Please read FAQ before…




About this website

  • 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
  • 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 a unique 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
  • getting same graphs give the possibility to easily compare different setups
  • extensions to 5.1, 7.1, 9.1.4, 22.2, Atmos,… systems will come, including DCP sound files
  • you may even use your phone and mic for the recording

Comment on this FAQ

Read the procedure before jumping on measurements ;)

Hereunder is a description of the whole procedure and you should find more detailed infos in all FAQ topics.

Unique advantages of this method

  • very complete for professionals but simple enough for amateurs !
  • a procedure that avoids mistakes to obtain valid graphs and easy to compare to others systems
  • MMM measurement is completed with other signals to check amplitude but also time and phase responses
  • MMM method gives results nearer to perception than other methods
  • the target correction is automatically calculated from your measurements and own datas
  • target can be finely adjusted
  • 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, much better for low frequencies
  • a full report is available in .png pictures
  • with a unique 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
  • getting same graphs give the possibility to easily compare different setups
  • extensions to 5.1, 7.1, 9.1.4, 22.2, Atmos,… systems will come, including DCP sound files
  • you may even use your phone and mic to do the record

Procedure

You can visualise measurements without login. To access to menus “My profile”, “Upload recording” and “My files”, you have to register and reply to registration mail.

When registering, a field “Company or Title” will be the main folder for all your measurements, and images  and subdirectories by dates. 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. Accepted are alphabets, numbers, underscore “_” and spaces (but no space at beginning or end and no consecutive spaces).

Page “Upload recording”

  • 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.
  • You may record directly to your computer combined to  a measurement mic, using the record function on page “Upload my record” or upload a recording done by other means.
  • Record in uncompressed format .wav mono, 16bits or 24 bits, 44.1 or 48kHz (aiff or flac will be accepted later)
  • start recording and then playing just after : there is a 5s starting silence so you have time to get to the listening position
  • kplace mic at listening position : sine sweeps and sync bongs. 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, sweep on left, sweep on right
  • «move mic to right » : place the mic about 30cm right to LP, sweep on left, sweep on right
  • «slowly move mic » : slowly move mic in a volume of about 1 cubic meter around main listening place : pink noise 20s on left channel, 20s on right channel, 20s of correlated noise on both channels. In larger rooms, you can move mic in a larger volume  : 1/5th of each dimension is a guideline.
  • 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).

Important :

  • 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° : https://www.minidsp.com/products/acoustic-measurement/umik-1 or http://www.audiophonics.fr/fr/micros-de-mesure/minidsp-umik1-micro-mesure-usb-omnidirectionnel-p-8269.html or https://www.amazon.com/miniDSP-UMIK-1-Measurement-Calibrated-Microphone/dp/B00N4Q25R8
  • 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 loudspeaker and mic when moving mic
  • correlated pink noise is identical on both channels
    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 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 ).

Fields “Company or title” and “Studio” are used to rank measurements in a structure \title\studio\date\, date being automaticaly written. So you can check evolution of performances in time and see any degradation. Important to avoid use of company name for which you have no rights, your account may be terminated without notice.

Field “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.

No accents or special characters in he fields « company » and « 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 “Target” : “Auto music” is recommended for HiFi and “Auto cinema” for home-cinema. For calibration of theaters or mixing rooms, SMPTE targets maybe be used.

“Timbre target” : the standard “Balanced” can be adjusted on user preferences to Sharp (brighter, 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” : low frequency target is defined by cutoff frequency and level : hereunder  “405” gives a +5dB level and a 40Hz cut. Level at 40Hz is +5-3=+2dB. Level at 130Hz is half max level so +5/2=+2.5dB.

Field to choose Test/Measure/Correction :

“Test” allows to verify if the record is valid to give correct measurements, see hereunder.

“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.

Comment on this FAQ

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…

http://www.ohl.to/audio/downloads/MMM-moving-mic-measurement.pdf

Other readings :

https://www.aes.org/e-lib/browse.cfm?elib=19477

https://mountaincrest.net/dev/wp-content/uploads/2018/12/Samsung_Harman_Onyx_AudioSolution_Whitepaper.pdf

MMM howto in video : https://www.youtube.com/watch?v=6RiuwqzjqlQ

https://www.erinsaudiocorner.com/loudspeakers/ interesting because for loudspeaker tests, MMM (called Moving Mic Average by Erin) is compared to Klippel’s results : it simply proves that MMM is very comparable to Harman’s In-Room prediction based on anechoic measurements.

Comment on this FAQ

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 simultaneous, opposite-phase pink. The pink noise is in conformity with SMPTE standard ST-2095-1, see also here : https://www.ohl.to/archives/395

Comment on this FAQ

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.

Comment on this FAQ

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

Comment on this FAQ

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

Comment on this FAQ

REW and ARTA are very nice measurement 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.

Loudspeakers.audio 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.

Comment on this FAQ

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…

Comment on this FAQ

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.

 

Comment on this FAQ

You need to have javascript enabled for your browser and also Active-X for Internet Explorer.

Some ideas, but might take some time :

  • calculate NBD IR, SM IR and LFX in conformity to Sean Olive metrics  but with measured performances instead of predicted ones (I’m a bit uncertain about NBD’s calculation which is not so clear to me)
  • import other calibration files
  • export corrections to other formats (json, pcm,…)
  • export EQ for IIR parametric filters
  • multichannel analysis for 5.1, 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)loudspeakers.audio

Comment on this FAQ

My own web site is www.ohl.to 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). On my website, I published the presentation of MMM  a measurement technique now used by many.

Comment on this FAQ

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.

Comment on this FAQ

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

Comment on this FAQ

06 march 2021 :

  • changed LF targets (+1dB on most targets, max now is +9dB instead of +8dB) so 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)loudspeakers.audio

Comment on this FAQ

Measurements

Category: Measurements

Analysis

Page p1 ESSENTIALS with graphs S1 to S6

In the “Test” page, you can see all form values used for calculation.

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.

Comment on this FAQ

Category: Measurements

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

Comment on this FAQ

Category: Measurements

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
  • please notice that the displayed mean value is the lowest of L and R values

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

Category: 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.

Comment on this FAQ

Category: How to record

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) :  https://audacity.fr/

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,…

Comment on this FAQ

Category: How to record

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).

Comment on this FAQ

Category: How to record
  • 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° : https://www.minidsp.com/products/acoustic-measurement/umik-1 or http://www.audiophonics.fr/fr/micros-de-mesure/minidsp-umik1-micro-mesure-usb-omnidirectionnel-p-8269.html or https://www.amazon.com/miniDSP-UMIK-1-Measurement-Calibrated-Microphone/dp/B00N4Q25R8
  • 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 loudspeaker and mic when moving mic
  • correlated pink noise is identical on both channels
  • 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 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 ).

Comment on this FAQ

Category: How to record

We recommend the MiniDSP Umik, 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 that 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 corresonding graph from IEC standards. You can see examples on measurement microphone websites such as GRAS or Bruel&Kjaer.

Comment on this FAQ

Category: How to record

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. The calibration file is the microphone response itself, not the correction !

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

Comment on this FAQ

Category: How to record

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.

Comment on this FAQ

Category: How to record

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.

Comment on this FAQ

Category: How to record

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.

Comment on this FAQ

Form "Upload recording"

To get graphs for your recording, you have to follow those 3 steps :

Comment on this FAQ

  • 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.
  • 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 “Target” : “Auto music” is recommended for HiFi/pro studios and “Auto cinema” for home-cinema. For calibration of theaters or mixing rooms, SMPTE targets maybe be used.
  • “Timbre target” : 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” : low frequency target is defined by cutoff frequency and level : hereunder  “406” gives a +6dB level and a 40Hz cut. Level at 40Hz is +6-3=+3dB. Level at 130Hz is half max level so +6/2=+3dB. Note that the LF target is for each loudspeaker but the black line shows both speakers together (L+R).

Comment on this FAQ

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

Comment on this FAQ

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 https://xiph.org/flac/format.html ).

Comment on this FAQ

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 or AES curves, those will be used as targets and no calculation adapted to your own setup.

To suit your needs, you can adjust the target with Timbre target and LF target :

  • LF target choice between 209 to 803
  • 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.

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

Comment on this FAQ

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.

Comment on this FAQ

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.

Comment on this FAQ

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.

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

Comment on this FAQ

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.

Comment on this FAQ

For some reason, you may get not measurement graphs (don’t forget to use refresh button before complaining !). In this case, just upload your files another time.

Comment on this FAQ

The method is very stable. To check it, just do some successive measurements and compare.

Comment on this FAQ

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.

My measurements

Category: 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.

Comment on this FAQ

Category: My measurements

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).

Category: My measurements

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 https://audio.online-convert.com/convert-to-wav

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, modify the column in a row type with “,” between values and place brackets at both ends, ie [0,0,1,0.05,……0,0]

 

 

Category: My measurements

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

 

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

Acoustics and EQ

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.

Comment on this FAQ

When you calculate corrections, you will get both types so you can decide for yourself. But please note that the linear phase correction is set to avoid any risk of pre-echo.

Comment on this FAQ

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 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 with presets now), completely configurable (near 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 configuration and on-site setup)

Load More