Please read FAQ before…

About this website

Read the procedure before jumping on measurements ;)

Hereunder is a short description of the whole procedure and you will find more detailed infos in FAQs.

Step 1 Register

Step 2 “Upload recording” page

  • Download test soundfile, choosing stereo or multichannel
  • Play the file in stereo on your loudspeakers and record simultaneously with your microphone.
  • Start recording and just after start playing : there is a 5s starting silence so you have time to get to the listening position
  • Follow instructions given by the voice of the soundfile
  • Fill the form and save, detailed instructions here
  • Upload your recording
  • If you have a mic calibration file, send it together with the recording

Step 3 Get your measurements

  • A few minutes after uploading your recording, your graphs will appear in All measurements  and also in My measurements
  • If  you have not yet bought a plan, you will only get a graph to check the validity of your recording


  • Without registering, you are a visitor and you can see Measurements and read FAQ.
  • Registering is completely free and gives you member status and access to pages “Upload recording“, “My files” and “My profile”.
  • When registering, you will have to fill a field “Company or Title”. This will be the name of the main folder for all your measurements, images and subdirectories.  The folder structure will be : \title\studio\date\ with dates automatically added and “Studio or reference” being a field to complete when uploading a recording. Having dates helps you to check evolution of performances in time and see any degradation.
  • Please note that the field “Company or title”, can only be related to one user and cannot be used later by another user (unless they are from same company, then please ask us). You cannot use a company name for which you have no rights : your account may be terminated without notice.
  • Accepted are alphabets, numbers, underscore “_” and spaces (but no space at beginning or end and no consecutive spaces).
  • Do not forget to finalize registration by clicking on the link in the mail you’ll receive.
  • When you are registered, you can use the “Upload recording” form to test if your recording file is accepted and valid.
  • If everything is ok, you can buy a plan to get the full results
  • very complete for professionals but simple enough for amateurs !
  • a procedure that avoids mistakes to obtain valid graphs and easy to compare to others
  • MMM measurement is completed with other signals to check amplitude but also phases
  • the MMM method gives results nearer to what is really perceived than other methods
  • the target correction is automatically processed from your measurement
  • target can be finely adjusted
  • advanced FIR correction is calculated for the processor or software you are using
  • this is the only method that is optimised for separated channels but also for channels driven simultaneously : this is much better in low frequencies
  • a full report is available in .png pictures
  • with just one sound file, a full set of analysis and graphs is calculated and gives a very complete view to understand the room and loudspeakers and their interaction, and a good base to calculate corrections to better your listening experience
  • various people getting exactly same graphs give the possibility to easily compare different setups
  • extensions to 7.1, 9.1.4, 22.2, Atmos,… systems will come, including DCP sound files
  • you may even use your phone and mic to record

MMM is a measuring method with a microphone moved by hand near the listening place : response obtained with this method gives a reliable representation of audibility.

The most complete infos about MMM method, I published it a long time ago…

Other readings :

MMM in video : interesting because for loudspeaker tests, MMM (called Moving Mic Average by Erin) is compared to Klippel’s results : it proves that MMM is very comparable to Harman’s In-Room or CEA-2034 prediction based on anechoic measurements.

The test track is a sequence of various signals that are choosen to permit all analysis seen on the graphs :

  • start sequence with synchronisation signal and identification frequency
  • log sine sweeps (Farina signal) for impulse responses, spectrograms, wavelets, distortion,…
  • short bursts for timing analyis
  • pink noises for MMM for independant channels and simultaneously. The pink noise is in conformity with SMPTE standard ST-2095-1, see also here :
  • if you set level so that the starting voice is at at reasonable level (like a person speaking in the room), the level of the sweeps will be near 75dBC, the pink noise per channel will be between 75 and 80dBC and the simultaneous pink noise on both channels between 80 and 85dBC.

Better measurements can give better corrections : generally when you EQ based on a measurement that is not conform to what we hear, the correction may worsen the listening experience ! That’s why we think MMM is a safe and reliable way to mesure and base correction.

If you do important changes to your room, you should redo measurements and calculate another EQ.

If you change your loudspeakers positions, you should redo measurements and calculate another EQ. Most changes will affect frequencies under 300Hz.

REW, ARTA, Rephase,… are very nice measurement or correction softwares. But both need some know-how to get valid results. This takes time. And it is difficult to compare between various audio sytems because so many parameters can be different. method is quicker, the process gives results that are valid and comparable to others users measurements. But this website and REW, ARTA,… can be nicely complementary.

It is not our job to recommend some brands. But generally the best loudspeakers come from companies that publish complete technical datas. Just check their websites…

Really recommended, complete and easy to understand:

Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms, by Floyd Toole’s

The best information comes from Audio Engineering Society, some information is free but you need to register and pay a yearly fee to have access to the whole library.

Very recommended about FIR correction is Denis Sbragion DRC documentation.

I would also recommend to read David Griesinger, Earl Geddes and JJ Johnston presentations, books and articles.


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Some ideas, but might take some time :

  • check and improve performance metrics
  • allow more time at start for recorded file
  • import other mic calibration files
  • export corrections to other formats (bin, stereo files,…)
  • calculate corrections for other sampling rates
  • export EQ for IIR parametric filters (?)
  • multichannel analysis for 7.1, 9.1.4, Dolby Atmos, DTS-X,…
  • multitone distortion analysis

If you have any good idea, just drop me a mail support(at)

My own web site is and I’m working in audio for many years, especially in pro audio. I have calibrated hundreds of professional studios, stereo to Atmos and DTS-X, for mixing rooms, movie theaters, broadcasters, post-production and mastering studios, but also some hifi systems. I’m a long time member of the AES and participating to the working group SC-04-08 (Working group on Measurement and equalization of sound systems in rooms). I also have been product manager for the launch of compactDisc in 1982 and later, Managing Director of Revox-Studer in France.

Quite a long time ago, I published on my website, the presentation of MMM  a measurement technique now used by many and which is the basis fof method.

You will find our conditions, privacy and RGPD rules in this page.

This web site is only to used for acoustics measurements and audio related topics. Uploading other kind of pictures is forbidden and the user may be banned without notice.

10 april 2021 :

  • improvements in accepted mic calibration files, ie last line can now be blank

9 april 2021 :

  • changed conditions for distance : now accept both “.” and “,” before decimals, max 4 characters instead of 3
  • if files are uploaded with a non valid form, blank fields are filled with default values in the calculation software

7 april 2021 :

  • correction of graphs S13 and S14 various gatings, that had an error with mic correction

31 march 2021 :

  • corrections for SMPTE, AES, B&K have been added for 5.1

30 march 2021 :

  • changes in LF target : in Auto Music mode, the real target being L+R, its value is set at +3dB above the choosen LFtarget. Also changed LFtarget values in form.

27 march 2021 :

  • modifications of performance rating calculation

23 march 2021 :

  • added B&K as target in the Upload Recording form

19 march 2021 :

  • bug correction : in some cases, measurements and/or correction files were L/R unbalanced by a few dB

13 march 2021 :

  • changed some “correction” graphs (with y axis +-18dB instead of +-9dB) so to better cope for various measurements and also, for clarification, deleted some graphs

08 march 2021 :

  • changes in the measurements list, down on page “Measurements” which was resetted to blank when starting new software version !
  • improvements in page 1 of graphs so that responses are better centered to 0dB

06 march 2021 :

  • changed some LF targets but you may have to fill again this field in the form
  • some bugs corrections

If you have questions or problems, or a just good suggestion, please send a mail to support(at)


The method is very stable. To check it, just do some successive measurements and compare. We encourage to also measure at different levels to check consistency of your setup and your corrections.


Page p1 ESSENTIALS with graphs S1 to S6

Real graphs

Ideal graphs (simulated)

Page 1, informations of measured system and SPL levels in dB weighted B and C
S1 Frequency response is smoothed to 1/20th octave under 200Hz and 1/6th above and represents the global balance of the loudspeakers measured with MMM method. Black curve corresponds to L+R in low frequencies. Green curve is the personnalised target caculated with measurement results and volume, distance and directivity.
S2 This response is more detailed because smoothed at 1/20th octave on the whole spectrum with a scale in conformity to CTA-2034 recommandations (25dB for a frequency decade). Indicated is Smoothess of InRoom response (SM_IR) for both L and R channels betwen 0 and 100%, 100% being the ideal value. And also indicated is Wide Band Deviation (WBD_IR) of InRoom response, 100% being ideal. At the moment, you cannot directly compare to those found in the AudioScienceReview website, here are real measured values while in ASR, the responses are estimated and not exactly calcutated the same way.
S3 Blue curve (left) and red (right) represent low frequeny response under 200Hz. Here we can see room modes near 35,60 and 100Hz. Those modes may be corrected by EQ, parametric or FIR, but dips at 55 et 70Hz are difficult to improve.
S4 Comparison of in-phase L+R and opposite-phase L-R : normally L+R should be much higher than L-R . But here at 90 or 120Hz, L-R is higher than L+R. This can lead to a sense of missing low frequencies because those frequencies are generally recorded in mono (L+R).
S5 RT60 is representing reverberation time in seconds. This measurement is not done in conformity to RT acoustics standard but gives a good indication of the sound field decrease. It is better that the curve shows no increase to the right (higher frequencies).
S6 ETC Energy Time Curve, shows the first 20 milliseconds, to display early reflections. It is recommended that both L and R curves stay under the recommanded AES limits in green. Here we see reflections at 8 and 9ms.
Page 2 for temporal aspects and phase, graphs S7 to S12
S7 and S8 show impulse responses. In an impulse response, it is mostly the high frequencies that are visible.  In this graph, we see a reflection at 4.8ms which correspond to a diffrence of distance of about 1.6m (4.8×0.34m).
S9 et S10 the step response is totally equivalent to impulse response, but with energy better dispatched on the frequency spectrum, it better shows the whole spectrum and it is easier to see some informations : here we see high frequencies starting before mids and lows (typical of a standard crossover).
S11 Phase : measurement being done at listening position, phase is retrieved from a frequency dependant window. The ideal response should be flat but we know that phase response is less important than amplitude.
S12 Group delay corresponds to phase variations : there is no clear limit of audibility but a response between the two green lines should be ok, it corresponds to +-0.5 periods. In above example, we se a peak at 1.2kHz due to the crossover.
Page 3 for other temporals, localisation and distortion, graphs S13 to S18
S13 et S14 Temporal evolution of frequency response : the time window starts at 2ms up to 100ms so we can see the evolution of some reflections.
S15 Pre-echo is a signal starting before the real signal (0ms) that may come from FIR equalisation : here some pre-echo can be seen near -7ms at -60dB.
S17 With wavelet display, we can see the spectrum of the pre-echo signal.
S16 Localisation : a well centered soundfield should stay near the green line for all frequencies but it depends on LR balance and loudspeakers distances.  Here we see a progressive shift to Left in low frequencies. ITD Interaural Time Difference and ILD Interaural level Difference are indicated on the graph.
S18 Total Harmonic Distortion (THD) : due to the short length of test signal sweeps, and depending on noise in the recording, distortion graphs may not allways be representative of the true distortion of measured loudspeakers. In this case, a measurement with stepped sine wave would be more effective (use REW or similar softwares).
Page 4, other temporals, graphs S19 to S24
S19 and S20 Spectrogram : this view is similar to S7 but detailled in frequencies : here are some reflections at 3, 8 and 9ms.
S21 et S22 Waterfall may give indication of room modes in low frequencies. In this case, a mode is seen at 40Hz.
S23 et S24 Wavelet visualisation
Comparable to spectrogram S13 but this kind of analysis gives better resolution in low frequencies.   Note that the horizontal scale is in periods. The 40Hz mode is clear. It is interesting to know that resonnant modes stay horizontal but reflexions are seen as oblique lines going up to the right.
Graphs p5 for temporal alignement
The perfect temporal alignement is when all crossings to level zero are at time 0 for all frequencies.

If field “Measure and correct” is validated, subdirectory “Correction” is created to contain.wav files for FIR correction, respectively Left and Right in linear phase and minimal phase : xxx-hyblinL.wav, xxx-hyblinR.wav, xxx-hybminL.wav, xxx-hybminR.wav. Those files can be directly used for corrections.

Pages p7 to p9 are also created.

p7 Correction
Separated C1 measurements L and R and also L+R (C2 black)
FIR corrections C3 calculated from L, R and L+R and hybrid corrections C4 (same in lower frequencies and separated above)
C5 Phase correction
Simulated responses of L, R and L+R after FIR correction : C7 for separated corrections and C8 for hybrid correction
p8 Simulated ETC energy-time curve after correction for pre-echo visualisation
p9 Simulated wavelets for pre-echo visualisation

For many people, most curves and graphs are not so easy to understand so some of you have asked about a simple performance rating. We have tested the ratings proposed by Sean Olive in AES papers 6113 and 6190 but for some reasons, it was not totally satisfying. Those ratings are based on anechoic room measurements extended to Predicted In Room results. With our method, we only measure InRoom values and we have to quantify performance based only on those real measurements.

We get the score from three main factors :

  • SM_IRR SMoothness of InRoom Response between 125 and 11500Hz : the proposal of Olive is not very intuitive (Pearson coefficient) and this value is not used by us
  • NBD Narrow Band Deviation of InRoom Response between 125 and 11500Hz (6.5 octaves) : measured surface difference between 1/20th octave curve and 1/2 octave curve, so it is not related to target and general slope
  • WBD Wide Bandwidth Deviation of frequency response from target curve : it is a value based on area difference (so related to variance) between the measured response and the target response between 125 and 11500Hz
  • LFD Low Frequencies Deviation is based on area difference between the measured response and the target response between 25 and 125Hz (2 octaves) but calculated on a linear frequency scale. We use a linear scale because we consider that problems at the high part of this frequency range are more audible and problematic than at the vey low frequencies. Note that this low frequencies target is flat under 80Hz, so it not not exactly same as the LF target defined in the Upload form
  • please notice that the displayed mean value is the lowest of L and R values or L, C and R in case of multichannel

It is important to understand that the rating is only based on measured amplitude response and is missing other factors that may influence audible quality : max levels, directivity, distortions, phase and time response, etc… So be carefull when you compare ratings of different systems, ithe highest may not be the best ! But compare numbers before/after equalisation/correction is certainly valid.


How to record

Download stéréo wav file LA2v1

This file can be written on a CD or DVD or streamed by a computer or added to a DAW session (Protools, Nuendo, Pyramix,…)

Play the file in stereo on your loudspeakers and record simultaneously with your microphone.

  • start recording and then playing just after : there is a 5s starting silence so you have time to get to the listening position
  • keep the mic at listening position : sync bong is followed by 3s left sweep and right sweep. Sweep signal is a sine signal which frequency is changing from very low to highest frequencies
  • voice «move mic to left » : place the mic about 30cm left to listening position, signal will be asweep on left followed by a sweep on right
  • «move mic to right » : place the mic about 30cm right to LP, sweep on left followed by sweep on right
  • «move mic to front» : place the mic about 30cm front to LP, sweep on left followed by, sweep on right
  • «move mic to rear» : place the mic about 30cm rear to LP, sweep on left followed by sweep on right
  • «slowly move mic » : slowly move mic in a volume of about 1m3 around listening place : pink noise 20s on left channel, 20s on right channel, 10s of correlated noise on both channels followed by 10s of pink noise with phase between channels. In larger rooms, you can move mic in a larger volume  : 1/5th of each dimension is a guideline.

The easiest way is to use the record function available directly on the page Upload recording.

You may record directly to your computer combined to  a measurement mic, using the record function on page “Upload my record”.

On a computer, there are some free softwares to play and record simultaneously : ie Audacity has an overdub function (in preferences) :

You can also use a portable recorder (Zoom or others) or your mobile phone with a usb mic : ie Iphone and Umik connected with a lightning-usb adapter.

Using a mobile phone, it is recommended to compare to measurements done another way to be sure than the recording feature of your phone does’nt reduce frequency response.

Record in uncompressed format .wav mono, 16bits or 24 bits, 44.1 or 48kHz (aiff or flac will be accepted later).

In all case be sure to avoid any process on the played sound file and on the recording : no E, compressor, limiter, filter,…

Record in uncompressed format .wav mono, 16bits or 24 bits, 44.1 or 48kHz (aiff or flac also accepted). The recording must start between 15sec to 1sec before the sync click (which is 5sec after the voice announcing “start to record now”. After the end of the sounds sequence, you can cut anywhere but avoid to leave more than 10 to 20 sec after (so that the file size keeps reasonable).

  • sound level must be high enough, ie the voice level should be comparable to someone speaking in the room, also avoid any clipping on the recording
  • notice that we have no responsibility due to too high levels : start with low levels
  • record without filter, compression or any process or EQ (unless you do a measurement with correction EQ).
  • the recommended mic is MiniDSP Umik, at a resonnable price under 100€, connected in usb so you need no preamp. This mic is delivered with an individual calibration file at 0 and 90° : or or
  • keep the mic vertically , pointing to ceiling, with the hand low on mic. Note that you have to send the corresponding calibration file (ie the one for 90°)
  • for left, right, front and rear position, better change slightly mic height
  • while sweeps, distance between positions is about 30cm for a studio or listening room and about 1m for a theater or large mixing room
  • with pink noise, the volume scanned by the mic is about 1mx1mx0.5m (Lxlxh) for a listening room or a studio up to 3mx3mx1m for a large room (preferably use a mic boom)
  • avoid any obstacle between loudspeakers and mic when moving mic
  • don’t move mic less than 30cm of obstacles (seat, table, console,…)
  • while moving mic, take care to constantly vary distance between mic and your body
  • the way to move mic is not very important but allways move slowly to avoid wind noises, ie 30cm/second
  • the whole stereo sequence is about 2mn 30 seconds
  • synchronisation is automatically done by software : no need to precisely cut the record by keep less than 15sec before voice start and keep the whole record up to the end (sequence can be cut anywhere after end of sounds, but not too long to avoid excessive upload time ).

We recommend the MiniDSP Umik 1, which is a usb microphone so no need for a preamplifier. This mic gets two individual calibration files, one at 0° and one at 90°. But other calibrated mic can do the job. Even better if you get a random incidence response for your microphone.

Preferably use the mic vertically and upload the 90° cal file. Ideally the right calibration would be for random incidence but more often you only get 0° and 90° responses. With a Umik the random incidence is nearly 3dB higher at 20kHz than the 90° response. It means that if you use the 90° calibration, you are a bit over-correcting and the real response of your loudspeakers is about 1dB lower at 10kHz and 3dB lower at 20kHz. See hereunder corresponding graph from IEC standards. You can see some examples on measurement mics websites such as GRAS or Bruel&Kjaer.

One type of measurement  may be spoiled by cheap microphones : because some microphones have have quite a high intrinsic noise, the distortions figures may be hidden into the mic noise floor. This can be improved if measuring at a higher sound level (but be careful with your ears and louspeakers).

If you have a calibration file, send it together with the recording in the version corresponding to your use (ie version at 90° if mic was used vertically, as recommended). If you have no calibrated mic, choose None/flat. Important : the calibration file is the response of the microphone, not the correction to apply !

For Umik, send the manufacturer cal file.The level reported in the graphs is calculated with the recommended 18dB gain of the Umik together with the sensitivity from calibration file (this is important for multichannel movie setups to be in conformity with SMPTE standards).

For “Other”, send a file named “calibration.txt” with first line at 0Hz and last line at 24000Hz, with a space between frequency and level such as :

0 -20
20 -0.1
1000 0
20000 0.1
24000 -20

If you have no calibration file, upload only the recording and no .txt file. The software will then use a flat calibration curve. Depending on your mic, the results above 5kHz may be wrong.

Start to play our sound file at low volume, adjust so that voice is at a realistic level. Following those advices, there is no risk for your loudspeakers.

For acoustic measurements, you really don’t need high sampling rate files to play or to record. So we choose to produce our sound files at 44.1 or 48kHz. For your recording, you can use any sample rate because the file will be resampled to 48kHz for analysis.

Form "Upload recording"

  • Fields “Company or title” and “Studio” are used to rank measurements in a structure title/studio/date, date being automaticcaly written. So you can check evolution of performances in time and see any degradation. Important : do not use of company name for which you have no rights, your account may be terminated without notice.
  • Field “Reference, Studio” : for home users with just one audio system, you may leave this field empty or fill with “home” or “listening room” or “without correction”, “with correction”, aso. For companies with more than one studio, you may enter the Studio name. You can hide your measurements to other users if you add “PRIVATE” to your reference or studio name,
  • No accents or special characters in he fields « company » and « Reference, Studio », because those fields are used as directory names. Allowed characters are alphabets, numbers, space and underscore (_). No space or _ at start or end.
  • Field to choose Test/Measure/Correction : “Test” allows to verify if the record is valid to give correct measurements, even if you bought no plan.
  • “New correction” is possibility to keep all values of the form and just change “Timbre target” and/or “LF target” to reprocess. You may want to change the record name so not to loose old corrections and be able to compare.
  • To fill other fields, check other FAQs.
  • It is important to fill the form and save it before to upload your recording ! Otherwise you won’t get your graphs done.
  • And remember that all your fields values are saved for next time, so less work for you for future measurements !





For stereo, the channels order is allways L R. But for multichannel setups, it is important to choose the real channel order so that the graphs show the exact channels : ie for 5.1, you can download L-C-R-LFE-Ls-Rs (movie order which is left, center, right, LFE, left surround, right surround) or L-R-C-LFE-Ls-Rs (SMPTE order). Also check which file you have downloaded : .flac files use SMPTE order (see ).

It really depends on your loudspeakers. In some cases, if you want to adjust the slope of the target curve, you could change the directivity at low or high frequencies set in the form.

For reference, here are the values used for calculation :

High frequencies, Omni/Dipole/Standard/Horn = 2/5/8/12

Low frequencies, Omni/Dipole = 2/5

Following fields are needed for the calculation of your optimised target curve :

  • Field “Target curve” : “Auto music” is recommended for home HiFi and pro music studios. For calibration of theaters or mixing rooms, SMPTE targets maybe be used with a choice depending on room size. Other possibilities are AES and B&K bin some cases.
  • “Timbre target” (only valid if “Auto music” is choosen) : the standard “Balanced” can be adjusted on user preferences to Bright (sharp, more high frequencies) or Warm (darker, more low frequencies). Difference to “Balanced” is a slope of approximatively +-0.3dB/oct (exact value depends on measurements,  distances and directivity) so nearly +2dB at 10kHz for “Sharp” and -2dB at 10kHz for “Warm”.
  • “LF target” (only valid if “Auto music” is choosen) : low frequency target is defined by cutoff frequency and level : hereunder  “409” gives a +9dB level and a 40Hz cut. Level at 40Hz cuttof freq is +9-3=+6dB. Level at 130Hz is half max level so +9/2=+4.5dB. Note that LF target for Auto Music is optimised for both speakers together (L+R). This means that to get a flat response for each separated channels, choose a value ending with 3 (ie 403 or 603), because L+R in low frequencies gives nearly +3dB compared to response of L or R alone.

Read more on this topic here.

The whole procedure gives a calculation of the so-called “house curve”. It is based on the recording itself (ie the calculated frequency dependant reverberation time) but also on parameters you entered in the form : room volume, distance, directivities. Note that if you choose SMPTE, AES or B&K curves, those will directly be used as targets and no calculation adapted to your own setup (in this case, LF and timbre target won’t be used)

To suit your needs, you can choose “Auto music” and adjust the target with Timbre target and LF target :

  • LF target choice between 208 to 902
  • Timbre target sharp/balanced/warm

Those choices you can set on the Upload recording form. The timbre target slopes are not fixed but depend on your measurement. If you know that your loudspeakers are quite neutral from a high quality manufacturer, and you are quite happy with the overall soud balance, we would suggest to choose the target that is the nearest to your measured curve, ie here the “balanced” one.

An information about the target slopes : with standard loudspeakers (dome in high frequency and closed or bass-reflex box in lows) and in a mid size room, at a distance of about 3m, the bright timbre target is nearly -1.05dB/oct, the balanced is -1.4dB/oct and the warm is -1.75dB/oct. The Harman slope (see Olive in AES papers) is also about -1.75dB/oct with a prefered LF target similar to LFtarget = 205 (AES convention paper 8994)

Another way to have an idea about the target slope, you can use target curve calculator.

At bottom of the form in page “Upload my recording”, depending on your plan, you can choose between some options :

  • “Test” gives the possibility to verify if the record is valid and if it will give correct measurements, see here.
  • “Measure” starts the analysis of uploaded file to write Measurement graphs.
  • “Measure and correct” adds calculation of FIR correction corresponding to your processor.
  • “New correction” keeps all values of the form, so you just change “Timbre target” and/or “LF target” to reprocess. You can change the record name and upload again so you will not rewrite on old corrections and keep possibility to compare.

Page TEST p6 shows graphs Rec1 to Rec5

Real measurement
Idéal curves (simulated)
Graph Rec2 shows detail of non synchonised record Synchronisation is correct when value “record time lag” corresponds to max and 3 black lines are aligned to peaks.
Unvalid Rec1 because lines are not aligned to parts on the graph. Valid : purple lines should be temporally separating parts.

In page “Upload recording“, you can upload a .wav, .flac or .aiff file for the recording and a .txt file for the mic calibration file. Max file size is 60MB.

In page “My files”, you can upload jpg and png picture files to your folders to show your room or loudspeakers.

If you want to keep your results private, just add PRIVATE in the “Reference, Studio” name in the Upload recording form. You can delete PRIVATE later from the name if you change your mind.

Depending on our server workload, it will generally take 5 to 10 minutes to get all your results. Be sure to use the refresh button to see the actual files.

For some reason, you may get not measurement graphs (don’t forget to use refresh button before complaining !).

First check : be sure that you have correctly FILLED all needed fields in the upload form and SAVED the form. Also check that your recorded sound file has a valid format, contains the start of sequence and does not start more than 15 seconds before the first voice.

After having checked, just upload your files another time.

My measurements

In “All measurements“, you will see pictures of all companies and studios. In “My measurements“, you will only see the files that are in your folders but all type of files, wav, txt, calibrations, images and correction files (.wav). And you have the possibility to upload, delete, move, rename and modify descriptions.

The folder “My measurements” will allways be available to a user even if its plan is over.

At the moment, convolution filters (for FIR correction) are only calculated for 48kHz sampling rate. If you need other sampling frequencies, you have to convert with an external tool (available in any DAW or tools such as sox).

It is recommended to also measure after EQ to check validity of the applied correction. You may also check at different sound levels.

At the moment, correction files are only calculated for following FIR processors and softwares in both linear and minimal phases versions :

  • generic 1024, 2048, 4096, 6144, 8192 and 16384 taps : mono 24bits 48kHz .wav files
  • MiniDSP 2x4HD 2048 taps : mono pcm 32bits float 48kHz .bin files
  • MiniDSP OpenDRC 6144 taps : mono pcm 32bits float 48kHz .bin files
  • QSC Qsys 8192taps : mono 24bits 48kHz .json files
  • Xilica Solaro QR1 or FR1 4096 taps : mono 24bits 48kHz .json files

For FIR corrections, the number of taps is the number of audio samples of the correction impulse, ie for 2048 taps, the length of the IR is 2048 samples (for any format, wav, pcm, bin, json,…).

If you need other formats (ie resolution or sampling rate), you can use an online audio converter such as

I don’t know if I will also calculate IIR (parametric) corrections : it is very depending on the actual processor (number of parametrics, Q type,…) and FIR processors are becoming cheaper and more common (ie MiniDSP, Xilica,…). Maybe for parametrics, I’ll just export a measurement file that could be used directly into REW so to calculate IIR correction.

If you bought a plan with correction included, the correction files are found in My measurements page in the correction folder with following names and in both formats .wav (48kHz 24bits) and .bin (pcm 48kHz 32bits float, big endian):

  • …-hyblin-L : linear phase hybrid correction for left channel
  • …-hyblin-R : linear phase hybrid correction for right channel
  • …-hybmin-L : minimal phase hybrid correction for left channel
  • …-hybmin-R : minimal phase hybrid correction for right channel

To use json files for Xilica Solaro or QSYS, you have to modify the json file in the correction folder : delete two first lines and first column, change the remaining column into a row with “,” between values and place brackets at both ends, ie [0,0,1,0.05,……0,0]

Important note : because correction can be positive in amplitude at certain frequencies, the software reduces the global level of the correction to avoid any overload. So when comparing before/after correction, you have to adjust level to match (with typical setup, you will generally need to reduce uncorrected level by about 3dB for a valid comparison).



Here is the folder structure for your files. Folder and files in red are private and only seen in “My measurements” so not seen by other users.

If you want to add pictures to display as folders thumbnails, you can use the “select” button at bottom of “upload recording” form. If you want a thumbnail for your  “Company or title”, keep the field “Reference, studio” blank. And if you want a thumbnail for a certain “Reference, studio”, just fill this field, save and upload the picture. Only .png and .jpg pictures are accepted.

If you cannot get your measurements !

I will try to add here possible reasons :

  • microphone calibration file may not be conform : ie take the umik file as it is, do not add any line even blank !
  • calibration file must also be in right order : frequencies must be increasing in all lines, lower frequencies are at top
  • too long a path is a problem because we use ftp copy for some files : so name of your company + name of studio or ref + name of your measurement file (or mic file if it is longer) must be less than about 180 characters

Another problem seen quite often : right level is measured much lower than left level ! This may come from your computer or software. When the level is quite high, a level compression is automatically set and the recorded level of right channel (and also center channel) is lower.

For any problem, please get in touch with support(at)

Acoustics and EQ

This question often comes when discussing electronic correction.
Let’s separate things :
– under 200 to 300Hz, room modes are allways present in any room (of normal size) and you generally have to smooth peaks, and avoid filling dips (don’t add energy : energy is there but it just cancels at some places)
– from 200 to 500-800hz, the position of the loudspeaker in relation to walls gives peaks and dips, but those acoustic problems can be improved with electronic EQ (above those frequencies, influence of reflections on walls are very dependent on exact listening position and fortunately, the auditory process has a frequency smoothing effect).
– above 600-700Hz, you may need to correct the loudspeaker if its response is not perfect !
So you need a method that shows only (and all) the defects that must be corrected (that’s why I think that MMM is a cool method).
It is important to note that a bad correction may be worse than no correction at all : so it is very important to be able to listen and compare with/without EQ (with levels exactly adjusted).

Hereunder is a comparison of 9 MMM measurements of the same loudspeaker in 9 different rooms (10 to 30 m2 surfaces) at distances between 2 and 3m. It clearly shows that :
– under 300Hz, the room is the key
– between 300 and 800Hz, the room and the loudspeaker are both important
– above 800Hz, the loudspeaker is the main factor. The response is the same but the slope depends of the distance and the room absorption (refer to Harman’s PIR Predicted In-Room)

Another comparison : same loudspeaker in anechoic room, and two other rooms (one room was measured 3 times) all measured MMM without any special acoustic correction. Here only the differences between measurements are shown with the base at 0dB being anechoic. You can clearly see that above 600Hz, delta is less than 2dB : MMM is measuring mainly the sound of the loudspeaker itself, not the room.

In the whole list of graphs, some are more usefull to check acoustic conditions :

Graph S5 in ESSENTIALS page shows RT60 reverberation time in the room which is high if the room is missing absorption. Another common problem is when the RT is really increasing in low frequencies, it means that absoprtion is too low in those frequencies, a problem that is sometimes not so easy to solve.

Graph S6 in ESSENTIALS page gives ETC Energy Time Curve, where you can see if some reflections within the first 20ms are happening. Remember that spund travels 34cm in 1ms, so you can figure where the annoying reflections come from.

Graphs S19 to S23 in ENERGY page, all show dissipation of energy with time but in different manners : spectrogram, waterfall and wavelets, which have various resolutions depending on frequency.

With all those measurements, you may think of acoustic treatments, positions and balance of absoption between frequencies.


EQ correction may change your experience and in some cases, you may need some time to become accustomed. But after some time, I’m quite sure that you wouldn’t want to go back without EQ.

Trinnov published an interesting paper about correction.

Which processors to use ?

I would recommend a FIR processor because it is easier and more precise for correction EQ (and it can also correct phase). Here are some possiblities I have worked with (prices are approximate and indicated without VAT) :

  • QSC QSYS, max 16384 taps per channel (with max total about 4×8192 for model Core110F), completely configurable, multichannel ie 8in/16out , very very complete features but a bit costly (about 3600€ for the smaller Core110F)
  • Trinnov, max (about) 4096 taps per channel (with total depending on stereo or multichannel model), high end product but you can only use Trinnov’s own software to mesure and EQ, not cheap (starts at 3600€ for stereo models)
  • MiniDSP OpenDRC, max 6144 taps per channel (with total of 2×6144), cheap and stable, only available with digital I/O, cheap, less than 300€
  • MiniDSP 2x4HD, max 2048 taps per channel (with total of 4096 for all channels), very cheap, near 200€
  • Xilica Solaro QR1 or FR1, max 4096 taps per channel (with max total of 6×4096 but FIR cannot be loaded into presets so cannot be changed on the fly, contrary to QSys), completely configurable (not far from QSYS) with choice of different I/O boards and nice possibilities to remote with Ipad or Android tablets, competitive price starts at about 900€ for QR1 (8 slots for I/O boards) and 1600€ for FR1 (16 slots) without I/O boards (boards are quite cheap). Note that those processor can ran at 96kHz but the FIR length is still 4096 taps so less efficient in the lows…
  • softwares are cheap (or free ! ie BruteFIR) but need more configuration and sometimes not so easy for every user

All above processor also have IIR parametric filters so you can adjust the LF EQ if you have not enough taps for a precise correction in very low frequencies. In France, QSYS, Xilica and Trinnov can be bought at Company 44.1 (with optional configuration and on-site setup)

When you calculate corrections, you will get both types so you can decide for yourself. Linear filters are named ie xxx-hyblin-L.wav and Minimal filters are named xxx-hybmin-L.wav.

Pre-echo artefacts cannot appear with minimal phase filters but note that the linear phase correction is calculated so to avoid any risk of pre-echo.

With more taps you have, the more precisely the correction will be in lowest frequencies. But a too precise correction may be offsetted with varying parameters : temperature or humidity, opened/closed doors or windows, changed furniture, aso. So practically, 2048 taps is already quite good and I don’t think that more than 16384 taps would be usefull. And generally, with all FIR processors, you can add some parametric EQ to finely ajust the lowest frequencies.

FIR filters can add delay to the signal passing through it. For symetrical FIRs, the delay is half the filter length, ie for a processor working at 48kHz and with 4096 taps, the delay is 2048/48000 second, near 43ms. This is unimportant when listening to music but can be a problem if you need video synchronisation (lipsync) and is a major problem if you work for dubbing or re-recording with musicians,… In those cases, better use the minimal phase correction FIR.

Our hybrid technologie allows a non symetrical Impulse Response, with a delay that is about half of the delay of a symetrical FIR. Be carefull if your system has, ie main front loudspeakers with FIR EQ (so with some delay) and a subwoofer with just parametric EQ (so without any delay), you certainly will have to add some delay to the subwoofer…