Systems and methods for suppressing noise and detecting voice input in a multi-channel audio signal captured by a plurality of microphones include (i) capturing a first audio signal via a first microphone and a second audio signal via a second microphone, wherein the first and second audio signals respectively comprises first and second noise content from a noise source; (ii) identifying the first noise content in the first audio signal; (iii) using the identified first noise content to determine an estimated noise content captured by the plurality of microphones; (iv) using the estimated noise content to suppress the first and second noise content in the first and second audio signals; (v) combining the suppressed first and second audio signals into a third audio signal; and (vi) determining that the third audio signal includes a voice input comprising a wake word.

In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

A remote microphone signal is obtained from a remote device, and a local microphone signal from a local device. A difference between strength of the remote microphone signal and strength of the local microphone signal is determined. An output audio signal is produced to drive a speaker in the local device. If the difference is greater than a threshold, then the local and remote microphone signals are applied to two input channels, respectively, of a two channel noise suppressor which produces the output audio signal, but if the difference is less than the threshold after a certain delay since the difference was greater than the threshold, then only the remote microphone signal is applied to a single input of a single channel noise suppressor which produces the output audio signal. Other aspects are also described and claimed.

A method and system for improving the intelligibility of an audio signal emitted from plural audio output devices in a space can help mitigate systems that have low intelligibility at installation due to reverberation effects; it can provide a quick post installation check of a system prior to a full-fledged STIPA (Speech Transmission Index for Public Address Systems) or equivalent test; and it allows for a quick verification if changes to the acoustical environment are expected to have major effects on STIPA tests.

A conversation support system supports conversation of passengers in an automobile. Seats of at least two or more rows are placed in the automobile, and the conversation support system includes a speech switching control part configured to designate a desired sound pickup and reproduce area based on designation by a first passenger sitting in a first seat, a first target speech emphasizing part configured to output a signal obtained by emphasizing speech emitted from the designated sound pickup and reproduce area to a loud speaker corresponding to the first seat, and a second target speech emphasizing part configured to output the signal obtained by emphasizing speech, which is collected with a microphone corresponding to the first seat and which is emitted from the first seat, to a loud speaker corresponding to the sound pickup and reproduce area.

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 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.

A computer-implemented method and system for performing testing of audio equipment in a conference room, the method executed by one or more processors, comprising: (a) commissioning the conference room with a set of audio video equipment, the set of audio equipment comprising one or more loudspeakers, one or more microphones, and audio signal processing equipment that includes at least an acoustic echo cancellation function; (b) determining an initial audio performance level in the conference room, and storing the initial audio performance level (IAPL); (c) determining that sound quality testing of the audio equipment in the conference room should be performed; (d) disabling the acoustic echo cancellation function in the audio equipment of the conference room such that an output from each of the one or more loudspeakers is not removed from a respective microphone output signal; (e) generating an electrical stimulus test signal and transmitting it to the one or more loudspeakers in the audio equipment of the conference room; (f) receiving an acoustic audio stimulus test signal generated by each of the one or more loudspeakers from each of the one or more microphones, and analyzing each of the received acoustic audio stimulus test signals to generate a current audio performance level (CAPL); (g) comparing the CAPL to the IAPL; and (h) determining if the audio equipment in the conference room passes or fails the sound quality test based on the comparison of the CAPL to the IAPL.

A remote microphone signal is obtained from a remote device, and a local microphone signal from a local device. A difference between strength of the remote microphone signal and strength of the local microphone signal is determined. An output audio signal is produced to drive a speaker in the local device. If the difference is greater than a threshold, then the local and remote microphone signals are applied to two input channels, respectively, of a two channel noise suppressor which produces the output audio signal, but if the difference is less than the threshold after a certain delay since the difference was greater than the threshold, then only the remote microphone signal is applied to a single input of a single channel noise suppressor which produces the output audio signal. Other aspects are also described and claimed.

A system and method are provided herein for dynamically adjusting delay in an audio distribution system, the method comprising: determining what audio processing devices comprise each of a plurality of audio data channels, wherein each of the plurality of audio data channels comprises a path from a digital audio receiving device to a back end audio playing device; obtaining digital audio processing delays for each of the audio processing devices for each of the audio data channels; determining which of the audio data channels has the greatest delay in processing and transmitting digital audio data signals (digital audio data processing delay) prior to broadcasting the digital audio data signals as acoustic audio signals; determining a difference between the audio data channel with the greatest digital audio data processing delay and each of the remaining audio data channels (delay difference per channel); and adding a delay to each digital audio word in each different audio channel that substantially equalizes the digital audio data processing delay between each of the different audio data channels.