Tag Archive for: soundcheck

Sequence Versioning in SoundCheck

Sequence Versioning is available in SoundCheck version 22 and later.
Sequence versioning makes it easy to track changes made to a sequence to ensure that colleagues and contract manufacturers are using the correct version. Comment fields make it easy to document changes, and the sequence history view provides a log of historical changes. Archiving functionality allows the sequence to be backed up while it is being developed, so changes can easily be rolled back while retaining the versioning information.

Watch the video demo of Sequence Versioning

 

Main SoundCheck Page

Soundbar Measurement over HDMI Connection

Soundbar Measurement - screenshot of final displaySoundbar measurement over HDMI is simple in  SoundCheck as the WASAPI driver option allows easy connection to the device under test via HDMI.  This test sequence is not exclusive to soundbars; it can be used to make audio measurements on any audio device that accepts audio over HDMI (e.g. TV, computer monitor, home theater speakers, etc.)

The sequence itself is quite simple, containing 4 steps: Stimulus, Acquisition, Analysis and Display. Frequency response, THD, Rub & Buzz and ePRB are shown on the final display. More important are the instructions for configuring your SoundCheck system’s Hardware and Calibration to support the HDMI connected device.

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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.

You can also learn more about the techniques demonstrated in this video in our June 2024 AudioXpress article on Measuring MEMS Microspeakers.

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.

 

SoundCheck 21 New Features

Check out our short demo video for a brief introduction to all the new features in SoundCheck 21. From exciting new algorithms for transient distortion detection, to security features to protect your sequences, it features something for everyone!

SoundCheck 21 New Features

Ready to try SoundCheck 21 for yourself?

If you are ready to upgrade your system to SoundCheck 21, or want to discuss how SoundCheck 21 can help your testing needs, contact your local sales engineer or sales@listeninc.com with any questions.

Already own SoundCheck 21, but aren’t on the latest version? SoundCheck 21.01, a patch for version 21 released in March 2023, and is a free update for all registered version 21 users. You can download it here. This patch fixes several minor bugs and is recommended for all users.

Learn more

The new enhanced Loose Particles algorithm correlates objective results to subjective analysis. To hear this algorithm for yourself, and to learn even more about how enhanced Loose Particles works, check out the full enhanced Loose Particles page.

To see some of the other great features, algorithms, and functionality from previous SoundCheck versions, check out our version timeline to learn more about Listen’s 20 years of innovation.

 

New Transient Distortion Measurement Algorithm

At Listen, we’re at the forefront of audio measurement research, and we’re always looking to improve on existing audio measurement techniques, even our own! Loose Particle detection has been a valuable production metric for analyzing transient distortion since we launched it in 2004. This new iteration uses our own original research to improve accuracy and reliability, show a clear correlation to audibility, and simplify limit setting. In addition to production line testing of speakers, headphones, drivers and other devices, it is also valuable for automotive Buzz, Squeak and rattle (BSR) measurements and measuring rattling components such as keys and buttons on a variety of devices. All is explained in the short video below.

Transient Distortion Detection Launch Video

Watch our launch video (broadcast date 05/25/2023) for the full details.

 

Ready to Measure your Transient Distortion?

If you have SoundCheck 21, you already have this new algorithm! Download our free complete end of line test sequence with enhanced Loose Particles to start using it right away.

If you don’t have SoundCheck 21, but have an older version, you can send us your recorded waveforms and we’ll analyze them for you and send you the results. Please contact sales@listeninc.com for a test sequence to record the waveforms.

No SoundCheck system at all? No problem! You can send us your speakers and we’ll test them for you. Or we may be able to arrange a system loan. Contact your sales engineer at sales@listeninc.com for more information.

Prefer to Read About It?

We know not everyone has 15 mins free to watch a video, although its hard to beat the benefits of a proper demonstration. So here’s a brief summary of the information about this new algorithm that is presented in the video:

What it does

The new algorithm measures transient distortion caused by loose particles that may become trapped in a device during manufacture and create an unpleasant sound when they vibrate in the finished product. This is measured in the time domain rather than the frequency domain as these artifacts appear randomly over time, and not periodically in the same way that harmonic distortion artifacts are presented. This algorithm has a couple of unique features:

1) It measures transient distortion separately from harmonic distortion which gives deeper insight into the failure mode and accelerates troubleshooting, particularly on the production line.

2) It’s easy to correlate with audibility as the algorithm removes the stimulus waveform to allow the user to listen to just the distortion artifacts. As well as enabling the user to truly understand how measured results correlate to listening, this also facilitates limit setting.

3) It’s reliable even in the presence of transient background noise since it relies on a cumulative event count rather than a single event triggering a fail. Limit setting is simple as it is not frequency dependent

 

Applications

  • Production line driver test
  • Production line finished product test (rattling buttons, keys, grills, and other components)
  • Buzz, Squeak and Rattle (BSR) / Impulsive distortion measurements in cars

 

How it works

Like Listen’s original 2004 transient distortion detection algorithm, enhanced Loose Particles relies on a time domain analysis of the waveform. However, rather than simply filtering and counting transients, it removed the stimulus, performs a time domain analysis of the remaining transients, then applies a unique Prominence calculation to evaluate each transient in the context of the surrounding waveform. A threshold is applied, and the number of transient events above that threshold are counted. The event count indicates whether a device has a transient distortion problem. Transient events caused by speaker manufacturing issues tend to repeat many times as particles bounce around in side the speaker, or components vibrate. In contrast, background noise events occur infrequently during the duration of the measurement. The threshold is set based on audibility, and the Loose Particle count limit can be set according to the environment to fine-tune the algorithm for the specific operating conditions.

 

Graphic demonstrating how the Transient Distortion algorithm (Loose Particles) works

4 Stages of the enhanced Loose Particles algorithm for transient distortion measurement: Response waveform, Loose Particle waveform, Prominence and enhanced Loose Particles.

 

 

For additional background, comparison to other methods, such as Crest Factor Analysis, and live demonstrations, please check out the video above.

 

Learn More

100 Things #61: Monitor Test Progress by Displaying Sequence Steps

SoundCheck includes powerful built-in tools for monitoring and troubleshooting. As a sequence runs, steps highlight with either green or red indicators, either showing a successful run or indicating an error. Display steps can be placed anywhere in a sequence, not just at the end, so you can receive visual data at specific sequence points. For even greater visual feedback, all steps can be configured to display as a sequence runs.

Monitor Test Progress by Displaying Sequence Steps

Learn more

Learn more about SoundCheck’s sequence writing capabilities.

 

Video Script:

Everyone knows that you can create powerful test sequences in SoundCheck, but did you know that SoundCheck also has some powerful built-in tools for monitoring the progress of your test sequence and troubleshooting in real time?

In SoundCheck, we create test sequences by dragging and dropping steps from the step template library into the working sequence, then double-clicking onto the step to edit. Using this simple method you can build up complex measurement sequences with hundreds, or even thousands of steps. But what do you do if your sequence doesn’t run as planned? I’m going to show you a couple of useful tools to help you quickly resolve any issues..

First, SoundCheck highlights the active step  in the sequence editor. Green indicates the step has successfully run or passed, red indicates some type of issue- for example, a step in the sequence has failed. This is a really simple check that quickly helps you identify where things are going wrong.

SoundCheck also supports multiple display steps. This is particularly useful in long sequences so you don’t have to wait for the sequence to run completely before viewing data. Here I’ve added a display step directly after the acquisition step to display the recorded waveform superimposed over the test stimulus. For troubleshooting, this assures that the sequence has properly recorded the response, and it can move onto analysis.

You also have the option to display the step as the sequence is running. This is enabled in ‘configure step’, and you can define how long to display the step. I’ll enable this on my acquisition step. Similar to using a separate display step, this option has the desired effect of viewing data in the sequence prior to analyzing it.

I use these 3 simple tricks a lot when I’m writing and de-bugging sequences. How do you de-bug yours? Do you have any tips and tricks to share? Let us know in the comments below.