At+t Corp.

Lucent Technologies Inc.

Jonathan B Allen

Donald J Youtkus

Mark R Sydorenko

Donald Kyle

James David Johnston

James Edward West

Deepen Sinha

John C Baumhauer, Jr

Sean Matthew Dorward

John C Songrady

Joshua M Shendelman

L David

Bryce R Daines

A monolithic second order gradient (SOG) microphone structure employs acoustic transmission lines wherein the acoustic phase delay along each of the acoustic transmission lines is in direct proportion to the length of each of the acoustic transmission lines and, where this is effected by the use of an acoustic impedance element placed within each acoustic transmission line that has an acoustic impedance related to the acoustic impedance of the associated acoustic transmission line. In one embodiment, the acoustic impedance element has a specific acoustic impedance substantially matched to the specific acoustic characteristic resistance of the acoustic transmission line. Various embodiments may utilize acoustic or electrical subtraction of the signals in the acoustic transmission lines to realize the desired directional sound pickup.

Directional microphone

A 'loudness balance' procedure is used to determine the amount of dynamic range for syllabic compression loss in an individual's cochlea. In particular, an individual is asked to adjust a set of tone volumes, which are played to them, until the tone volumes sound equally loud. Based on the adjustments performed by the individual, a compression ratio is determined for that individual over at least one frequency band. The determined compression ratio is then subsequently used by a telecommunications system to provide an individual-specific sound enhancement in a telephone call. As a result, the individual-specific sound enhancement particularly compensates for an amount of hearing loss unique to that individual.

Technique for determining a compression ratio for use in processing

A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to form noise masking thresholds for use in the compression of audio signals. These measures are employed in an illustrative subband, analysis-by-synthesis framework. In accordance with the illustrative embodiment, provisional encodings of the audio signal are performed to determine the encoding which achieves a loudness differential, between the original and coded audio signal, which is less than (but not too far below) the loudness uncertainty.

Tonality for perceptual audio compression based on loudness uncertainty

A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to determine the degree to which audio signals are 'tone-like' (or 'noise-like'). The degree of tone-likeness, referred to as 'tonality,' is used to determine masking thresholds for use in the compression of audio signals. Tonality, computed in accordance with the present invention, is used in conventional and new arrangements to achieve compression of audio signals.

Perceptual audio compression based on loudness uncertainty

An automated method for modifying a speech signal communicated over a telephone network by applying a gain factor in a telephone set before any side tone is added thereto. The gain factor is a function of a received signal indicative of the background noise at the location of the telephone set. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands. The individual modified subband signals may then be combined to form the resultant modified speech signal.

Background noise compensation in a telephone set

Methods and apparatus are disclosed for improving the clarity and intelligibility of incoming speech in a telephone connection. Improved sound quality is provided by automatically selecting, or enabling the selection of, an appropriate sound enhancement for a particular individual at a given telephone number. Sound enhancements applied to calls originating from or terminating at a given telephone station are determined on a call-by-call basis. The sound enhancement set at the beginning of a telephone call can be changed mid-call to a different sound enhancement. Thus, different occupants of a household may select their own sound enhancement, or may select no enhancement at all.

Method for customer selection of telephone sound enhancement

An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands. The individual modified subband signals may then be combined to form the resultant modified speech signal.

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Mountainside, NJ, United States of America

Jonathan B Allen