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.

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Complete End-of-Line Speaker Test (includes ePRB)

This sequence is an example of the many types of tests that can be performed quickly and simultaneously on a loudspeaker production line. It includes perceptual distortion measurement with the new enhanced Perceptual Rub & Buzz algorithm. A stepped sine sweep (StweepTM) 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 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, Loose Particle Envelope and Polarity. A Post-Processing step calculates the Ave. Sensitivity from 100 – 10kHz. A second analysis step analyzes the waveform from the impedance reference and outputs a curve of impedance versus frequency. Another 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 parameter are tested against limits in Limit steps. All these test parameters can be adjusted accordingly.

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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.

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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.

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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

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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.

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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.

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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.

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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.

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