Technical Resources
With over 100 combined years of audio measurement experience, our team has created a wealth of technical papers, sequences, articles and other useful information to assist you with your audio test needs. Please browse the collection below, or filter by type of resource.
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.
A New Method for Transient Distortion Detection
Transient distortion, or ‘loose particle’ measurement, is an important loudspeaker production line quality control metric that identifies and facilitates troubleshooting of manufacturing issues.
This paper introduces a new enhanced loose particle measurement technique that discriminates more accurately and reliably than current methods. This new method introduces ‘prominence’ after envelope detection, a new metric for audio measurements, that effectively isolates transient distortion in the presence of periodic distortion. This technique also offers the unique ability to listen to the isolated transient distortion waveform which makes it easier to set limits based on audibility and has widespread applications.
Authors: Steve Temme, Rahul Shakya and Jayant Datta, Listen, Inc.
Presented at 155th AES Conference (October 2023) New York, NY
Paper Introduction
Transient distortion, or ‘loose particle’ measurement, is a valuable quality control metric because it identifies non-periodic distortion, for example, rattling parts, separately from periodic distortion such as rubbing or buzzing parts. This facilitates troubleshooting of manufacturing issues. This paper introduces a new transient distortion measurement technique that is more accurate and reliable than current methods. In addition to improved performance, this new algorithm also aids understanding of the correlation between measurement results and audibility, since it is possible to isolate and listen to just the transient distortion artifacts. Although this analysis method was developed for measuring loose particles in loudspeaker drivers, it is also valuable for measuring rattling parts such as buttons, fasteners, and loose wires on various audio devices, and measuring impulsive distortion or Buzz, Squeak and Rattle (BSR) in automotive audio applications [1].
What is Transient Distortion? Why does it matter?
Transient distortion is caused by random clicking, popping, and other noises in the time domain. In a speaker or headphone driver, this might be caused by foreign particles such as glue or magnet fragments trapped in the gap behind the diaphragm or dust cap. In a device such as a smart speaker, transient distortion might come from a loose volume control button on the device that rattles when sound is played. In an automotive application, it could be characterized as buzz, squeak and rattle from loose wires, screws or fasteners in a car door that the loudspeaker is mounted in. In all cases, the sound is undesirable, so devices that exhibit such faults should be identified and rejected.
In the recorded time waveform, transient distortion faults appear as impulsive noises added on the stimulus wave. These impulses are not related to the frequency of the stimulus, but rather to the vibration caused by the displacement amplitude of the diaphragm. The transient distortion is more frequent and significant when the speaker is driven near or below its resonant frequency, where the displacement of the diaphragm is the greatest.
Although the sound – a random clicking, buzzing or popping noise – can sometimes sound similar to higher order harmonic distortion (Rub & Buzz), such defects are not clearly reflected in the frequency spectrum of the waveform. Figure 1 shows a waveform with transient distortion, and the corresponding frequency spectrum. The vertical black line represents the stimulus frequency and the orange broadband noise spectrum indicates the transient distortion. Transient distortion is best identified at the time the transients occur, unlike Rub & Buzz distortion which is best identified by the frequency at which it occurs [3].
The entire paper also covers:
Prior Measurement Methods
The new algorithm – comparison and results
Conclusions
In addition to this paper, please also check out the Enhanced Loose Particles Webpage and our detailed video explanation of how this algorithm works.
100 Things #90: Curve Smoothing
Curve smoothing in SoundCheck allows for non-destructive processing of data, resulting in smooth and easy to visually understand curves. Curve smoothing can lessen the effects of reflections in the test space, reduce noise, or make curves less jagged for publishing data. The smoothing post processing step in SoundCheck features an array of different, to facilitate different levels of the smoothing process, including various smoothing widths and windowing options.
Curve Smoothing
Learn more about SoundCheck post processing options
SoundCheck has a full suite of post processing capabilities including curve smoothing, resampling, resolution, curve arithmetic, and more. Read more details in our SoundCheck features and applications section.
Each sequence uses a stimulus configured to the device under test, and recommended hardware.
Video Script:
Curve smoothing, as its name suggests, is a useful post-processing option that turns your jagged lines into smooth curves. It may be applied to a curve for a number of reasons – to reduce the appearance of noise in the signal, to minimize reflections and other artifacts from the measurement environment, or simply to make a curve look better for presentation in sales and marketing literature. When smoothing is applied, the points of the curve are modified so that individual points that are higher than the immediately adjacent points are reduced, and points that are lower than the adjacent points are increased.
SoundCheck uses sliding-average smoothing also known as “boxcar” averaging where each point in the curve is replaced by the average of n adjacent points where n is a positive integer known as the smoothing width. SoundCheck supports standard 1/n octave smoothing widths from one octave to 1/24th octave as well as user defined log and linear values. In addition to a default rectangular window, a Hanning window may also be applied during the smoothing function. Smoothing is symmetrical at the midpoints of the curve but tapers to zero at the curve’s end-points. If the curve has uneven or non-standard spacing in the frequency domain, interpolation is used.
In addition to the standard Smoothing post-processing step, the smoothing function is also available in the Resolution post-processing step. This is useful when the final curve resolution is higher than or “not a mathematical factor” of the original resolution.
This feature’s been available in Soundcheck since it was launched in 1995. If you haven’t tried it yet, check it out!
100 Things #89: Apply Equalization To A Test Stimulus
Did you know you can equalize a stimulus in SoundCheck to remove the influence of hardware and components from your measurements? All of SoundCheck’s stimulus options can have EQ applied, include Stweep, waveforms, noise, and more. An EQ can also adjust a stimulus to focus on different frequencies, like boosting low or high frequencies for power testing. THD+N measurements benefit from this ability, as even applying a flat EQ curve to a Stweep smooths out frequency transitions.
Apply Equalization To A Test Stimulus
Learn more about SoundCheck stimulus flexibility
The stimulus is just one part of the completely flexible SoundCheck system. Learn more about SoundCheck’s features and applications.
If you want to try for yourself, our SoundCheck sequence library includes applications from measuring loudspeakers to microphones, VR headsets to cars, and more. Each sequence uses a stimulus configured to the device under test, and recommended hardware.
Video Script:
Did you know you can equalize any stimulus inside SoundCheck during its playback? In any test application, it is important to ensure that the inherent characteristics of the measurement hardware do not influence the measurement. For example, if you’re using a source speaker to measure a DUT microphone, you don’t want the loudspeaker’s frequency response to influence the measurement. You may also want to apply your own custom EQ curve to weight certain frequencies different, for example, boost low frequencies more than higher frequencies for power testing. You can import whatever EQ you prefer.
We can also equalize the source speaker using a reference microphone. First, we measure the speaker’s response, then invert it to give us the EQ curve. This curve can then be applied to any stimulus playing through the source speaker to correct for both magnitude and phase non-linearities.
When you check the ‘Apply EQ’ checkbox in SoundCheck’s stimulus step, the EQ curve is applied to the stimulus and saved to the memory list, ready for playback during the acquisition step.
This feature is available for all stimulus step types. For step-based stimuli such as Stweep and Multitone, where the stimulus doesn’t have all frequency components, EQ is applied only at those frequency points that are present. For Broadband stimuli like speech, music and Noise, ‘Apply EQ’ behaves like a time waveform filter.
There’s also another reason why you might want to use EQ in a step based stimulus such as a Stweep. When EQ is applied, even if there is no EQ curve, the transition from frequency step to frequency step is smoothed. This is particularly helpful for measurements such as THD+N, which are sensitive to ringing.
Naturally, ‘Apply EQ’ can be turned off if you want to characterize the speaker itself.
SoundCheck’s stimulus step provides many advanced options for a wide range of use cases. To learn more, check out our website or speak to your local sales engineer.
100 Things #88: SoundCheck Support Audio Over IP
SoundCheck supports testing audio over IP using Dante. Dante allows a connection between testing computers and devices over long distances, up to 100 meters. Audio over IP also supports large channel counts, which is perfect for multichannel testing across multiple rooms in a facility. SoundCheck flexible hardware compatibility means networked audio devices can be configured just like any other audio interfaces. In an R&D lab, multiple test labs can have data transmitted to a central SoundCheck system.
SoundCheck Support Audio Over IP
Learn more about Listen audio interfaces
Read on for more information and technical specifications of the AmpConnect 621, AudioConnect 2. Audio interfaces can be used with a variety of test hardware including Bluetooth interfaces, turntables, accelerometers, and more. Check out all of SoundCheck’s compatibility with audio testing hardware.
Video Script:
SoundCheck is known for its flexibility to work with any soundcard or audio interface, but did you know it also supports Audio over IP using Dante?
Dante by Audinate allows audio to be transmitted over a standard local IP network. This offers simplified connections where your audio interface is located a long way from your SoundCheck computer, for example if it’s in a test lab or anechoic chamber. Connecting via a Dante interface and CAT 5E or 6 ethernet cable allows data to be transmitted up to 100 meters or more using your existing ethernet infrastructure – something that would be impractical and expensive with standard audio cables. It also offers high channel counts, and the network can be expanded with a high-speed network switch.
The Dante Interface, for example the RME Digiface Dante, is connected to the SoundCheck computer via USB and it routes the audio to and from any Dante device connected to the network, such as this Lynx Aurora. It also tracks latency over the Dante network. SoundCheck’s hardware editor displays all devices routed through the Dante Controller as Dante Channels in the SoundCheck Hardware Editor, where they can be treated exactly the same as any other input or output channels to enable a full range of audio tests.
Let’s take a look at how this might work for a speaker test. The Dante equipped Aurora interface is our test hardware, providing the output signal to the speaker, the microphone power, and receiving the signal from the microphone. It’s connected to the network via its ethernet connection. At the other end, the networked Dante interface is connected to the USB port of the SoundCheck computer where it acts as a hub for any Dante-equipped devices – in this case the Aurora. These devices then appear as a single ASIO audio interface with a USB 3 connection to the SoundCheck computer. From this point, you can configure your audio test exactly the same way as usual, and the Dante controller will handle the signal routing and synchronization of all Dante devices, even if they are different.
Multiple Dante Audio Interfaces can be connected to increase channel count. This setup, for example, allows for 32 balanced line inputs and outputs through the Lynx Aurora(n) with an additional 12 balanced microphone inputs with phantom power through the RME 12Mic-D.
Audio over IP has many applications in both R&D and production environments. In the R&D lab, it’s a simple and cost effective way of transmitting data from a remote test lab to a central computer, or to enable a fully mobile audio setup that can be moved around the facility. In production applications it enables centralized data collection from many different production lines. Contact your sales engineer to learn more.
100 Things #87: Make Non-Coherent Distortion Measurements
Did you know that you’ve been able to make distortion measurements in SoundCheck with real-world signals such as speech and music since 2006? This is a valuable technique for testing modern devices with on-board DSP that filters out signals such as sine waves and noise. Non-coherent distortion measurements offer excellent correlation with perception and are easily implemented in SoundCheck. Steve Temme explains this technique in this short video.
Make Non-Coherent Distortion Measurements
Read more about making non-coherent distortion measurements
The 2006 AES paper on non-coherent distortion measurements is available to read from our technical papers library. This paper details all of the important considerations for making these measurements, including using a multitone versus music for a stimulus signal, understanding distortion measurement results, and more.
Video Script:
We talk a lot about harmonic distortion and transient distortion, but did you know SoundCheck also offers non-coherent distortion measurements? In fact, I believe we were the first audio measurement company to include this option.
Non-coherent distortion is a broadband distortion metric that includes harmonic and intermodulation distortion as well as noise. It offers better correlation to perception than harmonic or intermodulation distortion alone, and it can be used with real-world test signals such as speech and music as long as there is enough energy in the frequency range of interest. Otherwise, you might just be measuring background noise. I usually make these measurements in the nearfield to reduce background noise by placing the microphone close to the loudspeaker. This is particularly useful for the many modern devices that feature DSP that treats pure tones as noise and tries to filter them out.
Non-Coherent Distortion is a normalized cross-correlation measurement that determines the degree to which the system output is linearly related to the system input.
There’s a lot of complex math behind this – if you want to know more about that you can read our 2006 AES paper. Here, I’m just going to show you a quick demonstration.
Configuring non-coherent distortion in SoundCheck is a simple checkbox in the transfer function analysis editor.
I have a good speaker, and a speaker that exhibits some fairly significant distortion. Let’s look at the good speaker first. I’m going to play a short excerpt of Bird On A Wire at 90dB SPL by Jennifer Warnes – this song is widely used as a test track as it has good dynamic range.
And if you look at the results, you can see non-coherent distortion in percent per square root Hertz (spectral density) versus frequency. Since non-coherent distortion uses a broadband test signal for measurement, there is no direct correlation to harmonic or intermodulation distortion in percent. Typically the distortion level appears much lower than harmonic or intermodulation distortion because the test signal energy is spread out over the entire frequency range and not a single frequency for measuring harmonic distortion.
Now I’m going to play the same song on a speaker that I know shows some fairly heavy distortion
Now, looking at these results, you can see the non-coherent distortion is considerably higher than the good unit, especially at low frequencies.
So that’s it. Non-coherent distortion offers a way of measuring transducers with real-world test signals that correlates well to listener perception. To learn more, check out our AES papers on the subject, or download our free test sequence for non-coherent distortion measurement.
100 Things #86: Listen’s Latest Generation Audio Interfaces
Traditional sound cards have drawbacks for audio testing, such as physical controls that can easily be altered, cabling errors, and the need for manual calibration. Two interfaces in our next generation of audio testing hardware: AudioConnect 2, and AmpConnect 621, fix these common problems. Both interfaces feature high resolution audio inputs and outputs, TEDS compatibility, microphone power, and internal signal routing. AmpConnect 621 features a built in amplifier and impedance measurement. AudioConnect 2 is portable and can be fully powered over USB type C.
Listen’s Latest Generation Audio Interfaces
Learn more about Listen audio interfaces
Read on for more information and technical specifications of the AmpConnect 621, AudioConnect 2. Audio interfaces can be used with a variety of test hardware including Bluetooth interfaces, turntables, accelerometers, and more. Check out all of SoundCheck’s compatibility with audio testing hardware.
Video Script:
Since SoundCheck was launched in 1995, it has become a standard as an affordable and flexible audio test and measurement system using pro audio soundcards as an audio interface . But did you know that we also offer our own audio interfaces?
You may wonder why, since soundcards offer so many advantages, so let me share some secrets about why we designed our own hardware, and some of its lesser-known benefits.
When correctly calibrated, high-end soundcards are accurate and cost-effective, especially for high channel counts. But they do have some drawbacks, particularly when used on the production line. Common pain points with sound cards include:
- Cabling errors – it’s easy to connect something up wrong, or to have a faulty or loose cable
- People may accidentally adjust the controls on the soundcard, especially in a busy lab or factory environment – if someone fiddles with your gain control mid way through a production run, you’re going to have a problem!
- It’s not a big deal when you only have one soundcard to set up, but if you are configuring 30 or 40 production lines, it takes time to configure the channels and calibrate everything. And on top of that, your test integrity depends on this being done correctly.
Our latest generation hardware combines high resolution audio inputs and outputs with TEDS compatibility, microphone power, internal signal routing, and in some cases amplifiers and impedance measurement, to simplify your setup.
These are our two latest audio interfaces, AudioConnect 2 and AmpConnect 621. AudioConnect 2 is our ultra-portable, laptop-powered, low cost 2-channel interface. It has 2 input and 2 output channels and a headphone output. The inputs offer both constant voltage power for our SCMs, and constant current for IEPE microphones and all inputs support TEDS. It’s great for headphone measurements, or for a portable setup. AmpConnect 621 has six powered microphone inputs and two output channels, as well as a built in amplifier and impedance current sensor. This one’s great for multi-channel applications such as measurements with a 6-mic array, or when you need to drive a passive speaker or artificial mouth.
So first off, you can see that these are more than just a soundcard – we’ve put all the functionality you need in one box, with all the signals routed internally, This means that the only connection you have is one USB cable and that makes it hard to mess up your connections. So we can take a speaker testing setup that looks like this and replace it with this. And it actually costs less than all the separate components.
You’ll also notice something missing – buttons. There are no knobs or buttons on the front panel, although we do have level and overload indicators so you can be sure that everything is operating within maximum dynamic range. All control is via the software. This means that once you set your hardware configuration as part of your test sequence, no-one can deviate from that at all – either by adjusting knobs, or by incorrectly setting it up in the first place. This ensures that your test is being run correctly, and it’s particularly useful if you are relying on 3rd party manufacturers to run your tests. If you specify the hardware, software and test sequence, you can guarantee your product is tested to your exact standards with no room for error.
Another benefit of Listen hardware is fast setup with seamless plug and play operation. The calibration data is stored on the device’s firmware so when you connect your interface to SoundCheck, the system reads the calibration values automatically so there’s no need for manual calibration. The input channels also automatically populate with sampling rates, and the device self-test requires no additional cabling as all switching is internal. If you’re using TEDS microphones, you can also automatically read this data too.
So, as you can see, although we always have and always will support a wide range of soundcards, our own hardware offers some clearly defined benefits. Conversely, there are other situations, for example high channel count, where a sound card is the more cost-effective option. Our goal is test system modularity and flexibility, so sound card or audio interface – the choice is yours.
