A non-transitory computer readable medium storing a program causing a computer to execute a process is provided, the computer being included in a first information processing apparatus, and the process includes: (a) receiving notifications via a wireless communication channel from one or more information processing apparatuses other than the first information processing apparatus, each notification including conference identification information and information regarding a state of a sound-pick-up unit or a state of a sound-output unit of one of the one or more information processing apparatuses, the conference identification information identifying a teleconference to which the one of the one or more information processing apparatuses is to connect; (b) identifying an information processing apparatus as a nearby apparatus, the information processing apparatus being selected from the one or more information processing apparatuses that have transmitted the notifications received by the first information processing apparatus, the information processing apparatus having transmitted a notification including conference identification information that is the same as conference identification information of a teleconference to which the first information processing apparatus is to connect, the information processing apparatus being located at a distance from the first information processing apparatus, the distance being shorter than a predetermined distance; and (c) determining whether a howl is likely to occur by comparing a state of a sound-pick-up unit of the nearby apparatus that has been identified and a state of a sound-output unit of the first information processing apparatus or comparing a state of a sound-output unit of the nearby apparatus that has been identified and a state of a sound-pick-up unit of the first information processing apparatus.

An echo cancelling system includes a data transmitter circuit and an echo canceller circuit. The data transmitter circuit is configured to receive a first transmitted signal. The first transmitted signal has a first sampling rate. The echo canceller circuit is configured to generate a second transmitted signal according to the first transmitted signal. The second transmitted signal has a second sampling rate. The second sampling rate is greater than the first sampling rate. The echo canceller circuit is further configured to generate an echo cancelling signal according to the second transmitted signal. The data transmitter circuit is further configured to generate an output signal according to a received signal and the echo cancelling signal.

In some examples, an apparatus comprises: a housing; an internal speaker housed within the housing; an internal microphone housed within the housing; an interface; and a controller configured to: receive, using the internal microphone, ingress audio signals; output, using the internal speaker, first egress audio signals at a first power level when the internal microphone receives the ingress audio signals; detect that an external speaker is connected to the interface; based on detecting that the external speaker is connected to the interface, disable the internal microphone; and output, using the external speaker, second egress audio signals when the internal microphone receives the ingress audio signals, the second egress audio signals being output at a second power level higher than the first power level.

A method for selecting an audio signal for alignment to compensate for the latency that is introduced by content being sent, such as from an end device, over a network to a cloud based or other computing environment located remote from the end unit. Audio that is processed in the cloud is also being sent back to the end device. Selection may be accomplished using a loop back method, a Time Stamp (TS) method or a Ping method. The Ping method allows incoming and outgoing audio signals to be selected and processed in the cloud.

A sound generation system and related method include a user interface device and a processing device to obtain a specification of a three-dimensional space, obtain one or more sound tracks each comprising a corresponding sound signal associated with a corresponding sound source, present, in a user interface, representations representing one or more listeners and the one or more sound sources corresponding to the one or more sound signals in the three-dimensional space, responsive to a configuration of the locations of the one or more listeners or the locations of the one or more sound sources in the three-dimensional space in the user interface, determine filters based on the configuration and pre-determined locations of one or more loudspeakers, and apply the filters to the one or more sound signals to generate filtered sound signals for driving the one or more loudspeakers.

Even when a speech is small, the speech is allowed to be detected and an echo is allowed to be appropriately suppressed. Whenever a sample point of a reception signal transmitted through a receiving signal path that transmits a signal to a speaker is acquired, an optimal mask is sequentially generated or selected from base masks as one or a plurality of masks generated based on a learning signal based on a reception signal acquired within a predetermined period before a time point at which the sample point was acquired. Whenever the optimal mask is selected, whether a double-talk state is present is sequentially detected based on a result of comparing an input signal with the optimal mask. When detecting that a speech is not input to a microphone and the reception signal includes a speech, a process of suppressing an echo is sequentially performed on the input signal.

A first noise level, which is an estimated value of a magnitude of a noise component included in a first sound collection signal obtained from a first microphone which collects sound emitted from a first sound collection and amplification position is obtained, a second noise level, which is an estimated value of a magnitude of a noise component included in a second sound collection signal obtained from a second microphone which collects sound emitted from a second sound collection and amplification position is obtained, a ratio of a reproduced noisy sound level, which is an estimated value of a magnitude of noise at a position of a passenger at the second sound collection and amplification position in a case where the first noise level is reproduced from a second speaker placed at the second sound collection and amplification position, with respect to a second noisy sound level, which is an estimated value of a magnitude of noise corresponding to the second noise level at the position of the passenger at the second sound collection and amplification position is obtained, and a noise suppression amount is obtained so that a product of this ratio and the noise suppression amount becomes a constant set in advance.

A system that performs early reflections filtering to suppress early reflections and improve sound source localization (SSL). During music playback and/or when a device is placed in a corner, acoustic reflections from nearby surfaces get boosted due to constructive interference, negatively impacting SSL and other processing of the device. To suppress these early reflections, the device uses an Early Reflections Filter (ERF) that makes use of Linear Prediction Coding (LPC), which is already being performed during speech processing. For example, the device generates raw audio signals using multi-channel LPC coefficients and then uses single-channel LPC coefficients for each raw audio signal in order to generate a filter that estimates the reflections. The device then uses this filter to suppress the early reflections and generate filtered audio signals, thus resulting in better audio processing and better overall device performance.

A playback device is configured to: produce a first channel audio output of a first channel of audio content; produce a second channel audio output of a second channel of the audio content; receive captured audio content comprising (i) a first portion corresponding to the first channel audio output, (ii) a second portion corresponding to the second channel audio output, and (iii) a third portion corresponding to a voice command, wherein the captured audio content has a first signal-to-noise ratio; determine a set of signal components from at least one of the first channel or the second channel of the audio content; perform acoustic echo cancellation on a subset of signal components; determine an acoustic echo cancellation output; and apply the acoustic echo cancellation output to the captured audio content and thereby increase the first signal-to-noise ratio to a second signal-to-noise ratio that is greater than the first signal-to-noise ratio.

Systems and methods are disclosed for audio group identification for conferencing. For example, methods may include joining a conference call using a network interface; accessing an audio signal that has been captured using a microphone; detecting a control signal in the audio signal; and, responsive to detection of the control signal, invoking modification of an audio path of the conference call.