Does the measurement mic connected to input 1 should go BACK to output 1?, you don't mention here if it needs to be routed only to the inputs of the measurement software, because by default the mic we connect on input 1 will go ALSO to output 1 and that will create a feedback loop
You got it. I think I know what you're asking for, but can you be more specific? Are you talking about aligning mains to other subsystems (front fills, delays, subs etc.)? Or checking speaker crossovers within a single enclosure? Or simply how to read the phase graph?
This is by far one of the best TH-cam tutorials, on how to get started with OSM. And has also been a huge game changer for me, and my home theater as well as for my high end car audio sound system. Therefore, I now recommend everyone who in one way or another wants to learn how to get the most out of their sound systems to visit this particular TH-cam channel. Best regards from Sweden.
Thanks! Im implementing system tuning into my setup this year thanks to you! Ive also discovered the db meter feature in OSM, id like to keep the RTA mic at FOH after system tuning for a constant SPL reading during the show, but not sure how hard to gain the mic for an accurate reading, future video please!
Great Video Michael, Thank you so much for your effort and passion ... would be super helpful also to make a similar video about SMAART, and maybe a video about how to actually tune a sound system with Smaart in details adjusting EQ, Phase, Delaying speakers, an A-Z video explaining complete system tuning, much much appreciation and respect brother, Thank you.
Too much of theory.. make your videos short and simple in practicality. You bore me with your advertising more than the content I wish to hear from you. SOrry.. don`t take it personal Michael.
Hey there! Any idea why my scarlett 4i4 shows up as "focusrite usb audio" in open sound meter ? And also, when loop backing my generator signal it seems really unstable as in its jumping up and down in volume a lot more than on yours ? When i see your loop back volume in OSM its pretty steady and dosnt move much whereas mine moves quite a bit.. Im worried that my measurements arent correct because of that! It also seems really hard for me to get good coherence even in a relatively quiet room.. Hopefully you can anwser this!
I'm new to OSM and Smaart. Do you need the reference signal, or can you just use pink noise that can be generated from the console and simplify your setup by not having a loop thru?
If you only have one mic, in my case a sonarworks reference ID RTA mic, would you suggest tuning my left right by just tuning one side (having mic as close to the speaker as possible and running pink through just that one speaker) and then copying and pasting the eq to both sides or would it beneficial to run pink through both while being center to the two? Or would that create comb filtering?
Am confused about which is my reference signal spectrums on reall time on 😢😢😢 wen I was opan pink . I can find only mic response I can't understand reference 😕😕
I know this is an older one you did but thanks so much for stepping through this. Amazing job! I have now subscribed and working through many of your videos. Do you have any of your videos you would recommend that could guide me on what to do with the data? #1-how to read it; #2-how to apply changes to optimize systems. I have done some work with my home stuff and REW sweeps at 1m but need to move my efforts to PA work I do occasionally. Maybe pink noise is the best way? Biggest challenges I have are aligning top end (horns with compression drivers) to bass cabinets and then aligning all of this with subs. It’s older gear but all I have. Crossover types, optimal crossover points (acoustic also), slopes, phase alignment and driver alignment and gain for all components. Or maybe more simply stated, how to interpret the data and apply it to optimize the rig….across all the critical variables. Thanks again!
Great video Michael! Detailed and clear. I just started learning OSM and I'm liking it so far. I'm into home theater and want to learn better, smarter ways to analyze my hone theater system and to be able to tweak the system in real time. I want to learn how to do dual FFT measurements and see how updates changes my system in teal time. Your videos are informative and helpful. Thank you amd keep up the good work.
Open sound meter crashed on me tonight. I can send pink noise to monitors, meters no green signal meters on osm are not lighting up. Reference mic not picking up a thing. I'm wondering in this used evo 8 bit the dust is what I suspect?
Great video, very clear visual explanation of how the setup works. My 3 important getaways from this video: 1. Ensure that the speaker level is 20dB above the noise floor when measuring 2. In OSM, make sure that your levels are in the green and just/just not touching the orange and that they are as equal as possible 3. Reset the measurement delay, by clicking "apply estimated delay" everytime you move your microphone (or the speakers)
Thank you so much, Tim! Glad this was helpful for you. To be clear, every measurement you take won’t necessarily have the measurement vs the reference equal. I just like STARTING with what will be my show level with the mic in the middle of the audience to start equal, then go from there.
Hey- great video. Last couple of weeks I was searching for a video like this which is explaining step by step how to setup everything to get me started. Don't know why it took me that long to find your video. This is helping a lot- thx!
Great question. I would get your reference level to average between -18dBFS and -12dBFS, have your mic preamp +45 or +50dB of gain, then slowly bring up your speaker's level to have the measurement signal match your reference.
Great video. It is nice to see you push the technical part of audio engineering. this information is very valid and make a great differnce in your sound system
Hey ! First of all, thanks for your videos that help me a lot ! The setup you show here is with one speaker, in a live situation will you do each speaker (right and left) separately ? Or will you place the mic centered between the mains ? If yes at what distance ?
Hey, Jordan. Great question and glad you're liking the videos. If I had a simple left/right setup and only had a time for a few measurements, I would fire up just one of the sides and place my microphone in the middle of my audience and point it at the speaker and capture a measurement, then process accordingly. Then I would do the same with the other side.
It sounds like your interface is sending the mic's signal to the speakers. Make sure your direct monitoring feature is not on and the only signal that's feeding the speakers is the signal generator.
The routing on your interface is sending your measurement mic into the speaker. Make sure the only signal you're getting into your speaker is the signal generator.
Setting the delay means I'm delaying the reference signal to now be in time with the measurement signal. I can only get accurate phase data if they're synced up.
Please Excuse me if this comes across as a really stupid question but... Were would the Mixing Console be in the above chain? Computer > soundcard > *mixer > speaker?
Do you time align a 3way system the same procedure as you taught us in your chaos workshop? Tune the high and mid drivers @ onax 50%, then time align @ onax 75% , then add the sub???
Debating on the interface to purchase. I just purchased EMM 6 that you recommend. My question is, I don't own a reference monitor with xlr. All I have to explore with is a Klipsch system 2.1, male 3.5mm that I plug into my laptop. Any suggestion how I could possibly route it?
I finally was able to download and install. It took turning off my security, firewall, etc.. Then I didn't seem to have all the functions, buttons but found them accidentally by clicking around. I just need an interface and reference mics. Do you actually need 4 total for advanced operations? How much is considered a fair donation? 9.99 EUR by default. I hope I can get the basics and necessities down. Time will tell.
Hi Michelle. Thank you so much for your video. And i m using Tescam 2 Chanel interface. I did the same thing what you did. But when I am going level up from main pa i get feedback. I don't understand why it is happening. If you can explain to me it's really helpful to me. Thanks.
I received my Evo 8 today. Is there any to audio back to my laptop thru my laptop computer 2.1 computer speakers? Or am I just going to need a separate self powered speaker? I was hoping to figure out a way to use my computer speakers with sub but it doesn't look like there is way?
I have a question: When setting up to measure a bigger system (a line array for example) does the cable that is going to the speaker in this example going where? The mixing board?
I'm guessing if you're using a console plug into a channel with all processing bypassed rather than directly into the speaker correct? Looks like in Smaart there's signal split from the console and from the dsp.
I did my very 1st practice using two identical turbosound flashline monitors offsetting them, then taking the built-in delay on the with early arrival pushing it back in time 1.04ms. I need practice actually timing a sub to mains. It's still mind boggling needing to get them in phase and time?! 😮
Hi Michael. When you're in an arena with many many speakers, do you tune each and every one of them? If you do, how are they all connected to the mains and you still have control of each speaker? PS: live sound newbie here
I'm running a very similar setup; REW (Room EQ Wizard) along with a PreSonus AudioBox USB96 with the Behringer ECM8000. It works really well for me so far. I've not used Open Sound Meter; so I'll check it out.
Nice! Open Sound Meter just released a major update as well, so I think you'll really like working with it. REW is also fantastic, but I'm not near as fluid with it as OSM.
I have follow your tutorial... I am using u Phoria umc202HD and ECM8000 it is all I have... I cannot get the generator to engage... when the measurement layer is selected... it will engage when it is not selected. When I start the generator and then select the measurement layer it gives me a message saying ..... Device is not supported or busy...... Any help or ideas are much appreciated!
If I set up this gear and put the mic on the ground i could get data suitable for Tracebook right? Also, looping the signal back into the interface is just to confirm you have good data right? Once you do that, you can just take measurements ir do you compare the reference every time you take a measurement for some other reason?
Yup! I'd make sure and check out their measurement guidelines PDF for more specifics. You'll us the reference every time you take a transfer function measurement - th-cam.com/video/jQDLSWyR0qo/w-d-xo.html
This was great. Excited to see the explanation videos for impulse, phase, magnitude etc. When using this setup with a live PA, does the generator loop back run through the soundboard?
Thank you so much! When in a live environment the loopback still runs back into your audio interface. It still stands separate from your audio console. The reference signal (what's going to the speakers) run into either your console, DSP, or back of the speaker.
Hey i have a quick Question, can i use an TS Cabel instead of the TRS cabel for the loop back with 48V enabled (The Line Input should not have 48V but you will never know i guess...)? (Steinberg/Yamaha UR22Mk2)
@@MichaelCurtisAudio I have a bit of an issue. I think I’ve followed everything you’ve done as far as routing but my coherence never goes above 0.2. I’m using a window’s computer and my audio interface is the Scarlett 8i6
Make sure on the interface you're using that the direct monitor button is not pushed and that you do not have your microphone being monitored through your interface's software. Some interfaces also have a single knob that blends between the input and playback for monitoring. Make sure it's only on the playback side, not monitoring.
Another great info session!! Thanks Michael!! I Just need to download OSM Few questions! 1. On a venue, would you do your room tuning based on the mains only or would you include the subs as well? 2. Do you run your system full range or sub on a separate aux/matrix and how would you implement the subs when it comes to tuning? 3. Once you find your problem area, you eq it and remeasure?
You're very welcome! To answer your questions: 1. I do my mains first, then start integrating other subsystems like fills, delays, subs, etc. So yes, I do include the subs, for sure! 2. I do a "hybrid" approach. I run my sub as a separate matrix output, but is fed from my main LR mix. So, it's not on an aux that I can mix individual sources to, but is on a separate matrix send that I can turn up or down, delay, EQ. So, everything in my mix "could" potentially be in the subs, but I use EQ on the channel to shape tonal balance. 3. I'd like for you to clarify what you mean by problem area : ). EQ is just one tool of many that can solve differing problem types. I often find that many engineers try to solve TIMING problems with EQ, which is the wrong tool for the job.
@@MichaelCurtisAudio when I said problem area I meant like build up or holes in your frequency response! Expanding on the sub question, since you’re running it on a matrix off of the main LR, how do you setup the crossover frequency?
@@RaymanuelMuzik Got it! Thanks for the clarification. I first compare the trace I'm measuring in the room to the speaker's "near field" trace. I either snag that from tracebook.org or measure it on site myself. That gives me good, clean data to compare against Then I look at the impulse response and see if anything I'm seeing in the magnitude response is being "clouded" by reflections. I have a post about this on my old blog - producedbymkc.wordpress.com/2021/01/12/floor-bounces-and-eq/ That makes sure I'm not EQ-ing anything that I can't control in the first place : ) Secondly, I use a low shelf to fine tune the overall low end balance. This is variable due to the array's size and the low end buildup in a room. Third, I move to top end and use a shelf to bring that up to my target curve. If I'm able to divide the PA into zones I'll do HF EQ there as well. Bottom line - I never EQ an individual zone of boxes in a line array below 1kHz. There's no separation because of the spherical wavefront and overlap. When working with single point source boxes, I do the same low shelf and high shelf balancing act. Then if there's any 200-800Hz buildup, I'll use a filter across the whole array. But back to my earlier point, if the speaker has a fairly flat response in the near field, I know that if I'm seeing something very different when I'm in the far field that the change is likely due to room acoustics or floor bounces. When it comes to the sub matrix question: I also run my main LR mix through a stereo pair of matrices for the mains. So, the LR is a "container" for my mix, then my matrices fan it out to all of my PA zones, much like a system processor. I use the EQ on the PA L/PA R matrix to take care of any HPF for crossover to subs. I usually start with a second order Butterworth (-12dB per octave).
You're very welcome! Thank you for making wonderful, affordable, and intuitive software. Killer job on this latest update. Excited to dive more into it.
Plenty! I can visualize and confirm that the PA is doing as my design intended. When I use an analyzer like Smaart I'm not throwing away my ears, but giving myself yet another tool to make sure the system sounds great.
what do you think about using an Audio Interface with XLR outputs like Behringer UMC404HD? Would you achieve usable results if your Ref Signal is XLR output to XLR input?
I never said once that you shouldn't use your ears : ) My methodology simply STARTS with physics and quantifiable data, then I verify the results with my ears.
At the smaart classes I’ve been lucky to attend, they always say to use your ears after you have made an EQ decision and evaluate if it made things better or worse. I really respect the guys that have trained their ears to the point they don’t need a tuning rig to EQ a PA in a venue. Truly incredible to watch someone do this. For punks like me, being able to use a tuning rig has been a godsend, and it has helped train me to hear what is going on. I still have a long way to go to develop my ear, but it has been a huge help. And I imagine even for guys with better ears than mine, tuning rigs really help make the process go faster, which may translate to having more time for other things like being able to grab something to eat before showtime or getting a little bit more sleep. But my hats off to anyone who can do it all by ear.
@@billguitarvin Hey, Bill. Thanks for sharing here! I do think there's a subtle distinction to be made between setting the TONE of the PA vs aligning and verifying everything. Even if someone does have incredible ears and is able to quickly use EQ to dial in the tonality of a system to their liking very quickly, only a measurement system can tell you if your front fills are polarity inverted vs your mains. Like you mentioned, having a rig to speed up your workflow and aid with decision making is invaluable, but I don't believe an engineer can get to the point where they "don't need" the measurement system because they can do the entire gig on their own. All that being said, there are big time mix engineers who have still never touched a SMAART rig. If they're able to get great results, fantastic!
Michael is convinced that measuring everything is the only true way to run pro audio. I would assume that he is a very analytical person, needing data to be able to communicate. I would also assume that most audio engineers rely on sound fundamentals in order to mix effectively, with no measurement tools. In my home area, there are many qualified pro audio engineers, running Nationals on the regular. None of them use measurement mics to operate.
Hey Michael, I love your channel. Quick question for you. Does it matter if you are sending 48v phantom power to the reference input? Does that effect our results at all? For example, If I'm using a presonus audiobox with just two inputs and a single 48v phantom power switch that controls phantom power for both inputs, can I use this to both power my RTA mic and receive a clean reference signal?
Great question! Most audio interfaces have DC blocking, so it wouldn't hurt anything. However, if you want to be extra safe you can use a simple TS to TS cable (a guitar patch cable, basically) that would not allow phantom power to pass.
Yes, a simple TS would work! It would also make sure phantom power doesn't get passed to your interface outputs. I like carrying a TRS option just in case I need to use it as a patch cable on an aux out on a desk so I can keep the signal path balanced the whole way.
Is there a specific audio-related reason for using the DA-AD physical loopback solution, or is it just a hardware limitation of audio interfaces without loopback channel? In other words, could you just do the loopback in-the-box?
You could defiintely use the in-the-box loopback, but it doesn't account for the rountrip latency of your signal leaving and returning the interface. So, much won't change from a magnitude or phase response perspective, but your timing will be more accurate if you use the external physical loopback.
Impulse response, up down up starting with the 1st small peak or dip, or is it the 1st biggest peak or dip you start with to determine pos or neg polarity your speaker is going?
@MichaelCurtisAudio so that tells me I've been misreading it. So up is positive, down is negative correct? So if correct all my speakers are wired either in negative polarity or the pink noise generator is in reverse 🤔 OSM pink noise doesn't really indicate pos or neg, only invert the even channel and I'm only using a single channel 1 just like you instructed.
Hi there Noob here. Is this used to setup an EQ in your church building? What would you use this for and whats the best way to get your room EQ on your main fader so your sound sounds good and then proceed to do normal EQ on your instruments, vox etc. I hope i make sense. Still learning Thank you
Yup, this type of rig is exactly what you use to do the alignment and tone shaping of a sound system, EQ being one of your primary tools to shape the tonality. I usually end up applying the EQ on a Matrix, not the master fader itself, since that mix may end up going several places. If I had to pick one spot in the room to get a measurement of a simple LR mains rig, I would put the microphone halfway deep into the audience and directly in front of the left or right main, pointed at that speaker. Then I would apply EQ to get it to the desired tonality.
Good to know about Windows 11! I'm getting a Windows machine soon. But yes, it's a very powerful software and I hope Pavel gets some of the bugs worked out soon.
So basically the idea is you capture your reference (pink noise), save it as a capture, upload it to compare and use your measurement mic to make adjustments on your graphic EQ or delays to match the references as best as possible? Somewhat new to this and I mainly work in clubs and small theaters. Mainly looking for a fast way to check my PA before mixing a band. @michaelcurtis
Hey, Brian. You're close here, but the workflow's a little different than you've described. The concept you're thinking of having a reference trace to match is a target trace. The transfer function measurement here in OSM COMPARES a reference, the pink noise, and a measurement, what the microphone hears. So, pink noise is being both looped back into my interface as the reference, then also played through the speaker. The resulting graphs (IR, magnitude, phase) show me the DIFFERENCE between my reference signal and what my microphone is hearing. Hope that helps!
Hello mr Curtis, you will know from my question i am a newbie to all this analyzing stuff! so my questions are, when you get the data how do you use it to improve sound system quality??is that done by eq, and then you re-test?? i understand most of it but i get lost after all measurements are done. I must say i never thought of getting a tuning rig before, untill i saw your videos, i always thought it would be too expensive, and i thought it was only for big shows /set ups!!. i have an interface i use in my studio, i have cables, mic is on order. can i use a tablette? to run the software or must it be a laptop? Great videos thanks for all the information and thanks for your time
Hey, Zion. Thanks for the questions! You can run Open Sound Meter on an iPad if you would like. The app in the app store is reasonably priced, so you're not glued to a laptop. When you capture the data you're looking at several different things - the overall system response (which can be altered with EQ or turning up or down certain speakers in your rig), the phase and timing relationship between two different speakers, etc. Bottom line - a two channel analyzer is an X-Ray machine. An X-Ray technician has to know where to put the scanner to see what bones are broken. They have to know how to manipulate the images to see the health of the bones. So, I think your question is getting at "cool, there's data. Now what do I do with it?" The main things I'm doing with it is making sure all of my speakers "health" is well - they have consistent phase and magnitude response. Then I'm assessing the overall tonality of the areas where my speakers are covering. Is it consistent? Then I'm using it to make sure that my system time alignment is in order. I don't want a delay speaker to fire WAY too early compared to a main speaker, for example.
So in a live setting are you essentially reading the magnitude graph and EQing your room to balance out the dips in frequencies? Just wondering how you apply the information from the software to a live setting. Thank you!
Great question! Yes, after you have level set all your speakers to their zones, then it would be time to balance the tonality of the system using EQ. You're definitely taking the decay time and sonic signature of the room into account, but I always start with EQ'ing my system to a predetermined target curve, then listening and adjusting from there. It's tempting to be heavy handed with EQ and try to even out every little inconsistency, though. Less is more, work with broader EQ moves. Sometimes the peaks and valleys you're saying are from a floor bounce, which you can't EQ out. You can't solve a timing issue with a electronic solution. You should be close to where you need to be from the DESIGN and coverage first, then use EQ to take you over the finish line.
Do you think there is much to be gained from doing this at a small gig where you have only 2 tops and 1 sub or is this more for when you have a lot of speakers at bigger shows?
I think it's useful on any show, all the time. Setting up a measurement system will help you sniff out problems faster, make sure the polarity on your speakers match, and give you quantitative data when your ears are misleading. It's best to practice with a measurement setup on a small rig, get comfortable, then graduate into using it on larger and more complex setups.
Is the unique one tutorial that explains how to use this software, and how to wire up the entire connections. I've so much time looking for that. Thanks so much.l
REW would get you similar data, for sure. It uses a sine sweep instead of pink noise. You can use any audio interface you'd like, it doesn't have to be the Focusrite. If your speaker has no amp built in, then you would need an external power amp.
Um...wow! That was the clearest explanation of a basic measurement set up that I've ever heard! I will check out the rest of your channel as well. Thanks.
How do you make the connections for a stereo system? I have a Behringer X32 along with two Bose F1 Subs and 812 Array Loudspeaker stacked (loudspeaker/sub) left and right of the room.
This is a very informative vid thanks! I have one problem though. I seem to be getting a feedback loop somewhere. I think the mic input must be feeding into output 2 somehow but i did everything like you have and even have a very similar focusrite interface. Any suggestions?
Hi Michael, great video, I’m getting confused, on the output side of the interface, on goes back into the interface with the jack cable and would the other side jack to xlr go into the console on a spair input channel with no eq compression etc?
any reasion I can't use the usb on a X18r, input the cal mic into chanel 1, loop from aux 1 into chanel 2, and send it out the mains tru the speaker mangement system to the tops and subs? I'd rather not pull the umd404hd out of the studio.
The first reason is that the digital to analog and analog to digital conversion that your measurement is going through going out of your audio interface is not included in your transfer function. If your audio interface is linear, then that's not a big deal, but your measurement will be affected to a small degree since it's not truly apples to apples. The second is that you will not have the roundtrip conversion time on your audio interface accounted for with your delay measurements. The physical loopback has that full journey in there.
Thanks very much indeed, I'm just setting up a portable OSM rig for small PA gigs so your videos are exactly what I need to get me started. Looking forward to learning more. I'd like to recommend the interface I'm using for my portable rig, it's an ART USB Dual Pre which I bought for something else. It's a 2-in 2-out interface with phantom power, very compact, it needs no drivers and crucially it runs on USB power from the laptop (or a 9v battery) so it's a really simple physical setup.
@@MichaelCurtisAudio Actually I’m into car audio enthusiasts I want to learn car audio tuning. Whether with this software we can do these thing impulse response, time alignment, target setting, phase correction.
Hey, I'm Thinking of buying this setup. But I have some questions... Kinda tough to articulate but, when evertything is tuned how do you save and use it when dj's have to play and their mixer is hooked up on the main PA... I usually do the tuning on Driverack PA" and it saves the preset once I ring out the room etc. and everything that goes thru that sound procesor is eq-ed and so on. Do you have to run the programme simultaniously all night or it saves that preset on the actual hardware once you rout the signal thru it and to the amplifiers... I know, long question, really hard to express myself XD Thanks In advance.
I think I understand what you're asking : ). Every sound source (a band, DJ, podium mic, whatever) needs to pass through your system processor first, then go out to the speakers. So yes, you would do your tuning with the processor, save that, then sum down any other inputs into that processor.
@@MichaelCurtisAudio Yeah, I do that with my dbx... BUT... if I want to add this interface (interface you were showing us in the video) TO my existing setup (mixer-dbx-amps-speakers), is it possible so i can use it live on the gig to monitor everithing, put it in phase etc. If that's possible that would be AWESOME to see a video p.s. this video helped me a lot don't stop doing them for us enthusiasts :D (sry for my english)
@@minarellipower Now I understand! Yes, you could absolutely add this to your setups. You would run the output of the pink noise from the interface into one of the inputs on your dbx processor. Then from the input on your dbx you could route it to whatever speaker zones you're measuring. Thank you for the kind words!
I am trying to utilize this program for a car audio application and I am hoping you might be able to assist with some setup questions. Generally the measurement mic is setup in a 90 degree orientation at the listening position. Is this correct for the use of OSM and the calibration of the USB audio interface? Obviously in a vehicle situation the measurement mic location is skewed to one seat and is not only closer to speakers, but has different degrees of orientation. Would I need to choose a single reference speaker to calibrate the latency with or still use the entire sound system to generate the pink noise? I would assume that one speaker would need to be used as the reference to the static mic location so that impulse response on other individual speakers would have accurate timing? Also, it seems using one speaker as reference will not only time the latency of the USB audio interface, but will also include the latency of sound travel from the reference speaker to the microphone. I didn't see that you calculated distance into the latency for the speaker you used, so I am assuming that the mic would need to remain in that position since the latency was set in OSM for that distance. I have only used REW and a couple other programs for car audio tuning and I'm trying to branch out for better phase and impulse response, as well as live phase tuning for active speaker arrangements. I really appreciate the video and hopefully you can help me make sure I am setting this up properly for my purposes. TIA.
If you're doing a diffuse field measurement, yes you orient the mic 90° and face it upwards. You can totally do that in OSM. If you're trying to measure the phase response of a particular speaker in your car system, only have that single speaker play then retime your reference delay to sync to that speaker. You're right that the sync delay accounts for the wave propagation through air. I only set the delay time once since the mic stayed in that single position, but yes if you were to move the mic you would need to reset your delay.
@@MichaelCurtisAudio Thank you so much. What that tells me is that I am correctly applying what I am learning. 😂 I may not say it as eloquently as you, but I still understand exactly what you said. I feel like all of my time and efforts are somewhat paying off. It is because of very generous individuals like you that help us all be better and I truly appreciate your generosity.
Can you explain the routing of the referance through the console(say yamaha ql5) and back into the interface? If my mains and ff and delays are routed through a matrix , where do i send the reference signal in and out of and how do i eq those individual speakers? Thanks
Sure thing! The "looped back" reference signal. never enters the audio console. It passes out of an output on the audio interface then goes directly to an input. The measurement signal (which is another copy of the pink noise signal) goes into your audio console, then you're free to send that to whatever matrix needed to get to the appropriate speaker set.
@@ghighrolla711 It can be pre or post. All that matters is that the signal for measurement passing through the console doesn't go through any channel processing (EQ, dynamics, etc). Only have in line any processing you're wanting to apply to the whole PA, like using an EQ on a matrix output to shape the tonality of your mains.
You are so informative! Your advice has been very helpful. I want to be able to use my Open Sound Meter and 4ch interface to get Transfer Function measurements during performances. I know this sounds silly, but I'm confused about how to connect the interface to the mixer so that I don't have the measurement mic coming thru the speakers, but I can see what I do on the main outputs reflected in OSM. TIA.
Great question! Just make sure your main output on the interface isn't "monitoring" any input, just serving as a conduit for an output. That usually would be the left output and routing your signal generator to output 1 (left).
This was my question too. My 2i2 doesn’t have routing options so ch2 goes to the speaker and I get feedback with increased gain. Are you changing routing in ch 2?
Hey I’m using a Scarlett 2i2 interface, should I be hearing the measurement mic through my mixing console/speakers? My interface only has outputs left and right. Not sure if I routed this set up correctly. Thank you so much
Yes, you're correct there. The only signal that should be going to your speaker system is the pink noise. The microphone's output should only go to your audio analyzer software.
@@renesupersonic Yes. Take your left output and run it into your system processor or console to feed to your system. Take the right output and loop it back into input 2. Plug your microphone into input 1.
Many thanks for your videos ! This kind of knowledge need to be spread worldwide when we see how people work sometimes... Trying to practice in my studio setup (right speaker is sub, left speaker is main) any advices ? I tryied to move the microphone far from speakers but even if i gain stage again or increase the level of them, the coherence trace is lost after 1 or 2 meters from speaker How can we get a good coherence/phase trace and align correctly main/sub, even in large setup ?
@@MichaelCurtisAudio yes i did I think it come from boundaries effect in my room, even at high level But how can we do in bigger room, or live event with fly main speaker to get good coherence value?
@@vass1791 We just have to adjust our expectations and know how to read the coherence the data a little differently and look for patterns. If it's a bunch of smaller, very tight coherence drops it's probably due to room reverb. If they're wider and regular in pattern, that's a reflection and likely a comb filter. Just takes practice. Room reverb is something we can do very little about if we're not also in charge of acoustics/sound damping.
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Does the measurement mic connected to input 1 should go BACK to output 1?, you don't mention here if it needs to be routed only to the inputs of the measurement software, because by default the mic we connect on input 1 will go ALSO to output 1 and that will create a feedback loop
Hey man, would love that video about phase alignement using this software in a live setting!
You got it. I think I know what you're asking for, but can you be more specific? Are you talking about aligning mains to other subsystems (front fills, delays, subs etc.)? Or checking speaker crossovers within a single enclosure? Or simply how to read the phase graph?
@@MichaelCurtisAudio Mains to subs and maybe also adding some fills would be great!
Yea that would be awesome aligning mains to fill and subs
@@oli6843 plus 1
which software using
This is by far one of the best TH-cam tutorials, on how to get started with OSM. And has also been a huge game changer for me, and my home theater as well as for my high end car audio sound system.
Therefore, I now recommend everyone who in one way or another wants to learn how to get the most out of their sound systems to visit this particular TH-cam channel.
Best regards from Sweden.
Thanks a ton!
Thanks! Im implementing system tuning into my setup this year thanks to you! Ive also discovered the db meter feature in OSM, id like to keep the RTA mic at FOH after system tuning for a constant SPL reading during the show, but not sure how hard to gain the mic for an accurate reading, future video please!
Great Video Michael, Thank you so much for your effort and passion ... would be super helpful also to make a similar video about SMAART, and maybe a video about how to actually tune a sound system with Smaart in details adjusting EQ, Phase, Delaying speakers, an A-Z video explaining complete system tuning, much much appreciation and respect brother, Thank you.
Too much of theory.. make your videos short and simple in practicality. You bore me with your advertising more than the content I wish to hear from you. SOrry.. don`t take it personal Michael.
Hey there! Any idea why my scarlett 4i4 shows up as "focusrite usb audio" in open sound meter ? And also, when loop backing my generator signal it seems really unstable as in its jumping up and down in volume a lot more than on yours ? When i see your loop back volume in OSM its pretty steady and dosnt move much whereas mine moves quite a bit.. Im worried that my measurements arent correct because of that! It also seems really hard for me to get good coherence even in a relatively quiet room.. Hopefully you can anwser this!
I'm new to OSM and Smaart. Do you need the reference signal, or can you just use pink noise that can be generated from the console and simplify your setup by not having a loop thru?
If you only have one mic, in my case a sonarworks reference ID RTA mic, would you suggest tuning my left right by just tuning one side (having mic as close to the speaker as possible and running pink through just that one speaker) and then copying and pasting the eq to both sides or would it beneficial to run pink through both while being center to the two? Or would that create comb filtering?
Am confused about which is my reference signal spectrums on reall time on 😢😢😢 wen I was opan pink . I can find only mic response I can't understand reference 😕😕
I know this is an older one you did but thanks so much for stepping through this. Amazing job! I have now subscribed and working through many of your videos. Do you have any of your videos you would recommend that could guide me on what to do with the data? #1-how to read it; #2-how to apply changes to optimize systems. I have done some work with my home stuff and REW sweeps at 1m but need to move my efforts to PA work I do occasionally. Maybe pink noise is the best way? Biggest challenges I have are aligning top end (horns with compression drivers) to bass cabinets and then aligning all of this with subs. It’s older gear but all I have. Crossover types, optimal crossover points (acoustic also), slopes, phase alignment and driver alignment and gain for all components. Or maybe more simply stated, how to interpret the data and apply it to optimize the rig….across all the critical variables. Thanks again!
Great video Michael! Detailed and clear. I just started learning OSM and I'm liking it so far. I'm into home theater and want to learn better, smarter ways to analyze my hone theater system and to be able to tweak the system in real time. I want to learn how to do dual FFT measurements and see how updates changes my system in teal time. Your videos are informative and helpful. Thank you amd keep up the good work.
Why does my email address not work when signing up for the free guide and calculations spreadsheet?
Open sound meter crashed on me tonight. I can send pink noise to monitors, meters no green signal meters on osm are not lighting up. Reference mic not picking up a thing. I'm wondering in this used evo 8 bit the dust is what I suspect?
Great video, very clear visual explanation of how the setup works. My 3 important getaways from this video:
1. Ensure that the speaker level is 20dB above the noise floor when measuring
2. In OSM, make sure that your levels are in the green and just/just not touching the orange and that they are as equal as possible
3. Reset the measurement delay, by clicking "apply estimated delay" everytime you move your microphone (or the speakers)
Thank you so much, Tim! Glad this was helpful for you.
To be clear, every measurement you take won’t necessarily have the measurement vs the reference equal.
I just like STARTING with what will be my show level with the mic in the middle of the audience to start equal, then go from there.
Hey- great video. Last couple of weeks I was searching for a video like this which is explaining step by step how to setup everything to get me started. Don't know why it took me that long to find your video. This is helping a lot- thx!
You're very welcome!
does this work for live sound as well?
Hello, if I turn my microphone gain all the way up. It's red (clipping). Do i adjust the gain of my mic or match my reference gain to match it (red)?
Great question. I would get your reference level to average between -18dBFS and -12dBFS, have your mic preamp +45 or +50dB of gain, then slowly bring up your speaker's level to have the measurement signal match your reference.
Great video. It is nice to see you push the technical part of audio engineering. this information is very valid and make a great differnce in your sound system
Hey !
First of all, thanks for your videos that help me a lot !
The setup you show here is with one speaker, in a live situation will you do each speaker (right and left) separately ? Or will you place the mic centered between the mains ? If yes at what distance ?
Hey, Jordan. Great question and glad you're liking the videos.
If I had a simple left/right setup and only had a time for a few measurements, I would fire up just one of the sides and place my microphone in the middle of my audience and point it at the speaker and capture a measurement, then process accordingly. Then I would do the same with the other side.
What exactly would a speaker reading positive polarity look like in impulse response and phase? It's still alien to me. 👽
A speaker's phase response will be variable over frequency, although it will often "level off" in the high frequencies.
Hi Michael, I am getting feedback on the mic and unable to gain stage to the level of coherence. Any suggestions?
It sounds like your interface is sending the mic's signal to the speakers. Make sure your direct monitoring feature is not on and the only signal that's feeding the speakers is the signal generator.
All I'm getting is feedback into my speaker. Any suggestions?
The routing on your interface is sending your measurement mic into the speaker. Make sure the only signal you're getting into your speaker is the signal generator.
What are your thoughts on using the DBX Drive rack PA2?
If you have a smaller system that doesn't need a lot of discrete outputs to wrangle, it'll work great.
It's a single output driving the noise so how does one truly know if that noise is actually pushing pos or pulling neg on OSM?
That output must be compared to a reference signal to generate the impulse response data.
What is the meaning of setting the delay time? I don't really understand the purpose of this.
Setting the delay means I'm delaying the reference signal to now be in time with the measurement signal. I can only get accurate phase data if they're synced up.
Please Excuse me if this comes across as a really stupid question but... Were would the Mixing Console be in the above chain? Computer > soundcard > *mixer > speaker?
Do you time align a 3way system the same procedure as you taught us in your chaos workshop?
Tune the high and mid drivers @ onax 50%, then time align @ onax 75% , then add the sub???
I don’t see the pdf link that list the equipment needed for under $250.
I'd look at this updated video - th-cam.com/video/UF4eJuBAH3A/w-d-xo.html
When in a live setting the 1/4 inch to xlr that went to the speaker would be connected to my dbx pa?
That is correct.
Debating on the interface to purchase. I just purchased EMM 6 that you recommend.
My question is, I don't own a reference monitor with xlr. All I have to explore with is a Klipsch system 2.1, male 3.5mm that I plug into my laptop. Any suggestion how I could possibly route it?
Hi can this be used to measure pc to denon avr receiver (via hdmi) with 5.1. I’m willing to purchase hardware to learn. Thank you.
Can I use a dbx RTA-M reference microphone for this exercise?
Sure thing!
I finally was able to download and install. It took turning off my security, firewall, etc.. Then I didn't seem to have all the functions, buttons but found them accidentally by clicking around.
I just need an interface and reference mics. Do you actually need 4 total for advanced operations?
How much is considered a fair donation?
9.99 EUR by default.
I hope I can get the basics and necessities down. Time will tell.
Hi Michelle. Thank you so much for your video. And i m using Tescam 2 Chanel interface. I did the same thing what you did. But when I am going level up from main pa i get feedback. I don't understand why it is happening. If you can explain to me it's really helpful to me. Thanks.
I received my Evo 8 today. Is there any to audio back to my laptop thru my laptop computer 2.1 computer speakers? Or am I just going to need a separate self powered speaker? I was hoping to figure out a way to use my computer speakers with sub but it doesn't look like there is way?
Is there a similar video tutorial like this but with time aligning with subs.
Nathan Lively has a lot of videos on that.
I have a question: When setting up to measure a bigger system (a line array for example) does the cable that is going to the speaker in this example going where? The mixing board?
OSM stopped working again on my windows 11 laptop. This round I cannot get her back up and running? I even donated to the Russian! Any solutions?
I'm guessing if you're using a console plug into a channel with all processing bypassed rather than directly into the speaker correct?
Looks like in Smaart there's signal split from the console and from the dsp.
Thanks sir 🙏😭
I did my very 1st practice using two identical turbosound flashline monitors offsetting them, then taking the built-in delay on the with early arrival pushing it back in time 1.04ms. I need practice actually timing a sub to mains.
It's still mind boggling needing to get them in phase and time?! 😮
REW software will do the same thing.
Yup, but in a non-real-time fashion.
Hi Michael. When you're in an arena with many many speakers, do you tune each and every one of them? If you do, how are they all connected to the mains and you still have control of each speaker?
PS: live sound newbie here
I'm running a very similar setup; REW (Room EQ Wizard) along with a PreSonus AudioBox USB96 with the Behringer ECM8000. It works really well for me so far. I've not used Open Sound Meter; so I'll check it out.
Nice! Open Sound Meter just released a major update as well, so I think you'll really like working with it. REW is also fantastic, but I'm not near as fluid with it as OSM.
Sadly REW cant do dual channel measurement. Which is necessary on PA systems at creating Crossover/sub alignment/filter etc...
I have follow your tutorial... I am using u Phoria umc202HD and ECM8000 it is all I have... I cannot get the generator to engage... when the measurement layer is selected... it will engage when it is not selected. When I start the generator and then select the measurement layer it gives me a message saying ..... Device is not supported or busy...... Any help or ideas are much appreciated!
I would reach out to Pavel at Open Sound Meter. It sounds like it's a driver issue with the software?
It was ... I did Driver update and it is working now... Thank you!@@MichaelCurtisAudio
If I set up this gear and put the mic on the ground i could get data suitable for Tracebook right? Also, looping the signal back into the interface is just to confirm you have good data right? Once you do that, you can just take measurements ir do you compare the reference every time you take a measurement for some other reason?
Yup! I'd make sure and check out their measurement guidelines PDF for more specifics.
You'll us the reference every time you take a transfer function measurement - th-cam.com/video/jQDLSWyR0qo/w-d-xo.html
@@MichaelCurtisAudio thanks for the reply! always helpful!
I have a question. Can you load local music files in sound meter for testing?
This was great. Excited to see the explanation videos for impulse, phase, magnitude etc. When using this setup with a live PA, does the generator loop back run through the soundboard?
Thank you so much! When in a live environment the loopback still runs back into your audio interface. It still stands separate from your audio console. The reference signal (what's going to the speakers) run into either your console, DSP, or back of the speaker.
@@MichaelCurtisAudio thanks for the quick response. I’ve been pretty glued to the system tuning videos lately!
@@garends225 I'm glad they've been helpful for you!
Hi Michael. Excellent video, as always. Is there any chance I can get a cheat sheet for this workflow?
Hey i have a quick Question,
can i use an TS Cabel instead of the TRS cabel for the loop back with 48V enabled (The Line Input should not have 48V but you will never know i guess...)? (Steinberg/Yamaha UR22Mk2)
Absolutely!
@@MichaelCurtisAudio Thanks! Your Videos helped a lot! 🙌
Maybe you should do a Video about the Equal Loudness Curve. 😁
@@MichaelCurtisAudio I have a bit of an issue. I think I’ve followed everything you’ve done as far as routing but my coherence never goes above 0.2. I’m using a window’s computer and my audio interface is the Scarlett 8i6
Great video, let me ask you a question, shouldn’t the microphone be placed at listening position?
If I was trying to tune my studio monitors, yes. This demo is just an example with an extra speaker.
@@MichaelCurtisAudio is there any compatibility between this software and LM26 DSP as far as transferring calibration data
When following this tutorial to the T but when turning the chanel on speak i get feedback from rta mic what am I doing wrong?
Make sure on the interface you're using that the direct monitor button is not pushed and that you do not have your microphone being monitored through your interface's software. Some interfaces also have a single knob that blends between the input and playback for monitoring. Make sure it's only on the playback side, not monitoring.
@@MichaelCurtisAudio thank you 🙏!
I will try tonight.
Another great info session!! Thanks Michael!! I Just need to download OSM Few questions!
1. On a venue, would you do your room tuning based on the mains only or would you include the subs as well?
2. Do you run your system full range or sub on a separate aux/matrix and how would you implement the subs when it comes to tuning?
3. Once you find your problem area, you eq it and remeasure?
You're very welcome! To answer your questions:
1. I do my mains first, then start integrating other subsystems like fills, delays, subs, etc. So yes, I do include the subs, for sure!
2. I do a "hybrid" approach. I run my sub as a separate matrix output, but is fed from my main LR mix. So, it's not on an aux that I can mix individual sources to, but is on a separate matrix send that I can turn up or down, delay, EQ. So, everything in my mix "could" potentially be in the subs, but I use EQ on the channel to shape tonal balance.
3. I'd like for you to clarify what you mean by problem area : ). EQ is just one tool of many that can solve differing problem types. I often find that many engineers try to solve TIMING problems with EQ, which is the wrong tool for the job.
@@MichaelCurtisAudio when I said problem area I meant like build up or holes in your frequency response!
Expanding on the sub question, since you’re running it on a matrix off of the main LR, how do you setup the crossover frequency?
@@RaymanuelMuzik Got it! Thanks for the clarification.
I first compare the trace I'm measuring in the room to the speaker's "near field" trace. I either snag that from tracebook.org or measure it on site myself. That gives me good, clean data to compare against
Then I look at the impulse response and see if anything I'm seeing in the magnitude response is being "clouded" by reflections. I have a post about this on my old blog - producedbymkc.wordpress.com/2021/01/12/floor-bounces-and-eq/
That makes sure I'm not EQ-ing anything that I can't control in the first place : ) Secondly, I use a low shelf to fine tune the overall low end balance. This is variable due to the array's size and the low end buildup in a room. Third, I move to top end and use a shelf to bring that up to my target curve. If I'm able to divide the PA into zones I'll do HF EQ there as well. Bottom line - I never EQ an individual zone of boxes in a line array below 1kHz. There's no separation because of the spherical wavefront and overlap.
When working with single point source boxes, I do the same low shelf and high shelf balancing act. Then if there's any 200-800Hz buildup, I'll use a filter across the whole array. But back to my earlier point, if the speaker has a fairly flat response in the near field, I know that if I'm seeing something very different when I'm in the far field that the change is likely due to room acoustics or floor bounces.
When it comes to the sub matrix question: I also run my main LR mix through a stereo pair of matrices for the mains. So, the LR is a "container" for my mix, then my matrices fan it out to all of my PA zones, much like a system processor. I use the EQ on the PA L/PA R matrix to take care of any HPF for crossover to subs. I usually start with a second order Butterworth (-12dB per octave).
@@MichaelCurtisAudio thank you so much!! Learning a lot with your content!!
@@RaymanuelMuzik You're very welcome! Glad it's been helpful to you.
Thanks for the review! :)
You're very welcome! Thank you for making wonderful, affordable, and intuitive software. Killer job on this latest update. Excited to dive more into it.
We should do a video together on the channel once you get your course built. Would love to hear about it more in depth!
Too bad there's no way to show you these two measurement pictures on here.
What's different between tuning a pa by 👂 than using smaart to tune your pa ?
Plenty! I can visualize and confirm that the PA is doing as my design intended. When I use an analyzer like Smaart I'm not throwing away my ears, but giving myself yet another tool to make sure the system sounds great.
@@MichaelCurtisAudio so if you don't have a smaart rig what advice will you give someone tuning by ear ?
what do you think about using an Audio Interface with XLR outputs like Behringer UMC404HD? Would you achieve usable results if your Ref Signal is XLR output to XLR input?
XLR in to XLR out is no problem at all.
Whatever happened to using your ears, I did live sound engineering for 30 years without this gear ?
I never said once that you shouldn't use your ears : ) My methodology simply STARTS with physics and quantifiable data, then I verify the results with my ears.
At the smaart classes I’ve been lucky to attend, they always say to use your ears after you have made an EQ decision and evaluate if it made things better or worse. I really respect the guys that have trained their ears to the point they don’t need a tuning rig to EQ a PA in a venue. Truly incredible to watch someone do this. For punks like me, being able to use a tuning rig has been a godsend, and it has helped train me to hear what is going on. I still have a long way to go to develop my ear, but it has been a huge help. And I imagine even for guys with better ears than mine, tuning rigs really help make the process go faster, which may translate to having more time for other things like being able to grab something to eat before showtime or getting a little bit more sleep. But my hats off to anyone who can do it all by ear.
@@billguitarvin Hey, Bill. Thanks for sharing here! I do think there's a subtle distinction to be made between setting the TONE of the PA vs aligning and verifying everything. Even if someone does have incredible ears and is able to quickly use EQ to dial in the tonality of a system to their liking very quickly, only a measurement system can tell you if your front fills are polarity inverted vs your mains. Like you mentioned, having a rig to speed up your workflow and aid with decision making is invaluable, but I don't believe an engineer can get to the point where they "don't need" the measurement system because they can do the entire gig on their own.
All that being said, there are big time mix engineers who have still never touched a SMAART rig. If they're able to get great results, fantastic!
Michael is convinced that measuring everything is the only true way to run pro audio. I would assume that he is a very analytical person, needing data to be able to communicate. I would also assume that most audio engineers rely on sound fundamentals in order to mix effectively, with no measurement tools. In my home area, there are many qualified pro audio engineers, running Nationals on the regular. None of them use measurement mics to operate.
Hey Michael, I love your channel. Quick question for you. Does it matter if you are sending 48v phantom power to the reference input? Does that effect our results at all? For example, If I'm using a presonus audiobox with just two inputs and a single 48v phantom power switch that controls phantom power for both inputs, can I use this to both power my RTA mic and receive a clean reference signal?
Great question! Most audio interfaces have DC blocking, so it wouldn't hurt anything. However, if you want to be extra safe you can use a simple TS to TS cable (a guitar patch cable, basically) that would not allow phantom power to pass.
Hello! Very interesting and clear video, but why do you need a TRS cable to loop? A simple TS would work ? Or maybe I m missing something 😅
Yes, a simple TS would work! It would also make sure phantom power doesn't get passed to your interface outputs.
I like carrying a TRS option just in case I need to use it as a patch cable on an aux out on a desk so I can keep the signal path balanced the whole way.
Is there a specific audio-related reason for using the DA-AD physical loopback solution, or is it just a hardware limitation of audio interfaces without loopback channel? In other words, could you just do the loopback in-the-box?
You could defiintely use the in-the-box loopback, but it doesn't account for the rountrip latency of your signal leaving and returning the interface. So, much won't change from a magnitude or phase response perspective, but your timing will be more accurate if you use the external physical loopback.
Impulse response, up down up starting with the 1st small peak or dip, or is it the 1st biggest peak or dip you start with to determine pos or neg polarity your speaker is going?
It's usually the biggest and sharpest peak, assuming the speaker has sufficient high frequency content.
@MichaelCurtisAudio so that tells me I've been misreading it. So up is positive, down is negative correct? So if correct all my speakers are wired either in negative polarity or the pink noise generator is in reverse 🤔 OSM pink noise doesn't really indicate pos or neg, only invert the even channel and I'm only using a single channel 1 just like you instructed.
In OSM how are you suppose to know if your pink noise is is generating positive or negative?? The button is labeled "EVEN INV"
Hi there Noob here.
Is this used to setup an EQ in your church building? What would you use this for and whats the best way to get your room EQ on your main fader so your sound sounds good and then proceed to do normal EQ on your instruments, vox etc.
I hope i make sense. Still learning
Thank you
Yup, this type of rig is exactly what you use to do the alignment and tone shaping of a sound system, EQ being one of your primary tools to shape the tonality. I usually end up applying the EQ on a Matrix, not the master fader itself, since that mix may end up going several places.
If I had to pick one spot in the room to get a measurement of a simple LR mains rig, I would put the microphone halfway deep into the audience and directly in front of the left or right main, pointed at that speaker. Then I would apply EQ to get it to the desired tonality.
Love the app. But it didn't like windows 11 at all. I keep freezing . That's a bummer men Its so similar to smaart is not a joke.
Good to know about Windows 11! I'm getting a Windows machine soon. But yes, it's a very powerful software and I hope Pavel gets some of the bugs worked out soon.
So basically the idea is you capture your reference (pink noise), save it as a capture, upload it to compare and use your measurement mic to make adjustments on your graphic EQ or delays to match the references as best as possible? Somewhat new to this and I mainly work in clubs and small theaters. Mainly looking for a fast way to check my PA before mixing a band. @michaelcurtis
Hey, Brian. You're close here, but the workflow's a little different than you've described. The concept you're thinking of having a reference trace to match is a target trace. The transfer function measurement here in OSM COMPARES a reference, the pink noise, and a measurement, what the microphone hears. So, pink noise is being both looped back into my interface as the reference, then also played through the speaker. The resulting graphs (IR, magnitude, phase) show me the DIFFERENCE between my reference signal and what my microphone is hearing. Hope that helps!
Hello mr Curtis, you will know from my question i am a newbie to all this analyzing stuff! so my questions are, when you get the data how do you use it to improve sound system quality??is that done by eq, and then you re-test?? i understand most of it but i get lost after all measurements are done. I must say i never thought of getting a tuning rig before, untill i saw your videos, i always thought it would be too expensive, and i thought it was only for big shows /set ups!!. i have an interface i use in my studio, i have cables, mic is on order. can i use a tablette? to run the software or must it be a laptop? Great videos thanks for all the information and thanks for your time
Hey, Zion. Thanks for the questions! You can run Open Sound Meter on an iPad if you would like. The app in the app store is reasonably priced, so you're not glued to a laptop.
When you capture the data you're looking at several different things - the overall system response (which can be altered with EQ or turning up or down certain speakers in your rig), the phase and timing relationship between two different speakers, etc. Bottom line - a two channel analyzer is an X-Ray machine. An X-Ray technician has to know where to put the scanner to see what bones are broken. They have to know how to manipulate the images to see the health of the bones. So, I think your question is getting at "cool, there's data. Now what do I do with it?"
The main things I'm doing with it is making sure all of my speakers "health" is well - they have consistent phase and magnitude response. Then I'm assessing the overall tonality of the areas where my speakers are covering. Is it consistent? Then I'm using it to make sure that my system time alignment is in order. I don't want a delay speaker to fire WAY too early compared to a main speaker, for example.
Great video man.... Plz can u make a video sub and top phase alignment using OSM
Thank you so much! It is in the pipeline, for sure.
So in a live setting are you essentially reading the magnitude graph and EQing your room to balance out the dips in frequencies? Just wondering how you apply the information from the software to a live setting. Thank you!
Great question! Yes, after you have level set all your speakers to their zones, then it would be time to balance the tonality of the system using EQ. You're definitely taking the decay time and sonic signature of the room into account, but I always start with EQ'ing my system to a predetermined target curve, then listening and adjusting from there.
It's tempting to be heavy handed with EQ and try to even out every little inconsistency, though. Less is more, work with broader EQ moves. Sometimes the peaks and valleys you're saying are from a floor bounce, which you can't EQ out. You can't solve a timing issue with a electronic solution. You should be close to where you need to be from the DESIGN and coverage first, then use EQ to take you over the finish line.
Do you think there is much to be gained from doing this at a small gig where you have only 2 tops and 1 sub or is this more for when you have a lot of speakers at bigger shows?
I think it's useful on any show, all the time. Setting up a measurement system will help you sniff out problems faster, make sure the polarity on your speakers match, and give you quantitative data when your ears are misleading. It's best to practice with a measurement setup on a small rig, get comfortable, then graduate into using it on larger and more complex setups.
Is the unique one tutorial that explains how to use this software, and how to wire up the entire connections. I've so much time looking for that. Thanks so much.l
You are very welcome! Happy to help.
Is your set up the same results as rew and no focus right box? What if speaker has no amp enclosed?
REW would get you similar data, for sure. It uses a sine sweep instead of pink noise. You can use any audio interface you'd like, it doesn't have to be the Focusrite.
If your speaker has no amp built in, then you would need an external power amp.
Hey Michael. Just took my first trace thanks to this video. Keep up the good work!
Glad to hear that! Thanks for the encouragement.
My brain just popped. I hope one day I can get this down?! 😢
Stick with it!
Um...wow! That was the clearest explanation of a basic measurement set up that I've ever heard! I will check out the rest of your channel as well. Thanks.
Thank you so much!
Can I do this with a mono focusrite?
You need at least two separate inputs to do this.
@@MichaelCurtisAudio I figured that was the case unfortunately. Thanks!
How do you make the connections for a stereo system? I have a Behringer X32 along with two Bose F1 Subs and 812 Array Loudspeaker stacked (loudspeaker/sub) left and right of the room.
From a tuning standpoint? Or just the connections in general?
Connections
Top draw content @Micheal Curtis! Well done. I am amazed at how much I didn't know.
Glad this was helpful to you!
This is a very informative vid thanks! I have one problem though. I seem to be getting a feedback loop somewhere. I think the mic input must be feeding into output 2 somehow but i did everything like you have and even have a very similar focusrite interface. Any suggestions?
Just make sure in the mixer on the interface that you're not also sending the microphone input to the output feeding the speaker.
@@MichaelCurtisAudio you're exactly correct my friend. Walked away from it and came back with fresh eyes today and that's what was going on. Thanks!
Great video!! I’m subscribing
Thank you!
Thank you for taking time to make this video!
Hi Michael, great video, I’m getting confused, on the output side of the interface, on goes back into the interface with the jack cable and would the other side jack to xlr go into the console on a spair input channel with no eq compression etc?
Exactly! One output of the interface loops back into an open input while the other output feeds directly into the console or DSP.
@@MichaelCurtisAudio thank you very much Michael
any reasion I can't use the usb on a X18r, input the cal mic into chanel 1, loop from aux 1 into chanel 2, and send it out the mains tru the speaker mangement system to the tops and subs? I'd rather not pull the umd404hd out of the studio.
That sounds like it'd work to me. I'm almost positive the X18r has a routable signal generator, so that should work.
The XR18 does NOT have a built-in signal generator, the X32R does.
Is there any reason not to use the 'loop' function for the reference channel?
The first reason is that the digital to analog and analog to digital conversion that your measurement is going through going out of your audio interface is not included in your transfer function. If your audio interface is linear, then that's not a big deal, but your measurement will be affected to a small degree since it's not truly apples to apples.
The second is that you will not have the roundtrip conversion time on your audio interface accounted for with your delay measurements. The physical loopback has that full journey in there.
Thank you bro
Thanks very much indeed, I'm just setting up a portable OSM rig for small PA gigs so your videos are exactly what I need to get me started. Looking forward to learning more.
I'd like to recommend the interface I'm using for my portable rig, it's an ART USB Dual Pre which I bought for something else. It's a 2-in 2-out interface with phantom power, very compact, it needs no drivers and crucially it runs on USB power from the laptop (or a 9v battery) so it's a really simple physical setup.
You're very welcome! Make sure and check out my OSM video tutorial video as well.
That ART USB Dual Pre sounds awesome!
Great Video. Very informative...Thank You
You're very welcome!
thank you for sharing your knowledge
You're welcome!
Hi, what soft are you using.
Open Sound Meter.
@@MichaelCurtisAudio okay, but this software work like REW.
@@MichaelCurtisAudio Actually I’m into car audio enthusiasts I want to learn car audio tuning. Whether with this software we can do these thing impulse response, time alignment, target setting, phase correction.
Good Session ❤
which software using
Open Sound Meter
Can we please see a practical demo with OSM? Aligning subs to mains?
Later this year!
@@MichaelCurtisAudio wooo! Im aware youre a busy guy. All of these videos are great and ive watched most, several times. Much appreciated:)
Hey,
I'm Thinking of buying this setup. But I have some questions...
Kinda tough to articulate but, when evertything is tuned how do you save and use it when dj's have to play and their mixer is hooked up on the main PA... I usually do the tuning on Driverack PA" and it saves the preset once I ring out the room etc. and everything that goes thru that sound procesor is eq-ed and so on. Do you have to run the programme simultaniously all night or it saves that preset on the actual hardware once you rout the signal thru it and to the amplifiers...
I know, long question, really hard to express myself XD
Thanks In advance.
I think I understand what you're asking : ). Every sound source (a band, DJ, podium mic, whatever) needs to pass through your system processor first, then go out to the speakers. So yes, you would do your tuning with the processor, save that, then sum down any other inputs into that processor.
@@MichaelCurtisAudio Yeah, I do that with my dbx... BUT... if I want to add this interface (interface you were showing us in the video) TO my existing setup (mixer-dbx-amps-speakers), is it possible so i can use it live on the gig to monitor everithing, put it in phase etc. If that's possible that would be AWESOME to see a video
p.s.
this video helped me a lot don't stop doing them for us enthusiasts :D (sry for my english)
@@minarellipower Now I understand! Yes, you could absolutely add this to your setups. You would run the output of the pink noise from the interface into one of the inputs on your dbx processor. Then from the input on your dbx you could route it to whatever speaker zones you're measuring. Thank you for the kind words!
Hey Michael, do you have any tutorials on how to do the time alignment using open sound meter?
I currently don't, but have plans for that early next year.
@@MichaelCurtisAudio sweet...looking forward to it, been using holm impulse for time alignment but need something a little more....real time
Great video!!
I am trying to utilize this program for a car audio application and I am hoping you might be able to assist with some setup questions.
Generally the measurement mic is setup in a 90 degree orientation at the listening position. Is this correct for the use of OSM and the calibration of the USB audio interface? Obviously in a vehicle situation the measurement mic location is skewed to one seat and is not only closer to speakers, but has different degrees of orientation. Would I need to choose a single reference speaker to calibrate the latency with or still use the entire sound system to generate the pink noise? I would assume that one speaker would need to be used as the reference to the static mic location so that impulse response on other individual speakers would have accurate timing? Also, it seems using one speaker as reference will not only time the latency of the USB audio interface, but will also include the latency of sound travel from the reference speaker to the microphone. I didn't see that you calculated distance into the latency for the speaker you used, so I am assuming that the mic would need to remain in that position since the latency was set in OSM for that distance.
I have only used REW and a couple other programs for car audio tuning and I'm trying to branch out for better phase and impulse response, as well as live phase tuning for active speaker arrangements.
I really appreciate the video and hopefully you can help me make sure I am setting this up properly for my purposes.
TIA.
If you're doing a diffuse field measurement, yes you orient the mic 90° and face it upwards. You can totally do that in OSM.
If you're trying to measure the phase response of a particular speaker in your car system, only have that single speaker play then retime your reference delay to sync to that speaker. You're right that the sync delay accounts for the wave propagation through air.
I only set the delay time once since the mic stayed in that single position, but yes if you were to move the mic you would need to reset your delay.
@@MichaelCurtisAudio Thank you so much. What that tells me is that I am correctly applying what I am learning. 😂 I may not say it as eloquently as you, but I still understand exactly what you said.
I feel like all of my time and efforts are somewhat paying off. It is because of very generous individuals like you that help us all be better and I truly appreciate your generosity.
@@TickleFingers So glad to hear that! Keep up the good work.
Can you explain the routing of the referance through the console(say yamaha ql5) and back into the interface? If my mains and ff and delays are routed through a matrix , where do i send the reference signal in and out of and how do i eq those individual speakers? Thanks
Sure thing! The "looped back" reference signal. never enters the audio console. It passes out of an output on the audio interface then goes directly to an input. The measurement signal (which is another copy of the pink noise signal) goes into your audio console, then you're free to send that to whatever matrix needed to get to the appropriate speaker set.
@@MichaelCurtisAudio so the signal that goes into the console, goes to a channel then sent to the matrix? is it suppose to be pre fader?
@@ghighrolla711 It can be pre or post. All that matters is that the signal for measurement passing through the console doesn't go through any channel processing (EQ, dynamics, etc). Only have in line any processing you're wanting to apply to the whole PA, like using an EQ on a matrix output to shape the tonality of your mains.
You are so informative! Your advice has been very helpful. I want to be able to use my Open Sound Meter and 4ch interface to get Transfer Function measurements during performances. I know this sounds silly, but I'm confused about how to connect the interface to the mixer so that I don't have the measurement mic coming thru the speakers, but I can see what I do on the main outputs reflected in OSM. TIA.
Great question! Just make sure your main output on the interface isn't "monitoring" any input, just serving as a conduit for an output. That usually would be the left output and routing your signal generator to output 1 (left).
This was my question too. My 2i2 doesn’t have routing options so ch2 goes to the speaker and I get feedback with increased gain. Are you changing routing in ch 2?
@@jeffreystipe925 I had to make sure the MIX knob was set to PB and not to INPUT on my interface.
Hey I’m using a Scarlett 2i2 interface, should I be hearing the measurement mic through my mixing console/speakers? My interface only has outputs left and right. Not sure if I routed this set up correctly.
Thank you so much
In other words, should the measurement mic output ONLY be routed to my laptop/sound meter software?
Yes, you're correct there. The only signal that should be going to your speaker system is the pink noise. The microphone's output should only go to your audio analyzer software.
@@MichaelCurtisAudio is your set up still possible with a scarlett 2i2? the only 2 outputs of the interface are a Left and Right.
@@renesupersonic Yes. Take your left output and run it into your system processor or console to feed to your system. Take the right output and loop it back into input 2. Plug your microphone into input 1.
Looking for the video that tells me what to do with this data
Many thanks for your videos ! This kind of knowledge need to be spread worldwide when we see how people work sometimes...
Trying to practice in my studio setup (right speaker is sub, left speaker is main) any advices ?
I tryied to move the microphone far from speakers but even if i gain stage again or increase the level of them, the coherence trace is lost after 1 or 2 meters from speaker
How can we get a good coherence/phase trace and align correctly main/sub, even in large setup ?
Hey, Vass. Are you resetting your delay finder when you move your mic away from your speakers?
@@MichaelCurtisAudio yes i did
I think it come from boundaries effect in my room, even at high level
But how can we do in bigger room, or live event with fly main speaker to get good coherence value?
@@vass1791 We just have to adjust our expectations and know how to read the coherence the data a little differently and look for patterns. If it's a bunch of smaller, very tight coherence drops it's probably due to room reverb. If they're wider and regular in pattern, that's a reflection and likely a comb filter. Just takes practice.
Room reverb is something we can do very little about if we're not also in charge of acoustics/sound damping.
@@MichaelCurtisAudio Thanks for your advices, i'll practice it again