Abstract: A source device uses a profile of an audio output device (e.g., headphones or speakers) to adjust the acoustic output of the audio output device. A database of audio output device profiles is stored in a cloud or locally on the source device. The profiles may include electroacoustic measurement data characterizing the audio output device or processing parameters for the audio output device. A program running on the source device selects a profile corresponding to the connected audio output device. The profile is used by the software running on the source device to determine processing for an audio stream played by the audio output device. The processing provides equalization to modify the unique audio output device frequency response, and compensation for human perception of sound at different listening levels, and dynamic range adjustment to better match the capabilities of the audio output device.

Abstract: 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.

Abstract: 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.

Abstract: A speech/music discrimination method evaluates the standard deviation between envelope peaks, loudness ratio, and smoothed energy difference. The envelope is searched for peaks above a threshold. The standard deviations of the separations between peaks are calculated. Decreased standard deviation is indicative of speech, higher standard deviation is indicative of non-speech. The ratio between minimum and maximum loudness in recent input signal data frames is calculated. If this ratio corresponds to the dynamic range characteristic of speech, it is another indication that the input signal is speech content. Smoothed energies of the frames from the left and right input channels are computed and compared. Similar (e.g., highly correlated) left and right channel smoothed energies is indicative of speech. Dissimilar (e.g., un-correlated content) left and right channel smoothed energies is indicative of non-speech material. The results of the three tests are compared to make a speech/music decision.

Abstract: 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.

Abstract: Volume extension includes limiting the magnitude of Fast Fourier Transform (FFT) frequency bins which allows increases to the perceived level of audio content without causing distortion. A soft limit and smoothing is applied to each FFT bin is to prevent or reduce distortion while maximizing output volume. Frequency resolution is significantly improved compared to volume extension methods utilizing filterbanks and hard limiting, and distortion is reduced because no hard limiting occurs.

Abstract: 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.

Abstract: A speaker leak test system which quickly tests if a speaker enclosure is sealed properly. The speaker leak test system tests for air leaks in the speaker enclosure, which air leaks degrade the acoustic performance of the speaker. A vacuum (or pressure) source is connected through a vacuum hose to the speaker enclosure and vacuum is drawn. After a test vacuum level is reached, the mass air flow through the vacuum hose is measured. If the mass air flow does not exceed a pre-determined speaker dependent threshold for the speaker under test, the speaker is declared to be leak free. A bypass solenoid may reside in parallel with a mass air flow sensor to reach test vacuum quickly. This speaker leak test system is intended to be used in a prototyping and production environment and tests the speaker enclosure for leaks both quickly (e.g., less than five seconds) and accurately.

Abstract: A Voice Activity Detection (VAD) algorithm provides a simple binary signal indicating the presence or absence of speech in a microphone signal. The VAD algorithm includes a first step of noise suppression which both estimates and removes (i.e., filters) ambient noise from the microphone signal to create a filtered signal. The magnitude of the filtered signal is then compared to a threshold in order to produce a VAD output signal. The threshold is dynamic and may be derived either from the filtered signal itself, or from a noise spectrum estimate calculated by the noise suppression step.

Abstract: Methods, devices and computer program products facilitate the generation of high quality audio content. The audio content includes upper harmonics that are associated with a bass band of frequencies in order to enhance the perception of bass audio components that cannot be produced by the audio speakers. The upper harmonics are generated and processed in such a way that reduces the computational and memory requirements of the audio processing operations. The processed upper harmonics are combined with the original audio that are properly delayed to enhance the quality of the audio content. This Abstract is provided for the sole purpose of complying with the Abstract requirement rules that allow a reader to quickly ascertain the disclosed subject matter. Therefore, it is to be understood that it should not be used to interpret or limit the scope or the meaning of the claims.

Abstract: A Dynamic Noise Compensation (DNC) telephone speech enhancement algorithm addresses the issue of environment noise on the listener end of a telephone call. A single microphone proximal to the listener provides a sample of near end ambient noise level and of near end speech. A Voice Activity Detector (VAD) detects the presence of near end (listener) speech. The DNC algorithm adjusts the far end incoming speech level based on the near end ambient noise and the VAD ensures that the near end listener speech does not effect the incoming speech level adjustment.

Abstract: A spectral response shifting method allows a listener to shift a system response to match their preferences. The method includes a lookup table of coefficients for a number of Infinite Impulse Response (IIR) filter or Finite Impulse Response (FIR) filter which are selected by the listener to shift the response either toward bass or towards treble. In one embodiment, 5th order IIR biquads filters are used at ½ dB increments from ?6 dB to +6 dB. For example, when a listener selects the “more treble by 1 dB” setting, a set of coefficients is loaded providing ?1 dB in bass cut at the lowest frequency of the audible spectrum (20 Hz) and +1 dB of treble boost at the highest frequency of the audible spectrum (20 kHz). The frequency response at all other points is defined by interpolating between the highest frequency (20 kHz) and the lowest frequency (20 Hz).

Abstract: A speaker leak test system which quickly tests if a speaker enclosure is sealed properly. The speaker leak test system tests for air leaks in the speaker enclosure, which air leaks degrade the acoustic performance of the speaker. A vacuum (or pressure) source is connected through a vacuum hose to the speaker enclosure and vacuum is drawn. After a test vacuum level is reached, the mass air flow through the vacuum hose is measured. If the mass air flow does not exceed a pre-determined speaker dependent threshold for the speaker under test, the speaker is declared to be leak free. A bypass solenoid may reside in parallel with a mass air flow sensor to reach test vacuum quickly. This speaker leak test system is intended to be used in a prototyping and production environment and tests the speaker enclosure for leaks both quickly (e.g., less than five seconds) and accurately.

Abstract: A Voice Activity Detection (VAD) algorithm provides a simple binary signal indicating the presence or absence of speech in a microphone signal. The VAD algorithm includes a first step of noise suppression which both estimates and removes (i.e., filters) ambient noise from the microphone signal to create a filtered signal. The magnitude of the filtered signal is then compared to a threshold in order to produce a VAD output signal. The threshold is dynamic and may be derived either from the filtered signal itself, or from a noise spectrum estimate calculated by the noise suppression step.

Abstract: A Dynamic Noise Compensation (DNC) telephone speech enhancement algorithm addresses the issue of environment noise on the listener end of a telephone call. A single microphone proximal to the listener provides a sample of near end ambient noise level and of near end speech. A Voice Activity Detector (VAD) detects the presence of near end (listener) speech. The DNC algorithm adjusts the far end incoming speech level based on the near end ambient noise and the VAD ensures that the near end listener speech does not effect the incoming speech level adjustment.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: A system and method for minimizing the complex phase interaction between non-coincident subwoofer and satellite speakers for improved magnitude response control in a cross-over region. An all-pass filter is cascaded with bass-management filters in at least one filter channel, a[1d preferably all-pass filters are cascaded in each satellite speaker channel. Pole angles and magnitudes for the all-pass filters are recursively calculated to minimize phase incoherence. A step of selecting an optimal cross-over frequency may be performed in conjunction with the all-pass filtering, and is preferably used to select an optimal cross-over frequency prior to determining all-pass filter coefficients.

Abstract: A combined multirate-based Finite Impulse Response (FIR) filter equalization technique combines a low-order FIR equalization filter operating at a lower rate for equalization of a loudspeaker-room response at low frequencies, and a complementary low-order minimum-phase FIR equalization filter operating at a higher rate for equalization of the loudspeaker-room response at higher frequencies. The design of two complementary band filters for separately performing low and high frequency equalization, keeps the system delay at a minimum while maintaining excellent equalization performance. Splicing between the two equalization filters, for maintaining a flat magnitude response in the transition region of the two complementary filters, is done automatically through level adjustment of one equalization filter relative to the other. The present invention achieves excellent equalization at low filter orders and hence reduced computational complexity.

Abstract: Low-frequency bandwidth extension in the form of dynamic electrical equalization may be applied to loudspeakers so long as the excursion capability of their drive units as well as velocity limits of any port(s) or excursion limits of any associated passive radiator(s), and the power limits of the drive units are not exceeded. The bandwidth extension maximizes low-frequency bandwidth dynamically such that excursion is fully utilized over a range of drive levels, without exceeding the excursion limit. Additional limiting control is available for port air velocity or passive radiator excursion, and loudspeaker drive unit electrical power. The system applies to open back, closed box, vented box, and more complex box constructions consisting of combinations of these elements for loudspeaker designs using design parameters appropriate to each system.