I’ve known for years that engineers who make measurements wirelessly are using a special type of wireless specifically designed or the purpose. We are told that we need a digital wireless without a compander circuit to be able to measure accurately. Fine. No doubt those that know about such things are right but I wonder what sort of problems come up with using an off the shelf wireless rig.

Since as a general rule, we use omni mics for making audio measurements the path of least resistance is to start with a wireless lapel / omni lav mic. For this experiment, I’ll be using a Sennheiser Evolution series rig.

INSERT RESULTS HERE:

Obviously measuring things wrong and then making decisions and corrections based on bad measurement information is a waste of time at best.

I was curious how the different data point (the size of the sample window) affects a measurement.

In SpectraFoo Complete there are the following options to choose from.

In order to keep track of the difference (if any), I took a snapshot via the “TAKE SNAPSHOT” button at the bottom left side of the TF window.

In that same area is a button to display the snapshot overlay list.

This is what the overlay list looks like for this measurement process.

When you’re saving snapshots and then comparing them, it’s sometimes helpful to discontinue the measurement process and clear away any remaining data on the screen so you’re only looking at snapshots. These are the places to stop the measurement and clear the window.

All of the functions mentioned above can be achieved with key commands:

This is what it looks like when you turn off the measurement tool & clear the realtime data while showing all the snapshots.

Sound on Sound – Tape Maintenance article

When I first began working with transfer functions, I compiled a new PDF that includes only information related to transfer functions so I wouldn’t have to hunt for it in the full manual.

Here it is:
SpectraFoo4 FFT Only

Here is a link to the current full manual:
SpectraFoo Manual – Complete

Depending on your current configuration, it’s possible for Foo to update the main preference file to a state that is unstable. If Foo crashes and then refuses to reopen without crashing again, have no fear.

SOLUTION:

In OSX versions prior to 10.9, your library is located:

In Mountain Lion (10.9), your user library is hidden from you by default. Hold the option key while choosing the “GO” menu in the Finder, and choose “Library” from that menu.

Proceeed to:

Library/Preference/SpectraFoo Preferences

Proceed to:

Once inside the “SpectraFoo Preferences” folder you will find a file called:

SpectraFoo Prefs

Put that in the trash, empty and restart Foo.

There is no reason to delete the folder & the other files unless you are having issues while successfully running SpectraFoo which may be caused by another preference file. Once you restart SpectraFoo, the app will create a new SpectraFoo Pref file and things should be stable. Note that any chances you had made to Foo will be lost so it’s a good idea to make a copy of your SpectraFoo Preferences folder once you have things all configured to your liking.

Part of the magic of Foo is being able to make Window Sets and be able to move between custom screen layouts with keystrokes but I’ll cover that in another post.

Joel Lonky Using SpectraFoo Complete for Rob Zombie, Others

Metric Halo – setting up a designated Spectra Foo computer

Metric Halo – audio scrubbing in Spectra Foo

Metric Halo – Spectra Foo inputs

Spectrafoo : Lock scale with Groups

Metric Halo – THD measurements with Spectra Foo

Metric Halo – announces addition of Bob Katz’s K-System metering to Spectra Foo

When you start SpecraFoo for the first time, there are some configuration duties that must be completed before the system is ready to use.

Generator Use in 7.4
Quote:
“Transfer Function measurements can also be referenced internally to Smaart’s signal generator by enabling “Allow Multi-Device Transfer Function” option on the Transfer Function page of the main Options dialog and selecting the Generator as your reference signal source. In practice there are a few caveats to doing this, again mainly associated with delay times. For one thing, the latency of the measurement system from the output to the input becomes a factor in measuring absolute delay times. (This is generally not a factor when actively recording both input signals.) Another is that the latency through the measurement system may not be consistent from one measurement to the next, meaning you may need to measure the delay again each time you stop and restart the measurement and/or the signal generator. And finally if the measurement signal input and the output used to stimulate the system under test are not identically clocked, then the reference delay time will be subject to drift while measuring, and not just when restarting the measurement.”

prosoundweb.com thread – Smaart software generation loopback

soundforums.net – Virtual SMAART Training: Tell the crossover frequency and type

After spending a few months gathering, cataloging & regurgitating the information on this site, I realize that it might all be a bit overwhelming for a visitor. Welcome to my world! I am probably on my 10th+ attempt to read & comprehend Bob McCarthy’s book “Sound Systems: Design and Optimization”. Each time I understand a little bit more. I’m going to assume that if I don’t give up, I’ll get there.

If nothing else, watch the youtube videos available on the VIDEO page. I’ll be adding more links & possible producing some in house.

Write if you have questions.

Bruel & KJaer – Use of Weighting Functions in DFT / FFT Analysis Part 1
Bruel & KJaer – Use of Weighting Functions in DFT / FFT Analysis Part 2

Bruel & Kjaer – Dual Channel FFT Analysis part 1
Bruel & Kjaer – Dual Channel FFT Analysis part 2

Bruel & Kjaer to DPA – What you probably don’t know thread

Syn Aud Con – measurement mic shoot out

Until I read this , I hadn’t really thought about the randomness of pink noise but now I will have to find out more…

ACO Pacific – Extremely Random pink noise generator

“Many computer generated noise sources utilize 16 bit registers with 32 kHz sample rates. The resulting 2 second pseudo-random cycle has a limited bandwidth – typically 50 Hz to 16 kHz and the repeated 2 second “fold-over” of the registers produce a measurable “click” – a source of measurement error often requiring averaging 10’s or 100’s measurements. The 3025 has a useful bandwidth of 1.6 Hz to 39 kHz with 119 uHz line resolution.”

How random is software based pink noise generated in Smaart, Spectra Foo, etc…? What about hardware based pink noise generators?

ACO Pacific – windscreen study

coming soon

Sound Web thread on measurement mic stability

What I Learned In Smaart Class

I found this article reading over posts in the Rational Acoustics forum here:
Rational Acoustics – user forums thread on Smaart Class

Rational Acoustics forum thread regarding an issue using the Audix TM1 mic & Lectrosonics TM400 RF kit together

Here is a quote from this thread:

Sound Forum thread about the Audix TM1 / Lectrosonics TM400 kit – RF issue

QUOTE:

Karl Winkler
04-13-2012, 12:02 PM

I know I have asked this before but…. can the TM400 work with an audix TM1 ? or do you still need a small extension cable to break up the RF issue ?

At present, the same problem still exists between the TM1 and the HM transmitter (along with the older UH400TM transmitter). We have reached a standstill mostly because both engineering teams feel that their product is not the issue (ahem…) so yes, a short cable changes the RF properties of the combination so that the problem is reduced to the point where it is usable.

END QUOTE:

It appears that a short XLR cable between the Audix TM1 & the transmitter resolves the issue.

Why must one use an omni mic instead of a cardioid mic? After all, we all own cardioid mics but it is uncommon to own an omni.

The most popular (2) mics ever invented are the Shure SM58 vocal mic & SM57 instrument mic. Both have cardioid pickup patterns.

Wouldn’t it be swell to just one one of those?

Let’s take a look.

INSERT RESPONSE OF SM58
INSERT RESPONSE OF SM57
INSERT RESPONSE OF OMNI X

What would happen if we used an SM57 (the flatter of the two Shure mics) to tune a sound system?

INSERT SIDE BY SIDE TRACES

One of things you will notice when looking for a measurement mic is that there are no “LARGE” diaphragm options. When it comes to reproduction accuracy, the smaller the diaphragm the more accurate it can reproduce sound.

Here is a great article by DPA that explains why.

DPA University – Large Versus Small Diaphragm article

As a side note, over the years, I have heard engineers who favor large diaphragm mics suggest that one of the reasons is because a large diaphragm mic can reproduce lows better. This is simply not true. It IS true that cardioid mics suffer from what is called “PROXIMITY EFFECT”:
WIKI – Proximity Effect

I would argue that if you want “accurate” low end response, nothing can compete with a small diaphragm omni. One example is the Earthworks M50 with a frequency specification of (3hz to 50khz +1/-3dB)!

M Series measurement mics

Polar Response explained

Coverage Area explained

Phase explained

Frequency Response explained

Impulse Response explained

Earthworks Electronic Calibration File – webpage

Earthworks – Omni Application Guide white paper

The World Beyond 20khz – PDF by David Blackmer of Earthworks Audio Products

This PDF makes for an interesting read.

How Earthworks Measures Mics – PDF

Earthworks Audio website – Kickpad link

Here is the description from the website:

“Just Plug the KickPad™ into the mic line feeding your favorite cardioid kick drum mic and you will experience an incredible instant kick drum sound that is exactly repeatable, every time. The KickPad™ can also be used for vocals and bass. This will work with any device that has a 3-pin XLR. The KickPad™ is a passive device (requiring no Phantom Power) that can be used with either condenser or dynamic microphones. It works for both recording and live performance applications.”

What does it do really???

This evening I setup my measurement rig & patched one channel as a loop out / in with nothing more than a cable. The other channel had the KP1 Kickpad adapter inserted in the middle of two cables. Since I’m demoing Rational Acoustics Smaart 7 at the moment, I made measurements with both Spectra Foo & Smaart 7 to compare the two apps side by side. In case it’s not obvious, the upper trace of both measurements indicate the frequency response. The lower traces indicate the phase response.

Interestingly this frequency adjustment would be hard if not impossible to accomplish with a console’s channels trip EQ. Even 4 band parametric EQ wouldn’t achieve the same curve because flat filters aren’t typically available on audio consoles. I can think of only (2) flat filter options I’ve seen in the audio DSP world. All based on modern DSP (digital signal processing).

1. Harrison’s EQ plugins for Mixbus is unique in that instead of being able to adjust level, frequency & Q parameters, your interface is the equivalent of a pen:

Harrison Mixbus EQ plugin
Harrison Mixbus EQ – Youtube video
Harrison Mixbus Mastering EQ plugin
Harrison Mixbus Mastering EQ – Youtube video

2.Lake’s MESA Processing:

Mesa Filter

Sadly neither of these options are available to the general public in a form that can be added to a kickdrum channel on a live console so for those who want a simple passive EQ / PAD device designed specifically for use with a kick drum mic, VIVA LA KICKPAD!

The screenshot below shows a measurement made with both Spectra Foo Complete and Smaart 7 at the same time.

In order to get things to look comparable I had to adjust some of the default settings in Spectra Foo to match the default settings in Smaart 7. I’ve included a screen short of those settings below.

SIZE

Smaart 7 has a default minimum size which is rather large on a MacBook Pro 15″ Screen. Spectra Foo is infinitely size adjustable. I looked for a way to kill some of the side bars in Smaart but didn’t find any.

Here is a comparison of the two apps maximized for my MacBook Pro 15″ screen.

Some of the exciting features of Smaart 7 which are likely important enough to purchase it are as follows:

Live IR – The impulse response updates as the mic is moved
Delay Tracking – The delay time updates constantly

For live audio measurement purposes, this is pretty spectacular.

Some exciting features of Spectra Foo include the “studiocentric” tools. Keep in mind that Spectra Foo was born as a Pro Tools measurement plugin & became a stand alone app.

I see good reasons to have both apps & with (2) you can verify one against the other to make sure your rig is working properly.

A win/win situation.

In order to measure the latency & A/D DA conversion quality of an interface, you can loop out of the interface & return the signal to an input.

The following are latency measurements at the (4) different sample rates:

44.1k – 1.678 ms
48k – 1.542 ms
88k – .839 ms
96k – .771 ms

Obviously if you want the lowest latency possible, you should run at a sample rate of 96k.

Next let’s look at the frequency & phase response at each sample rate.

These videos are fascinating. An interview between Metric Halo’s BJ Butler & Stephan

Tech Talk #1: The digits in digital audio

Tech Talk #2: Fixed and floating point audio

Tech Talk #3: Levels

Tech Talk #4: Are all digital EQs the same? Part 1

Tech Talk #4: Are all digital EQs the same? Part 2

Tech Talk #5: Odds & Ends, Part 1

Tech Talk #5: Odds & Ends, Part 2

I’ve never much liked an open back guitar cabinet for my own rig but there are many classic designs that are open back designs.

Open back means that the speaker is acting like a dipole.

I would guess that the debate over which sounds better, digital or analog, will never be truly settled but one thing is clear.

Analog = instantaneous
Digital = delayed (latency)

You can route an analog signal thru a rack full of analog gear & the signal will still arrive at the final destination in (0) time. The electrons are traveling at almost the speed of light.

Due to the nature of digital conversion. Analog to Digital / Digital to Analog, a digitized signal cannot arrive in (0) time. There is inherent latency. The best we can hope for is to minimize the latency & use it when possible to our advantage.

One thing is for sure. Digital audio is here to stay & analog audio is a dying breed although there is hope when I see this in a grocery store check out line in 2014.

Some good news. The CD standard of 16 bit / 44.1k sample rate is fading. It can be argued that the 16/44.1 CD standard was never acceptable. A decision based on A/D D/A conversion technology at the time & cost savings.

I recently watched a video by Metric Halo’s own BJ Butler who stated that 24bit / 96k audio is as good as we need for capturing audio correctly. If you don’t want to watcht the whole thing, FF to around 17 minutes.

Metric Halo’s BJ Butler discusses sample & bit rates

@ a sample rate of 96k, latency drops by 1/2 compared to an audio device that is running at 48k. Even less latency @ 96k than at 44.1k. And the latency at 96k is small enough to be acceptable. 96k also sounds good so as long as the industry continues to move toward 96k sample rates as a standard, I think we will live through the growing pains of abandoning our analog roots & becoming digital sprouts:)

How much less latency are we talking about?
How do we measure the latency in and out of our gear?
How much latency is acceptable before it begins to cause issues?

For studio purposes where musician’s are monitoring themselves as they perform, the lower the latency amount the better. Some humans are bothered by even small amounts of latency when monitoring themselves (singers especially) where as some humans don’t seem to mind. What we do know is that when you’re trying to play or sing in time, the accuracy to which time is presented is important so I think we should be doing everything possible to avoid any unnecessary latency for monitoring purposes.