H03G9/025

MULTI-BAND LIMITER SYSTEM AND METHOD FOR AVOIDING CLIPPING DISTORTION OF ACTIVE SPEAKER
20230170866 · 2023-06-01 ·

A limiter system for an active speaker may include at least one lowpass filter configured to receive an input signal and output a signal lower than a crossover frequency, at least one highpass filter, configured to receive an input signal and output a signal higher than the crossover frequency, a first allpass filter configured to adjust the phase of the signal lower than the crossover frequency, a second allpass filter configured to adjust the phase of the signal higher than the crossover frequency, a first limiter, configured to receive and limit the signal from the first allpass filter, a second limiter, configured to receive and limit the signal from the second allpass filter, and a mixer, configured to mix the signal lower from the first limiter and the signal from the second limiter.

Audio processing device, system, use and method in which one of a plurality of coding schemes for distributing pulses to an electrode array is selected based on characteristics of incoming sound

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

Speech intelligibility enhancing system
11265660 · 2022-03-01 · ·

A speech intelligibility enhancing system for difficult acoustical conditions is disclosed, the speech intelligibility enhancing system comprising at least one ear plug (201) for insertion in an ear canal (218) of a person, the at least one ear plug being arranged with an ear canal facing portion (401) and an environment facing portion (402), and the at least one ear plug comprising an acoustically attenuating path (214; 214, 213) comprising a vent (214) coupling said environment facing portion (402) with said ear canal facing portion (401); and an electroacoustic path (202, 204, 209; 202, 203, 204, 208, 209, 210, 211, 212) comprising a microphone (202) at said environment facing portion (402), a variable gain (204) and a loudspeaker (209) at said ear canal facing portion (401); wherein said acoustically attenuating path (214; 214, 213) is arranged with a transfer function from said environment facing portion (402) to said ear canal facing portion (401) having a low pass characteristic having a low pass cut¬off frequency and said low pass characteristic attenuating sound by a nominal attenuation (Go) for frequencies below said cut-off frequency.

Method of controlling diaphragm excursion of electrodynamic loudspeakers

The present application relates in one aspect to a method of controlling diaphragm excursion of an electrodynamic loudspeaker. The method comprises dividing the audio input signal into at least a low-frequency band signal and a high-frequency band signal by a band-splitting network and applying the low-frequency band signal to a diaphragm excursion estimator. The instantaneous diaphragm excursion is determined based on the low-frequency band signal. The determined instantaneous diaphragm excursion is compared with an excursion limit criterion. The low-frequency band signal is limited based on a result of the comparison between the instantaneous diaphragm excursion and the excursion limit criterion to produce a limited low-frequency band signal which is combined with the high-frequency band signal to produce an excursion limited audio signal.

Psychoacoustics for improved audio reproduction and speaker protection

Psychoacoustic models may be applied to audio signals being reproduced by an audio speaker to reduce input signal energy applied to the audio transducer. Using the psychoacoustic model, the input signal energy may be reduced in a manner that has little or no discernible effect on the quality of the audio being reproduced by the transducer. The psychoacoustic model selects energy to be reduced from the audio signal based, in part, on human auditory perceptions and/or speaker reproduction capability. The modification of energy levels in audio signals may be used to provide speaker protection functionality. For example, modified audio signals produced through the allocation of compensation coefficients may reduce excursion and displacement in a speaker; control temperature in a speaker; and/or reduce power in a speaker.

MANAGEMENT OF BROADCAST AUDIO LOUDNESS
20170302241 · 2017-10-19 · ·

To control loudness during a junction between different types of broadcast content, such as a junction between programme and commercial or promotional content, representative loudness values for content respectively before (P) and after (C) the junction are received from a playout automation system. A time-varying gain control is applied before and after the junction in order to smooth loudness around the junction. The audio gain is smoothly increased prior to the junction to a gain (P+C)/2P times higher than the original gain value. Then, the gain is reduced shortly before the junction to a value (P+C)/2C times lower than the original gain value. After the junction, the gain is returned smoothly to the original value.

DYNAMIC SUPPRESSION OF NON-LINEAR DISTORTION

Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.

Speech processing using identified phoneme clases and ambient noise
09779721 · 2017-10-03 · ·

A wireless communication device is disclosed. The wireless communication device includes a processor, a memory, a transceiver configured to receive an audio signal, a codec to decode the audio signal, a dynamic range controller and a phoneme processor. The phoneme processor is configured to extract acoustic cues from each frame of the decoded audio signal and to identify a phoneme class in the each frame. The dynamic range controller is configured to apply dynamic range compression on the each frame based on the identified phoneme class.

Techniques for setting volume level within a tree of cascaded volume controls with variating operating delays

Techniques are disclosed for synchronizing gain adjustments across a cascaded network of audio gain stages having variant operating delays. In particular, a delay-synchronized volume adjustment system configured in accordance with an embodiment of the present disclosure includes a controller operatively coupled to the cascaded network of audio and configured to apply gain adjustments in a synchronized manner that accounts for operating delays that are inherent to each gain stage. In an embodiment, the controller synchronously adjusts each gain stage relative to a corresponding operating delay such that gain adjustments fully propagate at substantially a same point in time within a given acceptable tolerance, and thus, eliminates or otherwise mitigates perceivable volume shifts when mixing audio from two or more audio sources.

System for configuration and status reporting of audio processing in TV sets
11245375 · 2022-02-08 · ·

Systems are disclosed including a television (TV) set having included audio system. The TV set permits control over various functions, at least including audio volume, via a remote control. When the viewer activates the remote volume control, a graphic appears indicating the state of the volume control and, optionally the mute status. The graphic can be presented on the TV screen. In alternate embodiments, the graphic may be presented on a display of the remote, or even some other location, e.g., a different remote control. If a mute button is provided on the remote control, when the viewer activates the mute button, a graphic appears indicating the state of muting and, optionally, the volume control status. The TV set also offers control over various aspects of the audio system, including settings which go beyond volume up/down, generally through some sort of menu system.