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.
This sequence measures the self noise of a condenser microphone, using a spectrum analysis step and a power sum calculation to derive an RMS rating for the unit under test.
The sequence has several parts (some optional). The sequence intitally determines whether or not you have a high enough signal (signal being your microphone’s self noise) to noise ratio to accurately measure your microphone. It then takes a measurement from your microphone, creates a spectrum, accounts for the preamp gain, applies an A-weighting, and finally calculates the power sum. You are then prompted to enter the sensitivity of the microphone if it is known. The resulting display provides you with the equivalent input noise of your preamp, the self noise of your microphone in dBV, and also a result in dB(A) after factoring in the microphone’s sensitivity. You’re also provided with the waveform, a maximum voltage level, and the crest factor to check for sharp transients.
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.
When measuring noise cancelling headphones there are three important pieces of data to generate. 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 feature. 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 develop a diffuse environment with multiple speakers playing uncorrelated noise. This is a far more complicated arrangement and would require additional steps in the 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.
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.
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.
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.
This sequence demonstrates an alternative to the traditional SoundCheck single channel impedance measurement method. A white noise stimulus (10 Hz – 10 kHz) is played through the speaker while the signal across the amplifier terminals is recorded by Direct In 1 and the signal across the impedance box is recorded by Direct In 2. A transfer function analysis step is then applied to the recorded time waveforms to calculate the impedance curve. Subsequent post processing steps apply a frequency window, 1/24th octave smoothing and 1/24th octave resolution to 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 post processed impedance curves and results window.
The purpose of this sequence is to measure 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 to 40 kHz.
First, the near field frequency response is measured using a 1/12th octave stepped sine sweep by placing the microphone very close to the low frequency driver (less than an inch from the woofer). Then the far field frequency response is measured using a continuous log sweep with the Time Selective Response analysis algorithm.
- AES Convention– New York – October 16-19, 2019October 16, 2019 - 5:51 pm
- New Sales Director: Joe KushiAugust 12, 2019 - 2:51 pm
- AES International Conference on Headphone Technology – San Francisco – August 27-29, 2019August 12, 2019 - 2:15 pm
- AES International Conference on Automotive Audio – Munich – September 11-13, 2019August 12, 2019 - 1:55 pm
- ISEAT- Shenzhen, China – Nov 9-10, 2019August 12, 2019 - 10:10 am