Free Test Sequences
These SoundCheck test sequences can be downloaded free of charge. They form a convenient starting point for many common tests, and can easily be adapted for your specific requirements.
Loudspeaker Test Sequences
ANSI/CEA 2010-B 2014 Sequence for Subwoofer Maximum SPL
This sequence measures the maximum peak SPL of a subwoofer according to ANSI/CEA 12010-B 2014. In this test, 1/3 octave band limited tone bursts are presented to the subwoofer across a 3 octave range from 20 Hz to 160 Hz. At each frequency, the stimulus level is increased in +3 dB increments until the harmonic (and non-harmonic) distortion and noise (HD+N) exceeds the specified threshold. The level is then decreased by 3 dB and the test continues with level increments of +1 dB until the HD+N threshold is again exceeded. The peak SPL of the fundamental at the last passing test level is recorded and the sequence continues to the next frequency. Peak SPL values are weighted according to the power spectrum defined in the standard and the Average Weighted SPL and final Broadband Peak SPL calculated as specified.
Polar Plot (MDT-4000 Turntable) Sequence
This sequence measures the polar response of a loudspeaker in both the vertical and horizontal dimensions. It is designed to work with the Portland Tool & Die MDT-4000 turntable, and has all the necessary commands to automatically rotate it via RS-232. The sequence uses a log sweep stimulus with the Time Selective Response algorithm so that the measurements can be run in a non-anechoic environment. Note that the time window needs to be adapted to the user’s measurement space.
The sequence plays the stimulus and measures at 10 degree increments from 0 to 180 degrees. This process is repeated with the speaker positioned horizontally. The two results are mirrored to display full 360 degree polar plots for each axis. A directivity index curve is also calculated for each axis and is displayed at the end of the test.
Loose Particles Sequence
This sequence demonstrates how to use SoundCheck to detect loose particle defects in loudspeakers. Loose particles typically reveal themselves as randomly spaced impulses, so they may not be detected when performing frequency based measurements such as THD, even though they can be clearly heard as undesirable artifacts. The loose particle algorithm, which is an available function in all analysis algorithms, analyzes a time waveform to detect these impulses. The user sets a customized threshold level for detection.
Complete Test Sequence
The purpose of this sequence is to perform a full suite of basic measurements for a loudspeaker. A 500 mV stepped sine sweep from 20 kHz to 50 Hz is played through the speaker and measured via two channels of the audio interface. A calibrated reference microphone is connected to one of the channels, and an impedance reference is connected to the other.
A HarmonicTrak™ analysis step analyzes the recorded waveform from the reference microphone, and outputs frequency response, THD, Rub & Buzz, and various harmonic curves. A second analysis step analyzes the waveform from the impedance reference and outputs a curve of impedance versus frequency. A post processing step is used to estimate the characteristics of the impedance curve and calculates the max impedance, resonance frequency, and the Q of the resonance peak.
Measuring Loudspeaker Maximum Linear Sound Levels Using Noise to the AES75-2022 Standard
The AES75-2022 standard details a procedure for measuring maximum linear sound levels of a loudspeaker system or driver using a test signal called M-Noise. This is a complex procedure with many repetitive steps, which makes it time consuming to implement manually. This sequence automates the entire process, accelerating test time, minimizing operator intervention, and ensuring accurate and objective test results.
Polar Plot (Linear X Turntable) Loudspeaker Sequence
This sequence measures the polar response of a loudspeaker in both the vertical and horizontal dimensions. It is designed to work with the Linear X turntable, and has all the necessary commands to automatically rotate it via RS-232. The sequence uses a log sweep stimulus with the Time Selective Response algorithm so that the measurements can be run in a non-anechoic environment. Note that the time window needs to be adapted to the user’s measurement space. The sequence plays the stimulus and measures at 10 degree increments from 0 to 180 degrees. This process is repeated with the speaker positioned horizontally. The two results are mirrored to display full 360 degree polar plots for each axis. A directivity index curve is also calculated for each axis and is displayed at the end of the test.
Time Selective Response Sequence
This sequence demonstrates the capabilities of the TSR (Time Selective Response) algorithm in SoundCheck. Using TSR with a log sweep is a very fast and effective method for measuring frequency response and THD of a speaker in a non-anechoic environment. This example plays a quick log sweep from 20 Hz to 20 kHz. An analysis step then uses the TSR algorithm with time windowing to output frequency response, harmonics, THD, and impulse response curves.
Enhanced Perceptual Rub & Buzz Demo Sequence
This sequence demonstrates how SC20’s new enhanced Perceptual Rub & Buzz algorithm compares to normalized Rub & Buzz and subjective listening. Running this sequence on a batch of good and bad buzzing loudspeakers should help identify by measurement, audibly defective units and where to set production limits. Starting the sequence, the user is asked at what test level to play a stepped sine sweep (Stweep™) in 1/12th octaves, from 20 kHz to 50 Hz. The user is encouraged to try different test levels and change sweep parameters to find the optimum settings to catch buzzing loudspeakers. The loudspeaker is measured via two channels of the audio interface. A calibrated reference microphone is connected to one of the channels and an impedance reference built into the SC Amp or AmpConnect is connected to the other. A HarmonicTrak™ Analysis step analyzes the recorded waveform from the reference microphone, and displays both the enhanced Perceptual Rub & Buzz and normalized Rub & Buzz graphs.
Complete End-of-Line Speaker Test (includes ePRB and eLP)
This sequence is an example of the many types of tests that can be performed quickly and simultaneously on a loudspeaker production line. A stepped sine sweep (StweepTM) from 20 kHz to 50 Hz is played through the speaker under test and measured via two channels of the audio interface. A calibrated reference microphone is connected to one of the channels and an impedance reference built into the SC Amp or AmpConnect is connected to the other. A HarmonicTrak™ Analysis step analyzes the recorded waveform from the reference microphone and outputs Frequency Response, THD, Normalized Rub & Buzz, Perceptual Rub & Buzz (ePRB), Loose Particle Prominence and Threshold (eLP) and Polarity. A Post-Processing step calculates the Average Sensitivity from 100 – 10kHz. A second Analysis step analyzes the recorded waveform from the impedance reference and outputs a curve of impedance versus frequency. A Post Processing step performs a curve fit of the impedance curve and calculates the max impedance (Zmax), precise resonance frequency (f0), and the quality factor (Q) of the resonance peak. All measurements and parameters are tested against limits in Limit steps. All these limits can be adjusted to suit your own DUT.
RT60 Room Acoustics
The RT60 room acoustics sequence measures reverberation time and clarity of a room using multiple microphones to accurately characterize measurement environments. This is important for smart device testing as measurements of both speech recognition and audio output often need to be made in fully characterized rooms with known reverberation times and clarity. The method used in this sequence is fast, accurate, and made using fully calibrated signal paths. This sequence uses an omnidirectional speaker to play a Log Sweep from 250Hz – 15kHz and four microphones measure the impulse responses generated. These waveforms are analyzed using the Time Selective Response and room acoustics algorithms to calculate reverberation time (T20, T39, T60) and clarity (C7, C50 and C80) according to ISO 3382-1:2009.
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Anechoic Response of a Loudspeaker Without an Anechoic Chamber: Splice Sequence
This sequence measures the anechoic response of a loudspeaker in an ordinary room using both a near field and time-windowed far field measurement “spliced” together to cover the full bandwidth of the loudspeaker’s response from 20 Hz to 40 kHz. First, the microphone is placed very close to the low frequency driver (less than an inch from the woofer), and the near field frequency response measured using a 1/12th octave stepped sine). Next, the microphone is placed in the far field and the frequency response is measured using a continuous log sweep with the Time Selective Response analysis algorithm. An option is available for measuring a ported loudspeaker. The sequence displays both measurements on a graph, showing the overlap range where the measurements are equal. From this, the user selects the precise frequency at which to splice the two halves of the measurements together to obtain the full range free field response of the loudspeaker.
Triggered Record Using Chirp Trigger and WAV File (Version 17 and later)
This test sequence demonstrates SoundCheck’s Triggered Record – Chirp Trigger function for open loop testing of devices without analog inputs such as smart speakers, wearables, smart home devices, tablets and cellphones. A stimulus WAV file is created in SoundCheck and transferred to the device under test, where it is played back and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The Acquisition step is triggered by the chirp in the stimulus file. Chirp triggers are more robust than level and frequency triggers which are susceptible to false triggering due to background noise.
Measuring Max SPL versus Frequency
This sequence measures the Max SPL of a transducer versus frequency that a device can play back with acceptable distortion. It is particularly valuable for designers using DSP algorithms to optimize the performance of their speakers.
It characterizes the Max SPL of a transducer by setting limits on specific metrics (THD, Rub & Buzz, Perceptual Rub & Buzz, Input Voltage and Compression) and then driving the transducer at a series of standard ISO frequencies, increasing the stimulus level until the one of the limits is surpassed. The sequence begins by measuring the frequency response and impedance of the DUT. The user is asked if they wish to use the -3dB from resonance frequency as the test Start Frequency or manually enter another value. The user is then prompted to enter a Stop Frequency, initial test level and limit values for the metrics of interest. The sequence then plays the stimulus Start Frequency in a loop, increasing the level +3dB with each loop iteration until one of the limits is exceeded. The stimulus level is then adjusted -3dB and the sequence continues to a second loop which increases the stimulus level +0.5 dB with each loop iteration until the limit is exceeded. At this point, the limit results are saved to an Excel file, the stimulus frequency is incremented by a constant multiplication step and the process is repeated until the Stop Frequency is achieved. Every time the main loop is completed, the individual SPL and Stimulus Level x-y pairs are concatenated to master curves. At the end of the sequence, the Max SPL and Stimulus Level curves are autosaved in .dat format.
Triggered Record Using WAV File (Version 16.1 and later)
This sequence allows you to test devices without an analog input such as smart speakers, tablets, cellphones and MP3 players using SoundCheck’s frequency-based trigger functionality. This method offers improved accuracy over previous level-based triggering, especially in noisy environments. A stimulus WAV file is created in SoundCheck, and copied to the device under test, where it is played and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The stimulus WAV file to be used on the device under test (DUT) may be customized in the stimulus step.
Note that this sequence uses the level-based trigger available in SoundCheck 16.1 and later. If you are using version 16.0 or earlier, please see the level-based trigger sequence.
Smart Speaker – Embedded Microphone Test Sequence
This sequence demonstrates a method by which SoundCheck can measure the performance of a microphone embedded in a so-called “smart speaker”. This example assumes that the DUT is an Amazon Echo but it can be adapted for use with virtually any other type of smart speaker by substituting the Echo’s voice activation phrase WAV file (“Alexa”) with one specific to the desired make and model.
The sequence begins by playing a voice activation phrase out of a source speaker, prompting the DUT to record both the voice command and the ensuing stepped sine sweep stimulus. A message step then prompts the operator to retrieve this recording from the DUT’s cloud storage system. This is accomplished by playing back the recording from the cloud and capturing it with a Triggered Record step in the SoundCheck test sequence. The Recorded Time Waveform is then windowed (to remove the voice command) and frequency shifted prior to analysis and the result (Frequency Response) is shown on the final display step.
Smart Speaker – Embedded Loudspeaker Test Sequence
This sequence demonstrates a method by which SoundCheck can measure the performance of a loudspeaker embedded in a so-called “smart speaker”. This example assumes that the DUT is an Amazon Echo but it can be adapted for use with virtually any other type of smart speaker by substituting the Echo’s voice activation phrase audio file (“Alexa, play Test Signal One”) with one specific to the desired make and model.
The sequence begins by playing the voice activation phrase out of a source speaker, prompting the DUT to playback the mp3 stimulus file from the cloud, followed by a pause step to account for any activation latency. Following the pause, a triggered record step is used to capture the playback from the DUT. The Recorded Time Waveform is then frequency shifted prior to analysis and the results (Frequency Response, THD and Perceptual Rub & Buzz) are shown on the final display step.
We recommend reading our AES paper on this subject prior to continuing as it contains additional details on the test methods devised for this sequence.
Comparison of Wired and Wireless (Bluetooth) Speaker Response
This test sequence performs frequency response and distortion measurements of a Bluetooth speaker using both a wireless Bluetooth and wired stimuli; then compares the results. This sequence is configured for use with a Portland Tool & Die BTC-4149/4148 or BQC-4149/4148 Bluetooth interface.
Initially, the sequence prompts the operator to turn on the Bluetooth device under test and set it to pairing mode. BTC message steps will connect the Bluetooth device (operator selects the device from a list of detected Bluetooth devices) and connects Bluetooth audio. A 1 kHz test tone is transmitted, and if detected, the test sequence proceeds. A stepped sine sweep from 20 kHz to 100 Hz is played wirelessly to the Bluetooth speaker and measured via a calibrated reference mic.
Two post-processing steps convert the sampling rate and alignment of the response, then an analysis step calculates the frequency response and THD. The Bluetooth is disconnected, and the Bluetooth frequency response and THD curves are displayed on graphs. The operator is then prompted to connect the wired analog input into the Bluetooth speaker, and the same measurements are performed using the analog connection. Analog frequency response and THD curves are temporarily displayed on graphs, followed by graphs containing both Bluetooth and analog curves for comparison.
Triggered Record Using WAV File (Version 16.0 and earlier)
This sequence allows you to test devices without an analog input such as tablets, cellphones and MP3 players. A stimulus WAV file is created in SoundCheck, and copied to the device under test, where it is played and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The stimulus WAV file to be used on the device under test (DUT) may be customized in the stimulus step.
Note that this sequence uses the level-based trigger available in SoundCheck 16.0 and earlier. If you are using version 16.1 or later, please see the frequency-trigger based sequence which takes advantage of new functionality to offer more robust triggering.
Thiele-Small Parameters
Loudspeaker system performance can be quantitatively related to a set of electro-mechanical parameters. These parameters are known in the industry as Thiele-Small parameters. They were first introduced by A.N.Thiele and Richard H.Small in a series of famous articles published in the 1971-72 Journal of AES (Audio Engineering Society). Over the years these parameters have become standards in the industry, and are used by loudspeaker designers worldwide. This package contains SoundCheck sequences for measuring measuring Thiele-Small Parameters by Added Mass, Known Volume, Known Driver Mass methods.
Impedance Measurement – Dual Channel Method Using Math Post-Processing
This sequence demonstrates an alternative to the traditional SoundCheck single channel impedance measurement method. A stepped sine sweep from 20 Hz to 20 kHz is played through the speaker while the signal across the loudspeaker terminals is recorded by Direct In 1 and the signal across the sense resistor (impedance box) is recorded by Direct In 2. A heterodyne analysis step is then applied to calculate the fundamental response from both inputs and a math post-processing step divides Fundamental A (speaker terminal voltage) by Fundamental B (voltage across sense resistor). A post-processing step corrects for the value of the reference resistor before displaying the final impedance curve. The curve is then post-processed to calculate resonance frequency, maximum impedance and Q of the resonance peak. A set of arbitrary limits steps are also provided to generate pass/fail results.
Perceptual Rub & Buzz
This sequence uses the CLEAR algorithm for perceptual Rub & Buzz measurement to detect AUDIBLE Rub & Buzz. It uses a simplified auditory perceptual model to measure the loudness of Rub & Buzz distortion in phons rather than the more traditional dB SPL and % distortion units. These better identify whether distortion due to manufacturing defects can be heard by the listener than conventional measurements. In addition to a result which corresponds more accurately to the human ear, this new test method also offers two significant advantages for use on the production line. It is less sensitive to transient background noises than traditional methods, therefore is reliable in noisy environments, and it is much simpler to set limits than when using conventional distortion measurements. The sequence includes saved data that can be loaded from disk, so even if you don’t have a speaker handy you can still listen to the wav. file and see how SoundCheck displays the data.
Impedance Measurement – Dual Channel Method Using Transfer Function
This sequence demonstrates an alternative to the traditional SoundCheck single channel impedance measurement method. A pink noise stimulus (10Hz – 10kHz) is played through the speaker while the signal across the amplifier terminals is recorded by Direct In 1 and the signal across the sense resistor is recorded by Direct In 2. A transfer function analysis step is then applied to the recorded time waveforms to calculate the impedance curve. The curve is then post-processed to calculate resonance frequency, maximum impedance and Q of the resonance peak. A set of arbitrary limits steps are also provided to generate pass/fail results. The final display shows the impedance curve and a results window.
Headphone and Headset Test Sequences
Headphones Test Sequence
The purpose of this sequence is to test a stereo headphone. Both left and right earphones are measured simultaneously using a standard 1/12th Octave stepped-sine sweep from 20 Hz to 20 kHz.
The analysis is then performed using Listen’s HarmonicTrak™ algorithm that measures harmonic distortion and fundamental frequency response simultaneously. Then the diffuse-field and free-field corrected Fundamentals are calculated. The diffuse-field correction curve compensates for the overall frequency response from the diffuse-field (sound in every direction) to the eardrum and includes the effects of the head, torso, pinna, ear-canal and ear simulator. The free-field correction curve compensates for the overall frequency response from the free-field (sound at 0 degree incidence to the nose of the Head and Torso Simulator – HATS) to the eardrum.
Further post-processing of the signal compares left and right earphone responses to show the difference curve (magnitude and phase are available). The average sensitivity from 100 to 10 kHz for both left and right earphone is calculated and the total harmonic distortion displayed.
Measuring Hearing Protection Devices to ANSI S3.19-1974 Standard
This sequence is used to measure the NRR, or Noise Reduction Rating, of a hearing protection device to the ANSI S3.19-1974 standard. NRR is a numerical representation of the sound attenuation of a device. The sequence first measures the response spectrum of the unoccluded hearing protector test fixture, then makes a second measurement with the hearing protection DUT affixed. A signal generator virtual instrument generates the pink noise stimulus while an RTA virtual instrument simultaneously records the A and C weighted noise spectrums. The unoccluded and occluded measurements are analyzed with a series of post-processing steps according to the ANSI S3.19-1974 standard. The final display shows the NRR numerical value, RTA spectra of the left and right side of the unoccluded and occluded hearing test fixture, average attenuation level of the DUT, and the standard deviation of the DUT on the test fixture.
Prediction of Listener Preference of In-Ear Headphones (Harman Model)
This sequence, inspired by AES papers on statistical models to predict listener preference by Sean E. Olive, Todd Welti, and Omid Khonsaripour of Harman International, applies the Harman target curve for in-ear, on-ear and over-ear headphones to a measurement made in SoundCheck to yield the predicted user preference for the device under test. The measurements are made in SoundCheck and then saved to an Excel template which performs the necessary calculations to produce a Predicted Preference score using a scale of 0 to 100. The spreadsheet calculates an Error curve which is derived from subtracting the target curve from an average of the headphone left/right response. The standard deviation, slope and average of the Error curve are calculated and used to calculate the predicted preference score. The sequence also provides the option to recall data rather than making a measurement, which saves time for engineers who already have large quantities of saved data, and enables historical comparison with obsolete products.
Measuring Max SPL versus Frequency
This sequence measures the Max SPL of a transducer versus frequency that a device can play back with acceptable distortion. It is particularly valuable for designers using DSP algorithms to optimize the performance of their speakers.
It characterizes the Max SPL of a transducer by setting limits on specific metrics (THD, Rub & Buzz, Perceptual Rub & Buzz, Input Voltage and Compression) and then driving the transducer at a series of standard ISO frequencies, increasing the stimulus level until the one of the limits is surpassed. The sequence begins by measuring the frequency response and impedance of the DUT. The user is asked if they wish to use the -3dB from resonance frequency as the test Start Frequency or manually enter another value. The user is then prompted to enter a Stop Frequency, initial test level and limit values for the metrics of interest. The sequence then plays the stimulus Start Frequency in a loop, increasing the level +3dB with each loop iteration until one of the limits is exceeded. The stimulus level is then adjusted -3dB and the sequence continues to a second loop which increases the stimulus level +0.5 dB with each loop iteration until the limit is exceeded. At this point, the limit results are saved to an Excel file, the stimulus frequency is incremented by a constant multiplication step and the process is repeated until the Stop Frequency is achieved. Every time the main loop is completed, the individual SPL and Stimulus Level x-y pairs are concatenated to master curves. At the end of the sequence, the Max SPL and Stimulus Level curves are autosaved in .dat format.
Bluetooth Headset Measurement
This sequence tests the send and receive performance of a stereo Bluetooth headset with a built-in microphone using a mixture of analog and digital channels. The left and right earphones are measured simultaneously with a stepped sweep from 20kHz to 20 Hz using two Bluetooth profiles: A2DP and HFP. The mic is measured with a stepped sweep from 8kHz to 100Hz using the HFP profile.
A short 1kHz tone is pre-pended to the test stimulus which serves as a reference tone for resampling and frequency shift operations. Post-processing resampling and frequency shift precisely synchronizes the stimulus and response waveforms prior to analysis. In this case, the HarmonicTrak algorithm is used for frequency and THD analysis. A2DP frequency response and THD curves are displayed on the first display, followed by A2DP & HFP curves superimposed on a subsequent display. Lastly, the Bluetooth headset’s microphone is tested with HFP and its frequency response is shown on the final display along with the previously collected data.
Lightning Headphone Test (Open Loop Test)
This sequence tests a stereo headphone connected to a portable audio device such as a mobile phone or MP3 player. It is particularly useful for testing headphones with proprietary connectors such as the Lightning connector which otherwise can’t be tested in a conventional “closed loop” test configuration.
The test stimulus is created in SoundCheck, saved as a WAV file and loaded on to the portable device for playback. Both left and right earphones are measured simultaneously using a continuous log sweep from 20 Hz to 20 kHz. The sequence uses a short 1 kHz tone, pre-pended to the normal test stimulus to automatically trigger the test when playback of the test signal begins; it also serves as reference tone for any frequency shift calculations. Post-processing precisely synchronizes the stimulus and response waveforms, and then calculation of the measurement parameters proceeds as with any conventional headphone. In this case, analysis is performed using the Time Selective Response (TSR) algorithm which performs THD and fundamental frequency response analysis simultaneously in addition to producing an impulse response. The fundamentals are then post processed to derive the sensitivity of the left and right channels at 1 kHz.
Headphone Testing with SoundCheck ONE
This SoundCheck ONE template sequence contains all the essential steps for basic headphone measurements using SoundCheck ONE and AudioConnectTM. The sequence can be easily customized and saved for specific products by turning individual measurements on and off, and by adjusting settings within each sequence step such as stimulus range and level, tolerance limits, graphical displays, and data saving.
Please note that sequences in SoundCheck ONE cannot have steps added/removed or the layout modified – the full version of SoundCheck is required for this capability.
IEC-60268-7 Headphone Sequences
IEC-60268-7: Sound System Equipment – Part 7: Headphones and Earphones is an international standard intended to characterize the performance of headphones and earphones. The standard itself is a lengthy document, 9 Sections and 3 Annexes covering 46 printed pages. These SoundCheck sequences focus on the electro-acoustic tests which are detailed in Section 8 “Characteristics to be specified and their method of measurement”.
Five separate sequences are provided, each designed to measure specific characteristics. This approach provides the user with the flexibility to measure all or some of the characteristics of their headphone.
Active Noise-Cancelling Headphone Battery Life Test
This sequence is designed to measure performance characteristics of Active Noise Cancelling (ANC) headphones while monitoring the DC voltage and current provided to the headphone by its batteries.
The sequence first measures the passive attenuation of the headphone before moving into a loop. The loop plays a 2 minute pink noise stimulus at high output level to accelerate battery drain. During this stimulus period, a current measurement is made by Listen’s DC Connect. Immediately following the stimulus, battery voltage is measured followed by acquisition and analysis of audio parameters (response, THD and THD Normalized). The active attenuation of the headphone is then measured followed by a series of post processing and Autosave steps. The looping continues until no output is detected from the headphone, when the device shuts down due to insufficient battery capacity.
Noise Cancelling Headphones
When measuring noise cancelling headphones there are three important pieces of data to collect. Passive Attenuation is the amount of noise that is reduced at the ear simply by the headphones being worn. Active Attenuation is the amount of noise that is further reduced by turning on the device’s active cancellation circuits. Lastly, Total Attenuation is the combined reduction in noise from passive and active sources and is what the end user of the product will experience.
To calculate these metrics this sequence performs three separate measurements using a Head and Torso Simulator and a small speaker which serves as a noise source. The alternative to using the small speaker would be to use a diffuse background noise environment with multiple speakers playing uncorrelated noise. This is a far more complicated arrangement and would require additional steps in the sequence.
Headphone Test Sequence
This sequence performs a comprehensive headphone test on a stereo headphone. Both left and right earphones are measured simultaneously using a standard 1/12th Octave stepped-sine sweep from 20 to 20 kHz.
The analysis is then performed using the HarmonicTrak™ algorithm that measures harmonic distortion and fundamental frequency response simultaneously, and the diffuse-field and free-field corrected curves are calculated. The diffuse-field correction curve compensates for the overall frequency response from the diffuse-field (sound in every direction) to the eardrum and includes the effects of the head, torso, pinna, ear-canal and ear simulator. The free-field correction curve compensates for the overall frequency response from the free-field (sound at 0 degree incidence to the nose of the Head and Torso Simulator – HATS) to the eardrum. Further post-processing of the signal compares left and right earphone responses to show the difference curve (magnitude and phase are available). The average sensitivity from 100 to 10 kHz for both left and right earphone is calculated and the total harmonic distortion displayed.
Microphone Test Sequences
Microphone Polar Plot Substitution Method Using Outline ET250-3D
This sequence measures the directional response of a microphone and graphs the result as a polar plot. A log sweep stimulus is played from 100 Hz to 10 kHz at each angular increment, and the acquired waveform is analyzed using the Time Selective Response algorithm. This method allows the test to be performed in a non-anechoic environment by placing a window around the direct signal, eliminating the influence of reflections. Commands are sent automatically to the Outline ET250-3D turntable via an ethernet connection, instructing it to move in 10 degree increments after each measurement. The sequence measures the response every 10 degrees from 0 to 180 and mirrors the polar image, which simulates a full 360 degree polar and saves test time. The response at each angular increment is compared against the on-axis response to create a normalized curve. This removes the influence of the device’s frequency response and sensitivity, such that the polar plot only shows the directional response. The final display also contains a graph of the directivity index in decibels versus frequency.
Measuring Digital MEMS Microphones – Frequency, Sensitivity, and Power Supply Rejection (PSR) Performance
This test suite contains 3 sequences to enable comprehensive testing of digital MEMS microphones:
- Measurement of frequency and sensitivity using a calibrated source
- Measurement of frequency using the ‘substitution method’, a solution for measuring frequency response where the source speaker cannot or is not equalized
- Measurement of power supply rejection (PSR) performance
Microphone Acoustic Overload Point (AOP)
This sequence measures the acoustic overload point (AOP) of a microphone – the SPL required to produce 10% THD @ 1kHz from the microphone’s output.
A 1 kHz amplitude sweep from a calibrated source speaker is applied across a range from 120 dB SPL to 135 dB SPL.The recorded time waveform is then analyzed using the HarmonicTrak algorithm to produce a Fundamental and THD vs. Level curve. An intersection post processing function identifies where the THD curve intersects 10% and that level is is used in a limit step to produce a Pass/Fail AOP verdict.
Microphone SNR Measurement (Background Noise Method)
This sequence characterizes a microphone’s ability to passively and/or actively reject noise in the user’s environment. Unlike traditional microphone SNR measurements which calculate a ratio based upon a reference signal and the microphone’s noise floor, this method utilizes a signal (speech played from a mouth simulator) and noise (background noise played from two or more equalized source speakers) captured by both a reference microphone and the DUT microphone.
First a recording of the baseline ambient noise in the test environment is made and a 1/3 octave RTA spectrum is calculated from the recording. Next, the speech signal (mouth simulator) and noise signals (Left and Right speakers) are played consecutively and recorded separately using the reference microphone. A 1/3 octave RTA spectrum is calculated from each recorded time waveform. Next the same measurements are repeated using the DUT microphone. The resulting RTA spectra are then post processed to produce a signal gain spectrum and a noise gain spectrum which are then used to derive the SNR spectrum of the DUT mic. For best accuracy, the Signal and Noise spectra should be at least 5 dB above the ambient noise floor of the measurement environment.
Microphone Polar Plot: Substitution Method Using LinearX LT360 Turntable
This sequence measures the directional response of a microphone and graphs the result as a polar plot. A log sweep stimulus is played from 100 Hz to 10 kHz at each angular increment, and the acquired waveform is analyzed using the Time Selective Response algorithm. This method allows the test to be performed in a non-anechoic environment by placing a window around the direct signal, eliminating the influence of reflections. Commands are sent automatically to the LT360 turntable via an RS-232 connection, instructing it to move in 10 degree increments after each measurement. The sequence measures the response every 10 degrees from 0 to 180 and mirrors the polar image, which simulates a full 360 degree polar and saves test time. The response at each angular increment is compared against the on-axis response to create a normalized curve. This removes the influence of the device’s frequency response and sensitivity, such that the polar plot only shows the directional response. The final display also contains a graph of the directivity index in decibels versus frequency.
Smart Speaker – Embedded Microphone Test Sequence
This sequence demonstrates a method by which SoundCheck can measure the performance of a microphone embedded in a so-called “smart speaker”. This example assumes that the DUT is an Amazon Echo but it can be adapted for use with virtually any other type of smart speaker by substituting the Echo’s voice activation phrase WAV file (“Alexa”) with one specific to the desired make and model.
The sequence begins by playing a voice activation phrase out of a source speaker, prompting the DUT to record both the voice command and the ensuing stepped sine sweep stimulus. A message step then prompts the operator to retrieve this recording from the DUT’s cloud storage system. This is accomplished by playing back the recording from the cloud and capturing it with a Triggered Record step in the SoundCheck test sequence. The Recorded Time Waveform is then windowed (to remove the voice command) and frequency shifted prior to analysis and the result (Frequency Response) is shown on the final display step.
Open Loop Microphone Testing
This sequence demonstrates the two most common microphone measurements, frequency response and sensitivity, on a microphone embedded in a recording device. Typically, when measuring a microphone the response of the device can be captured simultaneously with the stimulus. However, with devices such as voice recorders and wireless telephone forming a closed loop can be cumbersome or impossible. This sequence demonstrates how to measure such a device by recording the signal on the device under test, transferring that recording to the computer running SoundCheck and then using a Recall step to import the recorded waveform and analyze it.
This specific sequence, v4, is an improvement on the prior versions. The v1 release required that the audio file containing the recorded response waveform be manually windowed outside of SoundCheck before being analyzed. The v2 release utilized a new feature in SoundCheck 14, using values from the memory list to semi-automatically trim the waveform before analysis. The v3 release completely automated waveform editing through the use of an intersection level and windowing post processing steps. Currently the v4 release uses the new Auto Delay+ algorithm, exclusive to SC18 and beyond. Auto Delay+ is capable of detecting and accounting for delays of -0.5 seconds to any positive delay, nullifying the need for windowing steps in the sequence. If you are interested in learning more about this algorithm please refer to the Analysis section of the SoundCheck manual.
Measuring Digital MEMS Microphones: Frequency, Sensitivity and Power Supply Rejection (PSR) Performance
This test suite contains 3 sequences to enable comprehensive testing of digital MEMS microphones.
The first measures the frequency and sensitivity and displays two graphs: absolute level in dBFS, and the same response curve but normalized to 0 dB at 1 kHz.
The second sequence uses the substitution method to test a digital MEMS microphone frequency response with a source speaker that is not or cannot be equalized. The MEMS microphone is simultaneously measuring with a reference microphone , and by subtracting the response of the reference microphone from the DUT microphone the response and sensitivity of the device under test is revealed.
Measuring Digital Microphone PSR (Power Supply Rejection)
The third sequence demonstrates a method for measuring a digital MEMS microphone’s power supply rejection performance (PSR). This sequence measures PSR at 217 Hz (the 217 Hz GSM TDM pulse often of concern) but is easy to modify to test at any frequency. A DC supply with a calibrated AC signal, simulating electrical interference is applied to the MEMS microphone. SoundCheck then records the audio from the DUT, analyzes it with a spectrum analyzer and extracts the RMS energy at the specific frequency of the simulated electrical interference and returns the PSR value. The setting of frequency, waveform type and amplitude of the simulated electrical interference is controlled entirely from within SoundCheck.
Microphone Polar Plot Sequence
This sequence measures the directional response of a microphone and graphs the result as a polar plot. A log sweep stimulus is played from 100 Hz to 10 kHz at each angular increment, and the acquired waveform is analyzed using the Time Selective Response algorithm. This method allows the test to be performed in a non-anechoic environment by placing a window around the direct signal, eliminating the influence of reflections. Commands are sent automatically to the LT360 turntable via an RS-232 connection, instructing it to move in 10 degree increments after each measurement. The sequence measures the response every 10 degrees from 0 to 180 and mirrors the polar image, simulating a full 360 degree test while saving time. The response at each angular increment is compared against the on-axis response to create a normalized curve. This removes the influence of the device’s frequency response and sensitivity, such that the polar plot only shows the directional response. The final display also contains a graph of the directivity index in decibels versus frequency.
Microphone Self Noise Test
The purpose of this sequence is to measure the self noise of a condenser microphone. To do this we use the spectrum analysis step and a power sum calculation to derive an RMS rating for the unit under test.
The sequence has several parts, a few of which are optional. The first section begins by prompting you to enter your mic preamp’s gain. If your preamp does not have a gain setting with labeled detents you may wish to use the “Amplifier THD + N” sequence located in C:\SoundCheck 18.1\Sequences\Electronics\ to determine this value. The following part tests the preamp’s self noise so that the system can determine whether you have a high enough signal (signal being the microphones self noise) to noise ratio to accurately measure your microphone.
Telephone Test Sequences
Receive Loudness Rating with ITU Real Speech Test Sequence
The purpose of this sequence is to measure the Receive Loudness Rating (RLR) following the ITU-T P.79 standard using a Head and Torso Simulator (HATS). First, real speech from the ITU-T P.501 standard is sent to the Device Under Test (DUT) speaker by an electrical interface. The HATS right ear captures the DUT‟s speaker response. SoundCheck calculates the frequency response and then RLR based on that recording.
Send Loudness Rating with ITU Real Speech Test Sequence
The purpose of this sequence is to measure the Send Loudness Rating (SLR) following the ITU-T P.79 standard. This sequence can be used with handsets, headsets, and conference call devices. First, real speech from the ITU-T P.501 standard is played out of a mouth simulator. The Device Under Test (DUT) microphone then captures the signal and transmits this back to SoundCheck. SoundCheck calculates the frequency response function in 1/3 octaves and calculates SLR based on that frequency response.
Wireless (Bluetooth) Test Sequences
Triggered Record Using WAV File (Version 16.1 and later)
This sequence allows you to test devices without an analog input such as smart speakers, tablets, cellphones and MP3 players using SoundCheck’s frequency-based trigger functionality. This method offers improved accuracy over previous level-based triggering, especially in noisy environments. A stimulus WAV file is created in SoundCheck, and copied to the device under test, where it is played and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The stimulus WAV file to be used on the device under test (DUT) may be customized in the stimulus step.
Note that this sequence uses the level-based trigger available in SoundCheck 16.1 and later. If you are using version 16.0 or earlier, please see the level-based trigger sequence.
Bluetooth Headset Measurement
This sequence tests the send and receive performance of a stereo Bluetooth headset with a built-in microphone using a mixture of analog and digital channels. The left and right earphones are measured simultaneously with a stepped sweep from 20kHz to 20 Hz using two Bluetooth profiles: A2DP and HFP. The mic is measured with a stepped sweep from 8kHz to 100Hz using the HFP profile.
A short 1kHz tone is pre-pended to the test stimulus which serves as a reference tone for resampling and frequency shift operations. Post-processing resampling and frequency shift precisely synchronizes the stimulus and response waveforms prior to analysis. In this case, the HarmonicTrak algorithm is used for frequency and THD analysis. A2DP frequency response and THD curves are displayed on the first display, followed by A2DP & HFP curves superimposed on a subsequent display. Lastly, the Bluetooth headset’s microphone is tested with HFP and its frequency response is shown on the final display along with the previously collected data.
Comparison of Wired and Wireless (Bluetooth) Speaker Response
This test sequence performs frequency response and distortion measurements of a Bluetooth speaker using both a wireless Bluetooth and wired stimuli; then compares the results. This sequence is configured for use with a Portland Tool & Die BTC-4149/4148 or BQC-4149/4148 Bluetooth interface.
Initially, the sequence prompts the operator to turn on the Bluetooth device under test and set it to pairing mode. BTC message steps will connect the Bluetooth device (operator selects the device from a list of detected Bluetooth devices) and connects Bluetooth audio. A 1 kHz test tone is transmitted, and if detected, the test sequence proceeds. A stepped sine sweep from 20 kHz to 100 Hz is played wirelessly to the Bluetooth speaker and measured via a calibrated reference mic.
Two post-processing steps convert the sampling rate and alignment of the response, then an analysis step calculates the frequency response and THD. The Bluetooth is disconnected, and the Bluetooth frequency response and THD curves are displayed on graphs. The operator is then prompted to connect the wired analog input into the Bluetooth speaker, and the same measurements are performed using the analog connection. Analog frequency response and THD curves are temporarily displayed on graphs, followed by graphs containing both Bluetooth and analog curves for comparison.
Automotive Test Sequences
In-Car Audio Measurements
This sequence tests the impulsive distortion, frequency response, and maximum sound pressure level of a vehicle infotainment system to the methods outlined in the Audio Engineering Society Technical Committee on Automotive Audio (TC-AA) in-vehicle measurements white paper. This white paper aims to define repeatable and defined car audio system measurements and in addition to the measurement methods, contains information on standardized test configuration, for example microphone and seat positioning. Please contact the TC-AA for more information on this project. This test sequence may, of course, be used with your own in-house physical configuration if adherence to the TC-AA guidelines is not essential.
This sequence includes one master sequence, three subsequences, plus a level check subsequence for calibration. The three individual subsequence tests have also been included in a separate folder so they may be run standalone. The standalone sequences each have their own Bluetooth connect and disconnect steps.
Automotive Max SPL
The sequence measures the Max Sound Pressure Level (SPL) of a car infotainment system in the vehicle’s interior
The sequence uses a 6 microphone array mounted at either the driver or passenger locations. A 30 second pink noise stimulus having an RMS level of -12 dBFS is played through the infotainment system and captured by SoundCheck’s Multi-channel Real Time Analyzer (RTA). The Multi-channel RTA produces 6 RTA curves which are then power averaged to produce a Max SPL Spectrum. The spectrum is then power summed to produce a single value for Max SPL.
Triggered Record Using WAV File and 6 Mic Array
This sequence allows you to measure a playback system without analog inputs using a 6 microphone array. Specifically, the sequence is designed to measure an in-car audio system. A stimulus WAV file is created in SoundCheck and transferred to the device under test (DUT) where it is played back and the response captured by SoundCheck using a triggered record function. The 6 recordings are batch analyzed to produce individual fundamental curves and the curves are post-processed to produce a single average curve from which an average sensitivity value is calculated.
Hearing Aid Test Sequences
Hearing Aid Frequency Response Test Sequence
This sequence follows the ANSI S3.22-1996 standard method for testing the frequency response of a hearing aid. An equalized stepped sine sweep from 8 kHz – 200 Hz is played at a level of 60 dBSPL through the anechoic box speaker, and the output of the hearing aid is analyzed with the Heterodyne algorithm to produce a frequency response. Next, the HFA (High Frequency Average) is calculated by averaging the response values at three frequencies (1000, 1600, 2500 Hz). The HFA is then subtracted by 20 dB. Two post processing steps are used to find the upper and lower frequency points at which the response curve intersects this calculated value (HFA – 20 dB). These are the high and low frequency cutoff points.
Hearing Aid OSPL 90 Test Sequence
This sequence follows the ANSI S3.22-1996 standard method for measuring the OSPL curve, the HFA value, and the Max OSPL value for a hearing aid. An equalized stepped sine sweep from 8 kHz – 200 Hz is played at a level of 90 dBSPL through the anechoic box speaker, and a broadband response curve is analyzed through the hearing aid. Next, the HFA (High Frequency Average) is calculated by averaging the values at three frequencies (1000, 1600, 2500 Hz), and this value is checked with a limit step. The Max OSPL is calculated by finding the maximum point on the broadband response. A limit is also applied to this value.
Release Time Hearing Aid Test Sequence
This sequence follows the ANSI S3.22-1996 standard method for testing the release time of AGC (automatic gain control) hearing aids. A 2 kHz sine tone is played at 90 dBSPL for 1 second and then immediately drops to 55 dBSPL for 2 more seconds. A band limited time envelope (1.5-2.5 kHz) is created and then run through a post processing step, which calculates the release time. It does this by calculating the time it takes the device to stabilize within 4 dB of its steady level.
AGC Hearing Aid – Reference Gain & EIN Test Sequence
This sequence performs two of the measurements from the ANSI hearing aid test standard S3.22-1996. The first part, in accordance with section 6.7 of the standard, helps the user set the reference test gain for the hearing aid, which is used for multiple measurements in the standard. The second part, from section 6.12, tests the equivalent input noise (EIN) of the device.
Smartphone and Tablet Test Sequences
Triggered Record Using Chirp Trigger and WAV File (Version 17 and later)
This test sequence demonstrates SoundCheck’s Triggered Record – Chirp Trigger function for open loop testing of devices without analog inputs such as smart speakers, wearables, smart home devices, tablets and cellphones. A stimulus WAV file is created in SoundCheck and transferred to the device under test, where it is played back and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The Acquisition step is triggered by the chirp in the stimulus file. Chirp triggers are more robust than level and frequency triggers which are susceptible to false triggering due to background noise.
Open Loop Microphone Testing
This sequence demonstrates the two most common microphone measurements, frequency response and sensitivity, on a microphone embedded in a recording device. Typically, when measuring a microphone the response of the device can be captured simultaneously with the stimulus. However, with devices such as voice recorders and wireless telephone forming a closed loop can be cumbersome or impossible. This sequence demonstrates how to measure such a device by recording the signal on the device under test, transferring that recording to the computer running SoundCheck and then using a Recall step to import the recorded waveform and analyze it.
This specific sequence, v4, is an improvement on the prior versions. The v1 release required that the audio file containing the recorded response waveform be manually windowed outside of SoundCheck before being analyzed. The v2 release utilized a new feature in SoundCheck 14, using values from the memory list to semi-automatically trim the waveform before analysis. The v3 release completely automated waveform editing through the use of an intersection level and windowing post processing steps. Currently the v4 release uses the new Auto Delay+ algorithm, exclusive to SC18 and beyond. Auto Delay+ is capable of detecting and accounting for delays of -0.5 seconds to any positive delay, nullifying the need for windowing steps in the sequence. If you are interested in learning more about this algorithm please refer to the Analysis section of the SoundCheck manual.
Audio Electronic Test Sequences
Amplifier THD+N Sequence
This sequence measures an audio amplifier’s Frequency Response, Gain, THD, THD+Noise, and Self-noise. It accomplishes this by playing a 1/3rd octave sine sweep through the amplifier. A HarmonicTrak™ analysis step calculates the fundamental frequency response curve as well as the distortion plots. The sequence then records and analyzes a spectrum of the amplifier’s self-noise.
MP3 Player (Multitone) Sequence
The principle is first to create a multitone stimulus waveform with SoundCheck and transfer it to the MP3 Player to be tested. This waveform is then played back, to be acquired and analyzed by SoundCheck. Both Left and Right channels are acquired at once. The acquisition is made easy by using a trigger on one input channel. The analysis is done by comparing the genuine stimulus and the play-back, using a cross-spectrum technique that provides the FRF, Cross-talk and Non-Coherent Distortion. The cross-talk between channels is measured by playing simultaneously two different multitones for Left and Right channels with interleaved frequencies, and making a cross-analysis between channels (e.g. finding in the right response the contribution of the left stimulus).
Due to inaccuracies of clock frequency, the play-back sampling rate is often different from the Soundcard recording sampling rate. Therefore, in SoundCheck, the recorded time Waveforms are frequency shifted as to obtain a perfect match between stimulus and responses. This frequency correction is performed on both channels before the Analysis steps. The exact playback sampling rate is displayed. The Sequence can be measure the MP3 player on the WAV or the MP3 file playback. The MP3 playback will yield a poorer FRF and more distortion, due to the lossy compression.
Triggered Record Using Chirp Trigger and WAV File (Version 17 and later)
This test sequence demonstrates SoundCheck’s Triggered Record – Chirp Trigger function for open loop testing of devices without analog inputs such as smart speakers, wearables, smart home devices, tablets and cellphones. A stimulus WAV file is created in SoundCheck and transferred to the device under test, where it is played back and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The Acquisition step is triggered by the chirp in the stimulus file. Chirp triggers are more robust than level and frequency triggers which are susceptible to false triggering due to background noise.
Triggered Record Using WAV File (Version 16.1 and later)
This sequence allows you to test devices without an analog input such as smart speakers, tablets, cellphones and MP3 players using SoundCheck’s frequency-based trigger functionality. This method offers improved accuracy over previous level-based triggering, especially in noisy environments. A stimulus WAV file is created in SoundCheck, and copied to the device under test, where it is played and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The stimulus WAV file to be used on the device under test (DUT) may be customized in the stimulus step.
Note that this sequence uses the level-based trigger available in SoundCheck 16.1 and later. If you are using version 16.0 or earlier, please see the level-based trigger sequence.
Triggered Record Using WAV File (Version 16.0 and earlier)
This sequence allows you to test devices without an analog input such as tablets, cellphones and MP3 players. A stimulus WAV file is created in SoundCheck, and copied to the device under test, where it is played and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The stimulus WAV file to be used on the device under test (DUT) may be customized in the stimulus step.
Note that this sequence uses the level-based trigger available in SoundCheck 16.0 and earlier. If you are using version 16.1 or later, please see the frequency-trigger based sequence which takes advantage of new functionality to offer more robust triggering.
General (non-application-specific) Test Sequences
Comparison of Effect of Different Masks on Speech Distortion
The purpose of this sequence is to compare the response of an artificial mouth when face masks of different construction are mounted on the mouth. The sequence assumes a lightweight disposable mask and a heavy multi-layer cloth mask. A stepped sinewave from 10 kHz – 100 Hz is played from the unoccluded mouth and the operator is then prompted to mount and measure the two masks over the mouth. The three mouth responses are then displayed on one x-y graph and the difference curves (unoccluded mouth vs masked mouth) are displayed on another. Finally, the average attenuation created by each mask across the measurement range is displayed on a table.
Note that the curve names are constructed by selecting the “Use Input Data Name” option on the Curves tab of the Analysis editors. The appended text in parentheses (No Mask, Disposable Mask, Cloth Mask) comes from the custom naming of the three Recorded Time Waveforms so if you wish to edit these, it can be done by editing the Waveform names in the Acquisition steps.
Waveform Windowing
This sequence demonstrates how SoundCheck’s Windowing post-processing function is applied to waveforms to remove measurement artifacts that might otherwise create false auto delay values and subsequent analysis errors.
This sequence uses data from a customer who was measuring the directivity of a hearing aid-type device by mounting it on a rotating HATS and using a short duration log sweep. The DUT does not have a perfect seal in the HATS ear and the devices signal processing produces a latency of around 35ms. When viewing the Recorded Time waveforms, both the leakage signal and the amplified signal can be seen. As the DUT approaches 180° the magnitude of the leakage into the HATS ear exceeds that of the amplified signal, creating false Record Delay values and subsequent analysis errors. This sequence applies a window to the Recorded Time Waveform to remove the early-arrival leakage, and calculates the true Record Delay values of the amplified signal, obtaining consistent analysis results at all angles of rotation. This sequence can be adapted to your other requirements, for example, removing early arrival signals from a waveform or editing out excessive delay in a waveform.
External Control of SoundCheck via TCP/IP
TCP/IP control of SoundCheck provides a powerful and expandable command format for controlling and interacting with SoundCheck via any programming language (C#, C++, MATLAB, VB.net, LabVIEW, Python, etc.), on any operating system, either locally or through a network. This is valuable for anyone wishing to control SoundCheck from an external program, e.g. as part of an overall test plan or for factory automation. Using this feature, a single computer can control multiple SoundCheck systems, simplifying production line measurements.
The ability to connect to, and control SoundCheck via TCP/IP first appeared in SoundCheck over 3 years ago, but for version 18, it has been enhanced with the ability to pass test configuration data into the memory list from external programs. This means, for example, that by externally storing parameters such as limits, test levels, and test signals, a single sequence can be used for multiple products, or testing the same product multiple times, simplifying sequence maintenance, and reducing test configuration time.
This application note and accompanying demo scripts walk you through how to use Python to:
- Control a simple loudspeaker test setup, launching SoundCheck and running a sequence
- Run a simple frequency response sequence from a command line interface, creating placeholder curves, values, results and waveforms in the MemoryList and pass values into the placeholders via external control.
- Read a WAV file and use it as a stimulus for performing an FFT Spectrum measurement in SoundCheck
Triggered Record Using Chirp Trigger and WAV File (Version 17 and later)
This test sequence demonstrates SoundCheck’s Triggered Record – Chirp Trigger function for open loop testing of devices without analog inputs such as smart speakers, wearables, smart home devices, tablets and cellphones. A stimulus WAV file is created in SoundCheck and transferred to the device under test, where it is played back and the response recorded in SoundCheck as if the stimulus were played directly from SoundCheck. The Acquisition step is triggered by the chirp in the stimulus file. Chirp triggers are more robust than level and frequency triggers which are susceptible to false triggering due to background noise.
Measuring Max SPL versus Frequency
This sequence measures the Max SPL of a transducer versus frequency that a device can play back with acceptable distortion. It is particularly valuable for designers using DSP algorithms to optimize the performance of their speakers.
It characterizes the Max SPL of a transducer by setting limits on specific metrics (THD, Rub & Buzz, Perceptual Rub & Buzz, Input Voltage and Compression) and then driving the transducer at a series of standard ISO frequencies, increasing the stimulus level until the one of the limits is surpassed. The sequence begins by measuring the frequency response and impedance of the DUT. The user is asked if they wish to use the -3dB from resonance frequency as the test Start Frequency or manually enter another value. The user is then prompted to enter a Stop Frequency, initial test level and limit values for the metrics of interest. The sequence then plays the stimulus Start Frequency in a loop, increasing the level +3dB with each loop iteration until one of the limits is exceeded. The stimulus level is then adjusted -3dB and the sequence continues to a second loop which increases the stimulus level +0.5 dB with each loop iteration until the limit is exceeded. At this point, the limit results are saved to an Excel file, the stimulus frequency is incremented by a constant multiplication step and the process is repeated until the Stop Frequency is achieved. Every time the main loop is completed, the individual SPL and Stimulus Level x-y pairs are concatenated to master curves. At the end of the sequence, the Max SPL and Stimulus Level curves are autosaved in .dat format.
