A web-based virtual meeting platform with echo suppression functionality that is hardware agnostic. The platform is configured in a client-server distributed architecture, wherein a plurality of user devices are connected to a server. A baseline for audio generated by a user device on the client-side of the client-server platform is established. Upon detection of a deviation from the baseline, including electro acoustic feedback and ambient noise, suppressive gain is applied to the audio input at the client-side of the client-server architecture. The suppressed audio stream is then transmitted to a cloud server and distributed to a plurality of user devices connected to the cloud service. The platform is accessible via an internet browser and is non-discriminatory as to the hardware of the device used to access the platform. Further, the platform does not distribute data in a peer-to-peer architecture.

A computer-implemented method dynamically mutes irrelevant sources of noise. The method includes identifying one or more sources of a noise in a vicinity of a listening device, where the listening device is associated with a user and the listening device includes a noise canceling function. The method also includes determining, for the user, a context, where the context represents a subject of a conversation related to the user. The method further includes calculating, for each of the one or more sources of noise, a relevance score. The method includes muting, by the listening device, each of the sources of noise where the associated relevance score is below a relevance threshold.

An audio transmission is received by a participant computing device. The participant computing device is one of a plurality of participant computing devices of a participant cohort that are co-located. Matched filters are generated based on the transmission that are configured to predict at least a portion of audio caused by playback of the transmitted audio signal. Each of the matched filters includes coefficients. An audio signal is captured with an audio capture device. The captured audio signal corresponds to audio produced by playback of the transmitted audio signal with audio output devices of devices of the participant cohort. A matched filter is identified that most accurately predicts the audio signal. A delay estimate is generated based on a predictive contribution of one of the coefficients of the matched filter.

A headset for voice communication is disclosed, the headset comprising at least one earphone having a speaker and one or more microphones. The headset is configured to be operated in a first mode in which an electronic noise cancelling circuit is configured to receive ambient audio via at least a first of the one or more microphones to implement an active noise cancelling function and to provide a noise cancelling audio signal to the speaker, and in a second mode in which ambient audio is provided as a hear-through audio signal to the speaker. The headset for voice communication is configured to detect whether a call is ongoing, and to provide a call signal in response to the detection. The headset comprises the electronic noise cancelling circuit, a voice activity detection unit configured to indicate when a user speaks, a switching element configured to switch the headset between operating in the first mode and operating in the second mode, wherein, when the headset is operated in the first mode and the call signal indicates that the user is not in a call, the switching element is configured to switch the headset from operating in the first mode to operating in the second mode when the voice activity detection unit indicates that the user speaks.

Disclosed herein is an audio lighting device, which includes a wireless module, an audio source control circuit, a light source control circuit, a speaker, and a light source board. The wireless module receives a wireless signal and outputs an audio source control signal. The audio source control circuit receives the audio source control signal and outputs a first analog signal. The light source control circuit receives the audio source control signal and outputs a second analog signal. The speaker receives the first analog signal and outputs audio signal. The light source board receives the second analog signal to control the brightness of the outputted light, in which the brightness of the outputted light is proportional to the volume of the outputted audio signal.

An audio device may be configured to produce output audio and to capture input audio for speech recognition. In some cases, a second device may also be used to capture input audio to improve isolation of input audio with respect to the output audio. In addition, acoustic echo cancellation (AEC) may be used to remove components of output audio from input signals of the first and second devices. AEC may be implemented by an adaptive filter based on dynamically optimized filter coefficients. The filter coefficients may be analyzed to detect situations in which the first and second devices are too close to each other, and the user may then be prompted to increase the distance between the two devices.

An audio system includes a plurality of near-field speakers arranged in a listening area. A plurality of cross-talk cancellation filters are coupled to the speakers. The speakers and the filters are arranged to provide first and second listening zones in the listening area such that audio from the first listening zone is cancelled in the second listening zone and vice versa. The system also includes at least one audio source providing audio content. Volume-based equalization circuitry receives an audio signal representing audio content for the first listening zone from the audio source and controls a volume adjustment applied to the audio signal to control a volume of audio in the first listening zone. The circuitry limits attenuation or amplification of a first frequency portion of the audio signal when a volume setting differential corresponding to a difference between volume settings for the first and second zones exceeds a predetermined value.

A signal processor for determining a plurality of drive signals for driving a plurality of loudspeakers to cancel a reverberation effect in a listening area, wherein the signal processor is configured to determine from one or more measured audio signals a plurality of measured physical coefficients in a basis of physical sound functions, such that a sum of the physical sound functions, weighted with the plurality of measured physical coefficients approximates the one or more measured audio signals, wherein at least half of the plurality of measured physical coefficients are zero, determine a residual error between the plurality of measured physical coefficients and a plurality of desired physical coefficients, estimate a transfer function describing a transformation from the plurality of desired physical coefficients to the plurality of measured physical coefficients, based on the determined residual error, and update the plurality of drive signals based on the estimated transfer function.

Multi-function apparatuses and methods associated with augmenting an analog audio signal are disclosed herein. In embodiments, a multi-function apparatus for performing a plurality of functions may include a microphone to receive a propagated analog audio signal and ambient noise; a receiver to receive a digitally streamed version of the analog audio signal; a harmonizer, that includes a plurality of processors, to generate a digital adjusted version of the analog signal including noise canceling signal to cancel some or all of the ambient noise, based on the propagated analog audio signal, the ambient noise, and the digitally streamed version of the analog audio signal; and a digital-to-analog converter to convert the digital adjusted version of the analog signal to an analog adjusted version of the analog signal. The analog adjusted version of the analog signal may then be outputted to augment the propagated analog audio signal. Other embodiments may be disclosed or claimed.

The disclosed technology contemplates use of a communication device in public without a user's voice being overheard by the public. In certain situations, public use of a communication device, even when the conversation is private, may not be appropriate. So, a preferred embodiment of the disclosed technology involves a system of enabling or disabling a communication device. Some embodiments may include computer hardware and memory with voice-recognition software that is configured to receive a user's spoken voice during a test conversation, analyze the voice for its hertz spectrums evident in the test conversation, and subsequently employ the data during active noise cancelation. Suitably, this voice recognized noise cancelation or active noise cancelation via voice algorithm can be used to actively noise cancel a user's voice in a manner that is specifically tailored to the user by directing the sound source to provide an inverted signal that is 180 degrees out of phase with the user's actual voice. In one embodiment, the voice algorithm allows a digital CPU to anticipate the manner, including tonality, rhythm, and cadence, of a user's speaking and predict an appropriately inverted signal so that noise is substantially canceled.