REW for the first time

arivel

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Hello everybody.
I just signed up because I would like some help from you.
I would like to measure the frequency response of a pair of audio cabinets but it would be my first time doing it. You understand then that I don't know anything about these things but you have to start.
I prefer to start in the simplest way possible so as not to complicate my life.
from what I read on the main page of REW it seems that the simplest way is to connect a USB microphone to the PC and install the REW software, nothing else, do you confirm? .
but is there a cheaper alternative to UMIK-1 without losing too much in performance?
 

arivel

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should the signal generated by REW coming out of the PC be connected directly to the amplifier via RCA?
 

FargateOne

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Did you take the time to read every references there ? Plenty of informations and a good way to learn.

 

skid00

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Research the 'moving microphone method' first. It's relatively easy, and less error-prone than a stationary microphone.

A laptop with RCA output, or USB into a DAC to feed your amplifier, is the easiest way.

You want to scan a single speaker at a time. Don't do both at once.

The UMIK-1 is the best low-cost option. Do NOT try to use a voice/music oriented mic, they don't have flat frequency response.

Just in case you don't know, most thread responders are just audio enthusiasts. Keep that in mind as you read their posts... :)

You can use REW's built-in 'EQ' function to correct for room modes (and tweak response to your listening preference). This is usually done from 10 or 20 Hz, up to 300 Hz or so. That depends on the size of your room. Using REW's EQ for higher frequencies can be problematic, as speakers radiate from all sides (which is what the mic will measure) but your ears/brain interpret the direct sound, and tune out reflections to some degree.
 

arivel

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Research the 'moving microphone method' first. It's relatively easy, and less error-prone than a stationary microphone.

A laptop with RCA output, or USB into a DAC to feed your amplifier, is the easiest way.

You want to scan a single speaker at a time. Don't do both at once.

The UMIK-1 is the best low-cost option. Do NOT try to use a voice/music oriented mic, they don't have flat frequency response.

Just in case you don't know, most thread responders are just audio enthusiasts. Keep that in mind as you read their posts... :)

You can use REW's built-in 'EQ' function to correct for room modes (and tweak response to your listening preference). This is usually done from 10 or 20 Hz, up to 300 Hz or so. That depends on the size of your room. Using REW's EQ for higher frequencies can be problematic, as speakers radiate from all sides (which is what the mic will measure) but your ears/brain interpret the direct sound, and tune out reflections to some degree.

Can you explain this sentence to me better? "mobile microphone method"
 
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arivel

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Hello.
I don't know yet which DAC I will use and I don't know if in the future I will decide to change it, however I am oriented towards an I2s input.
having said that is it a good idea to buy an amanero or xmos USB to I2s card? .
as far as the microphone is concerned, I focused on beheringer umc22 + ecm8000. they cost less but is it a good choice?.
do you know where i can find the calibration file?
 

JStewart

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I focused on beheringer umc22 + ecm8000. they cost less but is it a good choice?.
do you know where i can find the calibration file?

If you’re in the US you may find it economical to have that mic calibrated by these folks

I don’t own that mic so I may be wrong, but it doesn’t appear the manufacturer provides them.

You have probably already noticed that MiniDSP does provide calibration files for their Umik mics.
 

LCRLive

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Am also here back to the drawing board as initially to play around for fun I was just using the laptop's own crappy built-in mic and crappy built-in speakers to do some toying around measurements of a room for sh#*s and giggles.

But I have found a used Focusrite Saffire Pro 40 because of its really good converters as the interface connected as Firewire legacy setup to an old Sony laptop and my question is: since I did this because I do not trust Umik USB run of the mill interfacing, for now I am using the flattest frequency response Microphone that i can fine (until I can get a perfectly flat one) which I have plugged in through XLR through to input 1 on the rack interface and have setup the Saffire as ASIO.

But alot of youtube videos are not clear as to the Output/Input vs Timing Reference Output/Loopback inout and i have no idea why these are recommended by some to be different than others. That is for a Loopback to calibrate the soundcard system, why wouldnt it just be the same Output 1 to Input 1 for calibration? Im not clear on the reasoning for these and their proper configuration.

Also is there a reason why we can't use a relatively flat vocal mic for room measurements since the system calibration would account for it properly if we either acquire or make a mic calibration file? Would not the compensation for the mic be simiular to the compensation for variations in the sound interface variations? Because I would trust a pro audio mic over some random USB -based mic for a proferssional audio measurement any day.

And if thats no good then could somebody please reccomend some profesisonal flat frequency XLR microphone for room testing which also has a calibration file? Because I am not using this Umik USB stuff, it just seems like childs play. I want to do testing with proper pro audio equipment as I have started acquiring which is as follows:

Using a 4U Case with rack gear and a power rack conconditioner for being able to connect the following and also do measurements remotely all mobile:
-Focusrite Saffirte Pro 40,
-Crestron QM-AMP3X80MM Amplifier (very heavy duty) to sent the sweeps to whatever speakers are in the room
-A 'to be determined' profesisonal grade XLR-based Flat frequency response microphone (although now just using a Sennheiser e935 which is fairly flat with the exeption of a bit of being booseted by a couple db above 2000hz and up. (does not REW compensate after calibrating it)?
-Using an Old Sony Vaio Laptop with Legacy Firewire... seems to work connecting to the Saffire Pro 40 and am getting Mic levels (which is a massive relief so far because that part is usually the biggest nightmare trying to get various interfaces to work with whatever computers)

I need to make this work with this system, and if I need to then I would get a proper flat frequency mic with XLR connectivity if one exists for room testing, but if not I will have to find a flat as possible pro audio mic because I am not doing the USB mic thing which to me is not serious stuff. But I hope I dont still need an SPL meter after hooking up all this stuff and setting it up.

Also I am being told to make use of the headphone out of my laptop but once again, all this stuff reccomended so far like a USB mic and a built-in soundcard does not have proper A to D and D to A converters and doesn't seem like a pro grade way of doing all this to me. Would much prefer using a pro rack sound interface for all the conversion and outputting that to the rack amp which I can send to whatever speaker.
 
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LCRLive

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In addition this tutorial i have found on youtube which seems to be legit says I dont need an SPL meter if one has even an entry level interface since they are built on a high level but the REW manual help section says I would ideally need one and that "An SPL meter is still ideally required to provide a reference SPL figure against which to calibrate REW's SPL display". Very lost here.
 

John Mulcahy

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If you care what the SPL numbers show you'll need an SPL meter to calibrate them, but it won't have any effect on the shape of the graph, just the level it is drawn at. For someone trying to calibrate a home theater to a particular playback level, for example, SPL calibration is important, when trying to get a more even response it isn't.

There are two uses for a loopback cable in REW. One is to provide a connection that is used as a timing reference, the signal arriving on the loopback input is used to provide a reference for time = zero. The other use for a loopback is to calibrate the soundcard, that is done on the input that will be used for measurement since the channels may have slight differences in their response or behaviour, for example you won't spot monitoring being active if you use the right output and the left input, but if you then switched to the right input for measuring that monitoring would start causing problems.

For correct acoustic measurements (decay times and the like) mics should really have an omni response, many vocal mics don't since that isn't what you want from a vocal mic. Vocal mics may also have intentional response shaping and roll-offs at their frequency extremes. There is a comparison of some modestly priced measurement mics here and a comprehensive list of mics here. You can spend as much as you like on a measurement mic and if money is no object it's hard to go wrong with Earthworks, but it really isn't necessary.
 

arivel

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I have an ignorant question on the subject.
if i use an audio interface i guess i have to use the pc to supply the signal to the speakers via usb.
but then the signal must enter from the microphone to the PC but the USB port is already occupied by the player's output signal how does that of the microphone enter?
 

John Mulcahy

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Are you referring to using a USB mic and using a separate USB audio interface for signal output? Then you will need two USB ports, or a USB hub to connect to the PC's port if it only has one. If you are using a USB mic you could use the PC's headphone output for playback, or send audio over an HDMI connection if the system you are testing has an HDMI input. There is a guide on that kind of setup.
 

arivel

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not, I was referring to an XLR microphone coupled to an external sound card.
 

John Mulcahy

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I'm not sure what you are trying to say here. If you are using an audio interface connected by USB it sends and receives audio data to/from the host computer over its USB connection. Your mic connects to an input on the audio interface, one or more audio outputs of the interface connect to the system you are trying to measure.
 

LCRLive

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Yikes! All this nice gear and can't get a sweep.
Using an focusrite saffire pro 40 and on its own Saffire Mix control it says levels are going in and going ou in self monitoring my voice.. On REW preferences input one XLR mic and Output Left which is Output 1 on the Saffire going to Left output on the crestron amp. Monitoring my voice in to the mic works but there is no sweep. I have input 1 and output 1 selected both in preferences and in Measure.
 
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John Mulcahy

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Likely to be a Saffire mix control setting, but you can post a screenshot of the REW soundcard preferences to check all looks OK there. Mic monitoring must be disabled for measuring.
 

arivel

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I'm not sure what you are trying to say here. If you are using an audio interface connected by USB it sends and receives audio data to/from the host computer over its USB connection. Your mic connects to an input on the audio interface, one or more audio outputs of the interface connect to the system you are trying to measure.

I am trying to say that USB 2 is bidirectional but reception and transmission cannot happen at the same time so if I use an external audio interface for both the microphone and the speakers, I wonder how REW gets a signal to be processed
 
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John Mulcahy

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I am trying to say that USB 2 is bidirectional but reception and transmission cannot happen at the same time so if I use an external audio interface for both the microphone and the speakers, I wonder how REW gets a signal to be processed
USB data transfer is packet based, the interface and the USB driver take care of managing the interleaving of data transfers back and forth. USB audio interfaces work, just accept it and move on.
 

arivel

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the signal must be sent exclusively by REW via PC or can I use an alternative method?.
for example i have a philips cd723 player and an old marantz amplifier, can i use these tools and a cd containing pink noise?
 

LCRLive

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still nothing. Tried everything. I disabled input monitoring on the Saffire mix control. I have Input 2 (working from the mic when I do input monitoring to check) and I have the Left output which is Monitor output 1 selected. This is quite brutal.
 

LCRLive

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still nothing. Tried everything. I disabled input monitoring on the Saffire mix control. I have Input 2 (working from the mic when I do input monitoring to check) and I have the Left output which is Monitor output 1 selected. This is quite brutal.

Ok finally got it. I had to select DAW 1 as monitor output 1 in the extremely comlicated Saffire Mixcontrol panel. Major celebrations here.

So now to finally learn all the testing methods for loudspeakers and what tests I should actually run. I hear data you get from the factory for speakers is never what reality shows. But then again the person who told me this is not testing in an anechoic chamber. But I am hoping to know what tests I should take after I do all the crossover soldering and connecting (based on what a person i hired is designing and giving me for the crossover network and instructions). And is this the only testing software I will ever need? or do I need that dayton component testing kit too?

On the room influence, after i treat as much as I can, If I have a reference speaker and test that, whatever the room is doing to the readings imperfectly (will it really affect it that much if i am close-miking?), can I compare that to its own frequency response chart and then assume i can 'compensate back' negating what the room chart values are doing the same way for anything else I test? Like to take away or subtract mathematical values from the frequency or somehow overlay the room graph on top of the graph of the reference to get the mathematical compensation chart and then to be able to apply that to take away or add to the test charts that I generate for the newly built speakers? or does it not work that way?

The guy said as long as I have basic treatment in the room its fine and to apply as much as I can then to compare whatever I am referencing to and then I can just interpolate my own compensating based on negating whatever the room is doing to skew the results in other words.
 
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