Harman International Industries, Inc.

Harman Becker Automotive Systems Gmbh

Other

Her Majesty the Queen As Represented by the Minister of Industry Through the Communications Research Centre

Gilbert Arthur Joseph Soulodre

Kevin Eric Heber

Louis Thibault

Kirk I Bushen

William C Treurniet

A sampler module divides an audio signal into a series of sequential samples. A signal quality detector module identifies, over a plurality of samples at an outset of the audio signal, a spectral variance of a first range of frequencies of the audio signal below a predetermined threshold frequency as being consistently greater than a spectral variance of a second range of frequencies of the audio signal above the predetermined threshold frequency and determines a signal treatment indication responsive to the identification. A signal enhancer module sequentially receives and analyzes one or more sample components of the audio signal to identify lost parts of the audio signal in the one or more sample components of respective sequential samples, and generates, in accordance with the signal treatment indication, a corresponding signal treatment for each of the one or more sample components of respective sequential samples having a corresponding identified lost part.

Signal quality-based enhancement and compensation of compressed audio signals

A system for processing compressed audio includes a signal enhancer module configured to generate one or more signal treatments. The one or more signal treatments may be generated by the signal enhancer module based on analysis of the incoming audio signal. Alternatively, or in addition, characteristics of the incoming audio signal may be provided to the signal enhancer module for use in generating the one or more signal treatments. The one or more signal treatments may be added to file audio signals.

System for perceived enhancement and restoration of compressed audio signals

A sampler module may divide an audio signal into a series of sequential samples. A signal quality detector module may identify a consistent brick wall frequency of the audio signal spanning a plurality of the sequential samples at an outset of the audio signal and determine a signal treatment indication proportional to the brick wall frequency. A signal enhancer module may sequentially receive and analyze one or more sample components of the audio signal to identify lost parts of the audio signal in the one or more sample components of respective sequential samples, and generate, in accordance with the signal quality indication, a corresponding signal treatment for each of the one or more sample components of respective sequential samples having a corresponding identified lost part.

A multichannel compensating audio system includes first and second compensation channels to psychoacoustically minimize deviations in a target response, to psychoacoustically move the physical position of a speaker and/or to psychoacoustically provide a substantially equal magnitude of sound from a plurality of speakers in a plurality of different listening positions. The first compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a first compensated audio signal from a first audio signal. The second compensation channel may include a series connected delay circuit, a level adjuster circuit and a frequency equalizer circuit that generates a second compensated audio signal from a second audio signal. A first summing circuit is configured to receive at least the first audio signal and the second compensated audio signal and generate a first output signal for provision to a first speaker. A second summing circuit is configured to receive the second audio signal and the first compensated audio signal and generate a second output signal for provision to a second speaker. The first and second output signals may be output by the first and second speakers into a listening space and are acoustically perceived by a listener.

Multichannel audio system having audio channel compensation

An audio loudness control system includes a loudness adjuster module configured to adjust a level of an audio signal based on parameters associated with the audio signal. The parameters may include identification of a source of the audio signal. The parameters associated with the audio signal may be provided to the loudness adjuster in a message associated with the audio signal. The audio loudness control system may also include a database in communication with the loudness adjuster module. The loudness adjuster module may be configured to extract loudness related settings or parameters from the database that are associated with a source of the audio signal.

Audio loudness control system

A sound processing system receives an audio input signal that includes at least two different input channels of audio content. The sound processing system dissects the audio input signal to separate sources of audible sound included in the audio input signal into sound source vectors. Separation of the sources of audible sound into sound source vectors may be based on a perceived location of each of the sources of audible sound within a listener perceived sound stage. The sound source vectors may represent spatial slices across the listener perceived sound stage that may be individually and independently processed with the sound processing system. Following processing, the sound source vectors may be selectively assembled to form an audio output signal having output channels used to drive respective loudspeakers. Since the sources of audible sound are separated and independent, the audible sound sources may be included on any one or more of the output channels.

System for spatial extraction of audio signals

A device for processing a signal includes a processing module used to process data in a domain. The device includes domain conversion hardware and memory, both separate from the processing module. The memory receives and sends data represented in a first domain to the domain conversion hardware. The domain conversion hardware converts the first domain data into the second domain data that may be used or processed by the processing module. The second domain data is then accessed and used by the processing module for processing.

System for converting a signal

An audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.

System for modifying an acoustic space with audio source content

A system for processing compressed audio includes a signal enhancer module configured to generate one or more signal treatments. The one or more signal treatments may be generated by the signal enhancer module based on analysis of the incoming audio signal. Alternatively, or in addition, characteristics of the incoming audio signal may be provided to the signal enhancer module for use in generating the one or more signal treatments. The one or more signal treatments may be added to the audio signals.

A reverberant characteristic of an acoustic space is superimposed on an audio signal that is received by an apparatus. The apparatus decomposes the audio signal into an estimated original dry signal component and an estimated reverberant characteristic of the acoustic space. Estimation of the original dry signal component and the reverberant characteristic of the acoustic space is based on determination of an estimated impulse response of the acoustic space from the received audio signal. Once the audio signal is decomposed, the estimated original dry signal component and the estimated reverberant characteristic of the acoustic space may be independently modified by the apparatus. The modified or unmodified estimated original dry signal component and estimated reverberant characteristic of the acoustic space may be combined by the apparatus to produce one or more adjusted frequency spectra.

System for extraction of reverberant content of an audio signal

An apparatus for selectively extracting components of an input signal comprises a plurality of microphones for detecting an audio signal in a plurality of spaced locations. A plurality of signal samplers sample each spaced version of the audio signal. A microphone pattern processor derives a reference signal, a null signal and a core signal from the sampled signals. An adjustment frequency spectrum generator coupled to the null and reference signals generates an adjustment frequency spectrum signal that is applied to the core signal. An audio signal regenerator provides an output audio signal from the adjusted core signal. The apparatus can be used in a wireless device to pick up a speaker's voice in a noisy environment.

System for selectively extracting components of an audio input signal

A method and apparatus for extracting and changing the reverberant content of an input signal is provided. The method of estimating an impulse response for a reverberant system includes sampling a signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter. The method of creating a multi-channel signal using a signal with fewer channels from a reverberant system includes sampling the signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter, applying an inverse of the estimated impulse response to the signal to derive at least one of a direct signal component and a reverberant signal component, and recombining the direct signal component and the reverberant signal component to create a multi-channel signal.

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Kanata, Canada

Gilbert Arthur Joseph Soulodre