100 Things you didn’t know SoundCheck could do

100 Things #100: Production Line Audio Measurement With SoundCheck

Everyone knows SoundCheck is a versatile and flexible R&D audio test system. But did you know it’s also fast and cost-effective for production line audio measurement, and offers unrivaled integration with larger factory test environments?

End of line testing is nothing new to us. We started the global trend from human listeners and expensive hardware analyzers to software-based test systems back in 1995. Many of the measurements we introduced in the 1990s are still used today. Besides introducing newer and better measurement methods like perceptual algorithms, we’re driving the integration of audio testing within a larger factory test environment. Let’s take a look at some of the things that make SoundCheck great for end-of-line tests.

Production Line Audio Measurement with SoundCheck

 

Learn More About SoundCheck’s Production Line Features

Seminar Recording: External Control of SoundCheck. Detailed information about controlling SoundCheck as part of a large factory automation system.

Video: 100 Things #85: Integrate SoundCheck with your Database.

Video: 100 Things #11: External Control Using TCP/IP

Transitioning Audio Tests from R&D to the Production Line. An article by Steve Temme, reprinted from the March 2023 edition of AudioXpress.

 

Video Script: Production Line Audio Measurement

Everyone knows SoundCheck is a versatile and flexible R&D audio test system. But did you know it’s also fast and cost-effective for production line testing, and offers unrivaled integration with larger factory test environments?

End of line testing is nothing new to us. We started the global trend from human listeners and expensive hardware analyzers to software-based test systems back in 1995. Many of the measurements we introduced in the 1990s are still used today. Besides introducing newer and better measurement methods like perceptual algorithms, we’re driving the integration of audio testing within a larger factory test environment. Let’s take a look at some of the things that make SoundCheck great for end-of-line tests.

Most importantly, SoundCheck’s fast and reliable. Every test algorithm we’ve designed has speed and noise immunity at the forefront, from our unique stepped sine wave stimulus, Stweep, and Harmonictrak analysis back in 1995 to the second generation of perceptual distortion measurements in more recent years. And all our production line measurements, use the same stimulus to ensure fast throughput with simultaneous measurement of all parameters.

Soundcheck is hardware-agnostic, and compatible with many audio interfaces from our own custom designed all-in-one hardware to off the shelf soundcards. It even supports audio over IP with Dante. It works with any brand of amplifiers, microphones, couplers and test jigs. It’s also easy to control footswitches, PLCs, barcode readers and other production line equipment through a custom step in a test sequence. This gives you total flexibility, whether you are re-using existing hardware or building a system from scratch.

Both hardware and software are modular, so you can get the production functionality you need, without paying for anything that you don’t. Although a production system is significantly cheaper than an R&D SoundCheck system, it’s still fully compatible – you can create tests on an R&D system and send them to your production systems, or bring results from your production system back into an R&D system for detailed analysis.

However, it’s the seamless integration with custom factory test systems that really differentiates SoundCheck.

Full TCP/IP control lets SoundCheck communicate on any operating system, via any TCP/IP-supportive language including python, c-sharp and Labview. TCP/IP commands can trigger a test, pass the output back to an external program, or even pull in externally stored sequence parameters such as limits and stimuli. This allows the same test sequence to be used for many different products, reducing the sequence maintenance burden.

SoundCheck is just as flexible for saving data. Standard data formats include text, csv, Excel, TDMS, Matlab and SoundCheck’s open source binary file format. There’s also a plugin for WATS Test Data Management software. You can use an autosave step in your sequence to write curves, values, results, or waveforms directly to an SQL database each time a sequence is run, and industry standard tools can then be harnessed to run analytics over large data sets. If these options aren’t enough, all the data, curves, and other  items saved in SoundCheck’s memory list, can always be accessed directly via TCP/IP, so you can write your own customized program to collect exactly the SoundCheck data you need.

SoundCheck’s built-in security features provide peace of mind if you share your tests with manufacturing partners. Sequence protection locks and hides all the information in a sequence so that it can be run, but not viewed or altered. So you have confidence that your products are tested exactly how you intended. No-one can adjust the limits to achieve higher yield, and it removes the risk of  your tests being modified and re-purposed for use on other product lines. To add further security and measurement confidence, a sequence can even be configured to only run on a particular SoundCheck system, or block of system hardware keys.

These features let you bring the power of SoundCheck into pretty much any large automated test platform, no matter what software and operating system it is running on. Talk to your sales engineer to learn more.

 

 

100 Things #99: Calibrate Signal Paths with Any Interface

Calibrating signal paths is a critical part of any audio measurement, and SoundCheck offers the ultimate flexibility for calibrating audio interfaces. Whether you need two channels or sixty-four, analog or digital, each has its own unique configuration and there is no limit on the number of channels that can be calibrated. For example, its possible to have some channels calibrated with a 6-mic array for recording a response, while others are configured to measure motor vibration and RPM speed. Not only can you mix different devices, but each channel can be calibrated using different audio drivers so it’s no problem to combine something like a Bluetooth headset with analog ear simulators and a digital wav file. Learn more in this short video.

Calibrating Audio Signal Paths

 

Learn More About Calibrating Signal Paths in SoundCheck

Check out our calibration tutorials (section 2)

Read more about recommended audio interfaces to use with SoundCheck.

Learn more about AmpConnect 621 and AudioConnect 2, Listen’s self-calibrating audio interfaces

 

Video Script: Calibrate your Signal Paths with any audio interface

In any audio test and measurement system, your signal path begins and ends with your audio interface. Whatever software system and interface you’re using, it’s important to correctly calibrate all input and output channels to get accurate results

SoundCheck offers the ultimate flexibility for calibrating audio interfaces. Any number of channels can be calibrated, so whether you need two channels or sixty-four, each channel has its own unique configuration. This means it’s possible to have some channels calibrated with a 6-mic array for recording a response, while others are configured to measure motor vibration and RPM speed.

Not only can you mix different devices, but each channel can be calibrated using different audio drivers so it’s no big deal if you are combining something like a Bluetooth headset with analog ear simulators and a digital wav file.

This flexibility ensures your test system is future-proofed and can even calibrate hardware that doesn’t exist yet, so long as it conforms to digital audio standards. Over the years we’ve calibrated USB, Bluetooth, Dante, AVB, A2B and more, as well as the more standard types such as WDM, ASIO, Core Audio and WASAPI.

To calibrate an audio device, you need to measure both the Vp in and Vp out values as well as the latency at all the sample rates you will be using.

You can do this directly from the hardware editor itself. You’ll need an AC multimeter that’s accurate to at least 250Hz, and an adapter to insert it in the input / output chain of the audio interface during the calibration process. Should the need arise for field calibration, that can also be done using this method.

To avoid this step, when you purchase a 3rd party interface directly from Listen, we’ll determine the Vp values and the latency before it leaves our facility. All you need to do is enter the device values from the provided calibration sheet into the hardware editor, and you’re ready to start measuring.

Our own all-in-one audio test hardware takes this one step further with self-calibration. With both the 2-channel AudioConnect 2 and the 6-in, 2-out AmpConnect 621, hardware editor  values are measured during manufacture and stored on the device. These values are auto-populated in the hardware editor when it’s connected via USB, so you never need to manually calibrate these devices. If you swap hardware, the calibration is automatically updated.

To learn more about calibrating signal paths in SoundCheck, check out our online knowledgebase and user manual.

 

 

100 Things #98: MEMS Speaker Measurements

MEMS speakers are one of the biggest innovations in speaker technology in recent years. Offering full range performance with compact size and low power, they are rapidly being adopted for use in devices such as earbuds, hearing aids, smart glasses and more. With SoundCheck you can make exactly the same MEMs speaker measurements as you can with conventional mechanical speakers. Watch this short video where we demonstrate frequency response, impedance, and distortion measurements on the xMEMS Montara MEMS speaker.

MEMS Speaker Measurements

 

We’d like to thank Michael Ricci, Sr. Director of Electroacoustic Engineering at xMEMS for the technical guidance on Piezo-MEMS transduction.

Learn More About SoundCheck’s Advanced Features

Read more about more measurement features in SoundCheck.

Learn more about the Normalized Distortion Measurement technique mentioned in the video – we have a short video explaining this, or a longer (but rather old) technical paper.

More information is also available in the  SoundCheck Manual.

 

Video Script: MEMS Speaker Measurements

SoundCheck is one of the most widely used loudspeaker and microspeaker measurement systems in the world, but did you know that it can also measure MEMS micro-speakers? MEMS micro-speakers are rapidly becoming popular for devices such as hearing aids, earbuds, smart glasses and more as they offer full range performance with compact size and low power, and they are also SMT reflowable. They’re constructed in an entirely different way to conventional miniature speakers – rather than using inductive coils and magnets, they use a voltage driven capacitive actuator to provide full range performance.

I’m going to demonstrate a MEMS micro-speaker test using the xMEMS ‘Montara Plus’ full-range Piezo-MEMS microspeaker, that uses a monolithic solid state fabrication. These devices are entirely manufactured with MEMS processes in a semiconductor wafer foundry. When you’re testing these devices, the xMEMS provided driver circuit delivers Voltage bias and boost converter to step up the voltage as piezo-MEMS devices have a very high input impedance and draw very low current.

Here, I’m going to use xMEMS’ own charge amplifier. You’re also going to need to build the speaker into an earbud or make your own test jig in order to test it. I’m going to demonstrate using this test jig, which is actually the one that xMEMS uses for their own measurements, and we’re going to put an ear simulator coupler on it to simulate an in-ear measurement. Aside from that, the test setup’s very similar to what we would use for any other speaker. We have an AudioConnect 2 interface which will power the coupler, and that’s connected to SoundCheck for analysis.

So we have a test sequence that will play the stimulus and analyze the response. You won’t hear it as it’s all in the coupler. And here we can see the results.

Let’s start with the frequency response. You can see it has a very flat response at low frequencies, and then in the higher frequencies you have a resonance due to the piezoelectric material and the resonance of the coupler.

We can also look at the impedance. You can see here that it’s a very different shape from a conventional loudspeaker impedance. The values are much higher but it’s very linear, which makes it easy to compensate for.

We can also look at distortion. The total harmonic distortion is also very linear right up to where we get into the ear canal response.

And while we’re on the subject of distortion, I just want to use the measurements on this device to highlight the importance of using frequency normalized distortion measurement.

With this conventional distortion measurement, you can see the second and third harmonics plotted at their actual measured frequencies, along with the fundamental.

Frequency Normalized distortion measurement compares the harmonic levels to the fundamental level at their measured frequency before their ratio is plotted, rather than the fundamental level at the excitation frequency. This removes the effect of the non-flat frequency response from the distortion and makes it easier to see the peaks in the distortion response independent of the peaks and dips in the fundamental response. Here, you can see both regular THD, the orange line, and normalized THD, the blue line. And as you can see, you have a high Q here at resonance, but apart from that there is very little distortion, so you can focus your efforts on planning around this peak. If you were going by conventional distortion, you could be wasting your time trying to solve resonances you don’t have with this second bump on the graph here.

So that’s piezo-MEMS speaker measurements in a nutshell. Check out our website for more information on testing MEMS speakers, or if you want to learn more about normalized distortion measurement.

 

 

100 Things #97: Zwicker Loudness Measurement

Zwicker Loudness Measurement, an indication of overall perceived loudness level, is calculated in SoundCheck using the Zwicker Loudness post processing step. Instead of just measuring the absolute sound pressure level in dB SPL relative to 20uPa, the Zwicker Loudness algorithm takes into account how humans hear sound level using  the PEAQ international standard. This is an ITU-developed standardized algorithm for objectively measuring perceived audio quality as subjects would in a listening test.

Zwicker Loudness Measurement

Learn More About SoundCheck’s Advanced Features

Read more about more measurement features in SoundCheck.

More information is also available in the  SoundCheck Manual.

 

Video Script: Zwicker Loudness Measurement

Did you know that SoundCheck can calculate the overall perceived loudness level using a Zwicker Loudness post processing step ? Instead of just measuring the absolute sound pressure level in dB SPL relative to 20uPa, the Zwicker Loudness algorithm takes into account how humans hear sound level using  the PEAQ international standard. This is an ITU-developed standardized algorithm for objectively measuring perceived audio quality as subjects would in a listening test.

The input to this post processing step must be a spectrum of a complex signal in pascals or dBSPL. We can easily capture this in SoundCheck using an FFT or RTA broadband measurement using a calibrated Reference Mic signal path. To simulate the non-linearity of the ear, the Zwicker Loudness algorithm then filters these frequencies into auditory bands according to the bark scale – a frequency scale where equal distances correspond with perception. Once the spectrum is plotted on a bark scale, a frequency weighting is applied that correlates to human hearing. Finally, a level compression is applied and the loudness is output in Phons and Sones. The loudness spectrum can optionally be shown with the X axis either in Hertz or Bark.

Knowing the actual perceived loudness of a signal is extremely important for certain applications. For example, listeners that are trying to subjectively compare different headphones will be biased towards the louder one. If I want users to subjectively compare two different headphones, I need to make sure they are played back at the same level to avoid this bias. Looking at the 1kHz sensitivity of each headphone doesn’t take into account the difference in frequency response across the two devices. Often A-weighting is used to correlate measurements to human hearing, but a simple A-weighting curve makes a lot of assumptions such as what level of playback that will be used. Zwicker Loudness gives us a much more accurate perceived loudness, and enables us to precisely match the loudness, in phons, between the two devices regardless of level..

Zwicker Loudness is also widely used in communication testing for measuring loudness of both speech transmission, and ringtones. Check out our website to learn more.

 

100 Things #96: Laser Displacement Measurement of a Loudspeaker

Laser displacement measurement is a technique for measuring the peak displacement of a loudspeaker diaphragm at various power levels, frequencies or both. Did you know that SoundCheck can easily be configured to include a laser signal path? This makes it easy to correlate diaphragm displacement with electrical impedance and audio artifacts. In this short video, we demonstrate laser displacement measurements of a loudspeaker.

Laser Displacement Measurement

Get Our Free Laser Displacement Measurement Test Sequence

Ready to try it for yourself? You can read more and download this laser displacement measurement sequence here.

More information on configuring SoundCheck for use with lasers is also available in the  SoundCheck Manual.

 

Video Script: Laser Displacement Measurement of a Loudspeaker

Displacement lasers can be used to measure the peak displacement of a loudspeaker diaphragm at various power levels, frequencies or both. Did you know that SoundCheck can easily be configured to include a laser signal path? This makes it easy to correlate diaphragm displacement with electrical impedance and audio artifacts. Let’s take a look.

First, we create a Laser Signal Path in Calibration and once that’s done, a new calibrated device file for the instrument.  The sensitivity of most lasers is expressed in Volts per Millimeter and in this case, our laser’s sensitivity is 100 volts per millimeter.  After creating custom units, we can enter the sensitivity value, select a hardware channel and we’re ready to measure!

In this sequence, we’re using a stepped sine sweep starting at 1 kHz and ending at 20 Hz, and  we’re also simultaneously measuring the impedance and frequency response of our speaker under test.  The recorded time waveform from the laser can be analyzed just like any other waveform but there’s one additional post processing step required after analysis, converting the displacement level from RMS to peak.

As you can see, configuring SoundCheck for laser measurements couldn’t be easier. The resulting data can be used to study the displacement of the speaker under test and can even be used in conjunction with other SoundCheck measurements to calculate more advanced metrics such as Thiele-Small parameters. You can learn more about advanced speaker measurements on our website, www.listeninc.com.

 

100 Things #95: Time Domain Waveform Filtering

Time Domain Waveform Filtering in SoundCheck lets you apply any filter to a signal in the time domain instead of the frequency domain. This enables you to apply a filter, such as an A-weighting filter, without affecting the peaks or crest factor of the signal. Filters can also be applied to any waveform in the memory list, such as the stimulus, response, or any intermediate waveform. Watch this short video to learn how standard and custom waveform filters are used.

Time Domain Waveform Filtering

Learn More About SoundCheck’s Advanced Features

Read on about more measurement features in SoundCheck.

More information is also available in the  SoundCheck Manual.

 

Video Script: Using Time Domain Waveform Filtering in SoundCheck

Waveform filtering in SoundCheck lets you apply any filter to a signal in the time domain instead of the frequency domain. This is required when you want to apply a filter, such as an A-weighting filter, without affecting the peaks or crest factor of the signal, e.g. peak sound pressure level, A-weighted. It can also be applied to any waveform in the memory list, such as the stimulus, response, or any intermediate waveform.

Both standard and arbitrary filters are available. Standard filters include lowpass, highpass, bandpass and bandstop filters. You can select the cutoff frequencies, and control the slope of the filter using the filter order. SoundCheck’s standard filters are implemented as IIR Butterworth filters, and are ideal for most applications where you need to attenuate certain frequency ranges. For example, you can use a high-pass filter to remove some low frequency background noise or remove dc offset. Alternatively, you might use a lowpass filter to attenuate alias frequencies that could cause your amplifier to clip at very high frequencies that are not of interest.

You can also create your own arbitrary waveform filter by applying any curve from the memory list to the waveform. This can be used to apply weightings such as K-weighting for loudness or a bandpass filter  to a speech stimulus. Or you can even specify your own custom weighting or equalization, for example to see what happens to a customer’s speaker when they boost the bass.

 

100 Things #94: Road Noise and Active Road Noise Cancellation Measurements

Road Noise and Active Road Noise Cancellation Measurements are easy with SoundCheck. Everyone’s familiar with measuring headphone active noise cancellation with SoundCheck, but did you know it’s also great for in-car measurements of road noise and evaluation of road noise cancellation systems? Simply connect the USB-powered AudioConnect 2 to your microphones and laptop, and start making measurements. Watch this short video to see how easy it is with this compact and cost-effective package.

Make Road Noise and Active Road Noise Cancellation Measurements

Learn more

Read more about in-car road noise measurements.

Learn more about automotive testing using SoundCheck.

 

Video Script: Road Noise and Active Road Noise Cancellation Measurements

Everyone’s familiar with measuring headphone active noise cancellation with SoundCheck, but did you know it’s also great for measuring active road noise cancellation and road noise reduction in cars?

In this application, you want to measure the road noise at the location of the driver or passenger ears. A simple and cost-effective way to do this is to position two microphones near the outside of your ears or use a Head and Torso simulator. Here, we attached two SCM microphones to a Listen hat using some clips – we call this a low-budget HATS. This has the added advantage that if you are out on real roads, you can measure in the driver’s position.

Our configuration is simple. The microphones are connected to an AudioConnect 2 audio interface for microphone power and signal conditioning. This is a great application for this interface, as it’s USB powered, so you don’t need a power outlet in your car – you can just run it off your laptop. The AudioConnect 2 is connected via a single USB cable to the computer that is running SoundCheck for analysis.

As you can see, it’s a compact setup that easily fits on your dashboard or passenger seat.

To measure road noise, we simply need to drive at a fixed speed, remain silent – that means no talking, coughing, or children in the back seat –  and record for a fixed period of time. You can then look at both the sound pressure level and the frequency spectra of the road noise.

To evaluate a road noise reduction system, simply repeat the measurement with the noise cancellation turned on and subtract one result from the other to provide a value for the noise-reduction system of the car.

This is just one of many automotive audio measurements you can make with SoundCheck. Others include end of line QC, evaluation of components, tuning, Max SPL, impulsive distortion, Buzz, Squeak and rattle, POLQA analysis of communications systems and more. Check out the automotive section of our website for more information.

 

 

100 Things #93: Group and Batch Processing of Data Curves

Group and Batch Processing is a really neat feature in SoundCheck that saves huge amounts of time when processing data. Curves, values and waveforms can be grouped and processed together, and the analysis, post processing or statistics runs almost as quickly as on a single piece of data. This can be done during a sequence, or offline with previously collected data. It even extends to imported data – for example, if you want to run a POLQA analysis on a batch of recordings made in a different system, you can simply import the wav files and calculate scores for hundreds or even thousands of waveforms all at once.

Save Time Processing Data with Group and Batch Processing

Learn more

Read our Knowledgebase Article on using batch processing.

Learn more about the POLQA module in SoundCheck (video contains a demo of batch processing).

 

Video Script:

Audio test and measurement involves collecting and analyzing a lot of data. You might have multiple inputs and outputs, or you need to collect data not just once but over and over again. Perhaps you’re averaging measurements on a single unit over multiple runs, or testing multiple units in a production facility. Handling and processing all this data efficiently, in realtime, can be complex.

SoundCheck processes large groups of data quickly and easily with its group and batch processing capabilities. Curves, values and waveforms are grouped and processed together, and the analysis, post processing or statistics runs almost as quickly as on a single piece of data.

This is useful, for example, if you’re repeating a series of sequence steps on a single device, to calculate the deviation in its response at various positions, or if you’re averaging sensitivity values of a batch of 15 microphones for a spec sheet.

Groups of data can be analyzed and processed either within a test sequence or offline.

In a sequence, groups of data can be automatically created, saved in the Memory List and automatically analyzed together the same way every time the sequence is run. Here’s a simple example sequence where I capture recordings using a 6 mic array, group the recorded waveforms and use a single analysis step to get responses from each of the microphones. The same process can also be used in post processing or limit steps. SoundCheck also makes it easy to keep track of your data by allowing you to append your data names with Signal Path and Input data names.

Data processing outside of a sequence is known as “offline mode” – let’s take a look at an example. Here, I’ll group the frequency responses of 5 microphones I measured previously and calculate their sensitivity values at 1kHz in a single post processing step, rather than using 5 such steps. Note how fast it is in both cases!

SoundCheck’s batch processing capabilities even extend to imported data. For example, if you want to run a POLQA analysis on a batch of recordings made in a different system, you can simply import the wav files and calculate scores for hundreds or even thousands of waveforms all at once.

SoundCheck’s batch processing capabilities handle large amounts of data extremely fast, helping both R&D labs and production facilities to reduce test times. To learn more about SoundCheck’s extensive audio measurement toolkit, check out www.listeninc.com.

 

100 Things #92: Continuous Log Sweep with Time Selective Response Analysis

Did you know that SoundCheck was the first audio test system to implement the continuous log sweep stimulus, way back in 2001. Also known as a frequency log sweep, or Farina sweep, this stimulus is used with time selective response (TSR) analysis. TSR analysis allows reflections to be windowed out, making it great for loudspeaker simulated free field measurements and room acoustics measurements. It’s also valuable as a smart trigger for robust open loop measurement testing. Watch this video for a quick overview.

Continuous Log Sweep with Time Selective Response Analysis

Learn more

Read on about stimulus and analysis capabilities in SoundCheck.

 

Learn more about Simulated Free Field Measurements

Short Video Demonstration of free field measurements without an anechoic chamber

Full-length Demonstration of free field measurements without an anechoic chamber

Article explaining simulated free field measurements (reprinted from Voice Coil Magazine)

The Original 1992 paper introducing the Simulated Free Field Measurement Technique

 

Learn more about room acoustics measurements using the Log Sweep Stimulus

Full-length Demonstration of Room Acoustics measurements

 

Video Script:

Did you know that SoundCheck was the first audio test system to implement a continuous log sweep stimulus? We introduced it back in 2001,  shortly after Angelo Farina’s landmark AES paper on the subject. Let’s take a look at how it works and how it’s used.

A continuous log sweep, sometimes known as a frequency Log sweep or Farina sweep,  is a continuous sine sweep with equal time and energy in every octave. Since it sweeps slower at low frequencies but speeds up as the frequency increases,  it’s a great choice for fast measurements. It differs from a conventional stepped sine stimulus, in that the continuous log sweep plays across all frequencies in the range with a defined sweep rate per decade, whereas the stepped sine sweep “steps” through different frequencies across the range.

Both stimuli can measure frequency response and harmonic distortion, but the analysis methods differ. A continuous log sweep uses a time selective response, or TSR analysis. This involves calculating an impulse response and applying a user-defined time window that can isolate or  remove any reflections caused by the test environment. A stepped sine requires a HarmonicTrak analysis. Only the continuous log sweep with TSR analysis can window out reflections, allowing a simulated free field measurement even when you are not in a fully anechoic environment.

Let’s take a look. In the TSR analysis step, we’ll enable this checkbox here to output an impulse response to the memory list so we can view it. It can be displayed either on a linear or logarithmic scale.  The window size at the top is where we define the start and stop points of the window that’s applied to the impulse response. We can look at this in SoundCheck to help us decide which points to use. Here, we can clearly see a large impulse that has been autodelayed to 0 seconds to show the direct sound from our sound source. And because we’re in a non anechoic environment, just a normal room, you can see reflections from the walls, floor, ceiling, table etcetera.  in the impulse response. We can adjust the window to remove them, and you can see the frequency response updates. 

This technique is very powerful, but like all techniques there are tradeoffs. So Log TSR analysis might not be the best option for all applications. The measurement resolution is affected by the window size – as the window size narrows,  the frequency resolution reduces, and you can see the effects on the frequency response. This is particularly noticeable at the lower frequencies where  the lack of resolution can make the data inaccurate if the window is too small. We need to be careful to configure the window size to capture the direct sound but be wide enough to get the greatest frequency resolution, without any reflections due to the test environment.

TSR Analysis  offers significant benefits for several applications. We use it for the high frequency measurements in a loudspeaker simulated free field measurement, which we can then splice together with the low frequency Stepped Sine Sweep stimulus measurement. It’s also valuable for room acoustics, for example, for calculating RT60 and clarity measurements. And if you’re running open loop tests, our cross-correlation smart trigger uses a continuous log sweep to provide a way of triggering an open loop measurement that is extremely robust and far less susceptible to false triggers than other methods. 

To learn more about the applications of a continuous log sweep stimulus, check out the technical papers and demo videos on our website.

100 Things #91: Measurement of Intermodulation Distortion

Intermodulation Distortion measurements are a great alternative to harmonic distortion for measuring narrowband devices such as hearing aids and communication devices. In such devices, harmonic distortion measurements tend to underestimate the distortion as the higher-order harmonics fall outside the pass band of the device. In this short video, Steve Temme demonstrates and explains the two IM distortion measurement options in SoundCheck – intermodulation distortion and frequency distortion and discusses how they can be used for low frequency speaker measurements, narrowband devices and microphones.

Measurement of Intermodulation Distortion

Learn more

Read on about more analysis capabilities in SoundCheck.

Video Script:

Although harmonic distortion is perhaps the most commonly measured distortion metric, it’s often not ideal for measuring narrowband devices such as hearing aids and communication devices. These products often have a high frequency cut-off around 3-5 KHz, so the higher-order harmonics fall outside the pass band of the device, so harmonic distortion measurements often underestimate the distortion.

A useful alternative we offer in SoundCheck is intermodulation distortion. Intermodulation distortion relies on the interactions between two simultaneous pure tones to produce measurable intermodulation products. These measurements actually present a more realistic representation of real-world signals such as speech and music that are rich with intermodulation products than the single tone used in harmonic distortion

SoundCheck offers two intermodulation distortion measurement options – Intermodulation Distortion and Difference Frequency Distortion. For Intermodulation Distortion, we superimpose a sweeping frequency tone against a fixed frequency tone. For Difference Frequency measurements, we use a stimulus consisting of two sweeping tones separated by a specified frequency interval, which can be a fixed difference or a fixed ratio. These are fully customizable.

In both cases, the two signals interact to produce intermodulation products. With Intermodulation Distortion, these are equal to the sum and difference of the upper frequency and integer multiples of the lower frequency. Difference Frequency distortion, only considers the components that are the difference and multiples of the difference, between the excitation frequencies.

Each type has its own specific applications. For example, Intermodulation distortion is mostly used for loudspeaker measurements, particularly at low frequencies, and Difference Frequency distortion is ideal for testing narrowband devices as the frequencies can be chosen so that the intermodulation products mostly fall within the pass band. This is easy to do in SoundCheck – simply configure your two test stimuli, and select your analysis – either Intermodulation Distortion, or Difference Frequency Distortion – in the analysis editor.

Intermodulation distortion is also a valuable technique for measuring microphones. Usually, the harmonic distortion from the source speaker playing the test tone is greater than the harmonic distortion that you are trying to measure from the microphone. However, if separate test tones are fed individually to two separate loudspeakers, the loudspeaker’s harmonic distortion has no influence on the measured intermodulation frequency components, enabling accurate measurement of the microphone’s intermodulation distortion.

To learn more about intermodulation and other types of distortion, check out our website, and stay tuned for a new in-depth seminar on distortion.