Systems and methods for identifying audio events in one or more audio streams include the use of a cascade audio spotting system (such as a cascade keyword spotting system (KWS)) to reduce power consumption while maintaining a desired performance. An example cascade audio spotting system may include a first module and a high-power subsystem. The first module is to receive an audio stream from one or more audio streams, process the audio stream to detect a first target sound activity in the audio stream, and provide a first signal in response to detecting the first target sound activity in the audio stream. The high-power subsystem is to (in response to the first signal being provided by the first module) receive the one or more audio streams and process the one or more audio streams to detect a second target sound activity in the one or more audio streams.

Systems and methods for identifying audio events in one or more audio streams include the use of a cascade audio spotting system (such as a cascade keyword spotting system (KWS)) to reduce power consumption while maintaining a desired performance. An example cascade audio spotting system may include a first module and a high-power subsystem. The first module is to receive an audio stream from one or more audio streams, process the audio stream to detect a first target sound activity in the audio stream, and provide a first signal in response to detecting the first target sound activity in the audio stream. The high-power subsystem is to (in response to the first signal being provided by the first module) receive the one or more audio streams and process the one or more audio streams to detect a second target sound activity in the one or more audio streams.

In some embodiments, an audio system can be monitored by determining a sound suppression mode, and sampling information representative of an input signal and an output signal resulting from processing of the input signal in the sound suppression mode. The monitoring method can further include providing a display representative of the sampled information.

In some embodiments, an audio system can be monitored by determining a sound suppression mode, and sampling information representative of an input signal and an output signal resulting from processing of the input signal in the sound suppression mode. The monitoring method can further include providing a display representative of the sampled information.

A method, system, and apparatus for noise cancelation is disclosed, which may be used in a wireless unit (WU). The WU may include a processor, a memory, a user interface, internal microphones and internal speakers. A removably connected headset may include microphones and speakers. The WU may receive a first ambient noise from headset microphone(s), which may generate a first signal based on the first ambient noise. The WU may receive a second ambient noise at internal microphone(s), which may generate a second signal based on the second ambient noise. The WU may calculate an estimate of ambient noise based on the first and second signals, calculate a signal for noise cancellation based on the estimate, cancel estimated ambient noise from an audio output signal based on an application of the signal for noise cancellation, and send the audio output signal to speakers of the headset or of the WU.

A method, system, and apparatus for noise cancelation is disclosed, which may be used in a wireless unit (WU). The WU may include a processor, a memory, a user interface, internal microphones and internal speakers. A removably connected headset may include microphones and speakers. The WU may receive a first ambient noise from headset microphone(s), which may generate a first signal based on the first ambient noise. The WU may receive a second ambient noise at internal microphone(s), which may generate a second signal based on the second ambient noise. The WU may calculate an estimate of ambient noise based on the first and second signals, calculate a signal for noise cancellation based on the estimate, cancel estimated ambient noise from an audio output signal based on an application of the signal for noise cancellation, and send the audio output signal to speakers of the headset or of the WU.

Utilizing an adaptive multichannel technique to mitigate reverberation present in received audio signals, prior to providing corresponding audio data to one or more additional component(s), such as automatic speech recognition (ASR) components. Implementations disclosed herein are “adaptive”, in that they utilize a filter, in the reverberation mitigation, that is online, causal and varies depending on characteristics of the input. Implementations disclosed herein are “multichannel”, in that a corresponding audio signal is received from each of multiple audio transducers (also referred to herein as “microphones”) of a client device, and the multiple audio signals (e.g., frequency domain representations thereof) are utilized in updating of the filter—and dereverberation occurs for audio data corresponding to each of the audio signals (e.g., frequency domain representations thereof) prior to the audio data being provided to ASR component(s) and/or other component(s).

Utilizing an adaptive multichannel technique to mitigate reverberation present in received audio signals, prior to providing corresponding audio data to one or more additional component(s), such as automatic speech recognition (ASR) components. Implementations disclosed herein are “adaptive”, in that they utilize a filter, in the reverberation mitigation, that is online, causal and varies depending on characteristics of the input. Implementations disclosed herein are “multichannel”, in that a corresponding audio signal is received from each of multiple audio transducers (also referred to herein as “microphones”) of a client device, and the multiple audio signals (e.g., frequency domain representations thereof) are utilized in updating of the filter—and dereverberation occurs for audio data corresponding to each of the audio signals (e.g., frequency domain representations thereof) prior to the audio data being provided to ASR component(s) and/or other component(s).

An audio system includes: a speaker; a microphone that generates a microphone signal based on sound output from the speaker; a mixer module configured to generate a mixed signal by mixing the microphone signal with an audio signal; a filter module configured to filter the mixed signal to produce a filtered signal and to apply the filtered signal to the speaker; and a detector module configured to determine a howling frequency in the microphone signal attributable to sound output from the speaker, where the filter module is configured to decrease a magnitude of the filtered signal at the howling frequency.

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.