What is phase response? Why is it useful in audio testing? Phase response is one of the most fundamental measurements used to describe the behavior and linear performance of a transducer, such as a loudspeaker. It describes the phase relationship between a transducer’s input signal and its acoustic output as a function of frequency. This short educational video provides a clear engineering explanation of phase measurements and their role in electroacoustic testing.
Watch the Video Here:
Video Transcript:
Phase response is another fundamental way to describe the behavior of a transducer, such as a loudspeaker.
While frequency response tells us how much output a speaker produces at each frequency, phase response tells us when that output arrives relative to the input signal.
Ideally, all frequencies would have the same time delay like they were emanating from a point source. This would result in a minimum phase response.
In reality, physical and electrical factors such as mass and compliance, enclosure behavior, resonances, and crossover networks introduce frequency-dependent phase shifts. This means different parts of the spectrum arrive at different times, and the phase is not linear across the frequency range.
This is important because the sounds we listen to, like music and speech, contain many frequencies simultaneously. When those frequencies are no longer time-aligned, transients become smeared, clarity is reduced, and spatial imaging is degraded.
To measure phase, typically we play a test signal – usually a 20Hz to 20kHz continuous sine sweep or log chirp – through a speaker, and capture the acoustic output with a calibrated precision microphone in a controlled environment. This is transmitted, via an audio interface to a computer where FFT-based transfer functions are used to compare the input signal to the output. This extracts both the magnitude and phase components of the system’s transfer function.
We then plot phase response in degrees against frequency. Like frequency response, ideally this is consistent over the frequency range, but in practice it varies.
Phase measurements are essential for aligning drivers, designing and optimizing crossovers, configuring microphone arrays, and ensuring coherent system behavior.
Taken together, frequency response and phase response provide a complete description of a transducer’s linear performance.
How do you measure USB headsets? In this short video, we demonstrate how to measure USB headsets using SoundCheck. You’ll see how to configure digital input and output channels, set up the headset in the hardware editor, and perform measurements including left and right earphone frequency response, THD, sensitivity, and microphone performance. This method can be used for any USB audio device, making it easy to accurately evaluate both playback and recording performance.
Today I’m going to demonstrate how we can measure a USB headset using SoundCheck. We’ll test both the left and right ears and the microphone.
SoundCheck supports both analog and digital inputs and outputs, so to test a USB headset, all we need to do is plug it into our computer’s USB port and configure a new digital channel in the hardware setup, then we can test it just like a conventional headset.
Let’s review our setup. We’ll use a head and torso simulator to capture the headset’s response, and I’m powering that with a SoundConnect 2 power supply. The microphones within HATS transmit the signal to the computer via the AmpConnect 621 Audio Interface, and that also has a built-in amplifier to power the HATS mouth, which we need for the microphone test.
First, I’ll configure the USB headset in SoundCheck. We do this in the Hardware Setup. You can see here we already have the input and output channels of the AmpConnect 621 configured – it’s full plug and play so that happens automatically.
I’ll plug the USB headset into the USB port of my laptop. In the hardware editor, I’ll right click to create an input channel. I’ll set it to use the Windows WDM driver, and now it shows me all the available devices. We see the headset mic there – let’s set the type to digital. Since it’s digital we don’t need to enter anything in the calibration field. I’ll leave the sampling rate as the default, as my audio interface is using this value. We’ll also do the same for the outputs. We’ll set the driver and pick the device, there’s 2 channels so we’ll set these to left and right, and again it’s a digital device.
First we’ll measure the headphone response, and we’ll use the USB headset test sequence that’s available for free on our website.
When we open our sequence, it prompts us to re-link the signal paths used in the sequence. Let’s add them to system calibration.
And now we need to assign hardware channels and calibration files. It’s warning me that the signal paths I just added don’t have associated hardware channels and calibration data, so we’ll click through all of those so we can add them. Here’s our signal paths.
We’ll assign signal paths digital out 1 and 2 to USB Headset out L and R. Since these are digital outputs the calibration is unity digital out
Now we’ll do the same for the input headset sequence. We’ll set Digital In 1 to the USB headset and calibration to unity digital in
Let’s just do a quick check and make sure all our signal paths are correct. It’s looking good
So now we’re all set up and ready to measure.
Let’s hit start.
The sequence plays our test stimulus, and it gives us the left and right frequency response, and also the THD for the left and right earphone. Over here we see the waveforms, the left and right recorded response, and down here we calculate our average sensitivity. Sensitivity is a useful production line test to confirm that the average level across a bunch of headphones is reasonably stable. If the sensitivity swings dramatically in either direction from sample-to-sample, it can indicate a quality problem in manufacturing. We also show the left/right tracking – that’s the difference in sensitivity between the left and right ear over the frequency range.
Now we have some results, let’s take a look behind the scenes at what our test sequence is doing.
We’re using a Stweep that sweeps from 20 kHz down to 100 hertz at 12th octave resolution. This is a compound stimulus with a very short, 1kHz pilot tone, before the Stweep. The pilot tone isn’t analyzed, but we use it in this frequency shift step here to align the time and phase of the response signal with the stimulus before we analyze it.
This feature ensures accurate analysis. It compensates for any unknown phase differences that are introduced from the stimulus and response being played through different hardware devices – the stimulus is played through the laptop via USB, and the response is recorded through our audio interface.
Next, our analysis step compares the stimulus waveform with the frequency-shifted version of the response waveform to get the fundamental. There’s also a post-processing curve averaging step that calculates the average sensitivity.
Lastly, the display step that defines what graphs and results are shown. So that’s the earphones, now let’s test the microphone. We’re already connected and set up, so I’ll tell the sequence to continue and measure the microphone.
So that’s the earphones, now let’s test the microphone. We’re already connected and set up, so I’ll swap over to the microphone test sequence and measure the microphone.
We heard our test stimulus coming out of the HATS mouth, and being recorded by the headset microphone. And here we can see the frequency response and sensitivity of the microphone.
And if we look at the sequence, we can see it’s very similar to the headphone. We’re using an 8 Khz to 100Hz test stimulus, again with a trigger tone to align our stimulus and response waveforms. We have our frequency shift calculation, just like before. Then our analysis step to calculate frequency response, and also a post-processing step to calculate sensitivity. And of course we have a display step to determine what is shown on screen.As you can see, because SoundCheck supports any combination of analog and digital audio devices, it’s very simple to measure a USB headset. All you need to do is make sure you have it set up as a hardware device in SoundCheck, and use a compound stimulus and frequency shift step to align your waveforms before analysis. You can measure any other USB connected audio device in the same way. Check out our website to learn more and download the SoundCheck test sequence.
We’re excited to announce the launch of a brand new instructional series Audio Basics. In this series we walk you through the key measurements used to characterize audio transducers, explaining what they are, what they tell us about performance and why they matter. We’ll also look at how each parameter is measured, and how the results are used in both development, and production.
The first Audio Basics video is all about frequency response.
What is frequency response? Why is it useful in audio testing? Frequency response is one of the most fundamental measurements used to describe the performance and sound of a transducer, such as a loudspeaker, headphone, or microphone. This short educational video explains what frequency response is, how it is measured, and why it is one of the most important tools in electroacoustic testing.
Watch the Video Here:
Video Transcript:
Frequency response is one of the most fundamental ways to describe the way a transducer, such as a loudspeaker, sounds.
A speaker converts an electrical signal into an acoustic one, and its frequency response describes the magnitude of that conversion as a function of frequency.
In an ideal world, for any given input voltage, it would produce a constant output sound pressure level at each frequency.
However, in reality, deviations from this ideal response come from physical limitations, enclosure resonances, diaphragm breakup and more, and might look more like this.
We can see parts of the spectrum are emphasized and others are attenuated, and this is what gives a speaker its unique sound.
A boosted bass might be described as ‘warm sounding’, while a boost in the higher frequencies may make it seem more bright. What we describe subjectively as “bright,” “muddy,” or “neutral” largely maps back to frequency response.
Measuring frequency response enables objective comparison, informs design tradeoffs, and provides the baseline for equalization, calibration, and system optimization.
Measuring frequency response is simple. We play a test signal – usually a continuous sine sweep or log chirp, from 20Hz to 20kHz – through a speaker and capture the response with a calibrated precision microphone in a controlled environment. The response in dB is then plotted against frequency to show the frequency response graphically.
This is used in transducer R&D to design a signature sound, and also in production to ensure that the speaker is within spec.
https://www.listeninc.com/wp/media/2026/08/Frequency-Response-Thumbnail-scaled.png14402560Renee Lucashttps://www.listeninc.com/wp/media/logo.pngRenee Lucas2026-08-07 10:37:522026-08-28 09:01:33Audio Basics: Frequency Response
Last week, industry expert and Listen President Steve Temme presented the latest recommendations for automotive audio measurement and the evolving methods used to evaluate modern in-car infotainment systems at Harman’s North American Headquarters in Novi, Michigan. Recorded ahead of the AES Automotive Conference, the presentation discusses the work of the AES Technical Committee on Automotive Audio (TC-AA) and the industry’s efforts to standardize automotive acoustic measurements.
This session provides a rare opportunity to gain valuable knowledge in the rapidly evolving in-car audio and infotainment industry. Whether you’re developing OEM infotainment systems, benchmarking competitor vehicles, performing automotive acoustic research, or implementing standardized measurement procedures, you’ll gain practical insights into the current state of automotive audio testing.
Steve unpacks the latest work from the TC-AA, highlighting new standardized testing models for frequency response, Max SPL, and impulsive distortion. He also showcases advanced characterization techniques, such as seat-to-seat spectral uniformity and psychoacoustic distortion methods.
How do you measure hearing protection devices? In this short video, we demonstrate how to measure hearing protection using SoundCheck in accordance with the ANSI S3.19-1974 standard. You’ll see how to set up a head and torso simulator, calibrate the system, and perform repeated unoccluded and occluded measurements using pink noise to accurately calculate attenuation and Noise Reduction Rating (NRR). This method can be used for a wide range of hearing protection devices to ensure reliable and standardized performance testing.
Watch this Video Demo of Measuring Hearing Protection
Video Transcript: How to Measure Hearing Protection
In this short video, we’ll measure the performance of hearing protection devices and calculate the Noise Reduction Rating using the ANSI S3.19-standard.
Our setup is simple. We have a head and torso simulator to wear the hearing protectors and capture the signal, a speaker to play the test stimulus, and a calibrated reference microphone for speaker calibration. The stimulus and response signals are transmitted to and from the computer using the AmpConnect 621 audio interface which also contains an amplifier to power the speaker. I also have a SoundConnect 2 to provide 200V polarization voltage to the ears since I’m using an older HATS.The newer versions have pre-polarized ear simulators so they can be powered directly from Listen’s audio interfaces.
We’ll play a pink noise test signal and capture the response with the ears unoccluded. We’ll then put on the hearing protectors and make the occluded measurement. We’ll re-position and re-measure several times and take an average to account for fit variations. We’ll then compare the occluded and unoccluded data to calculate the noise reduction ratio, or NRR.
My reference microphone is already calibrated, and connected to input 1. Now I need to calibrate the ear simulator input signal paths and source speaker in the calibration editor.
Let’s go to Setup > Calibration and go to Input Signal paths > Ear Sim L. I have connected this to Input 2, now I’m ready to calibrate the ear. I’m going to use a B&K 4231 calibrator with the coupler UA 1546, hit the calibrate button and save the sensitivity value. Now we’ll repeat that for the other ear, Ear Sim R which is connected to Input 3..
Next, I need to calibrate the source speaker. I’ll place the reference mic as close to one side of the hearing protector test fixture as possible. We’ll navigate to the output signal paths tab inside the Calibration editor, and select Source speaker. I’ll make sure it is assigned to Output 3 because that’s the amplifier output of the AmpConnect 621 where we have it connected. Now I’ll click Calibrate using Reference mic as the input. We’ll use a level of 80dBSPL from 10kHz-100Hz, the same frequency range that we’ll use in the sequence. It’s not quite flat, so we’ll iterate a couple more times until the EQ’d response is flat. This looks good. Now, everything is calibrated, we’re ready to run the sequence.
Let’s hit start.
First, the sequence wants to know how many trials I want to do. We’re going to make this measurement several times, removing the DUT and placing it back so that we can average the response to account for fit variation. We’ll run it 5 times for the purposes of this demo.
First, we’ll capture the unoccluded response – this means measuring the sound levels without any hearing protection in place. The test sequence will play pink noise through the speaker and record the sound spectrum at both ears of the fixture. Let’s hit enter.
Let’s save this measurement.
Now the sequence is asking me to place the hearing protection device onto the test fixture and we’ll continue to repeat the test to measure the occluded response. Now we’ll just follow the instructions and do it four more times.
Next, the sequence performs postprocessing steps on these unoccluded and occluded measurements to calculate the attenuation at different frequencies, average attenuation, standard deviation and finally the Noise Reduction Rating (NRR) – you can see its 26.6 on the left and 32.1 on the right. These measurements provide a complete picture of the device’s performance.
And that’s it! Using this pre-written sequence – which you can download from our website – it’s quick and easy to accurately measure attenuation and determine the Noise Reduction Rating of any hearing protection device to the ANSI standard.
How do you measure hearing aids? In this short video, we demonstrate how to test hearing aids using SoundCheck with standard ANSI S3.22 measurement techniques. You’ll see how to set up an anechoic test box and coupler, and configure SoundCheck for key measurements including frequency response, input/output linearity, OSPL 90, and release time. This method can be used to accurately evaluate hearing aid performance and ensure compliance with industry standards.
Today, I’ll demonstrate a hearing aid measurement using SoundCheck with some of our standard hearing aid test sequences. We’ll use some of the tests outlined in the ANSI S.22 hearing aid test standard – frequency response, input/output linearity, OSPL 90, and release time.
First let’s take a look at our hardware setup.
It’s a little more involved than something like a loudspeaker measurement because hearing aids have some unique test requirements. For a start, we need an anechoic test box – this one’s a B&K 4232 but you could use anything similar. It has a speaker integrated inside, which we’ll play our test signal out of. We’ll put our hearing aid in this coupler with a ½” pressure microphone. We also have a SCM reference free-field microphone which is also going to go in the test box. The ½” mic needs LEMO power, so it’s connected to a SoundConnect 2 power supply, and the mic input then goes into the AmpConnect 621 Audio Interface. The AmpConnect also provides power for the SCM mic and amplification for the speaker in the test box.
If I were running the complete suite of ANSI S.22 tests, I would also need a DC power supply and current monitor but since the tests we’re running today don’t require voltage regulation or current monitoring, we’ll just power the hearing aid from its internal battery.
I’ve already calibrated the source speaker located inside the test box using my SCM reference microphone pointed at the source speaker. The coupler is configured with a ½” microphone capsule, and we’ll attach our hearing aid to the coupler’s input – we have some tack in here to get a good seal. Now we’ll put it in the test box so its microphone port is located at the same position I had the reference microphone. Let’s close the box, and we’re ready to test.
Let’s start with Frequency Response. The hearing aid should be set to its reference test gain. We’ll play an equalized stepped sine sweep from 10 kHz – 100 Hz at a level of 60 dBSPL through the test box speaker, and analyze the output of the hearing aid with the Heterodyne algorithm to produce a frequency response. Next, the HFA, or High Frequency Average is calculated by averaging the response values at three frequencies – 1000, 1600, and 2500 Hz. We then subtract 20dB from the HFA, and two post processing steps are used to find the upper and lower frequency points at which the response curve intersects the HFA -20 value. These are the high and low frequency cutoff points. Limits can be applied to all of the curves and values of this sequence. Let’s run it.
And here you see the Fundamental curve falls outside of the upper and lower limits at a few frequency points, the low frequency cutoff is 217 Hz and the high frequency cutoff is a little over 5 kHz.
Our next test is input/output linearity. We’ll leave the hearing aid at reference test gain, and use an amplitude sweep from 50-90 dBSPL played through the test box speaker. This is repeated at four frequencies (1 kHz – 4 kHz) in 1 kHz increments, and the output of the hearing aid is plotted vs. the input on the graph. This test is useful for characterizing any automatic gain control or compressor in the hearing aid, and is used to attenuate the hearing aid’s output above a certain level so as to not further damage the user’s hearing. Let’s run it.
The effect of the automatic gain control is clearly visible at all four frequencies. The x-axis represents the stimulus level and the y-axis the output level from the hearing aid. We can see that the gain limiting kicks in when the input is in the 60-75 dB range, depending on the frequency and note how the overall level of the 4 kHz curve is 10 dB lower than the others because the frequency response of the hearing aid is around 10 dB lower at that frequency relative to the others.
The OSPL 90 test measures the hearing aid’s performance with an input signal of 90 dBSPL. For this test we need to set the hearing aid gain to its maximum. An equalized stepped sine sweep from 8 kHz – 200 Hz is played at a level of 90 dBSPL, and a broadband rms response curve which includes harmonics as specified in the standard, is analyzed from the recorded time waveform of the hearing aid’s output. Next, the HFA (High Frequency Average) is calculated by averaging the values at three frequencies (1000, 1600, 2500 Hz), and the Max OSPL is calculated by finding the maximum point on the broadband response curve. Let’s take a look.
The OSPL-90 curve looks much like the frequency response curve from our earlier measurement and considering that the input to the hearing aid is a calibrated sweep at 90 dB, the amount of gain in our measurement is between 25 and 40 dB across most of the sweep range. The high frequency average from this measurement is 126.7 dB and the maximum value on the curve is 129.7 dB.
Lastly we’re going to measure the release time of the hearing aid. Again, we want the gain set to max, and we’ll play a 2 kHz sine tone at 90 dBSPL for 1 second, then immediately drop it to 55 dBSPL for 2 more seconds. A band limited time envelope to improve the measurement signal to noise ratio – from 1.5-2.5 kHz – is calculated and then run through a post processing step, to calculate the release time – that’s the time the coupler SPL remains within 4 dB of the steady value for the 55-dB input. [run test]
Here we can see that the hearing aid output initially drops down to around 85 dB but after 45 ms the release time has stabilized and the hearing aid maintains a steady output level of 115 dB.
So that’s four quick examples of tests from the ANSI S.22 hearing aid standard. Of course, once you have the equipment for these, there are many other hearing aid tests that you can run using the same test setup, and you can configure tests for the complete ANSI S.22 standard in SoundCheck. In fact we’ve already done it, and we sell the test sequence package with all 20 measurements as an add-on to SoundCheck so you don’t even have to program it yourself! Please contact us to learn more.
https://www.listeninc.com/wp/media/2026/04/Hearing-Aids-TN-1080p.png10801920Julius Wijonohttps://www.listeninc.com/wp/media/logo.pngJulius Wijono2026-04-29 16:09:242026-08-28 14:35:48Hearing Aid Measurement
How do you measure devices that can’t be directly connected to an audio interface? In this video, we demonstrate how to measure open loop devices such as smartphones and smart speakers using SoundCheck. You’ll learn how to generate a stimulus file, play it back from the device under test, and use triggered recording with a microphone and audio interface to capture the response for analysis. This flexible technique allows accurate measurement of frequency response, distortion, THD, and Rub & Buzz – even when playback and recording are asynchronous.
Video Transcript: How to Measure Open Loop Devices
Devices like smartphones and smart speakers can be challenging to test as they can’t be directly connected to conventional interfaces to make audio measurements. This means playback and recording are asynchronous.
To make accurate measurements, we use a technique called open loop testing. This lets us manually load the stimulus signal onto the device and play it using a triggered record. This means SoundCheck starts recording the response once it hears a certain sound. This might be a level threshold, or a specific frequency, or even a chirp. Once it hears that trigger it will begin the acquisition, record the stimulus, and then analyze it as normal.
Let’s take a look. I’m going to use a standard test sequence that comes with SoundCheck, but first let’s look at our test configuration.
We’re using SoundCheck to create the stimulus file, and we’ll load that manually onto our device – you can send it by email or download via the cloud, or even store it in the device firmware. We’ll then run the sequence, and play the test signal from the phone. When the trigger tone is detected, the software will start recording using an SCM microphone which is connected to the software via an AudioConnect 2 audio interface, and analyze the response.
Let’s go ahead and run the test.
The sequence is giving me the option to create the stimulus files ready to load onto the device, but I’ve already done that so I’ll just decline and let the sequence continue….
Now it’s listening for the trigger. I’ll play the sound…. And you can see, it went ahead and captured the response – we have our recorded time waveform here. Over here we have our fundamental, and also the second and third harmonics. Down here we have THD and Rub & Buzz, and over here we have perceptual Rub & Buzz. You can see we’ve got some distortion across the spectrum, and some Rub & Buzz at low frequencies – that’s not really a surprise – phone speakers tend to be small and inexpensive so these issues are common. Once you have the waveform captured you can run any analysis or calculation on it, so if you wanted, you could add other measurements to this test.
Now that we’ve seen it in action, let’s go back to the sequence and learn a bit more about triggered recording. We’ll start with the stimulus.
As you see, we still have a standard stimulus step from XHz to YkHz, but when we look at it closer we can see that it has two sections. First we have a frequency log chirp that tells SoundCheck when to start recording. This is much more robust than simple level or tone triggers, especially if you’re in a production environment where you might have background noise or even pick up test tones from other lines. And you can see here we’ve set it so that this portion of the stimulus does not get analyzed.
The second portion of the stimulus is our test signal – a standard frequency stepped sweep from 10kHz down to 100Hz and we have ‘analyze’ set to ‘yes’ for that one.
The acquisition step must also be configured for triggered record. Here, we make sure the ‘triggered record’ checkbox is checked, and then select the type of trigger that we’re using. We used ‘level and cross-correlation’, which looks for the specific chirp that we used – that one is the most robust but we also offer simpler level and frequency triggers.
So, as you can see, open loop measurements are simple with SoundCheck. This method can be applied to any device that doesn’t have a direct connection like smart speakers, smart glasses, phones or even robots. Check out our website to learn more.
https://www.listeninc.com/wp/media/2026/03/Open-Loop-Demo-TN-scaled.png14402560Julius Wijonohttps://www.listeninc.com/wp/media/logo.pngJulius Wijono2026-03-11 15:53:332026-08-28 14:29:19Open Loop Test Measurement
Recent transducer development, particularly piezo-MEMS loudspeakers, have brought renewed attention to Normalized Distortion (also known as Frequency Normalized Distortion), because it is essential to accurately characterize their behavior.
Normalized Distortion is a measurement algorithm first proposed in a 1993 AES paper by Listen president Steve Temme during his time at Brüel & Kjær. Although it has been available in SoundCheck since 1996 and used by some engineers for nearly 30 years, interest in the technique has surged recently. This renewed focus is driven both by the industry’s increasing emphasis on correlating measurements with audibility – an area Listen has long prioritized – and by the emergence of new microspeaker technologies that cannot be fully evaluated using conventional distortion measurements.
In this seminar, we’ll first explain distortion mechanisms and talk about correlating measured results to what we hear using masking curves. We’ll then examine how normalized distortion differs from the conventional approach and show why it’s essential for characterizing modern piezo-MEMS loudspeakers.
Learn how to average measurement data the right way in SoundCheck. In this video, we’ll walk through curve averaging, averaging multiple curves, and waveform RMS calculations to get meaningful results from your data.
Additional Resources for SoundCheck
Detailed explanations of how to use SoundCheck can be found in the SoundCheck manual.
Video Transcript: How to Average Data in SoundCheck
There are several ways we can average data in SoundCheck. The sequence step that you use depends on your input data and what you’re actually trying to get out of it. Let’s look at some examples.
First, let’s find the average of a single curve. We can do this using a post-processing curve average step. Here, my input is curve one. My operation is average. And when I output this value, it’ll give me the average value of all of the magnitude values from my curve as the Y axis, and also the average phase data as the Z axis on the value.
Now let’s find the average of a number of curves. We use this all the time, for example finding the power average of multiple microphones in an array. Here I have eight different curves and I want to find the average not of one but of all eight. Instead of a curve average step here, I’ll use an average curve step. First, I need to put all these curves into a group. I’ll just call it curves. And then in this post-processing step, I’ll choose the group that I’m interested in, which could be either curves or waveforms, and the averaging type. I have the option of complex, which will include phase data, or power, which won’t. I’ll use power for now. Rather than outputting a single value like we had before, this generates a curve that’s the average of all of my curves. And I can even put this on my display. And you’ll see that this curve here, the average curve, is now on my graph alongside the other curves.
We can also average waveform data. Here I have a one volt RMS sine tone. I might want to know what the average of this is here. If we use a curve average step, because this is a waveform and there’s positive and negative data, the average is going to be about zero, which isn’t very interesting to me. Instead, we could use the waveform RMS step. I can input my waveform. And this gives me an output of one volt RMS, which is as expected since that’s what I have set for my waveform here.
So that’s 3 easy ways we can find averages in SoundCheck. Check the manual for more details.
https://www.listeninc.com/wp/media/2026/01/Average-Data-Blog-TN.png10801080Julius Wijonohttps://www.listeninc.com/wp/media/logo.pngJulius Wijono2026-01-06 05:00:152026-08-27 13:59:38How to Average Data in SoundCheck
Understand when to use SoundCheck’s FFT Analyzer and RTA live versus in a sequence. In this video, we’ll show how live multi-instrument FFT/RTA measurements are ideal for real-time adjustments and troubleshooting, while sequence-based analysis steps enable automated, repeatable measurements, averaging over time, and advanced calculations like frequency response.
Additional Resources for SoundCheck
Detailed explanations of how to use SoundCheck can be found in the SoundCheck manual.
Video Transcript: SoundCheck’s FFT Analyzer and RTA: Live or in a Sequence
In SoundCheck, you can run an FFT or an RTA either live via the multi-instrument or through the analysis step in a sequence.
Let’s outline the differences.
Taking an FFT or an RTA via the multi-instrument is always an input-only measurement and it displays the live curves and changes over here in the graph. So, as I’m talking, you’ll see that these are displaying the current measurement on my reference microphone. This is very helpful for live adjustments – for example if I’m changing the DSP on a device and I want to see how it behaves, or checking the seal on some headphones. When you’re looking at it live, you can change settings on the fly, so using these controls I can see what happens if I switch from average to max hold, min, or change the windowing or weighting directly in the instrument. Most of this can be done in the analysis step as well, but you might have to rerun the acquisition or open an editor and click apply before you’re able to see the results, whereas here you can see the changes instantly.
The instrument is also very helpful for troubleshooting. If something is wrong with a measurement,you can just open up an FFT or an RTA to troubleshoot what’s happening and then solve it.
If we instead look at the FFT and RTA in a sequence, I can use an acquisition step set to record only. I can then run that reference microphone recording through an FFT spectrum analysis step and an RTA spectrum analysis step to get the FFT and RTA spectrum response for my reference mic. Instead of seeing live changes, I can define the full time length I’m recording and then average across that full recording. This is useful if you want to test something over a set number of seconds and just see the total average there without any additional user interaction. It’s much easier to automate.
There are some calculations we can do in the live multi-instrument. For example, I can look at the power average of multiple inputs and do some basic operations like subtraction and addition. One of the options we have here that’s exclusive to the live multi-instrument is a masking curve. I can play a signal here, add a masking curve, and I can then quickly see what parts of my spectrum are being masked by a dominant tone. That’s only available in the live multi-instrument.
Virtual instruments can be added to sequences in an acquisition step. When used this way, both the instrument and analysis steps output the curves to the memory list of the sequence once the step is complete. Post processing, limits, or statistics can be used for further calculations on the measurements.
There are two common use cases where an analysis step would be required rather than using the virtual instrument: Using previously recorded waveforms and measuring a frequency response.
For example, if I have wave file recordings from a past measurement, or an open-loop test, I can recall those and add them directly into an analysis step.
We also use an RTA analysis step to measure the frequency response of a device under test. The virtual instrument only calculates an RTA Spectrum of the input signal path, but the analysis is able to calculate spectra for both the Stimulus and Analysis waveforms. This then allows a frequency response calculation. The FFT analysis step will only measure a spectrum for the response waveform, but using the Transfer Function analysis step you can perform complex FFT on the stimulus and response waveform to output a frequency response.
https://www.listeninc.com/wp/media/2025/12/FFT-Analyzer-and-RTA-Blog-TN.png10801080Julius Wijonohttps://www.listeninc.com/wp/media/logo.pngJulius Wijono2025-12-23 05:00:172026-08-27 13:49:53SoundCheck’s FFT Analyzer and RTA: Live or in a Sequence
