Abstract: A first mobile communication device that includes a first microphone, a first speaker, and a first delay unit. The first microphone is configured to (i) receive, during a conference call, a first user first microphone signal from a first user, and (ii) output a first microphone digital signal to the first delay unit. The first user first microphone signal represents audio content outputted by the first user. The first delay unit is configured to delay, by a delay period, the first microphone digital signal to provide a delayed first user first device digital signal. The first mobile communication device is configured to output, to a mixer, the delayed first user first device digital signal. The delay period is determined based on measurements executed by at least one mobile communication device out of the first mobile communication device, a second mobile communication device and a third mobile communication device.

Abstract: A first mobile communication device that includes a first microphone, a first speaker, and a first delay unit. The first microphone is configured to (i) receive, during a conference call, a first user first microphone signal from a first user, and (ii) output a first microphone digital signal to the first delay unit. The first user first microphone signal represents audio content outputted by the first user. The first delay unit is configured to delay, by a delay period, the first microphone digital signal to provide a delayed first user first device digital signal. The first mobile communication device is configured to output, to a mixer, the delayed first user first device digital signal. The delay period is determined based on measurements executed by at least one mobile communication device out of the first mobile communication device, a second mobile communication device and a third mobile communication device.

Abstract: Noise suppression in audio signals received by a voice activated device is supported with motion sensors. Motion sensors may provide an indication of movement or motion information, such as the linear or rotational displacement of the voice activated device. A noise reduction unit, which is held in idle mode to conserve power, may be activated in response to an indication of movement. When activated, the noise reduction unit may adapt to environmental noise from the new location or orientation, and switch back to idle mode. When speech is subsequently detected in audio signals, the noise reduction unit has already adapted to the noise and accordingly may reduce noise in the audio signal with no delay when activated. Additionally or alternatively, motion information may be used by the noise reduction unit to quickly adapt to noise in the audio signal.

Abstract: A first mobile communication device that includes a first microphone, a first speaker, and a first delay unit. The first microphone is configured to (i) receive, during a conference call, a first user first microphone signal from a first user, and (ii) output a first microphone digital signal to the first delay unit. The first user first microphone signal represents audio content outputted by the first user. The first delay unit is configured to delay, by a delay period, the first microphone digital signal to provide a delayed first user first device digital signal. The first mobile communication device is configured to output, to a mixer, the delayed first user first device digital signal. The delay period is determined based on measurements executed by at least one mobile communication device out of the first mobile communication device, a second mobile communication device and a third mobile communication device.

Abstract: A method for speech level adaptation, the method includes: (A) Receiving, by a voice interactive intelligent personal assistant, multiple input audio signals that includes first and second groups of input audio signals. (B) Buffering the multiple input audio signals. (C) Searching for a voice trigger in the first group of input audio signals. When finding the voice trigger then (D) Determining a linear gain factor to be applied on the second group of buffered audio signals, (E) Applying the linear gain factor on the second group of buffered audio signals to provide the output audio signals; and (F) applying a speech recognition process on the output audio signals to detect an audio command embedded in the output audio signals.

Abstract: A mobile communication device that comprises a microphone, a speaker, an input output port and one or more modules; wherein the microphone is configured to receive, during a conference call, (a) an acoustic signal from a first person, and (b) a cross echo from a speaker of another mobile communication device that participates in the conference call, the cross echo is generated as a result of the acoustic signal from the first person; and wherein the one or more modules are configured to cancel the cross echo based, at least in part, of the acoustic signal of the first person.

Abstract: Provided is a wearable audio input/output device. The wearable audio input/output device is provided with a frame, a first audio collection unit, a second audio collection unit, and a first speaker. The frame can be mounted to a user. The first audio collection unit is provided on the frame. The second audio collection unit is provided on the frame. The second audio collection unit is arranged such that the distance between the first audio collection unit and mouth of the user is less than the distance between the second audio collection unit and the mouth of the user. The first speaker is provided on the frame between the first audio collection unit and the second audio collection unit.

Abstract: Provided is a wearable audio input/output device. The wearable audio input/output device is provided with a frame, a first audio collection unit, a second audio collection unit, and a first speaker. The frame can be mounted to a user. The first audio collection unit is provided on the frame. The second audio collection unit is provided on the frame. The second audio collection unit is arranged such that the distance between the first audio collection unit and mouth of the user is less than the distance between the second audio collection unit and the mouth of the user. The first speaker is provided on the frame between the first audio collection unit and the second audio collection unit.

Abstract: Methods and systems are provided for enhancing listening intelligibility in electronic devices. A vibration sensor may be used to generate feedback corresponding to vibrations caused by the outputting of the acoustic signals, and the feedback may be used in adjusting the listening intelligibility stage. In some instances, a microphone may be used to obtain audio input corresponding to ambient noise affecting intelligibility of audio outputted, as acoustic signals, via a speaker, to a user. The audio input may be used to control a listening intelligibility stage applied to audio content when the acoustic signals are generated for outputting by the speaker. In particular, the listening intelligibility stage may comprise application of dynamic time-scale modifications.

Abstract: A method for detecting whispered speech of a user of a mobile computerized device, the method comprises: detecting, by the mobile computerized device, whispered speech context; and attempting to detect, by the mobile computerized device and by using whispered speech detection parameters, the whispered speech; and wherein at least one whispered speech detection parameter of the whispered speech detection parameters differs from at least one corresponding non-whispered speech parameter that is used for detecting non-whispered speech.

Abstract: Methods and systems are provided for enhancing speech intelligibility in electronic devices. During outputting of acoustic signal via an electronic device, measurement of forces applied by user of the electronic device against the device (or enclosure thereof) may be obtained. The force measurements may be used to assess and/or estimate the listening intelligibility experienced by the user. Further, the force measurements may be used to control or adjust a listening intelligibility stage applied during generation and/or processing of the acoustic signals that are outputted via the electronic device. In some instances, an audio input, corresponding to ambient noise affecting intelligibility, may be obtained, and may be used to control or assist in controlling the listening intelligibility stage.

Abstract: A method for detecting whispered speech of a user of a mobile computerized device, the method comprises: detecting, by the mobile computerized device, whispered speech context; and attempting to detect, by the mobile computerized device and by using whispered speech detection parameters, the whispered speech; and wherein at least one whispered speech detection parameter of the whispered speech detection parameters differs from at least one corresponding non-whispered speech parameter that is used for detecting non-whispered speech.

Abstract: Methods and systems are provided for enhancing speech intelligibility in electronic devices. During outputting of acoustic signal via an electronic device, measurement of forces applied by user of the electronic device against the device (or enclosure thereof) may be obtained. The force measurements may be used to assess and/or estimate the listening intelligibility experienced by the user. Further, the force measurements may be used to control or adjust a listening intelligibility stage applied during generation and/or processing of the acoustic signals that are outputted via the electronic device. In some instances, an audio input, corresponding to ambient noise affecting intelligibility, may be obtained, and may be used to control or assist in controlling the listening intelligibility stage.

Abstract: Methods and systems are provided for enhancing listening intelligibility in electronic devices. A vibration sensor may be used to generate feedback corresponding to vibrations caused by the outputting of the acoustic signals, and the feedback may be used in adjusting the listening intelligibility stage. In some instances, a microphone may be used to obtain audio input corresponding to ambient noise affecting intelligibility of audio outputted, as acoustic signals, via a speaker, to a user. The audio input may be used to control a listening intelligibility stage applied to audio content when the acoustic signals are generated for outputting by the speaker. In particular, the listening intelligibility stage may comprise application of dynamic time-scale modifications.

Abstract: Methods and systems are provided for acoustic echo cancellation in electronic devices. The echo cancellation may comprise applying, as a first step, echo cancellation filtering to an acoustic input obtained via an acoustic input element (e.g., microphone), and applying, as a second step, echo suppression to the acoustic input, wherein the echo suppression comprises suppressing residual echo in the acoustic input. The echo cancellation filtering may comprise identifying and/or filtering out echo components, both linear and nonlinear, in the acoustic input, with the echo components corresponding to an echo signal caused by an acoustic output outputted via the acoustic output element (e.g., speaker). A sensor signal, generated by a vibration sensor that detects vibrations in the electronic device including vibrations caused by the outputting of the acoustic output, may be used as reference signal in the echo cancellation filtering and/or the echo suppression.

Abstract: A method and apparatus for filtering a reference signal by a series of taps having coefficients, comprising dividing the taps into at least one layer of a plurality of groups, each group comprising a plurality of taps, as a first layer, forming a plurality of coefficients arranged in at least one group, as at least one subsequent layer and forwarding multiply-accumulate results of each group in each layer sequentially to the at least one group in the at least one subsequent layer to generate an output signal, and adjusting the coefficients based on an error signal formed as a difference between the output signal and a received input signal related to the reference signal and an interim error signal as a difference between the input signal and a summation of the multiply-accumulate results of at least one layer.

Abstract: A method for Comfort Noise Generation (CNG) comprising the steps of recording information of Background Noise (BGN); generating white noise samples; and generating Comfort Noise (CN) by applying coefficients that are extracted from said information of BGN on White Noise (WN) samples.

Abstract: The method and system use a speaker available in an audio end device when not in use as a microphone. The speaker is utilized for enhancing the sound quality received by the person or device at the near end, i.e., the user of the device, or the speaker at the far end, i.e., the speaker with whom the near-end speaker is communicating. Exemplary applications are shown for such utilization, including single or dual microphone near-end automatic listening enhancement, acoustic echo cancellation or suppression with or without double talk detection, and microphone array noise reduction.

Abstract: A method of maintaining signal convergence of a transmitter and a receiver of an adaptive differential pulse code modulated (ADPCM) communication system during discontinuous transmission, by generating synchronized comfort noise frame in the transmitter and in the receiver during quiet periods and mutually updating the receiver's decoder and encoder on their states.