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.

An audio encoding device and an audio decoding device are described herein. The audio encoding device may examine a set of audio channels/channel groups representing a piece of sound program content and produce a set of ducking values to associate with one of the channels/channel groups. During playback of the piece of sound program content, the ducking values may be applied to all other channels/channel groups. Application of these ducking values may cause (1) the reduction in dynamic range of ducked channels/channel groups and/or (2) movement of channels/channel groups in the sound field. This ducking may improve intelligibility of audio in the non-ducked channel/channel group. For instance, a narration channel/channel group may be more clearly heard by listeners through the use of selective ducking of other channels/channel groups during playback.

The invention relates to a hearing aid comprising a first microphone configured to receive a first acoustic signal and to convert the first acoustic signal to a first electrical audio signal, a speaker configured to emit an acoustic output signal into an ear of a user of the hearing aid device, a first analog-to-digital converter for converting the first electrical audio signal into a first time-domain input signal, a first input unit comprising a first analysis filter bank which is configured to convert the first time-domain input signal to a number NI,1 of first input frequency bands wherein the number NI,1 of first input frequency bands is determined by said first analysis filter bank, a first frequency band bundling and allocation unit which is configured to bundle adjacent first input frequency bands and to allocate first frequency bands to be processed to a number NP,1 of first processing channels, a memory unit which is configured to store data indicating which of the first NI,1 input frequency bands are subject to a likelihood of feedback that is above a threshold, a signal processing unit is configured to process the first frequency bands to be processed in the number NP,1 of first processing channels, and wherein the number NP,1 of first processing channels is smaller than the number NI,1 of first input frequency bands, and wherein the first frequency band bundling and allocation unit is configured to generate a first bundling and allocation scheme which determines the bundling of the first NI,1 input frequency bands and the allocation of the first frequency bands to be processed to the first NP,1 processing channels wherein said first bundling and allocation scheme depends on the likelihood of feedback to occur in at least one of the first NI,1 input frequency bands.

Various embodiments describe techniques for switching microphones in a multiple microphone system. The techniques incorporate sampling audio signals from multiple microphones, determining a microphone that has the greatest incoming amplitude during the analysis window, and switching the microphone to that greatest amplitude microphone. The transition point for switching microphones may be determined when either the amplitude of the incoming signal is within an error bound of zero or at a zero-crossing in the input amplitude stream.

Described herein is an audio device with a microphone which may adapt the audio output volume of a speaker by either increasing or decreasing output volume based on an audio input volume from a user and a distance from the user to the audio device. The audio device may also adapt its output volume to lower the audio output based on detecting one or more interruptions including occupancy and acoustic sounds.

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, computer program product, and computer system for receiving, by a computing device, a speech signal from a speaker via a plurality of microphone zones. A temporal cue based confidence may be determined for at least a portion of the plurality of microphone zones. A power cue based confidence may be determined for at least a portion of the plurality of microphone zones. A microphone zone of the plurality of microphone zones from which to use an output signal of the speaker may be identified based upon, at least in part, a combination of the temporal cue based confidence and the power cue based confidence.

A gaming headset system for use in a game event. A headset (HS) has a mouth microphone (MM), and a headphone with over-the-car car-cups (EC) each having inside a loudspeaker transducer (LT) and a feed-back microphone (FB). A feed-forward microphone (FF) is placed on an exterior part of the car-cups (EC). An ambient noise and speech suppression system (ANSS) serves to attenuate ambient noise and suppress intelligibility of ambient speech sound reaching the gamer (GM) in response to input signals indicative of sound captured by one or more of: 1) the mouth microphone (MM). 2) the feed-forward microphones (FF), 3) the feed-back microphones (FB), and 4) one or more sources (GSP_I. AUDM) located external to the car-cups (EC). The ambient noise and speech suppression algorithm comprises an active noise cancellation algorithm part (ANC) and a masking noise signal algorithm part (MSK) which in combination provides an effective active attenuation of noise and speech reaching the gamer and ensuring that even weak speech sounds will not be understood by the gamer. E.g. the ANSS algorithm may use microphones near the game speaker or game commentator and/or the audience for the ANC algorithm part and/or the masking noise signal algorithm part.

Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

A speech intelligibility system. Embodiments comprise a talker unit, a listener unit and an earpiece. The talker unit includes a microphone to receive audible speech content and to produce electrical signals representative of the speech content, and a transmitter coupled to the microphone to produce wireless transmissions containing the speech content. The listener unit includes a receiver to receive the wireless transmissions and to produce electrical signals representative of the speech content. At least one of the talker unit and the listener unit includes an amplifier to amplify spectral components of the speech content within a frequency range having a lower end between about 800 Hz and 1,700 Hz and an upper end between about 7,000 Hz and 11,000 Hz. The earpiece is coupled to the listener unit and includes a speaker to produce audible speech content having the amplified spectral components and a tube to direct the audible speech content from the speaker toward a user's ear canal.