An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.

An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.

Audio processing with audio transparency can include receiving a user content audio signal and receiving a microphone signal. The microphone signal can contain sensed sound of a user environment. Strength of the sensed sound can be increased based on strength of the user content audio signal, to reduce a masking of the sensed sound during playback. The sensed sound and the user content audio signal can be combined in a composite output audio signal used to drive a speaker. Other aspects are also described and claimed.

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

Novel methods and systems for creating and using user profiles for dialog boost and sound equalizer adjustment to compensate for various ambient sound situations. For creating profiles when not at the ambient sound, a synthesized/pre-recorded ambient noise can be mixed with the media to simulate the noise conditions.

Novel methods and systems for creating and using user profiles for dialog boost and sound equalizer adjustment to compensate for various ambient sound situations. For creating profiles when not at the ambient sound, a synthesized/pre-recorded ambient noise can be mixed with the media to simulate the noise conditions.

Disclosed herein are method, system, and computer program product embodiments for performing the continuous tuning of received audio input from an earpiece or microphone, wherein the audio input is independently altered in the frequency domain for output to an earpiece worn by a user as well as separately for an additional audio output to an external connected speaker, for an optimal experience.

A signal processing technique the noise suppressing performance of which is more improved than conventional one is provided. A first component extraction unit 14 extracts a non-stationary component ^φ.sub.S.sup.(A)(ω, τ) derived from a sound coming from a target area and a stationary component ^φ.sub.S.sup.(B)(ω, τ) derived from an incoherent noise from a power spectrum density ^φ.sub.S(ω, τ) of the target area through processing of time average. A second component extraction unit 15 extracts a non-stationary component ^φ.sub.N.sup.(A)(ω, τ) derived from an interference noise and a stationary component ^φ.sub.N.sup.(B)(ω, τ) derived from an incoherent noise from a power spectrum density ^φ.sub.N(ω, τ) of a noise area.