I hope my attached mockup makes what I want crystal clear.

I tried hacking foobar2000's "Core.cfg" to insert 0 values for each frequency band, but I don't know how to recalculate the file's checksum (assuming my hacking was correct), so no-go.

I'd love playing with this. My current setup sounds fine to my ears, but I would like to see what this program thinks of it and if correction can even make things sound better.

I was always about playing absolutely bitperfect and without any EQ, but maybe this changes my mind...

I have a mic connector on my mainboard, play music via a USB connection and I have 2 denon calibration mics that came with my previous receiver.

Off course this is not a real calibrated measurement microphone, but could it be somewhat useful for this task?
I'd like to know if I can get this to work, before buying a dedicated calibrated microphone.

Are there any clues how "off" this method can be? Apart from the microphone I use the onboard soundcard's microphone input. Does that make a huge difference or is the mic by far the most determining factor?

Thanks in advance!

Well I tested with my denon mic. Turns out that after EQ'ing everything sounds flat, no life in it. Low notes sound like they are EQ'ed out, not pleasurable to listen to, very "cold" sound.

Maybe I'm doing something wrong. In the examples you see the lows in the graph fade out to the  unhearable range. In my measurements they are quite high even in de 20hz range.

When I turn the slider up to the point the lows are unaffected (quite high) it restores, but it also gets back more to the original tone of the setup, so I think I am eliminating a part of the calibration I just done.
Also i put the mic on a one-foot tripod at ear height and put the children to bed (that helps a lot!)

Is there someone with a little more experience that can help me out?

At this moment I am already getting more pleasant result by playing with the settings, but it's quite difficult. Maybe there are some tips or do & don'ts to consider :)

I'm getting the hang of it....

This plugin is fantastic  :o  :o  :))  :))

I thought I had tuned my set quite good. After turning off the plugin, everything seems more boomy. Also the highs are much more dampened.
Love this thing. Soon I'm getting myself a good measurement microphone. Only need to find out now how to import the calibration file in this tool... I have no idea yet.

Quote from: mfast on 2014-12-21 23:08:09

Interesting, what about a VST multichannels version?

You can use multiple instances of Room EQ VST in your DAW.

That is obviously not the case in foobar2000, which is OK by me. I just have a question about it.
I just started using the Room EQ plugin in foobar2000 in combination with a 5.1 channel setup. No problem, I'd like to know if Room EQ in this situation only affects the front L+R channels?

I'm getting the hang of it....

This plugin is fantastic  :o  :o  :))  :))

I thought I had tuned my set quite good. After turning off the plugin, everything seems more boomy. Also the highs are much more dampened.
Love this thing. Soon I'm getting myself a good measurement microphone. Only need to find out now how to import the calibration file in this tool... I have no idea yet.

A little update. I did use a calibrated mic short after my post, but found that the sound after calibration was still not quite how I like it after listening more thoroughly to it. The boomyness was fixed by putting the outer rings of the bongs into my CM10 s2 speakers. It fixed that perfectly.

After trying with and without calibration it somehow not quite satisfies me. Maybe I like the signature of the B&W speakers the way they are, but then again calibrated should be de point to start. Tried many different settings but still I miss something. The low end is a bit too suppressed. Maybe this comes from a slight hum from my aquarium pumps (50hz). I don't know if they influence the measurement? It's barely noticable, but to a microphone maybe more??

Are there any suggestions or alternatives to this to fool around with?

Meh, another update without changelog.  ::)

Hiyas, can I measure a purposeful profile using builtin laptop mic or do I need an external studio microphone?

Hiyas, can I measure a purposeful profile using builtin laptop mic or do I need an external studio microphone?

The built-in laptop mic has a non-linear frequency response and its directional pattern is not suitable for this application. You need a conventional measurement microphone. They are not expensive. Look at this model: https://www.parts-express.com/dayton-audio-umm-6-usb-measurement-microphone--390-808

Thanks k2k for this wonderful piece of software, very user friendly. I recently bought a UMIK-1 for setting up the foobar DSP.

As someone who is new to room correction, I have a few questions;
What volume in decibels (approx.) should my speakers output while running the measurement?
How many measurements should I run if I have only a single static listening position?
Does mic placement make a big difference in terms of height and angle?

What volume in decibels (approx.) should my speakers output while running the measurement?
[...]
Does mic placement make a big difference in terms of height and angle?

Is there any reason not to start with your listening volume and the listening position? 

(If height makes a difference in terms of sound, then mic placement will placement will make a difference too, of course.)

Thanks k2k for this wonderful piece of software, very user friendly. I recently bought a UMIK-1 for setting up the foobar DSP.

As someone who is new to room correction, I have a few questions;
What volume in decibels (approx.) should my speakers output while running the measurement?
How many measurements should I run if I have only a single static listening position?
Does mic placement make a big difference in terms of height and angle?

REW suggest 75db + (take care for clipping if going up).

Moving the mic 1" will change your measurements. There's no consensus about mic placement up or forward, so do what suits you.

I take one or two measurements before changing something in my room. There's chances of having weird measurements or noise during a sweep.

Does the software offer the option to correct only below the Schroeder frequency of the room, as per the recommendations of Toole and some others who assert that 'correcting' above that may be detrimental.  Below Schroeder, room modal effects dominate; above it, reverberant effects dominate.   For 'typical' rooms it's circa 300Hz. 

Does the software offer the option to correct only below the Schroeder frequency of the room, as per the recommendations of Toole and some others who assert that 'correcting' above that may be detrimental.  Below Schroeder, room modal effects dominate; above it, reverberant effects dominate.   For 'typical' rooms it's circa 300Hz.

If you don't wish to correct the frequency response above 300 Hz, use the manual correction of the curve. Use the left mouse button to draw arbitrary curves, or use the right mouse button to draw straight lines. The statement about the influence of the reverberation effect is correct, however, Room EQ applies an advanced multipoint measurement algorithm that takes this effect into consideration. You may want to compare the different types of correction with your own ears (use the "Save preset" and "Load preset" buttons to conveniently switch the plots).

News
New version released ( v2.6.9 ).
Minor improvements in DSP algorithms.

News
New version released ( v2.7.0 ).
The new version includes a new "High resolution" feature.
The "Stereo balance" switch is replaced by a slider.

NO TEST SIGNAL OUTPUT

Hello, I am trying to use Room EQ with Foobar 1.4.3 and ASIO interfaces, however I cannot have the test signal out.

My Foobar DSP set-up requires 96kHz sampling rate and an active X-over plug-in (Thuneau Allocator) because I am using multi-amps.
So, MathAudio Room EQ is followed by Thunea Allocator that takes stereo input and gives 6 channels output.

I have been able to configure Room EQ for getting mic signal from ASIO analog input, however the default output stream and also the default digital output look not sending the audio to the next DSP element in the chain and no signal output at all.

To check the problem, I did the same using the VST version of Room EQ. In this can, I can get the test signal output, but I cannot configure the mic input.

Anyone can help me?

Many thanks and kind regards, Andrea

Hello, I am trying to use Room EQ with Foobar 1.4.3 and ASIO interfaces, however I cannot have the test signal out.

Hello Andrea,

The Foobar2000 version of Room EQ allows you to measure your room. You don't need VST for measurement.
1. Select the "Room Measurement" radio button to activate the Room Measurement mode.
2. Select the correct mic input (left or right) by means of the "Mic input" radio buttons.
3. Click the "In/Out" button and select your input device.
4. Look at the "Mic signal indicator" and speak to the microphone. If everything is done correctly, the indicator will show you the presence of the mic signal. After that you can click the "Start measurement" button to measure your room.

Quote from: AndreaT on 2019-04-26 11:19:04

Hello, I am trying to use Room EQ with Foobar 1.4.3 and ASIO interfaces, however I cannot have the test signal out.

Hello Andrea,

The Foobar2000 version of Room EQ allows you to measure your room. You don't need VST for measurement.
1. Select the "Room Measurement" radio button to activate the Room Measurement mode.
2. Select the correct mic input (left or right) by means of the "Mic input" radio buttons.
3. Click the "In/Out" button and select your input device.
4. Look at the "Mic signal indicator" and speak to the microphone. If everything is done correctly, the indicator will show you the presence of the mic signal. After that you can click the "Start measurement" button to measure your room.

Hello K2K, the mic is there as I clearly stated in my post. The mic is detected and well working.
Is Room EQ that looks not generating any test signal.

Pressing the Foobar In/Out button there is also the option to select the OUTPUT.
I first used the default, then I tried also the other reasonable good outputs, but NO SIGNAL OUT.

It is clear that with my configuration, Room EQ does not send its signal to the next DSP component in the audio chain.

Regards

Herein attached more details about my case
Thanks and regards

Herein attached more details about my case

Room EQ sends the test signal directly to the selected output device. In your case it sends the sweep sound to "Dispositivo uscita digitale". Is it the correct output device? Room EQ can be chained with other foobar2000 components for playback only. If you wish to process the test sound by any DSP plug-ins, you cannot do it in Foobar2000, but you can do it in any DAW. For example, you can use demo versions of Reaper and Room EQ VST to process the sweep sound by another plug-in and measure your room. You can download Reaper from https://www.reaper.fm  You can also find some instructions for setting Reaper up at https://mathaudio.com/room-eq-faq.htm Save the results of your measurement as a SNR file. After that you can use it for playback in Foobar2000 together with any other Foobar2000 DSP components.

Hello K2K, many thanks. I did as you instructed and I got it working.
Regards, Andrea

Hello K2K, what a nice software. I use foobar2000 to stream music from a windows 10 tablet to a network player with 2 speakers and 2 subs. Can I use mathaudio roomcorrection for this setup, or does it only work with the soundcard of the tablet? The output menu in mathaudio doesn't show  the in foobar2000 connected network player. Can't find this use case here and want to ask this before I buy the measurement microphone.

Hello K2K, what a nice software. I use foobar2000 to stream music from a windows 10 tablet to a network player with 2 speakers and 2 subs. Can I use mathaudio roomcorrection for this setup, or does it only work with the soundcard of the tablet? The output menu in mathaudio doesn't show  the in foobar2000 connected network player. Can't find this use case here and want to ask this before I buy the measurement microphone.

Hello audioclassic,

First of all it is necessary to check whether or not your tablet is powerful enough to work with Room EQ. You may contact us at https://mathaudio.com/contact.htm and I will send you a SNR file. You can run Room EQ with this file on your tablet.  If you can play music without any interruptions, your tablet can perform all necessary calculations in real time. Alternatively, you can make the SNR file yourself by means of the sound card of your tablet (the quality of the microphone has no meaning because you will not use this SNR file for room correction).

You also need to click the "In/Out" button on the Room EQ's panel and check whether you can see your network player in the list of output devices. If the player is present, click the "Room measurement" button and check whether you can hear the sweep sound. If you can hear the sound, your tablet can be used to measure your room. If you don't see your network player in the list of output devices, you can temporarily use another computer (laptop of desktop) to measure your room and make the SNR file. Then you can use that SNR file on your tablet to correct your room.

Thanks for the answer K2K, I already tried to make a SNR file with the tablets microphone and speakers and that worked fine, so the tablet can handle it. But the network player, although connected to foobar2000 and working fine with music streaming doesn't show up in the output menu of mathaudio on the tablet, and not on my other big Wacom i7 tablet and not on my workstation laptop. So the conclusion will be that it can't be done this way, I think?

Thanks for the answer K2K, I already tried to make a SNR file with the tablets microphone and speakers and that worked fine, so the tablet can handle it. But the network player, although connected to foobar2000 and working fine with music streaming doesn't show up in the output menu of mathaudio on the tablet, and not on my other big Wacom i7 tablet and not on my workstation laptop. So the conclusion will be that it can't be done this way, I think?

Is it possible to temporarily use another audio or USB input of your network player to connect it to your laptop for measurement? If not, you can try to temporarily use demo versions of Reaper and Room EQ VST for measurement. It is possible that Reaper can "see" the audio interface of your network player. You can download Reaper from https://www.reaper.fm You can also find some instructions for setting Reaper up at https://mathaudio.com/room-eq-faq.htm

The network player has no other computer connection than  wifi or network cable. The usb is for music files only.  And Reaper doesn't show / see the network player either (although windows 10 does). So its not gonna work I ques.

The network player has no other computer connection than  wifi or network cable. The usb is for music files only.  And Reaper doesn't show / see the network player either (although windows 10 does). So its not gonna work I ques.

It is strange that Foobar2000 can work with the driver of your network speakers but Reaper cannot do it. Open Reaper, go to Options -> Preferences -> Audio -> Device and click the "Audio system" drop list. It will show you the list of available audio drivers. Select the driver of your network speakers if it is available. If it is not available, I don't see any simple and reliable methods to overcome this problem...

Hello Audioclassic, if you are still having problems with Reaper, I suggest you to use the free CARLA DAW/VST host.
I am using it and it is excellent.

Strange as it is, the driver can be seen in windows 10 and in foobar20000, but not in Mathaudio and not in Reaper. I quess the Carla Daw/Vst host is for Linux, I am just a stupid windows 10 user. A possibility for room equalization with dlna is difficult to find. I found somewhere that maybe in Jriver there is a possibility with plugins. I hate to leave foobar2000, but I will research this further.
Thanks for the responses.

Hello Audioclassic, ... of course you should download the appropriate Carla for your platform.
However, if you are using Windows 7 or 10, I would now suggest to use Cantabile 3 free edition (registration required, but absolutely free).
I moved to it a couple of weeks ago and I found it better for this application (easier to use and lighter in terms of resources used).

Hello Audioclassic, ... of course you should download the appropriate Carla for your platform.
However, if you are using Windows 7 or 10, I would now suggest to use Cantabile 3 free edition (registration required, but absolutely free).
I moved to it a couple of weeks ago and I found it better for this application (easier to use and lighter in terms of resources used).

It looks like Cantabile doesn't work with DLNA too. Audioclassic uses network speakers and he needs DLNA.

I found a workaround for using dlna and mathaudio. I read you can make a wav file of a measurement sweep in REW. Using  2 computers you can than measure in one with foobar and mathaudio and set a delay of 15 seconds. With the other computer you play the measurement sweep wav file in foobar exactly at the right time. I don't know how good a REW sweep measument file is for Mathaudio but it does work. Nevertheless its about time someone creates a room eq software that is ready for dlna😊. A lot of people will be using dlna at the moment. I have a Panasonic network player with 2 KEF speakers and under each speakers a REL subwoofer. I guess a lot of people now have a setup like this. Thanks for all the answers.

I found a workaround for using dlna and mathaudio. I read you can make a wav file of a measurement sweep in REW. Using  2 computers you can than measure in one with foobar and mathaudio and set a delay of 15 seconds. With the other computer you play the measurement sweep wav file in foobar exactly at the right time. I don't know how good a REW sweep measument file is for Mathaudio but it does work. Nevertheless its about time someone creates a room eq software that is ready for dlna😊. A lot of people will be using dlna at the moment. I have a Panasonic network player with 2 KEF speakers and under each speakers a REL subwoofer. I guess a lot of people now have a setup like this. Thanks for all the answers.

There is a free program that streams audio from PC to DLNA: https://www.streamwhatyouhear.com/
Hopefully it could send the sweep sound from Room EQ to your network speakers. You may want to try it.
The WAV sound from REW produces a substantial measurement error if you use it with Room EQ.

OK, the REW  WAV file did work but the result doesn't sound good indeed. I tried stream what you hear before, but this gives a delay that makes it unusable for mathaudio (and for youtube film)

If the developers read this: please keep the presets when upgrading the component. fb2k went completely silent until I could search through my files to find it.

If the developers read this: please keep the presets when upgrading the component. fb2k went completely silent until I could search through my files to find it.

Foobar2000 automatically resets the preset when the size of the new preset is not equal to the size of the old preset. We recommend to save the old preset as a SNR file before upgrading the component. The SNR file helps you to manually restore your preset.

News
New version released ( v2.7.2 ).
The new version allows one to redo the last measurement if an occasional noise has happened during the measurement.
Please don't forget to save your preset as a SNR file before upgrading the component.

I am having problems with the software working on my computer. When Mathaudio is enabled in the DSP and opened, there is no response on the meter to show the microphone is picking up sound. The in/out button is not visible either. I can see that the Umik-1 is functioning in the recording properties of Windows 7. Clicking the button for a sweep does not function, no sound output from Mathaudio. If I am playing a song and click the sweep button it outputs the test signal but does not record. Output was changed with WASAPI, KS, ASIO, and direct Sound. 32bit DLL is loaded in Foobar. 

Is this not working because the Umik-1 is not recognized by MathAudio?

My chain is as follows; Computer - Berkeley Alpha USB - Tri-Vista 21 DAC - Pre Amp - Amplifiers - Speakers.

Software; Windows 7 Pro, Latest Foobar2k, MathAudio, Umik-1. Exclusive access checked in sound properties and tried unchecked.

All help appreciated!

Thanks Mark

I am having problems with the software working on my computer. When Mathaudio is enabled in the DSP and opened, there is no response on the meter to show the microphone is picking up sound. The in/out button is not visible either.

The "In/Out" button is absent in VST/AU versions of Room EQ. Make sure that you use the Foobar2000 version of Room EQ.

Quote from: racerxnet on 2019-09-21 17:28:20

I am having problems with the software working on my computer. When Mathaudio is enabled in the DSP and opened, there is no response on the meter to show the microphone is picking up sound. The in/out button is not visible either.

The "In/Out" button is absent in VST/AU versions of Room EQ. Make sure that you use the Foobar2000 version of Room EQ.

Thank You for the response. I found out exactly as you noted, I loaded the incorrect DLL file. I can now try out the software.

Mark

Is there a way to minimize the room EQ window while using the playlist from Foobar, or am I thinking wrong. Once I close MathAudio while playing a song it stops. I know Dirac Lives playback engine does not interfere with or stay on top of the playlist window. After I save a correction file and start Foobar, do I have to open MathAudio on top to enable the use of the file created?

Mark

Is there a way to minimize the room EQ window while using the playlist from Foobar, or am I thinking wrong. Once I close MathAudio while playing a song it stops. I know Dirac Lives playback engine does not interfere with or stay on top of the playlist window. After I save a correction file and start Foobar, do I have to open MathAudio on top to enable the use of the file created?

Click the "Apply" button in the DSP Manager.

Hi,

A newbie question on MathAudio Room EQ/foobar.

The tutorial says "Place the microphone at your ear height. Set it vertically." which implies pointing to the ceiling...correct?
Therefore I should use the UMIK-1 microphone *_deg90 calibration file...

I just wanted to re-confirm my understanding...

Thanks!

The tutorial says "Place the microphone at your ear height. Set it vertically." which implies pointing to the ceiling...correct?
Therefore I should use the UMIK-1 microphone *_deg90 calibration file...
I just wanted to re-confirm my understanding...

Yes, you’re right.

Hello,

Another noobie question:  If I am doing the EQ for one MLP (main listening position), should I just take 1 measurement sample, or do a multi-point measurement?

If I should do a multi-point measurement, should I cover the 50cm2, or 90cm2 (90cm2 is larger than the MLP seat)?
(https://i.ibb.co/7yk209B/MLP2.jpg)

**Mine is a HiFi Stereo setup.

Thanks for any pointer, and help understanding RoomEQ's difference between single measurement vs. multi-point measurement.

Hello Boxerfan88,
personally I do measurements only at the Golden Spot (MLP).
I also tested the multi points, but with my room geometry doesn't work better.
So, if you want to clear this matter, you should test both and then take the better result.
Regards, Andrea

Another noobie question:  If I am doing the EQ for one MLP (main listening position), should I just take 1 measurement sample, or do a multi-point measurement?
If I should do a multi-point measurement, should I cover the 50cm2, or 90cm2 (90cm2 is larger than the MLP seat)?
Thanks for any pointer, and help understanding RoomEQ's difference between single measurement vs. multi-point measurement.

A multi-point measurement is always better than a single-point measurement. Among other things, a single-point correction makes the sound more dependent on the location of your head.
The smaller the multi-point measurement area, the better the accuracy. The measurement area should be as small as possible, but it should cover all possible locations of your head. In other words, the size of the measurement area depends on your habits.

Hello AndreaT, k2k, thank you both for your very valuable inputs. I tried various combinations and settled on the small multi-point measurement @ MLP, as it sounded the best to my ears.

The measurement that I settled on was taken along the edges of 60cm x 30cm square @ MLP, 6 multi-point measurements.

(https://i.ibb.co/Psp2tvq/measurement.png)

@boxerfan88 : The Golden Spot should always be in. That's why the basic matrix is 9x9 = Golden Spot plus 8 points surrounding it.

@K2K: Sorry, I do not agree in full with you. In my case I have a precise listening point, my head is always placed there. So, in my case, the optimal correction would probably be a very small MLP (15 x 15 cm square area total) that simply doesn't make sense to consider.
I confirm that in my case the MLP equalization considering 90 x 90 cm area gave worse result than the single point equalization at the Golden Spot.
Regards, Andrea

Sorry, I do not agree in full with you. In my case I have a precise listening point, my head is always placed there. So, in my case, the optimal correction would probably be a very small MLP (15 x 15 cm square area total) that simply doesn't make sense to consider.
I confirm that in my case the MLP equalization considering 90 x 90 cm area gave worse result than the single point equalization at the Golden Spot.
Regards, Andrea

The length of the sound wave is equal to about 3 cm at frequency of 10 kHz. Consequently, if you move your measurement microphone a few centimeters far from its initial position, the superposition of the direct and reflected waves will be different (due to the difference of their phases), and the frequency response will be different too. The multipoint measurement minimizes this problem and helps to improve the accuracy of the measurement. If you are absolutely sure that you don't move your head, you may want to use a very small area for the multipoint measurement.

HI,
I installed mathaudio room eq: it sends lost of cracks in the speakers! (this is in room eq mode, with a calibration properly done). I tried with a second PC, and it's the same. Its very upsetting, because removing the component in foobar doesn't stop it. I had to remove foobar completely and re-install!!
any idea why?
thanks

HI,
I installed mathaudio room eq: it sends lost of cracks in the speakers! (this is in room eq mode, with a calibration properly done). I tried with a second PC, and it's the same. Its very upsetting, because removing the component in foobar doesn't stop it. I had to remove foobar completely and re-install!!
any idea why?
thanks

Such cracks are theoretically possible on 'very' old computers if you use the "High resolution" mode in Room EQ. Try to use the "Normal resolution" mode instead. Try also to increase the buffer size in Foobar2000. If you remove Room EQ from Foobar's DSP panel, don't forget to click the "Apply" button to apply your changes. Room EQ cannot affect the performance of Foobar2000 if you removed the Room EQ component.

Hi K2K:

Can you please tell me, if the Room EQ software will only work through my laptop line audio and/or USB outputs?

I ask because I have a home-network based audio system. Far across the house is my file server. Over my in-home network, I share my music file folders to Foobar2000, which is running on a laptop in my listening room.  I use the BubbleUPnP plugin to turn a Chromecast Audio device into a UPnp/DLNA renderer. The digital output from the Chromecast Audio feeds a stand-alone DAC, which in turn feeds audio to my stereo amplifier.

The system works beautifully, is of very high audio quality (24/96), and I do not have to worry about having cables attached to my laptop.

I want to use the Room EQ software to improve my sound, but have read a few comments here that confuse me. Others are claiming that Room EQ + Foobar will NOT work over DLNA? Is this true for all situations? Or...is there a method you can please advise me to take in order to make Room EQ function together with my streaming setup?

Worst-case condition, I can get an active USB cable, take the Room EQ audio from my laptop USB port and run it 6 meters over to my DAC, eliminating the Chromecast Audio streaming part of my chain.

Any advice from you or others here on the forum is greatly appreciated! 

Thanks,
Papi Chulo

Can you please tell me, if the Room EQ software will only work through my laptop line audio and/or USB outputs?
I ask because I have a home-network based audio system.

The problem of network based systems lies in the fact that they substantially delay the sound. This fact creates a problem for the Room EQ's measurement subsystem. If your DAC has a USB input, try to temporarily use a USB cable to measure your room. After that you should be able to use your network system exactly as you did it before.

If your DAC has a USB input, try to temporarily use a USB cable to measure your room. After that you should be able to use your network system exactly as you did it before.
[/quote]

Thanks for the quick reply! Yes, I have a USB input on the DAC for measurement. This seems like a simple fix to my problem.

1) My measurement mic is a USB type. Will I be able to use it and the USB DAC simultaneously for measurement?

2) I guess I’ll need to install a driver for my USB DAC. Is there any particular one that you recommend? WASAPI, ASIO, etc?  I’m running Windows 10.

Thanks again!!!

1) My measurement mic is a USB type. Will I be able to use it and the USB DAC simultaneously for measurement?
2) I guess I’ll need to install a driver for my USB DAC. Is there any particular one that you recommend? WASAPI, ASIO, etc?  I’m running Windows 10.

Simply connect your USB microphone and your USB DAC to your PC. Then click the "In/Out" button on the Room EQ panel and select both these devices in the corresponding drop-lists. If your USB DAC is not present in the list of the available output devices, install a conventional Windows driver for your USB DAC (not ASIO).

@k2k : Hello, I see that has been released a new version 2.7.3.
However, on the official web site there is no Release Note or Change Log (and never was).
Do you know what's changed?
Regards, Andrea

News
New version released ( v2.7.3 ).
This update is our response to those users who asked us about the possibility of trying B&K and Harman target curves. Some explanations are given at the beginning of the Room EQ FAQ: https://mathaudio.com/room-eq-faq.htm

Is there a mic calibration file for the Neumann KM84 (Cardioid) or KM83 (Omni), and if so, where can I find them?

Is there a mic calibration file for the Neumann KM84 (Cardioid) or KM83 (Omni), and if so, where can I find them?

Both these mics are not good for the room measurement. You may want to look at this one (https://www.minidsp.com/products/acoustic-measurement/umik-1) (USB) or this one (https://www.bhphotovideo.com/c/product/747384-REG/Nady_CM_100_CM_100_Reference_Measurement.html/reviews) (phantom powered).

News
New version released ( v2.7.3 ).
This update is our response to those users who asked us about the possibility of trying B&K and Harman target curves. Some explanations are given at the beginning of the Room EQ FAQ: https://mathaudio.com/room-eq-faq.htm

@k2k : Hello K2K, thanks for the clarification, however I couldn't understand how then restore the default target curve in case the loaded one is not liked.
Regards, Andrea

@k2k : Hello K2K, thanks for the clarification, however I couldn't understand how then restore the default target curve in case the loaded one is not liked.
Regards, Andrea

Simply click the "Bright" button or the "Neutral" button.

Many thanks for your kind support. (:->)

News
New version released ( v2.7.4 ).
The new version includes a new graphical user interface.

Hello @k2k , thanks for the update.
Unfortunately on the official website there are no release notes... Do you know where can I get the "release notes"?
Reading the new documentation (very little) there are no more references to uploading of target curves.
Do you know if this last feature is still present or gone away?
Kind regards, Andrea

Hello @k2k , thanks for the update.
Unfortunately on the official website there are no release notes... Do you know where can I get the "release notes"?
Reading the new documentation (very little) there are no more references to uploading of target curves.
Do you know if this last feature is still present or gone away?
Kind regards, Andrea

Yes, this feature is still present. You can read about it at the beginning of the Room EQ FAQ: https://mathaudio.com/room-eq-faq.htm
We often publish some information about new releases on this forum.

Hello @k2k, many thanks for your prompt reply and for directing me to the Room EQ FAQ.
Kind regards, Andrea

News
New version released ( v2.7.5 ).
Minor improvements in GUI.

Hello @k2k, today I upgraded your component from the v2.7.1 to the last v2.7.5.
Very good to have now integrated the measuring options.

However, the Signal Output selection is limited and, in my opinion, has also a bug.

LIMITATION FOUND
As per attached screeshot, in my DSP Chain, your Room EQ is not the last component.
My last component is Thuneau Allocator (and it must be the last one). It is a X-over and phase arbitrator.
So, I need to inject your sweep signal into it, or, in other words, to have your Signal Out routed into it, simply following the DSP chain.

ASIO PROBLEM
I am using a professional MOTU 896mk3 DAC that makes use of the MOTU ASIO driver.
Well, even selecting any usable ASIO output, I cannot get any sweep out from within Foobar.

Instead everything is working as expected setting up an appropriate DAW project within Cantabile and using the VST2 demo plug-in.

Kind regards, Andrea

Hello @k2k, today I upgraded your component from the v2.7.1 to the last v2.7.5.
Very good to have now integrated the measuring options.
However, the Signal Output selection is limited and, in my opinion, has also a bug.

The measurement subsystem of the Foobar2000 version of Room EQ doesn't work with ASIO drivers. It works with conventional Windows drivers. As far as I understand your DAC includes such drivers and you can use them for measurement.

The Foobar2000 version of Room EQ cannot 'inject' the sweep sound to another Foobar2000 component. You can chain the Room EQ component with other Foobar2000 components for playback only, not for measurement.
The VST version of Room EQ can "inject" the sweep sound to another VST plug-in, but I don't know whether the Thuneau Allocator is available as a VST plug-in.

Hello @K2K, thanks for the details, I was unaware of this Foobar limitation.
So, I will keep going as in the past, using Cantabile and the VST demo version for measurement.
PS: Thuneau Allocator is a VST2 plug-in and I am using it within Foobar via the VST wrapper.
Kind regards, Andrea

Hello @k2k,
my audio system is all time playing at 24bit 96kHz and I have ESL loudspeaker capable (measured with REW) to play up to 30..35kHz.
Also my subwoofer systems can go down to approximately 15Hz.
So, my question is: is there any option to have Room EQ serving the range 15Hz to 35kHz?
Or, is it eventually possible to use REW to measure the response as wide as I want and then to load it into Room EQ?
Thanks and regards, Andrea

Hello @k2k,
my audio system is all time playing at 24bit 96kHz and I have ESL loudspeaker capable (measured with REW) to play up to 30..35kHz.
Also my subwoofer systems can go down to approximately 15Hz.
So, my question is: is there any option to have Room EQ serving the range 15Hz to 35kHz?
Or, is it eventually possible to use REW to measure the response as wide as I want and then to load it into Room EQ?
Thanks and regards, Andrea

Room EQ works in the audio frequency range of 20 Hz to 20 kHz. Room EQ doesn't linearize the frequency response at infrasound and ultrasound frequencies.

Hello @k2k, I have another question for you: is there any option to read / open the .snr config file?
Is it possible to convert to any standard format file readable by REW?
I think it would be useful to have the option to manually modify specific response points.
Kind regards, Andrea

No, you cannot read the SNR file or use it in another EQ program.
You can easily modify any point of the target curve with your mouse:
left button - arbitrary points,
right button - straight lines,
mouse wheel - single point on one of the plots (left or right).

Hello @k2k, my intention is not to use the SNR file with other EQs.
I just would like to better have the control of what going on during the equalization.
I know the possibility of using the mouse ... but you should know that it has many limitation, low accuracy and practically no way to see exactly what I am doing where... I mean, I cannot dial in precise value I get from another computational model (this just for example).
I understand I am not the average user... I am now using Room EQ for "audio experiments" searching for improvements over my legacy DRC solution. To be honest, so far I didn't find out a really winning set-up. I mean, I found Room EQ an excellent tool, basically very fast to setup, but at the same time too limited for getting an absolute fine tuning that would allow it to win (get absolute better audio performance) against my legacy DRC.
Anyway, thanks so much for this Foobar component and all the best for your future developments.
Kind regards, Andrea

Hello @k2k, do you know Dirac Live?
They state to use an innovative "Mixed-phase room correction technology" surpassing  any prior or competing solution.
If you know about, would you kindly give us your opinion or compare your solution with it?
Thanks and regards, Andrea

Hello @k2k, do you know Dirac Live?
They state to use an innovative "Mixed-phase room correction technology" surpassing  any prior or competing solution.
If you know about, would you kindly give us your opinion or compare your solution with it?
Thanks and regards, Andrea

We don't discuss competitors.
Key features of our technology are listed at https://mathaudio.com/why-room-eq.htm

hello , i cannot acces to https://mathaudio.com from at least 3 days.
i am alone?

hello , i cannot acces to https://mathaudio.com from at least 3 days.
i am alone?

I've just tested https://mathaudio.com from multiple countries. Everything works. Please send me your IP address and I will check it in our firewall. You can see your IP address at https://www.whatismyip.com/  I've sent you a personal message with my email address.

Quote from: Melomane on 2020-01-17 20:37:28

hello , i cannot acces to https://mathaudio.com from at least 3 days.
i am alone?

I've just tested https://mathaudio.com from multiple countries. Everything works. Please send me your IP address and I will check it in our firewall. You can see your IP address at https://www.whatismyip.com/  I've sent you a personal message with my email address.

it is à problem with my internet access provider, sorry.

Hello K2K.

First off, congratulations on a truly wonderful software program, and THANK YOU for making it available to FB2K users at no cost. It is easy to use, very powerful and IT WORKS!!! I had a really bad room resonance (+15 dB peak at 50 Hz!) and Room EQ was able to null it out. My system sounds so much better now!!!

I have a few questions for you:

1) I'm using an Dayton UMM-6 USB mic with calibration file, and measuring my room with a normal 2-channel stereo setup. Your instructions say to point the mic straight up at the ceiling from listening position, but my mic calibration file is specifically calibrated at ZERO degrees. In this case, should I not be pointing the mic horizontally at the space midway between my two speakers? Will I get a more accurate reading this way? I don't understand why I should be pointing it at the ceiling! High frequency response (>10 kHz) of the microphone must drop off considerably @ 90 degrees off axis, no???

2) Is it possible to use Room EQ simultaneously with another DSP EQ program running on FB2K? In my case, I have open baffle speakers paired to subwoofers. There is a significant response dip where the subwoofer and OB speakers "crossover" - so I use a DSP equalizer to boost that dip to get a flatter response.

If I have that DSP EQ program running as normal, can I then use Room EQ and do a room measurement while both are running? What about after I apply room EQ to my system - can I have the two programs running at the same time on FB2K? Will they interact or cause problems?

I think I read somewhere that if concurrent DSP programs are being used, Room EQ should be the LAST (or bottom) program in the "active DSP list" in FB2K? Is this correct?

Thanks again for giving us such a wonderful program. It is truly a fantastic addition to my sound system!

Papi Chulito
Northern Arizona, USA

Hello Papi,
1) The high frequency response will drop off a little in any case (pointing the microphone at the point between your speakers doesn’t ideally solve the problem). On the other hand, pointing the microphone at the ceiling makes the microphone equally sensitive to the wall reflection, and this factor is very important. Ideally, a 90 degrees calibration file is preferable, but a zero degrees calibration file normally doesn’t produce an audible distinction.
2) The Foobar2000 version of Room EQ has its own audio output for the sweep sound. In other words, the sweep sound is not processed by any other Foobar2000 components that are present in the chain. You may want to try to avoid usage of the additional boosting equalizer. Simply move down the vertical slider of Room EQ to compensate for the dip (or increase the cut-off frequency of your subwoofer to avoid the dip).

Hello Papi,
(snip)...
...You may want to try to avoid usage of the additional boosting equalizer. Simply move down the vertical slider of Room EQ to compensate for the dip (or increase the cut-off frequency of your subwoofer to avoid the dip).

Hi K2K, and thanks for your response. Unfortunately, I have a problem that will persist, regardless of your excellent suggestion.

My main speakers start a steep roll-off at about 200 Hz. I tried adjusting the crossover frequency on my woofer as high as it would go...and to my surprise it only goes to ~125 Hz, then drops off at 24 dB/octave. Even with the crossover F set up high, there is a rather deep suck out between 125 and 200 Hz...a critical zone for upper bass guitar.

You are correct in that I could drop the correction curve WAY down to the bottom of that trough. I tried that, and as you can imagine, the resulting sound was "artificial" and lifeless. Like many other users, I have found that DSP corrections sound best when used sparingly...I set my correction curve (Bruel and Kjaer) to allow for a bit of ripple below the line - this makes the system sound the most realistic to me. Human hearing is complex...it almost appears humans want some deviation from a perfect curve to sound "real."

So how did I solve the problem? After placing the correction curve to where I liked it, I noted the center F and depth of the remaining troughs. I used another DSP program called Easy Q to apply a boost where appropriate. I am happy with the result! Easy Q seems to work together with Room EQ without any negative artifacts...at least none that I can hear. I'm using a Intel Core 5 that appears to handle the double DSP processing duties without bogging down in the least.

Is it OK for me to continue using the two DSP programs together as I outlined?

Thanks again! 

    

Is it OK for me to continue using the two DSP programs together as I outlined?

Hi Papi,
There is no problem with your setup.
You can use free MathAudio Headphone EQ component for Foobar2000 instead of a conventional equalizer. Headphone EQ can work as a conventional equalizer and it 'always' uses 64-bit floating-point filters - even if you use 32-bit Windows or 32-bit CPU.

Quote from: Papi Chulito on 2020-03-12 04:47:53

Is it OK for me to continue using the two DSP programs together as I outlined?

Hi Papi,
There is no problem with your setup.
You can use free MathAudio Headphone EQ component for Foobar2000 instead of a conventional equalizer. Headphone EQ can work as a conventional equalizer and it 'always' uses 64-bit floating-point filters - even if you use 32-bit Windows or 32-bit CPU.

K2K,

OK, great! Thanks for the tip on your headphone EQ program. I'll try it out!

My only "dream wish" at this point would be for the Room EQ and Headphone EQ programs to be integrated, so that the test tone sweep for Room EQ comes from the Headphone EQ program. This way, I could dial in the lift I need at my sub/main speaker crossover point, and then see the result by sweeping with Room EQ...to make final corrections for room reflections, bass resonant peaks, etc. I think using Room EQ + a "standard" DSP EQ in consort is a better sounding approach than just bringing the Room EQ correction curve down so low that it takes the "life" out of the sound. Just my opinion!  

That combo of integrated programs would be the ULTIMATE solution for every possible application an audiophile might have!

Thanks again!

My only "dream wish" at this point would be for the Room EQ and Headphone EQ programs to be integrated, so that the test tone sweep for Room EQ comes from the Headphone EQ program. This way, I could dial in the lift I need at my sub/main speaker crossover point, and then see the result by sweeping with Room EQ...to make final corrections for room reflections, bass resonant peaks, etc.

You can do it right now by means of using demo versions of VST plug-ins for measurement, saving the resultant presets and loading that presets to the Foobar2000 versions of Room EQ and Headphone EQ. The process is as follows:
1. Install demo version of Reaper (you can download it from https://reaper.fm)
2. Download demo versions of Room EQ VST and Headphone EQ VST from https://mathaudio.com/download.htm
3. Add Room EQ VST to Reaper as it is shown at https://mathaudio.com/images/reaper.png
4. Add Headphone EQ VST to Reaper in series with Room EQ VST to process the sweep tone with the Headphone EQ VST.
5. Measure your room and adjust all the desirable settings of both Room EQ and Headphone EQ.
6. Save the presets of both Room EQ and Headphone EQ as .snr and .hdp files correspondingly.
7. Add both Room EQ and Headphone EQ components to Foobar2000 and load the .snr and .hdp files to the corresponding components.
8. Click the "Apply" button.
That's all.

I just would like to better have the control of what going on during the equalization.
I know the possibility of using the mouse ... but you should know that it has many limitation, low accuracy and practically no way to see exactly what I am doing where... I mean, I cannot dial in precise value I get from another computational model (this just for example).

I second Andrea's wish. I wanted to register Mathaudio for a looooong time, but held back, because it lacks the control one has with other systems. Mathaudio EQ is a "black box" correction. This alone would be fine for me, but I also would like to have control over:

* target curves in a numerical way (for example by importing a custom target curve)
* be able to test different multi-point measurements by
   a) making as many measurements as I want (which are then stored and can be accessed later)
   b) then separately choosing which of those measurements to use for the multi-point calibration

This would give a very effecient way to test if enlarging the multi-point measurement area and / or adding more spots will be benefitial for a specific setup or not.

I wanted to register Mathaudio for a looooong time,

MathAudio Room EQ for Foobar2000 is free. You don't need to register it.

I also would like to have control over:
* target curves in a numerical way (for example by importing a custom target curve)

You can already do it. See the first question at our FAQ https://mathaudio.com/room-eq-faq.htm  You can download bruelandkjaer.targetcurve and manually edit its numerical values. You can also import other target curves in the ".targetcurve" format.

I second Andrea's wish.

Room EQ's filters include many hundreds of coefficients. All of them are interrelated. Changing one coefficient totally change the response of Room EQ in hardly predictable way. You can easily get unpleasant instability issues if you change the coefficients manually. This is the case when an automatic method is much more accurate than a manual one.

* be able to test different multi-point measurements by
   a) making as many measurements as I want (which are then stored and can be accessed later)
   b) then separately choosing which of those measurements to use for the multi-point calibration

You can make as many sets of measurements as you wish and store them as SNR files. After that you can use the "Load preset" button to load them one by one and compare the sound.

Thanks for the answer.
I did not mean Mathaudio for foobar, but the general version which works systemwide.

Great that target curves work now. This is relatively new, was not there when I tested the last time.

My wish on influence was not to be able to configure filters -- I know that a Blackbox approach is given here. My wish was to be able to quicker and easier manage measurements for multiple point calibrations. But it seems this is now also possible. I'll give it a try! :-)

Very curious to see how my tests will turn out! :-)

Thanks for offering the foobar version for free!

Hi K2K, thank you very much for the beautiful software that I use with Foobar that allows you to easily have an excellent sound. With my system, applying the neutral curve the result is very good, but I miss a bit of bass. In your graph the neutral curve appears to be contained from 20 Hz to 20K within 5 decibels, while in the graphs of the bruel & kjaer 1974 Moller curve the curve appears to be within 6 decibels. Maybe that's why it seems to me that a little bit of bass is missing, they should be increased by about a decibel? Thanks again for the wonderful software

Hi K2K, thank you very much for the beautiful software that I use with Foobar that allows you to easily have an excellent sound. With my system, applying the neutral curve the result is very good, but I miss a bit of bass. In your graph the neutral curve appears to be contained from 20 Hz to 20K within 5 decibels, while in the graphs of the bruel & kjaer 1974 Moller curve the curve appears to be within 6 decibels. Maybe that's why it seems to me that a little bit of bass is missing, they should be increased by about a decibel? Thanks again for the wonderful software

Hi Enri-audio, if you you wish to try the Bruel & Kjaer curve, you can download it from our web site. See the first question at Room EQ FAQ: https://mathaudio.com/room-eq-faq.htm

Hi K2K, thanks for the bruel & kjaer curve! I made my hi-fi system with 2 differently positioned subwoofers, two front speakers and two others that play sideways upwards. By controlling the amount of indirect sound there is an almost holographic stereophony. With your system I can simply equalize this complex system, the result is truly excellent, thanks again!
I noticed that if I measure the room with the volume of the sweep a little high, even without any distortion, I get a sound with less low frequencies. if I take the measurement at a lower volume, the graph of the frequency response shifts very slightly downwards but it seems practically the same, yet the bass goes well and the listening is extraordinary. Can you tell me why? Maybe the sweep volume should be similar to the volume used when listening to music? Thank you

Hi K2K, thanks for the bruel & kjaer curve! I made my hi-fi system with 2 differently positioned subwoofers, two front speakers and two others that play sideways upwards. By controlling the amount of indirect sound there is an almost holographic stereophony. With your system I can simply equalize this complex system, the result is truly excellent, thanks again!
I noticed that if I measure the room with the volume of the sweep a little high, even without any distortion, I get a sound with less low frequencies. if I take the measurement at a lower volume, the graph of the frequency response shifts very slightly downwards but it seems practically the same, yet the bass goes well and the listening is extraordinary. Can you tell me why? Maybe the sweep volume should be similar to the volume used when listening to music? Thank you

Hi Enri-audio, dynamic speakers are non-linear devices. They always produce some non-linear distortion even if you don't hear it. Usually the non-linear distortion increases with the volume and affects the frequency response of speakers. Normally the sweep volume should be similar to your listening volume. Too high sweep volume can reduce the accuracy of the measurement. If you wish to accurately measure the non-linear distortion in your audio system, you can use free MathAudio THD Meter which can be downloaded from https://mathaudio.com/download.htm (you need a demo version of Reaper https://www.reaper.fm or any other DAW to use the THD Meter).

Hi k2k, I noticed that by measuring the room, decreasing or increasing the buffer of the input and output card, there is a minimal effect on the sound of the equalization. For now in tests that I made a 512 S buffer it sounds better. Can you recommend the optimal values to use?
Should the buffer of Foobar2000 also be changed, or is it bypassed?
Also I would like to know if there is a stand-alone version of Mathaudio to buy, it would be very useful to me.
Thanks and bye!

Hi k2k, I noticed that by measuring the room, decreasing or increasing the buffer of the input and output card, there is a minimal effect on the sound of the equalization. For now in tests that I made a 512 S buffer it sounds better. Can you recommend the optimal values to use?
Should the buffer of Foobar2000 also be changed, or is it bypassed?
Also I would like to know if there is a stand-alone version of Mathaudio to buy, it would be very useful to me.
Thanks and bye!

Hi  Enri-audio,
The quality of the sound does not depend on the size of the buffer at all, however, if the buffer is too small, old computers may produce pauses between buffers that sound like loud crackles. Foobar2000 and your audio card may use different sizes of buffers - this fact doesn't affect the sound. If you don't hear the loud crackles, the size of the buffer is sufficient and you should not worry about it.
Room EQ can be used as a system wide equalizer on your computer if you need it: https://mathaudio.com/system-wide-eq.htm

Hi k2k, thank you very much for the possibility that Mathaudio provides as a system equalizer. For the difference in sound of the equalization depending on the buffer, I confirm that I have done hours of listening tests on a hi-end system, which confirm that the 512s setting gives the best results. I repeated the measurements several times; I don't know if this is the only parameter that varies, but for now I have no other hypothesis. I tried starting from 32s to more than 1000, the best results are always at 512 s. In any case, what do you recommend? For example, can the size of the buffers influence the synchronization of the signal, between what comes out of the audio system and what is processed by the software? In this case, should we have the lowest buffer, but compatible with the processing capacity of the PC? Or what could be the reason? Thanks again

Hi k2k, thank you very much for the possibility that Mathaudio provides as a system equalizer. For the difference in sound of the equalization depending on the buffer, I confirm that I have done hours of listening tests on a hi-end system, which confirm that the 512s setting gives the best results. I repeated the measurements several times; I don't know if this is the only parameter that varies, but for now I have no other hypothesis. I tried starting from 32s to more than 1000, the best results are always at 512 s. In any case, what do you recommend? For example, can the size of the buffers influence the synchronization of the signal, between what comes out of the audio system and what is processed by the software? In this case, should we have the lowest buffer, but compatible with the processing capacity of the PC? Or what could be the reason? Thanks again

Hi  Enri-audio,
I have no idea why the buffer of 512 samples can sound better than the buffer of 1024 or 2048 samples. Large size buffers help to avoid interruptions of sound on old slow computers, that is why Foobar2000 uses so large default buffer size (1000 ms). Small size buffers help to reduce the latency of audio systems which is important for direct monitoring, live performances and watching video. If you use Foobar2000 for playing music, the latency is not important, so you can safely use a large buffer size. Room EQ processes all buffer sizes without any distinction - every bit of the digital stream remains the same. Try to set your audio card to the largest possible buffer size to minimize the load on the CPU.

 hi k2k, i bought a license to use mathaudio as vst on cubase, where should i put the code to activate it? thank you

 hi k2, i bought a license to use mathaudio as vst on cubase, where should i put the code to activate it? thank you

Also I can't use it with Cubase. If I use mathaudio as a system equalizer, it works if I play an audio file, but not with Cubase. if i load it as cubase vst plugin, it beeps even though i put the key code in the registry.

Also I can't use it with Cubase. If I use mathaudio as a system equalizer, it works if I play an audio file, but not with Cubase. if i load it as cubase vst plugin, it beeps even though i put the key code in the registry.

Hi Enri-audio, the registration button is located in the "About" dialog. If you have any other questions concerning the VST version of the plug-in, please contact us directly at https://mathaudio.com/contact.htm 

News
New version released ( v2.7.7 ).
v2.7.7 uses newer versions of Foobar2000 SDK and Windows SDK.
Windows XP and Windows Vista are no longer supported.

Hello @k2k,
thanks for the news!

However, I am not so expert on the matter of SDK and I would like to know what's the improvement achieved?
I mean, I am running Foobar on Windows 10, and does this v 2.7.7 run better? Less memory usage? Less CPU load?

If I am happy with the v 2.7.6, why I should upgrade it?
Thanks and regards, Andrea

Hello @k2k,
thanks for the news!

However, I am not so expert on the matter of SDK and I would like to know what's the improvement achieved?
I mean, I am running Foobar on Windows 10, and does this v 2.7.7 run better? Less memory usage? Less CPU load?

If I am happy with the v 2.7.6, why I should upgrade it?
Thanks and regards, Andrea

Hello Andrea,
v2.7.7 is compiled with a newer compiler which knows and uses the advantages of the latest types of CPU. Most probably v2.7.7 puts less load on the CPU if you use a last-generation CPU chip, though we did not make such measurements. We updated the plug-in to make it more compatible with future releases of Windows and Foobar2000. We wish to be sure that the plug-in will not stop working after the next Windows update. Our previous ten-years-old compiler was optimized for Windows XP and nobody guarantees its compatibility with future updates of Windows 10. That is why we decided to move to a newer technology.
The new version of Room EQ doesn't bring any changes in the sound, so there are no urgent reasons to upgrade.

Hi, any idea why the measurement sweep signal sounds crackling if Windows default format set on 48 kHz and no crackling on 44.1 kHz using Lexicon Alpha? (I want to show it with a screen recording, but in that no crackling.)

Hi, any idea why the measurement sweep signal sounds crackling if Windows default format set on 48 kHz and no crackling on 44.1 kHz using Lexicon Alpha? (I want to show it with a screen recording, but in that no crackling.)

Nothing matters, either (now default) WASAPI shared or exclusive output used, as soon as the Room Measurement radio button selected the system sounds distorted (with 48 kHz default formats) or noisy even without playback (with 16 bit default formats). The only usable setting is 24 bit and 44.1 kHz.
Standard USB audio device drivers used by Windows 10 for Lexicon Alpha.

Just out of curiosity (because it's a obsolete HW) I tested (v2.7.7) with my workplace PC's VT1705 VIA codec and even the Start Measurement button not working (foobar v1.6.1 default - WASAPI shared - output).

Hi, any idea why the measurement sweep signal sounds crackling if Windows default format set on 48 kHz and no crackling on 44.1 kHz using Lexicon Alpha? (I want to show it with a screen recording, but in that no crackling.)

I don't know the reason of this problem, but the accuracy of measurement doesn't depend on the sampling rate. You can safely use 44.1 kHz for measurement.

Just out of curiosity (because it's a obsolete HW) I tested (v2.7.7) with my workplace PC's VT1705 VIA codec and even the Start Measurement button not working (foobar v1.6.1 default - WASAPI shared - output).

Click the "In/Out" button and select the necessary audio output. Make sure that nothing takes exclusive control of the audio device's driver (including Foobar2000).

I don't know the reason of this problem, but the accuracy of measurement doesn't depend on the sampling rate. You can safely use 44.1 kHz for measurement.

Yes, fortunately accuracy not affected, and at least one default format still works. The whole thing just occurred in the first place because with this device the Windows default is 24/48. It's only a suspicion that one reason might be the rework for the new foobar (beside driver or HW reasons), but without previous version I cannot test this.

Yes, fortunately accuracy not affected, and at least one default format still works. The whole thing just occurred in the first place because with this device the Windows default is 24/48. It's only a suspicion that one reason might be the rework for the new foobar (beside driver or HW reasons), but without previous version I cannot test this.

Most probably this is a driver problem. If you set your audio card to the default sampling rate of 48 kHz, the audio player should be able to switch it to 44.1 kHz when necessary. If the driver doesn't allow switching the sampling rate, it may result in the "crackling". Try to find and install the latest driver for your audio interface. If the driver is not designed specifically for your version of Windows, it is better to set it to the default frequency of 44.1 kHz because 44.1 kHz is the sampling rate of majority of audio files in existence. You can check your audio driver like follows. Set the audio driver to 48 kHz, remove Room EQ from Foobar2000, switch Foobar2000 to a conventional audio driver (not WASAPI) and try to play a conventional audio file with the sampling frequency of 44.1 kHz. If you hear the "crackling", most probably this is a driver problem.

I've just found out that Foobar2000 v1.6.1 uses a new default WASAPI output, so you cannot use it to test DirectSoud drivers. In any case I think that the problem is caused by the audio driver. A lot of audio cards have problems with automatic sample rate switching. This is not a large problem because you can manually set your card to 44.1 kHz for the measurement.

I've just found out that Foobar2000 v1.6.1 uses a new default WASAPI output, so you cannot use it to test DirectSoud drivers. In any case I think that the problem is caused by the audio driver. A lot of audio cards have problems with automatic sample rate switching. This is not a large problem because you can manually set your card to 44.1 kHz for the measurement.

Yes that's what I wanted to emphasize. But I wrongly misinterpreted in your latest change log that foobar's change to WASAPI shared output is one reason for the modernization of the latest version. And that's why I maintained the possibility that Room EQ device access through foobar might be one reason. The driver is the latest, but as I found out only the USB device node using manufacturer components. There is no problem when I play 44.1/48 kHz sources with WASAPI exclusive which demands sample rate change from the device. Anyway, thanks a lot. If I want, I will do DirectSound tests with previous foobar version, and next week I hope I can start the measurements and corrections with your free plugin which I very much appreciate.

foobar2000 1.6.2 (https://www.foobar2000.org/changelog) fixed the problem most likely with this: "Added workarounds for wrong audio mix format reported by certain buggy soundcard drivers. (beta 3, beta 4)". Room EQ does an awesome job with cleaning my most compromised 100-500 Hz range.

MathAudio Room EQ of Main Speakers with Sub

All, what is the best approach for using MathAudio Room EQ with main speakers and a sub? I am using Foobar2000 as the music player (outputting via ASIO or other direct means) and use two DACS: Meridian Explorer and, lately, Khardas Tone board. Also, measuring with a miniDSP UMIK-1. I run the main speakers full on, then via pre, feed the sub which I can pick up where the main speakers roll off. These are Nautilus 804s with overall frequency response is from 45Hz-20kHz +/- 2dB (the crossovers are 350Hz from bass to midrange and 4kHz from midrange to tweeter). The sub is a Def-Tech Super Cube II with frequency response 14-200 Hz and continuously variable low-pass 40-150 Hz crossover. Will set both sub and speaker volumes with pink noise to around 75db (or comfortably loud setting).

This is what I am thinking:

• equalize for main speakers and room
• equalize sub and room probably starting with low-pass setting as dictated by the results, but, figuring somewhere around 50 hz
• equalize mains, subs, and room together
• iterate results around crossover frequency

Does this make sense or is there a better approach to this method and sequence?

If I need to, I can rewire the main speakers to have them be filtered by the sub (introduces the need to to buy or cut my current nice speaker cables into two pieces each; would prefer not to do that).

• equalize for main speakers and room
• equalize sub and room probably starting with low-pass setting as dictated by the results, but, figuring somewhere around 50 hz
• equalize mains, subs, and room together
• iterate results around crossover frequency

Does this make sense or is there a better approach to this method and sequence?

If I need to, I can rewire the main speakers to have them be filtered by the sub (introduces the need to to buy or cut my current nice speaker cables into two pieces each; would prefer not to do that).

The third approach looks preferable. By the way, the frequency response of cheap cables will be corrected by Room EQ, so you don't need to buy too expensive cables.

K2k, I was afraid someone would say that.

Is the right approach to go speakers first, then sub, then both? Or just go speakers, then speakers and sub?

PS my UMIK-1 just came in today.

Is the right approach to go speakers first, then sub, then both? Or just go speakers, then speakers and sub?

There is not much sense in measuring your sub separately. Measure your sub and your speakers simultaneously. Try to adjust the crossover of your sub to make the total frequency response of your audio system as flat as possible. After that move down the vertical slider of Room EQ as it is described at https://mathaudio.com/room-eq.htm

So, I rewired the speaker cables, now using the sub's high-pass cross over. Using pink noise, I level matched the speakers to 75 db, then just the sub. Then brought them together and lowered overall level down to 75 db. I then did some measurements (added the calibration curve as indicated) and iterated the sub high pass filter from about 65 to around ~40 hz. I then brought the sub volume up slightly and fiddled with the crossover a bit. Got it as somewhat flat (okay hilly) as I could. Save that. Then used the neutral curve and brought down the overall level about 8-10 dbs. Saved settings. I did play around with flattening the curve more for a flatter (or neutral curve) response, and I could not really detect any muffled or muddy sound. So, not sure where I will end up with the overall attenuation setting using the neutral curve. Tomorrow I will do some fine tuning and multiple measurements around the sitting position. I am seriously going to have to electrically tape the sub volume and crossover knobs plus the slight change to the balance on the pre so they don't get mistakenly changed. So, a good start, May redo some of the steps and levels and measurements, too. A lot of variables to optimize. So far, I think I heard some improvement in sound, but, I've not done critical listening yet. 

Thank you to K2K for some great work, for making this available to the Foobar2k community, and for your helpful directions and responses (you care about your work and the attentive community). I've also been using the headphone version with my Philips X2s (hope the curve for the X2s is the same as the X2HRs...should be close, anyway). Again, great work.

As many times as I read the instructions, not until I tried it with headphoned and then room eq (measurements w/new calibrated microphone) was I able to see that  it was very intuitive, quite easy, and maddening fun. 

Well, interesting day:

* Blew the tweeter on my right B&W Nautilus 804. Not sure when, but, started seeing drop off on right channel high-frequency  and knew something was wrong. Ear to the tweeter, no sound. Uuuuggh. This happened to me a long time ago, replaced the tweeter, but, now they are tough to find. B&W support indicates low-stock (they say contact them). I emailed them to see if I can buy one (or two). Lesson learned. Don't calibrate at too high a level (be careful of any pops, etc...although I had none). Well, for this exercise, I do have the calibration curves from the night before and earlier in the morning. So the first attached image file is the curves for the N804 calibration (pre-tweeter blow) with the DefTech SuperCube II. Mainly addressed the peaks around 30hz and higher one at 60-70hz.

* Okay, moved my N804s to the rear and pulled up my KEF LS50s. Very nice speakers and I can see fairly linear response; they  just don't provide the fuller sound of the N804s (even with the DefTech Sub). In terms of imaging also think the N804 s provide a better sound stage (maybe its the fuller sound). So, the second attached image file is curves for the Kef LS50 calibration. So, I brought down the overall measurement level as don't need to blow one of these UniQ drivers. I did numerous measurements around the sitting position: nine left mike and nine right mike (not sure it matters if it is one microphone, but the extra measurements don't hurt). These came out pretty good. I again see peak (room) resonances of (slight) 30hz and a larger one around 60hz I I did not see these as pronounced with the N804s, but, they were there. I had iterated the sub volume and filter to bring up the 30hz. So, now brought down the Neutral Room EQ curve to compensate for those two pronounced peaks. Outcome looks good, Sounds good.

Question: for both of these calibrations, are these approximately the proper levels for the equalization? Should they be broght a bit lower? I could go back and further change the sub crossover and volumes and remeasure. 

* Next, I attached a picture of my Foobar2000 setup. Been evolving it from scratch for 10s of years. Tabs for playlists, biography on left. Album cover below that. Main playlist window with song information including codec, bit encoding, frequency, bitrate, dynamic range average for album and specific for song, etc (the measurement meter is included in the setup). Below that, tabs with spectrum analyzer, equalizer, video (for youtube playlists & links), lyrics, selection specifics, & console. To the right, dynamic range plot of the song selection. Below that are tabs for vu meters, coverflow, and Shpeck visualizer.  Song play info also displayed in the status bar at bottom (including HDCD, anyone?).

* Foobar2000 andMathAudio are a lot of processing for an older lap laptop (updated to latest Win10 and patches). I can see that the added MathAudio algorithm, even in normal resolution mode, tends to slow and delay a few things during song start-stops, or selections within Foobar, plus moving cursor around the windows desktop. But, these are manageable.

* Lastly, I added a picture of the entire setup. This is a pretty wide open room (moved ping pond table off to far side), drop ceiling, hard walls, carpeted floor. Speakers about 4-5 feet off the back wall, everything centered to the sofa chair and parallel with the back walls.

Question: for both of these calibrations, are these approximately the proper levels for the equalization? Should they be broght a bit lower? I could go back and further change the sub crossover and volumes and remeasure. 

I'd prefer a lower position of the vertical slider. Try different positions of the vertical slider and use the "Bypass" radio button and the "Bypass signal volume" slider to compare the corrected sound with the original sound as it is described in item 10 at https://mathaudio.com/room-eq.htm  Listen to the processed sound at least 30 minutes, then switch to the bypass mode: this method helps to clearly hear the contrast between processed and original sounds.

It looks like your first pair of speakers has problems with HF crossover (left speaker, 2000 Hz) and LF crossover (right speaker, 200 Hz).

Set your listening volume to measure your speakers.

Thanks again. I read in audiophonids.fr the following: 'Positioning this slider too high can cause too little improvement in frequency response, and too low a position can lead to excessive excursions of your speaker membranes as well as nonlinear distortion.' So, a little confused here. You recommend bringing the MathAudio curve to a lower level (to the valleys across the spectrum). So, does this level create unnatural excursions, distortions, and phase issues? Or is a moderate attenuation a better more natural approach? I hear the difference in sound with my unprocessed setup coming across a bit more punchy while the neutral equalized a bit more, well, neutral. I would like to have the best imaging while not introducing distortion.

Thanks again. I read in audiophonids.fr the following: 'Positioning this slider too high can cause too little improvement in frequency response, and too low a position can lead to excessive excursions of your speaker membranes as well as nonlinear distortion.' So, a little confused here. You recommend bringing the MathAudio curve to a lower level (to the valleys across the spectrum). So, does this level create unnatural excursions, distortions, and phase issues? Or is a moderate attenuation a better more natural approach? I hear the difference in sound with my unprocessed setup coming across a bit more punchy while the neutral equalized a bit more, well, neutral. I would like to have the best imaging while not introducing distortion.

The current setting of the vertical slider (-6 dB) allows your speakers (the first pair) work from 30 Hz. You can make them work from 20 Hz if you set the vertical slider to -30 dB, however, the large resultant excursion of the sub speaker diafragm can produce a substantial amount of upper harmonics. It is always better to not overload your speakers to avoid a muddy sound. I'd prefer the slider position of -12...-13 dB (in your case). If you usually use your speakers on a very small fraction of their maximum power, -15..-17 dB can sound even better. Ideally, you may want to try a few positions of the vertical slider and to find a position which correspond to a most natural and transparent sound.

I can see that the added MathAudio algorithm, even in normal resolution mode, tends to slow and delay a few things during song start-stops, or selections within Foobar, plus moving cursor around the windows desktop.

When you will finish creating  EQ curve, you can create impulse for convolver and use convolver instead of equalizer on playback. It will be much easier for CPU.

Well, now listening to the 2nd pair (LS50s) of speakers until/if I can get the tweeter on the N804s replaced.

Did not mean to indicate I would try to compensate to 20hz. That's  the limiting capability of the sub so I am fine with that.

On the 2nd pair (Ls50s), I was thinking of how much attentuation to apply across the rest of the resonse.

On the last suggestion of creating an impulse curve for a convolver. I will have to learn about that (new to me). I've seen links to convolver plugins, but, any link you can point me to will be appreciated. Thank you.

When you will finish creating  EQ curve, you can create impulse for convolver and use convolver instead of equalizer on playback. It will be much easier for CPU.

- Convolvers use much more CPU than Room EQ.
- Convolvers usually use a rectangular windows that add a specific distortion on low frequencies. The best convolver-based solutions use advanced window functions (Hann, Hamming, Kaiser, etc) that partially reduce the distortion of the rectangular window. In any case a convolver cannot accurately reproduce the frequency response of Room EQ.
- Convolvers add a substantial delay to the sound. Less delay = less accuracy.
- Convolvers operate at a single sampling rate. Theoretically, you need to record impulses for all sampling rates and switch them when necessary to be able to play audio files with different sampling rates. Practically, the convolver-based solutions often use resamplers to solve this problem. Resamplers distort the sound.
- Some convolvers add aduible pre-echo to the sound. Pre-echo came to our life with lossy codecs (MP3, Vorbis, etc) and it doesn't make our life better :D
We would be happy to use a convolver in Room EQ if it could be any better than our current algorithm :D

On the 2nd pair (Ls50s), I was thinking of how much attentuation to apply across the rest of the resonse.

You may want to try -20,  -22 and -24 dB. Most probably -24 dB is the best solution is this case because the original frequency response of your speakers is good enough and "ideal" correction looks possible. In any case, it is better to try all options to be sure that my guess is correct.

- Convolvers use much more CPU than Room EQ.
- Convolvers usually use a rectangular windows that add a specific distortion on low frequencies. The best convolver-based solutions use advanced window functions (Hann, Hamming, Kaiser, etc) that partially reduce the distortion of the rectangular window. In any case a convolver cannot accurately reproduce the frequency response of Room EQ.
- Convolvers add a substantial delay to the sound. Less delay = less accuracy.
- Convolvers operate at a single sampling rate. Theoretically, you need to record impulses for all sampling rates and switch them when necessary to be able to play audio files with different sampling rates. Practically, the convolver-based solutions often use resamplers to solve this problem. Resamplers distort the sound.
- Some convolvers add aduible pre-echo to the sound. Pre-echo came to our life with lossy codecs (MP3, Vorbis, etc) and it doesn't make our life better

What convolver are you talking about? Aren't you confusing convolver with FIR filter? I use https://hydrogenaud.io/index.php?topic=85107.0 You can test it yourself.
It uses significantly less CPU, than MathAudio Room EQ. It doesn't add distortion by itself, it doesn't add pre-echo (it is accurate down to ~-150 dB). And it will accurately reproduce frequency response of Room EQ. It operates at single sample rate,yes. So resampler should be used in DSP chain before it. But resamples do not "distort the sound" unless these are really shitty resamplers. And what do you mean by "Less delay = less accuracy"? Absolute delay doesn't matter when just listening to music.

What convolver are you talking about? Aren't you confusing convolver with FIR filter? I use https://hydrogenaud.io/index.php?topic=85107.0 You can test it yourself.
It uses significantly less CPU, than MathAudio Room EQ. It doesn't add distortion by itself, it doesn't add pre-echo (it is accurate down to ~-150 dB). And it will accurately reproduce frequency response of Room EQ. It operates at single sample rate,yes. So resampler should be used in DSP chain before it. But resamples do not "distort the sound" unless these are really shitty resamplers. And what do you mean by "Less delay = less accuracy"? Absolute delay doesn't matter when just listening to music.

I am not confusing convolver with FIR filter because there is no difference between convolver and FIR filter  :)  Simply compare the mathematical description of convolver with the mathematical description of FIR filter — they are the same: http://musicweb.ucsd.edu/~trsmyth/filters/FIR_Convolution.html
We call it as a convolver when we think in time domain, and we call it as a FIR filter when we think in frequency domain.

The DSP theory states that FIR filters cannot be as fast and accurate (simultaneously) as IIR filters. That is what I meant. Sorry, I don’t discuss any specific products of other developers.

IMO, all real-time resamplers substantially reduce the quality of the sound. Just look at their distortion: https://src.infinitewave.ca/ Best resampling algorithms cannot process the sound "on the fly" because they use too much CPU. That is why Room EQ supports all sampling rates without resampling.

You may not agree with me because I am a mad audiophile :)

You are right that the absolute delay of the audio signal doesn’t matter when we don’t watch video or play musical instuments.

IMO, all real-time resamplers substantially reduce the quality of the sound

Opinions are useless in this case. Results of ABX test are needed.
https://hydrogenaud.io/index.php?topic=3974

TOS 8. All members that put forth a statement concerning subjective sound quality, must -- to the best of their ability -- provide objective support for their claims.  Acceptable means of support are double blind listening tests (ABX or ABC/HR) demonstrating that the member can discern a difference perceptually, together with a test sample to allow others to reproduce their findings.  Graphs, non-blind listening tests, waveform difference comparisons, and so on, are not acceptable means of providing support.

The DSP theory states that FIR filters cannot be as fast and accurate (simultaneously) as IIR filters.

Now, let's go to practice... Here is archive containing 5 files (original; processed by MAREQ; processed by convolver (https://hydrogenaud.io/index.php?topic=85106.0); preset for MAREQ; impulse for convolver) -  https://www.dropbox.com/s/utx3ub1fjgm0pop/mareq%20vs%20convolver.zip?dl=1
Can you find any significant differences between two processed files at least in pictures? Can you hear these differences in ABX test?
Difference in processing speed: ~60x for MAREQ vs ~1000x for convolver (as reported by fb2k's decoding speed test).

P.S. I am not trying to doubt usefulness of Mathaudio Room EQ.

Convolvers use much more CPU than Room EQ.

Do you have numbers that can prove this statement?
Are you aware of fact that there is various ways to implement convolver?
(Uniform or non-uniform partitioned convolutions are one of them)
They all use FFT in one way or another.
I really hope you do not use direct convolution approach.

Quote from: k2k on 2020-12-17 13:14:05

IMO, all real-time resamplers substantially reduce the quality of the sound

Opinions are useless in this case. Results of ABX test are needed.
https://hydrogenaud.io/index.php?topic=3974

We don“t use resamplers in our products because all of them add a measurable distortion to the sound. If your resampler satisfies you, it is fine. I don't see any reasons to argue further.

Quote

Convolvers use much more CPU than Room EQ.

Do you have numbers that can prove this statement?
Are you aware of fact that there is various ways to implement convolver?
(Uniform or non-uniform partitioned convolutions are one of them)
They all use FFT in one way or another.
I really hope you do not use direct convolution approach.

If we try to replicate an IIR by means of an FIR (convolver) without sacrificing such important characteristics as frequency response, phase response and digital noise, FIR filters use more CPU than IIR filters.

For example, a standard second-order IIR filter uses 5 multiplications. If we replicate it by means of a FIR filter, we cannot implement it with 5 multiplications even if we use a FFT. The problem lies in accuracy and digital noise. FIR filter is always less accurate if we use the same number of double-precision multiplications in FIR and IIR.

So you are saying that your software uses IIR? What order of IIR? and are you using second order sections? How to get linear phase out of it?

archive containing 5 files (original; processed by MAREQ; processed by convolver; preset for MAREQ; impulse for convolver) -  https://www.dropbox.com/s/utx3ub1fjgm0pop/mareq%20vs%20convolver.zip?dl=1

Actually, binary comparison reported differences between processed files at ~ -52 dB.
With longer impulse differences are at ~ -108 dB -  https://www.dropbox.com/s/fnmxkwpgqy6qh89/mareq%20vs%20convolver%20new.zip?dl=1 And processing speed for convolver with long impulse is ~140x, still noticeably faster than MAREQ.

So you are saying that your software uses IIR? What order of IIR? and are you using second order sections? How to get linear phase out of it?

Yes, it uses high-order IIR filters. The order is variable.
A resonating room can be considered as a sort of an IIR filter with its poles and zeros. Consequently, the audio system can be considered as two serially connected IIR filters: (1) Room EQ and (2) room. We linearize the phase of the system, not Room EQ only. Room EQ is a sort of an "inverse" IIR filter which tries to correct both amplitude and phase problems of another IIR filter (room). A linear-phase filter is much worse in this sense because it corrects the amplitude response only and unavodably adds some pre-echo to the sound.

You can make as many sets of measurements as you wish and store them as SNR files. After that you can use the "Load preset" button to load them one by one and compare the sound.

Mmh, I can not make measurements and store them individually nand afterwards combine them as multi-measurements. That would be great to test out different multi-measurement configurations.

So what I think would be great is to make any number of measurements. For example a set of L/R measurements for a specific spot. Save that (manually named or just automatically numbered) on the hard disk. _Then_ after the measurement process is complete one would choose (multi-selection) the wanted measurements and Mathaudio would generate the filter accordingly. So one could make 9 measurements for example and then try if the result will improve when placing additional measurements inside the grid, or alternativly outside the grid (so enlarging the grid).

Because Mathaudio Room EQ is a black box solution this would give the users (who want that) a bit of extra parameters to test. Of course one can also do this manually and make different sets of measurements, but a large number of measurements would need to be repeated.

One question: When Mathaudio uses IIR filters it can't create a linear phase. So how das Mathaudio "take care of the phase"?

One question: When Mathaudio uses IIR filters it can't create a linear phase. So how das Mathaudio "take care of the phase"?

Non-causal IIR filters can create a linear phase if necessary. The problem lies in the fact that a standard room never has a linear phase response. MathAudio Room EQ tries to correct the phase response of the room. The term "linear phase filter" looks very attractive, however, linear-phase filters don't reduce the phase distortion of the room. More important, they add some audible pre-echo to the sound. The pre-echo is a very serious problem of linear-phase filters. It is possible that some people like the pre-echo. I'd compare it with the "echo" button on some cheap PC speakers. Some people like it. But it is important to understand that such sound is not honest. Never ever use linear-phase filters if you need a honest sound.

Thanks for the explanation about phase.

What do you think about the idea of allowing to make and save a bunch of single measurements which then can be combined to create a new filter. That would be a huge time-saver and allow much better to fine-tune the correction.

Could be like that:
GUI Option (tick box) → save single measurements (I would just auto-number those or present a field which would allow to override the auto-number)
Then when opening an SNR file one would not choose a single SNR, but use multi-select to choose all wanted measurements to be combined for a new filter. Possibly the single measurements could be named .SNR1 to show that they are meant to be combined for the final filter.

Another additional wish / idea I have would be to show in the GUI which .SNR file is active and to be able to switch to a second one with one click. This has two use-cases:
1) finding the best filter for a given setup
2) quick option to allow to change between two filter setups (e.g. for single-place listening and one for multi-seat correction).

Idea how to implement that: Implement two buttons in the GUI  which show the SNR name in / above those buttons). A click on the button switches to the relevant SNR file. To load the SNR files one could add a button "Load 2nd SNR file" or something like that.

What do you think?

What do you think about the idea of allowing to make and save a bunch of single measurements which then can be combined to create a new filter.
Another additional wish / idea I have would be to show in the GUI which .SNR file is active and to be able to switch to a second one with one click.

You ideas are correct and we already thought about these features. The problem lies in the fact that we wish to keep the Room EQ's interface as simple and understandable as possible. We tried to draw a new interface and it looks too complicated. Besides that professional DAW's already include preset switches for VST plug-ins. It is possible that the Foobar2000 developer is going to include a preset switch too. Such a preset switch could work with all Foobar components including Room EQ.

2) quick option to allow to change between two filter setups (e.g. for single-place listening and one for multi-seat correction).

It is possible that the Foobar2000 developer is going to include a preset switch too.

foobar2000 already supports saving and switching DSP chain presets. For fast switching DSP Switcher can be added to Toolbar, or DSP chain(s) can be assigned to button(s) in Toolbar or to keyboard shrortcut(s).

foobar2000 already supports saving and switching DSP chain presets.

Great, I was not aware of it.

Another additional wish / idea I have would be to show in the GUI which .SNR file is active and to be able to switch to a second one with one click. This has two use-cases:
1) finding the best filter for a given setup
2) quick option to allow to change between two filter setups (e.g. for single-place listening and one for multi-seat correction).

I've just tested the Foobar's "DSP chain presets" feature and it perfectly works with Room EQ, so you can use it to switch between two or more Room EQ presets.

I do use the 'DSP chain presets' and have the drop down to pick the correct setting on my main Foobar2000 window. However, I believe that doesn't carry to the specific MathAudio Headphone or Room EQ preset. Those presets I change for various headphones, room. and speaker settings. Small effort given that the two Foobar2000 versions of MathAudio are provided at no cost.

I do use the 'DSP chain presets' and have the drop down to pick the correct setting on my main Foobar2000 window. However, I believe that doesn't carry to the specific MathAudio Headphone or Room EQ preset.

Foobar2000 is able to memorize the specific Room EQ and Headphone EQ presets by means of the "DSP chain presets" feature. I've tested this feature and it is very convenient. Foobar2000 already works like a professional digital audio workstation (DAW): professional DAWs are able to memorize all specific settings of audio plug-ins. It is important to "apply" the current preset by means of the "Apply" button before saving it as a "DSP chain preset".

Then you've corrected my post and appreciate the update. I will start to save specific DSP chains with varying MathAudio settings. Thank you for that.

Quick question: is there a way to create a windows startup command or shortcut that calls the a specific saved DSP chain? Just curious.

Quick question: is there a way to create a windows startup command or shortcut that calls the a specific saved DSP chain? Just curious.

I never tried the foobar's command-line options, so I don't know whether it is possible or not.

Ok, so...does MathAudio support kernel streaming in Foobar? I use kernel streaming with the M2Tech Young driver with Hiface. When I check the In-Out/Output Device, it shows "SPDIF Interface-Young 1.0", but not the kernel streaming option.

Ok, so...does MathAudio support kernel streaming in Foobar? I use kernel streaming with the M2Tech Young driver with Hiface. When I check the In-Out/Output Device, it shows "SPDIF Interface-Young 1.0", but not the kernel streaming option.

Yes, kernel streaming is supported because Foobar2000 supports KS: https://www.foobar2000.org/components/view/foo_out_ks  
Please note that the "In/Out" menu lists the devices that can be used for measurement only. This menu doesn't list all the devices that can be used for the playback.

News
New version released ( v2.7.8 ).
A minor improvement in the signal processing algorithm.

I have just started using the MathAudio RoomEQ plugin for Foobar on my PC, output going to Dx3Pro via USB. As the output level going to the DX3Pro is now significantly lower, I am wondering if it is OK to increase the signal to the DAC by upping the Preamp gain in the Foobar settings? The screenshot below shows Foobars VU meters playing at a very low level now. Should I just up the preamp settings such that the VU meters are at a more normal level. (Prior to the use of the MathAudio plugin, I had the preamp settings at - 6db as I read somewhere that this is best to avoid any chance of clipping.) Unfortunately my knowledge of digital clipping/gain matching between Foobar and DAC is minimal (at best!)...
Anyway, any help or advice would be most appreciated. Thanks in advance!

I have just started using the MathAudio RoomEQ plugin for Foobar on my PC, output going to Dx3Pro via USB. As the output level going to the DX3Pro is now significantly lower, I am wondering if it is OK to increase the signal to the DAC by upping the Preamp gain in the Foobar settings? The screenshot below shows Foobars VU meters playing at a very low level now. Should I just up the preamp settings such that the VU meters are at a more normal level. (Prior to the use of the MathAudio plugin, I had the preamp settings at - 6db as I read somewhere that this is best to avoid any chance of clipping.) Unfortunately my knowledge of digital clipping/gain matching between Foobar and DAC is minimal (at best!)...
Anyway, any help or advice would be most appreciated. Thanks in advance!

Ideally, you should set all Foobar's gain/volume controls to 0 dB. After that adjust the position of the "Room EQ gain" slider in accordance with the instruction (see step 15 at https://mathaudio.com/room-eq.htm ). Use the volume knob of your audio amplifier to adjust the comfortable volume. All digital volume controls on your PC reduce the signal-to-noise ratio, so it is better to not use them at all (set all of them to 0 dB).

Great, many thanks for your help. Have been reading up on digital volum, gain matching etc, so think I now have a clearer picture of what is going on. One thing that I am still wondering: presumably if I leave Foobars Preamp gain at 0, as opposed to the -6db previously, even if some tracks are over compressed/too high (gain wise), any digital clipping that may occur will only cause a lack of fidelity ie there is no chance of damage to tweeters, as with analogue clipping in an amplifier? (Not too fussed about a few badly mastered/levelled tracks sounding rough but definitely not keen on any risk to speakers!)
Also, thank you so much for a truly awesome product. I am what I like to call a "budget audiophile" and given all I needed to buy was a second hand UMIK-1, RoomEQ is by far the best improvement to my system in the last ten years. Took me a couple of tries to get my twin subs and Tannoy V12s dialed in, volume-wise, but I have to say that the improvement in clarity is fantastic. Also, thankfully my amplifier, a Lazarus H-1A has plenty of headroom to run it. Reckon my system is now the best I've heard... And all for well under £1k.

Thank you! 👍

Great, many thanks for your help. Have been reading up on digital volum, gain matching etc, so think I now have a clearer picture of what is going on. One thing that I am still wondering: presumably if I leave Foobars Preamp gain at 0, as opposed to the -6db previously, even if some tracks are over compressed/too high (gain wise), any digital clipping that may occur will only cause a lack of fidelity ie there is no chance of damage to tweeters, as with analogue clipping in an amplifier? (Not too fussed about a few badly mastered/levelled tracks sounding rough but definitely not keen on any risk to speakers!)
Also, thank you so much for a truly awesome product. I am what I like to call a "budget audiophile" and given all I needed to buy was a second hand UMIK-1, RoomEQ is by far the best improvement to my system in the last ten years. Took me a couple of tries to get my twin subs and Tannoy V12s dialed in, volume-wise, but I have to say that the improvement in clarity is fantastic. Also, thankfully my amplifier, a Lazarus H-1A has plenty of headroom to run it. Reckon my system is now the best I've heard... And all for well under £1k.

Thank you! 👍

If you adjust  the "Room EQ gain" slider in accordance with the instruction, digital clipping will not happen.

Fantastic. Thanks for your help. Had read several conflicting posts on various sites re digital gain/clipping... Your advice (and revisiting step 15) has made it much clearer...and simpler.
Many thanks again. 👍

Loving the improvement in clarity using RoomEQ generally but now thinking adding a third sub as I have a BK XXLS400 that I use with my AVR as it goes deeper than the two 10" subs that I use for stereo which don't reach much below 40Hz. I would be placing the Bk between the two 10" subs (which have both left and right main speakers on top of them.) Just wondering what I should set the BK crossover at prior to measuring with RoomEQ. I would like to make sure that the BKs location is not audibly noticeable, so wondered if setting it's crossover somewhere between 40 and 80Hz might be best? Anyway, thanks again. Any advice most appreciated.

Use Room EQ to measure the frequency response of your room with a few different settings of your sub. Compare the resultant "unprocessed" frequency responses. The best setting of your sub corresponds to the flatter "unprocessed" frequency response.

Thanks. Settled on a 40Hz crossover for the central BK sub as there was no real gain from having it any higher. Am I right in thinking that the phase setting is irrelevant as RoomEQ accounts for phase differences? It has a dial from 0-180 and I just left it in the middle of the range. Am loving the introduction of deep bass without the integration faff prior to MathAudio. Thanks again! 👍

Thanks. Settled on a 40Hz crossover for the central BK sub as there was no real gain from having it any higher. Am I right in thinking that the phase setting is irrelevant as RoomEQ accounts for phase differences? It has a dial from 0-180 and I just left it in the middle of the range. Am loving the introduction of deep bass without the integration faff prior to MathAudio. Thanks again! 👍

The correct phase setting can improve the "unprocessed" frequency response near the cut-off frequency, so you may want to make a few measurements with different phase settings and choose the phase setting which corresponds to the best "unprocessed" frequency response.

Thanks. After a bit more tweaking of levels, crossovers and phase settings I have arrived at the RoomEQ setup shown in the screenshot below. Presumably it is best to try to keep the lowering of the slider to a minimum (to maximise SNR)? Not quite sure what the best compromise is as, if RoomEQ is set to normal resolution, the slider can generally be set much higher. Presumably this compromise of SNR vs response flatness, is really down to personal preference and audio hardware capabilities? Reckon will spend the weekend playing with settings and see what I prefer. Many thanks for all of your help.

Thanks. After a bit more tweaking of levels, crossovers and phase settings I have arrived at the RoomEQ setup shown in the screenshot below. Presumably it is best to try to keep the lowering of the slider to a minimum (to maximise SNR)? Not quite sure what the best compromise is as, if RoomEQ is set to normal resolution, the slider can generally be set much higher. Presumably this compromise of SNR vs response flatness, is really down to personal preference and audio hardware capabilities? Reckon will spend the weekend playing with settings and see what I prefer. Many thanks for all of your help.

Your "unprocessed" frequency response became much better than the previous one. Yes, the best position of the vertical slider depends on the audio hardware capabilities. If you move the vertical slider down, you improve the frequency response. If you move the vertical slider up, you reduce the amplitude of the speaker diaphragm movement (too large amplitude results in non-linear distortion if your speakers are not powerful enough). The best position of the vertical slider corresponds to the maximum 'fidelity' and 'transparency' of the sound. Try a few positions of the vertical slider to find the best one. See steps 10 and 11 at https://mathaudio.com/room-eq.htm  for more details.

Thanks. This is pretty what I thought. I would like to think that my main speakers will be okay as they are actually made for high power/PA applications and therefore made to handle high excursion. I am actually more concerned for the two small subs underneath them as they are relatively low power and made hi-fi/home audio use....Anyway, have been testing at fairly high volume (due to my neighbours still being stuck in Holland post Covid). Anyway, absolutely loving the sound with a bottle of wine... and no sign of driver over excursion as yet. Many thanks. 👍

... Quick update... Settled on the -15db slider setting as it seems like a good compromise and still gives me a frequency response down to 20Hz. Minor tweaking of the BK sub phase has tightened up the bass too. Quite a stunning difference pre and post MathAudio...
Many thanks for your help and an amazing piece of software.
My shout for beers if any of you are in Edinburgh any time.
Nice one. 👍

My shout for beers if any of you are in Edinburgh any time.
Nice one. 👍

Thanks!   :D

STRANGE PROBLEM WITH ROOM EQ 2.7.8 IN "HIGH RESOLUTION" MODE & "SOLUTION" (PROBABLY NOT THE BEST...)

Hello @k2k , I am experiencing a strange problem when playing .WV (WavPack) files having Room EQ v2.7.8 in the DSP chain: I am having 1 or 2 micro-pauses every second (having Foobar output buffer lenght at its default).

This problem occurs only when 1) I am playing .WV file, 2) I have the DSP chain running above 48kHz rate (to be honest, I never tested over 96kHz, so I should say that I have the problem at 88.2 and 96kHz rate) and 3) Room EQ is in "high resolution" mode.

The original WV files I am using are all recorded at 44.1kHz and I up-sample to 96kHz using the "Resampler DSP component" as the first component of the DSP chain. These WV files are perfect and the WAV and FLAC derived from those can be perfectly played.

The problem (the micro-pauses) does not occur setting Room EQ in "normal resolution" mode and/or playing any other audio format I regularly use (as FLAC, APE, MP3, SACD.ISO) in "high resolution" mode.
As well as the problem does not occur increasing the Foobar Output Buffer to 1000ms (in my case)

You can find more details about the troubleshooting of this problem in my original thread " BUG: No way to play WavPack !!! - Problem with FB 1.6.9 and 1.6.8" at https://hydrogenaud.io/index.php?topic=121991.new#new

Happy to support your further debugging / testing as you will instruct me.
Looking forward for a definitive solution (if possible)...
Regards, Andrea

Quote from: Papi Chulo on 2019-10-25 04:33:55

Can you please tell me, if the Room EQ software will only work through my laptop line audio and/or USB outputs?
I ask because I have a home-network based audio system.

The problem of network based systems lies in the fact that they substantially delay the sound. This fact creates a problem for the Room EQ's measurement subsystem. If your DAC has a USB input, try to temporarily use a USB cable to measure your room. After that you should be able to use your network system exactly as you did it before.

Network based systems are very common these days. Would it be possible to measure the delay first and then do the normal measurements? Cheers.

STRANGE PROBLEM WITH ROOM EQ 2.7.8 IN "HIGH RESOLUTION" MODE & "SOLUTION" (PROBABLY NOT THE BEST...)

Hello @k2k , I am experiencing a strange problem when playing .WV (WavPack) files having Room EQ v2.7.8 in the DSP chain: I am having 1 or 2 micro-pauses every second (having Foobar output buffer lenght at its default).

This problem occurs only when 1) I am playing .WV file, 2) I have the DSP chain running above 48kHz rate (to be honest, I never tested over 96kHz, so I should say that I have the problem at 88.2 and 96kHz rate) and 3) Room EQ is in "high resolution" mode.

The interrupted audio stream is the result of the CPU overload. Room EQ consumes more CPU in the "High resolution" mode. Room EQ also consumes more CPU when it processes the high-sample-rate audio. Make sure that the buffer size is no less that 500...1000 ms. If this doesn't help, you have three options: (1) use the "Normal resolution" mode, (2) don't use upsampling, (3) use a more powerful computer. In my opinion the option (2) is preferable.

Network based systems are very common these days. Would it be possible to measure the delay first and then do the normal measurements? Cheers.

A delay of 200 ms (or smaller) is acceptable. We tested two network based systems a few years ago and the delay of each of them was too large (more than 500 ms). Unfortunately we did not test modern network based audio systems, so I don't know whether or not they can be used for Room EQ measurement.

A delay of 200 ms (or smaller) is acceptable. We tested two network based systems a few years ago and the delay of each of them was too large (more than 500 ms). Unfortunately we did not test modern network based audio systems, so I don't know whether or not they can be used for Room EQ measurement.

I tried to measure the delay and it is the 250-300ms range over Powerline. With a more direct network connection it could be even less. Maybe UPnP devices can just be added for Room Measurement?

I tried to measure the delay and it is the 250-300ms range over Powerline. With a more direct network connection it could be even less. Maybe UPnP devices can just be added for Room Measurement?

You can try to use VST or VST3 versions of Room EQ for room measurement because they can handle the delay of up to 500 ms. A demo version of Room EQ VST/VST3 is enough for measurement. You can use it with a demo version of Reaper: https://reaper.fm   See the instruction at https://mathaudio.com/images/reaper.png  Use the "Save preset" button in Room EQ VST/VST3 to save the result of your measurement, then use the "Load preset" button to load it to the Foobar's version of Room EQ.

Quote from: a3aan on 2022-01-13 22:19:18

I tried to measure the delay and it is the 250-300ms range over Powerline. With a more direct network connection it could be even less. Maybe UPnP devices can just be added for Room Measurement?

You can try to use VST or VST3 versions of Room EQ for room measurement because they can handle the delay of up to 500 ms. A demo version of Room EQ VST/VST3 is enough for measurement. You can use it with a demo version of Reaper: https://reaper.fm   See the instruction at https://mathaudio.com/images/reaper.png  Use the "Save preset" button in Room EQ VST/VST3 to save the result of your measurement, then use the "Load preset" button to load it to the Foobar's version of Room EQ.

Thanks, I figured it is easier to rebuild my asio usb audio signal path.

Is there a way to ensure that RoomEQ loads the last used configuration file? I have different files for different amplifiers and for some reason RoomEQ always loads the same file. I would have thought that the default would be the last used one. Thanks in advance. 👍

Is there a way to ensure that RoomEQ loads the last used configuration file? I have different files for different amplifiers and for some reason RoomEQ always loads the same file. I would have thought that the default would be the last used one. Thanks in advance. 👍

Simply click the "Apply" button (see the question No 18 at https://mathaudio.com/room-eq-faq.htm ).

Thanks. But I can't see an "Apply" button!

Tbh This is such a minor issue that it doesn't really matter. It is no problem to load the relevant configuration file when Foobar starts.

Thanks. But I can't see an "Apply" button!

The "Apply" button is located on the Foobar's DSP Manager. See the right red arrow at the following image
https://mathaudio.com/images/dsp-manager.png

Doh. Sorry!

News
New version released ( v2.7.9 ).
The left mouse button can be used for drawing straight lines. The right mouse button is not used any more to comply with the VST3 standard.

Nice one, thanks!

Any particular update instructions? Do you need to uninstall the older version first or anything?

Nice one, thanks!

Any particular update instructions? Do you need to uninstall the older version first or anything?

You don't need to uninstall the old version. The instruction is located inside the foo_room_eq.zip file.

News
New version released ( v2.8.0 ).
Some changes are made to ensure compatibility with future versions of Windows.
New feature: Room EQ works as a simple conventional mouse drawing EQ if necessary: just load the simple_eq.snr (https://mathaudio.com/simple_eq.snr) preset and mouse draw any EQ curve you want.

Fantastic...

Many thanks!
:))

foo_room_eq version:
MathAudio Room EQ 2.7.9 (foo_room_eq) (2022-02-09) ?
https://mathaudio.com/download.htm

Have tried to update several times and after update Foobar still shows version 2.7.9.

Any ideas?

Have tried to update several times and after update Foobar still shows version 2.7.9.
Any ideas?

Oh, sorry, the Windows versions were not updated on the web site. I've just updated them. I apologize for the inconvenience.

Still coming up with 2.7.9 for some reason...
Any ideas?

Still coming up with 2.7.9 for some reason...
Any ideas?

I've just downloaded foo_room_eq.zip from our web site. The version is correct. Please make sure that you downloaded the file from our official website https://mathaudio.com/download.htm

Hate to say it, but still shows 2.7.9...

Has anyone managed to get 2.8?

Link in previous post works for me...

(https://i.imgur.com/uB3ziz1.png)

Weird. Doesn't work for me. Have tried several times with that link. Tried uninstalling first too. No luck.

Very strange indeed.

Try to clear your browser cache or use a different browser (?)

Weird. Doesn't work for me. Have tried several times with that link. Tried uninstalling first too. No luck.
Very strange indeed.

Calculate the hash of the downloaded file and compare it with the correct hash:
SHA1:  c452cbab5e24bb55375498af3cb750eca02cf866
MD5:  c7e4364c33f60b6767f099cf943bd65f
CRC32:  fc63ea39
If the hash is correct then the file is correct.
SHA1 can be calculated as follows: copy the downloaded foo_room_eq.zip file to C:\ directory and run the Windows command
certutil -hashfile “C:\foo_room_eq.zip” sha1

Cleared my browsers cache and it worked!

Many thanks for your help.
Much appreciated! 👍🏿

News
New version released ( v2.8.1 ).
The new version is compatible with both x32 and x64 versions of Foobar2000.

What is added or changed in version 2.8.1
Thanx

What is added or changed in version 2.8.1
Thanx

The new version supports the x64 version of Foobar2000 (Foobar2000 v2.0 beta is available in both 32-bit and 64-bit versions).

having crackling noise problems when using MathAudio Room EQ on two different windows installations on two different foobars.
Is there any fix for this? older versions didn't seem to suffer from this issue.
PS: audio is not clipping.

having crackling noise problems when using MathAudio Room EQ on two different windows installations on two different foobars.
Is there any fix for this? older versions didn't seem to suffer from this issue.
PS: audio is not clipping.

Crackling can be caused (theoretically) by extremely low size of audio buffers. Make sure that you don't use the minimum possible size of buffers in your audio interface. If this is not the case, please send us more information about your setup: type of audio card, type of CPU, version of Windows, size of audio buffers in Foobar2000. You can find our contact information at https://mathaudio.com/contact.htm

Greetings,  Just came across this, looks very interesting ,going  to give it a try.  I have not made it all the way through this thread yet, but working my through it.  I am ordering a Mic today, looks like it comes with a 1.5M USB cable.  To position it at my listening location, I would need a to have a 25' cable.  I can order an active 32' USB extension, any problem with that?

Thanks, looking forward to the results.

Greetings,  Just came across this, looks very interesting ,going  to give it a try.  I have not made it all the way through this thread yet, but working my through it.  I am ordering a Mic today, looks like it comes with a 1.5M USB cable.  To position it at my listening location, I would need a to have a 25' cable.  I can order an active 32' USB extension, any problem with that?

Thanks, looking forward to the results.

Most probably it should work, but you may want to ask the microphone manufacturer to be on the safe side.
Please post your results on this thread because this information can be interesting to other users.

Sent question miniDSP, still waiting for a response.  I did speak with support at PartsExpress, they did not think it would be an issue.  Will post when/if I get a response from miniDSP.

Finally got a response from miniDSP - "Unfortunately we've never tested this specific cable but we don't think it would be an issue. It's a standard active cable that typically solve the length limitation seen on USB.
Hoping this info helps and feel free to contact us if you have further questions. "

 Everything has ben ordered, will see how it works.

Received the UMIK-1 and Active USB cable today, works fine.  I did a quick test using REW SPL meter to check response with and without the Active USB cable, no difference.  Will begin setting up MathAudio tonight!

So I finished my first pass at setting up MathAudio, a significant change for the better.  Much cleane at all levels/frequencies than how  I was configured.

Previous config - From my Server, I run to my DAC, then into a Pre/Pro (with Audyssey) running in "All Channel Stereo into a 5 channel Amp.  I am running 2 sets of speakers and 2 Subs. 

With MathAudio running, Audyssey is turned off, running the Pre/Pro again in "All Channel Stereo".  Bass response is much cleaner, very crisp.  Mids and Highs are balanced, overall just a much cleaner sound.  The biggest change I can hear is listening to Brahms Piano Concerto #1.  The start of this is very intense, great deal of percussion/low end and the rest of the Orchestra just cranking.  Before it was just noise, could not really make out individual instruments.  Now, clean/distinct.  I still have more listening to do, perhaps a bit more tweaking, but all I can say is using this with Foobar is superior to using Audyssey in my Processor.

Received the UMIK-1 and Active USB cable today, works fine.  I did a quick test using REW SPL meter to check response with and without the Active USB cable, no difference.  Will begin setting up MathAudio tonight!

Do you use a 32' active cable with your UMIK-1 microphone? Is it a USB 3.0 cable?

Yes I used the 32' active USB cable, it is USB2.0, not 3.0.  Here is link to the cable - https://www.monoprice.com/product?p_id=6149

Yes I used the 32' active USB cable, it is USB2.0, not 3.0.  Here is link to the cable - https://www.monoprice.com/product?p_id=6149

Thank you.

HI,  I repositioned my speakers and want to re-run the measurements and room correction.  Do I need to clear the existing config/profile or just re-take the measurements and will that produce a new room profile?  I have saved my current config as a preset, just want to make sure I don't mess anything up.

HI,  I repositioned my speakers and want to re-run the measurements and room correction.  Do I need to clear the existing config/profile or just re-take the measurements and will that produce a new room profile?  I have saved my current config as a preset, just want to make sure I don't mess anything up.

You don't need to clear the existing profile. Simply run the new measurement and your old data will be erased automatically.

Great, thanks.

I am using Mathaudio for Foobar since a while, with different speakers sets, and I am very happy about the results, especially with regard to imaging which becomes much more stable and focused than before. I assume this has to do with the phasing capabilities of Mathaudio. Unfortunately I could not find much on the internet about how Mathaudio operates with regard to phase and delay.

Now, I am in the process of designing a 4 ways horn loudspeaker system.
One of the drawback of multiways horn system is the phase  and delay issues at  the crossover points, which ideally would require measuring of frequency response and phase around the crossover point and then design a passive crossover which corrects the phasing/delay issues.
Now since this type of filters are quite complicated to implement (and most times implies a trial and error approach) I wonder if Mathaudio would be capable to address the phasing issue in the described scenario.
In other words, if I implement a simple Linkwitz-Riley second order passive crossover, will Mathaudio be able to address the phasing and delay issues among the 4 elements (bass, midbass, midhigh, high) of the loudspeaker ?
Keep in mind that the phase rotation/delay is remarkable between some elements, such as the woofer which is in direct radation, and the mid bass horn which is approximately 160cm long.

Room EQ 'partially' compensates for disadvantages of crossovers. In other words, use best possible crossovers to get the best possible sound.

I am planning to correct the crossover related phase shifts with rePhase, which generates a convolve file.
Is it possible to use both the convolve file and Mathaudio from foobar ?
The idea is to use the convolve file generated by rephase just to correct the xover phase shift and let Mathaudio take care of all the rest (in room frequency response and room induced phase shifts).

I am planning to correct the crossover related phase shifts with rePhase, which generates a convolve file.
Is it possible to use both the convolve file and Mathaudio from foobar ?
The idea is to use the convolve file generated by rephase just to correct the xover phase shift and let Mathaudio take care of all the rest (in room frequency response and room induced phase shifts).

Foobar2000 allows one to use two or more DSP effects in a single chain. I am afraid that measuring the crossover induced phase shifts is not a simple problem.

I have been reading a paper regarding sweeps for room correction, which states that due to average rooms (in household environment) having  a Q factor or 20 or more the time duration of the sweep is very important in order to achieve an accurate correction and to minimize drops in displayed amplitude and shifts in displayed frequencies.
The paper suggests ideal sweep times of at least one minute to maximize correction accuracy, in household environment.
I wonder if in a future release of this wonderful piece of software will be implemented the option to set the duration of the sweep, or if there is some workaround so that an external sweep of the desired duration can be used within the Mathaudio measurement tool.

I attach a link to study, which is in Italian, but Google translate does a pretty good job of translating the paper.

http://mariobon.com/Glossario/___Teoria_dei_segnali/Sweep.htm (http://mariobon.com/Glossario/___Teoria_dei_segnali/Sweep.htm)

I have been reading a paper regarding sweeps for room correction, which states that due to average rooms (in household environment) having  a Q factor or 20 or more the time duration of the sweep is very important in order to achieve an accurate correction and to minimize drops in displayed amplitude and shifts in displayed frequencies.

The statements in that paper are true for the specific measurement technique which is mentioned in the paper. Room EQ uses a different technique which allows it to use relatively short sweep signals without sacrificing the measurement accuracy.

News
New version released ( v2.8.3 ).
The new version avoids the "SHGetFolderPath Error" message on French version of Windows 11.

News
New version released ( v2.8.4 ).
A bug is fixed which affected the correct work of the vertical slider through external MIDI controllers in VST and AU versions.

Hi. Just wondering if it is possible to load preference/target curves in to RoomEQ?
I would like to try some that I have downloaded. I have them in both csv and txt format. Any help would be most appreciated. Thanks!

Hi. Just wondering if it is possible to load preference/target curves in to RoomEQ?
I would like to try some that I have downloaded. I have them in both csv and txt format. Any help would be most appreciated. Thanks!

You cannot load target curves in .csv and .txt format. However, you have the following options:
- You can use the default "Bright" or "Neutral" target curves and edit them as you want.
- You can use B&K or Harman target curves and edit them as you want. See the question 1 at https://mathaudio.com/room-eq-faq.htm
- If you need a conventional EQ with a specific curve, you can load the https://mathaudio.com/simple_eq.snr preset and edit it as you want.
You can also use microphone calibration curves in .csv and .txt format.

Great. Thanks for your help!

Happy to help.

Quick question re volume control. Apologies if this has been covered already...
Is the best policy to have Foobars volume maxed out and then trim the RoomEQ gain by clicking on the red "clipping dot" so as to avoid clipping?

Thanks in advance. 👍🏼

Quick question re volume control. Apologies if this has been covered already...
Is the best policy to have Foobars volume maxed out and then trim the RoomEQ gain by clicking on the red "clipping dot" so as to avoid clipping?

Thanks in advance. 👍🏼

Yes. This method guarantees the maximum possible signal-to-noise ratio on the output of your DAC.

News
New version released ( v2.8.5 ).
Support of MIDI commands is added to "Save preset" and "Load preset" buttons in the VST/VST3/AU versions.