Standardizing the Immersive Audio Playback Chain

Immersive Audio Listening Room at Genelec: Customized room for listening to spatial audio

Immersive Audio Listening Room at Genelec

If you follow my posts, you’ll know that spatial audio measurement is a big research area for us right now. As immersive audio takes off, it’s important that designers of spatial audio systems have the tools they need to accurately measure spatial sound characteristics such as localization and envelopment. This will enable consistency in playback so that artists and mixers know that the effects that they are creating sound as they intended, no matter what the consumer’s playback equipment.

To complicate the issue, having designed the playback system, manufacturers also need to take into account the myriad ways in which their customer may configure their speakers to ensure consistent reproduction. This is no simple task when room acoustics also comes into it and few consumers have a perfect listening space.

While we are some way from having universal measurement techniques and standards, I witnessed an interesting insight into how Genelec is addressing the issue of customer configuration at the AES/ASA/BAS Boston chapter meeting at their facility in Natick earlier this week.

First we received a demonstration in their newly-configured immersive room where they demonstrated several soundtracks. This included correlated and uncorrelated pink noise at low frequencies to illustrate bass reproduction. They also played a couple of musical pieces including the musical score from a video game (which was surprisingly impressive) and a piece of Indian music. These were both very immersive, with sound coming from multiple directions. What I found interesting about this was that in the 9-seater listening room, there was really only one sweet spot or seat where you received the full impact of the immersive experience. The effect in the other seats was not uniform and depended on the seat’s position relative to the center spot. This will certainly present challenges to sound designers!

In a separate room, we saw how Genelec helps their customers ensure that their equipment is correctly configured. This was impressive too! Their bespoke software uses a microphone and network adaptor to make measurements in the room and upload them to a server. The speaker-room interactions are diagnosed, and their software returns a comprehensive report including frequency response, time response and time-frequency analysis including wavelet analysis. It also provides an electronic file that equalizes and time-aligns the speakers to compensate for the room characteristics and exact speaker placement. This ensures that Genelec’s customers are hearing the sound as they intended, regardless of their room configuration.

This is a great start, and it’s encouraging to see spatial audio playback system vendors following through to ensure the end user’s experience is as intended. However, this is only one part of the puzzle. There is still a need for the industry to agree on useful metrics that allow manufacturers to design and evaluate their systems to Dolby Atmos and other spatial audio specifications. This will allow the creative effects of spatial audio to transcend individual manufacturers and allow consumers hear exactly what the recording engineer intended, regardless of their chosen brand of playback equipment. If you make spatial audio playback equipment, what metrics do you think show the most potential for evaluating spatial sound effects? We’d love to hear from you.

AudioConnect 2 Audio Measurement Interface Demonstration

What makes an audio measurement interface different from any other audio interface? This is a good question. As you probably know, SoundCheck works with most audio interfaces – this is advantageous as our customers can use hardware they already own, keeping the overall system cost down.

However, if you don’t have an audio interface and need to get one at the same time as your test system, there are many advantages to purchasing dedicated test hardware such as the AudioConnect 2, or for multi-channel applications, the AmpConnect 621. They include many features specifically designed for audio test, such as microphone power and internal switching. This saves money on other components. They are designed for production and field measurements so they are ruggedly constructed, and unnecessary features such as front panel controls are eliminated to avoid accidental adjustment. Perhaps more importantly, Listen’s audio interfaces are configured for full plug’n’play automated setup that hugely simplifies the setup process and virtually eliminates the possibility of incorrect configuration.

Watch this short 5 minute video to learn more.

 AudioConnect 2 Audio Measurement Interface Demonstration Video

Learn more about the AudioConnect 2 Interface

Check out our product information page, and another short video that we made explaining the philosophy around offering our latest generation audio interfaces.

More Information?

Curious? Contact our sales team for more information and pricing.

Audio Measurement Troubleshooting: 10 Time-Saving Tips

Audio Measurement TroubleshootingAudio measurement troubleshooting can be challenging. All too often we plug and unplug things, tweak settings and re-test, often changing several variables at once. A consistent and logical approach greatly accelerates the process.  Here, we share the process that our experienced team of support engineers has developed to assist customers. Follow these 10 audio measurement troubleshooting tips to quickly get to the root of your problems.

  1. Check your cables. Cables are the single biggest cause of setup problems! Make sure all your cables are properly connected and in the right place. Next substitute alternate cables one at a time – cables often get broken inside or damaged, even though they pass visual inspection. Make sure you understand the different types of cables and when you should use each – particularly where they look similar, e.g. single-ended and balanced cables. It’s a good idea to label your cables to make it quick and easy for anyone to replicate your measurement setup and quickly fix things if anything becomes unplugged.
  2. Check everything is plugged in! It sounds obvious, but if there is no output signal or not what you expect (e.g. it just looks like noise), check that everything is plugged in and turned on – not just the computer, but all audio interfaces, amplifiers, hardware, microphone power supplies etc.
  3. Run a self-test. Always run a self-test before making measurements to confirm that your measurement system and cables are working correctly. This facility is usually built into your measurement system and confirms that all components are working as expected.
  4. Make one change at a time and document everything. Don’t move cables, tweak the sequence and change the microphone position all at once – you won’t know what the cause of the problem was! Change one thing at a time, and keep a checklist of what you have done (this will be useful if you need to call customer support).
  5. Check your audio interface configuration. If your audio interface is incorrectly configured, nothing else will work. Ensure sound monitoring is switched off to prevent feedback, and make sure you are using the full dynamic range. Try making a loopback (output to input) measurement on the audio interface using the appropriate SoundCheck self-test. It should have a flat frequency response.  If your frequency response looks noisy it is likely that you have insufficient gain. You should have unity (0dB) gain when looped back on itself and very low THD. Latency should be consistent and repeatable and must be ≥ 0 Record Delay. There are many places where audio interface settings can be incorrectly configured  – check your hardware setup, ASIO control panel and mixer (if applicable), device front panel (if applicable) and Windows audio device settings.
  6. Calibrate your signal paths. Calibrating your signal paths is critical to accurate measurements and you should always have a multimeter and an acoustic calibrator handy! Calibration should be carried out at regular intervals as well as when you first run the test, particularly if there is a change in atmospheric conditions. Good measurement microphones and electronics are typically very stable but loudspeakers e.g. mouth simulators are generally very non-linear and their performance changes with temperature. It’s  always a good idea to warm up a loudspeaker before testing it by playing pink noise for example at a reasonable level. Factories should re-calibrate every day or at the beginning of every shift.
  7. Measure twice. Always measure something at least twice on the first measurement to make sure that you get the same result. This helps identify problems with the setup or background noise. It’s far better to discover this on the first measurement than at the end of a day of data collection! A golden unit, e.g. a demo speaker that you have measured many times, is a good sanity-check if measurements appear unexpected. A golden unit is advantageous over calibration as it also takes into account the fixturing and test environment.
  8. Check your units and your resolution. Make sure you are measuring in your intended units. For example the difference between dB Pa and db SPL is 94dB. If you are not careful you could blow something up. It’s also important to check your resolution. Lower resolutions provide faster measurements, but it’s important to ensure that you are using enough. If you keep increasing the resolution until the curve doesn’t change, you can easily identify the lowest acceptable resolution to test at.
  9. When in doubt, look at your recorded time waveform. Use your software’s oscilloscope function to see if your recorded waveform looks noisy, has drop outs, or gets cut-off too soon.  Look at the peaks of the recorded waveform and Max FSD in the memory list to see if it looks unexpectedly flat – it may be overloaded and clipping. These problems are very difficult to see in the frequency domain.
  10. Minimize background noise. Background noise is the biggest cause of unrepeatable measurements, so minimizing this improves your test environment. Easy ways to minimize background noise impact include positioning the microphone closer to the source, increasing the test level, increasing the duration of the test signal, and using repeated averages of the test signal. Doubling the averages or signal duration should increase the signal to noise ratio by 3 dB.

We hope you found these audio measurement troubleshooting tips useful. Follow our blog for more handy hints for audio measurement.

Directional Audio Measurements with the MDT-4000 Turntable

Have you ever wondered about the thought process that goes into designing a new audio test product? Our sales and support teams worked closely with Portland Tool & Die during the design of the MDT-4000 turntable for directional audio measurements to ensure it addressed all the pain-points that our customers had with other brands – speed, accuracy, portability, control and more. In this short video, designer Kris Hett demonstrates these features and you can see how seamless integration is with SoundCheck.

 Directional Audio Measurements with the MDT-4000 Turntable

Learn more about the MDT-4000 Turntable

Check out our product information page, and our comparison of the MDT-4000 specifications with other popular turntables

Free polar plot test sequence for use with the MDT-4000 Turntable. This gets you up and running quickly with polar plots, and can be used as a base for creating your own measurement sequences.

More Information?

Curious? Contact our sales team for more information and pricing.

Does audio test feel like Groundhog day?

It doesn’t with SoundCheck. Advanced test automation enables repetitive and time-consuming  tests on speakers, headphones, microphones, smart devices, communications devices and more to be run automatically.

Let’s take a look at an example. The AES75 standard for measuring Max SPL takes considerable time and operator involvement to run – there are multiple iterations of running a test at different levels, examining results, adjusting levels and repeating. Then again for the next speaker. Groundhog Day, right?

In SoundCheck, the entire sequence is automated! It makes measurements, objectively compares them, increases the levels, and repeats the measurements to the failure point. It then automatically reduces the level and repeats measurements repeated until the Max SPL value is determined. This leaves you free to kick back, listen to some tunes or work on something else!

 

Check out this short video for a demonstration of the sequence in action:

 

This is made possible by the advanced sequence writing options in SoundCheck, including looping. This short video demonstrates how to automate measurements with sequence looping to free up your time for more exciting tasks.

Want to know more? Request a demo and one of our engineers will be in touch.

The Missing Measurements – Challenges of Measuring Immersive Audio

The University Atrium and Presentation Space that was Simulated in the Immersive Listening Room (Photo courtesy of Acentech)

Earlier this week I attended the AES Boston section meeting at Acentech’s state-of-the art facility, lured by the promise of a demonstration of their 3D acoustic listening simulation. I was not disappointed! After some drinks and conversation with other attendees, we broke into smaller groups to experience their immersive listening room. Although only the size of a large conference room, this room simulates much larger spaces such as lecture halls, restaurants and offices.

We listened to a simulation of a University atrium and presentation space. There was a dining space on a balcony above it, and offices and classrooms above and behind that. We heard how it would sound both with and without diners during a presentation, and if you closed your eyes it was easy to believe that you were actually there. This impressive auralization was created by first recording a presenter in an anechoic environment, then applying sophisticated predictive models to apply reflections, reverberations and background noise to account for the room configuration and construction materials. It was then reproduced in the immersive room on an Ambisonic playback system. This model permits not only evaluation of  architectural features, but also experimentation with different materials for walls, windows, furniture etc. Architects, designers and sound consultants can evaluate and adjust the acoustics of a room at the design stage, ensuring acoustic perfection in the finished product.

What I found surprising was the lack of feedback to the model based on real measurements. It seems that there’s a prime opportunity to make actual measurements in the room once it is built, and compare this real-life data to the model. This was also apparent when I spoke to another attendee, a recording engineer, about mixing in Dolby Atmos. We discussed how he could master something until it was perfect on his Dolby Atmos system, but there was no way of knowing whether other listener’s Dolby Atmos systems would reproduce the sound in the same way he intended.

Spatial audio measurements are still in their infancy, and they are a complex combination of perceptual measurements such as localization, intertwined with room acoustics. Until we devise reliable and accurate ways of measuring spatial audio, we have no way of measuring room simulations to compare them to actual measurements, and no way of comparing the fidelity of various Dolby Atmos playback systems. This is a source of frustration for many product design engineers who are accustomed to having good measurement data to drive their product development.

Recently, I’ve been investigating how a binaural head perceives virtual source location and envelopment, and we’ve developed some rudimentary ways to measure sound localization and envelopment using interaural level difference and interaural cross-correlation measurements with a head and torso simulator. What are your thoughts on spatial audio measurement and what measurement techniques and metrics you are working on? Let me know in the comments and please reach out for me if you’d prefer an offline discussion.

Learn more about Acentech.

Learn more about stereo soundfield measurements for quantifying sound localization and envelopment.

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.

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.