A method, apparatus and computer program including: obtaining a spatial audio signal from a plurality of microphones; dividing the obtained spatial audio signal into at least a first component and a second component; applying a first audio signal optimizing system to the first component and applying a second audio signal optimizing system to the second component; and enabling a signal including the optimized components to be provided to a speaker for rendering.

In an approach to adaptive sound masking, one or more computer processors analyze a surrounding of one or more users and stores in a database. The one or more computer processors receive a request from the one or more users for adaptive sound masking. The one or more computer processors analyzes a surrounding environment associated with the one or more users and storing a first information associated with the surrounding environment in a database. The one or more computer processors generate a cognitive sound mask base on the first information. The one or more computer processors produce a sound cone based on the cognitive sound mask and directing the sound cone at a distracting sound. The one or more computer processors adapt the sound cone based on changes to the surrounding environment.

A set of signal measures is sent, wherein each signal measure in the set of signal measures corresponds to a respective audio signal received by a playback device in a media playback system and is processed based on a first set of audio processing algorithms. A plurality of signal measures is identified in the set of signal measures. Audio signals corresponding to the identified plurality of signal measures are processed by one or more devices in the media playback system to improve a signal measure of each of the audio signals. The audio signals are processed based on a second set of audio processing algorithms. The processed audio signals are combined into a combined audio signal.

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.

An electronic device and a controlling method therefor are provided. The electronic device includes a speaker, a microphone, and an audio processor configured to adjust a size of a signal of a predetermined frequency band in an input audio signal, determine whether to adjust the size of the audio signal wherein the size of the frequency band was adjusted based on the output level of the speaker, and output the audio signal processed based on whether the adjustment was performed through the speaker. The audio processor is further configured to perform acoustic echo cancellation for the sound signal using the input audio signal or the audio signal of which size was adjusted based on whether the adjustment was performed, based on receiving a sound signal including the output audio signal through the microphone.

A real-time audio signal processing system includes an audio signal processor configured to process audio signals using a modified generalized eigenvalue (GEV) beamforming technique to generate an enhanced target audio output signal. The digital signal processor includes a sub-band decomposition circuitry configured to decompose the audio signal into sub-band frames in the frequency domain and a target activity detector configured to detect whether a target audio is present in the sub-band frames. Based on information related to the sub-band frames and the determination of whether the target audio is present in the sub-band frames, the digital signal processor is configured to use the modified GEV technique to estimate the relative transfer function (RTF) of the target audio source, and generate a filter based on the estimated RTF. The filter may then be applied to the audio signals to generate the enhanced audio output signal.

A method, apparatus and computer program, the method comprising: including enabling an output of an audio mixer to be rendered for a user where the user is located within a sound space, wherein at least one input channel is provided to the audio mixer and the at least one input channel receives a plurality of microphone output signals obtained by a plurality of microphones recording the sound space; determining that a first microphone records one or more sound objects within the sound space; and in response to the determining, enabling one or more of the plurality of microphone output signals to be, at least partially, removed from the at least one input channel to the audio mixer.

Method and system for maintaining accuracy of voice recognition are described herein. The audio system reproducing sound using a loudspeaker array that is housed in a loudspeaker cabinet may selection from a number of sound rendering modes and changing the selected sound rendering mode based on the current playback volume set on the audio system. The sound rendering modes include at least one of: a number of free space modes and a number of complex modes. Other aspects are also described and claimed.

A system for enhanced processing of voice-based signals in a voice-controllable sound-generating system (SGS) is provided. An SGS audio source may communicate electronic SGS audio signals to both (a) one or more speakers, which output corresponding SGS sound waves and (b) an audio countering system. A microphone may detect sound waves and output corresponding audio signals including: (a) distorted SGS audio signals corresponding with SGS sound waves and (b) additional audio signals originated from other sources, e.g., including voice-based commands. The audio countering system may (a) receive the electronic SGS audio signals from the SGS audio source; receive signals from the microphone representing the microphone-detected sound waves, and (c) use the electronic SGS audio signals received to cancel or counter the distorted SGS audio signals included in the microphone-received audio signals, to thereby enhance any voice-based commands included in the received audio signals.

Systems, apparatus, and methods for processing audio signals associated with conferencing devices communicatively connected in a daisy-chain configuration using local connection ports included on each device are provided. One method involving a first conferencing device comprises receiving auxiliary mixed microphone signal(s) from at least one other conferencing device via at least one local connection port, each auxiliary signal comprising a mix of microphone signals captured by the at least one other conferencing device; determining a gain adjustment value for each auxiliary mixed microphone signal based on a daisy-chain position of the at least one other conferencing device relative to the position of the first conferencing device; adjusting a gain value for each auxiliary mixed microphone signal based on the corresponding gain adjustment value; generating a loudspeaker output signal from the gain-adjusted auxiliary mixed microphone signal(s); and providing the loudspeaker signal to the loudspeaker of the first conferencing device.