Audyssey Laboratories, Inc.

Jeffrey Clark

A minimum level for a stimulation signal used in room correction processing is determined by measuring background noise. The stimulation signal is repeated a number of times and resulting responses are recorded. The recording responses are averaged, and the average is subtracted from each recorded response to obtain the background noise present in each recorded response. A stimulation signal to background noise ratio is computed from the stimulation signal and background noise and compared to an SNR threshold to determine if the stimulation signal level is sufficient to support the room correction processing. The background noise may be AC hum introduced electronically into the response signal, acoustic noise introduced by AC ventilation systems or noise emitting devices (refrigerators, etc), and it may be structure-born noise introduced by shaking the microphone, e.g. a bus drives by, shaking the floor the microphone is standing on.

Background noise measurement from a repeated stimulus measurement system

A method for equalization contouring provides a reduction of equalization in certain frequency regions either by user control or by automated selection of frequency, without introducing artifacts. A control curve smoothly scales the magnitude of the equalization in the areas where less equalization is desired to obtain a contoured equalization. The control curve varies by frequency and may be defined specifically for every sampled frequency value of the equalization, may be a continuous function of frequency, or may be a function of control points at a select number of frequency points. The control curve may also have automatic inputs, e.g. a machine-detected cutoff frequency of a speaker may be used to determine a control point in the control curve. As another example, the reverberation time (e.g. RT60) may be used to determine a control point in the control curve. The result is a contoured equalization curve without sudden steps.

Equalization contouring by a control curve

An attenuation control before digital signal processing provides digital headroom. A pre-digital signal processing volume control receives a volume control signal and provides a pre-digital signal processing first attenuation of an input signal before digital signal processing. A master volume control provides a master volume control attenuation after the digital signal processing. The first attenuation may be the same as the master volume control attenuation, or it may be less or more than the master volume control attenuation, depending on the application and desired headroom. The first attenuation and master volume control attenuation may be linearly related or non-linearly related, or not a mathematical function (non-continuous, etc) at all. Changes in the attenuation may be ramped to avoid discontinuities in the signal, preferably over a period between 10 and 100 milliseconds, and are preferably applied as close to simultaneously with master volume changes as possible.

Volume curve adjustment for signal processing headroom

A method for measuring acoustic impulse response of a Device Under Test (DUT) addresses sample rate variations by determining clock rate differences between the DUT and test system, and making adjustments to compensate for the different clock rates. An interrogation signal is generated with two 'events' spaced a known number of samples apart, at the nominal sampling rate of the DUT. The interrogation signal is played through the DUT and recorded by a measurement system. The number of samples between the two 'events' is measured in the sampled signal to determine the shift introduced by the different clock rates. The adjustment is then applied to either the original frequency response measurement signal by adjusting the sampling rate of a frequency response measurement signal to compensate for the different clock rates, or to the measured signal before averaging, to align the samples and prevent cancelling.

Method for asynchronous impulse response measurement between separately clocked systems

A filtering method approximates a target Finite Impulse Response (FIR) (or transversal) filter and reduces computational requirements by eliminating high pass filtering required by known multi-rate filters. An input signal is copied into two identical signals and processed in parallel by a full-rate path, and by a reduced-rate path. Parallel filters are computed and applied in each path, the reduced-rate signal is up-sampled, and the two signals summed. The high pass filter required by known multi-rate filters is eliminated and the low pass filter in the prior art is implicit in a down sampling. Linear phase FIR filters are used for down and up sampling, resulting in constant group delay. Added benefits include the option of zero added latency through the filtering and the constant group delay added to the target FIR. The user may choose criteria such as minimum resolution in each band.

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Alhambra, CA, United States of America

Jeffrey Clark