A dynamic optical microphone system may include an acoustic microphone that receives an audio signal and a laser microphone that transmits a laser beam and receives optical feedback from a human struck by the laser beam. The system may include a depth sensor that determines a distance to the human and a camera that tracks human faces. A processor may be communicatively coupled to the acoustic microphone, laser microphone, depth sensor, camera, and a memory storing computer executable instructions. The processor may determine a direction to a human, direct the laser beam at a voice box of the human, determine a distance to the human using the depth sensor, adjust an intensity of the laser beam based on the distance, receive optical feedback and isolate a voice signal through the optical feedback from background noise in the audio signal.

Systems and methods for adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

A workspace assembly includes a plurality of sound sensors including at least first and second sound sensors located in first and second spaces within a facility, respectively, at least one communication device located within one of the first space and the second space, and a processor in communication with the sound sensors and the communication device, the processor adapted to compare at least some characteristic of the sounds in at least a subset of the plurality of spaces to identify a particular space from which the sound emanates and to compare at least some characteristic of the sound emanating from the particular space to at least one threshold characteristic value and to provide a signal via the at least one communication device based on the results of the comparison.

A method, system and computer program product includes detecting a volume level for audio input of a first user in a multi-user conference call, and automatically adjusting a volume level for a second user receiving audio output of the first user based on at least one of preferences of the second user, historic data between the first and the second user, and geographic characteristics of the audio input of the first user.

Systems and methods for adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using suprathreshold and/or threshold hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

A method and system to limit voice-filter bandwidth based for a device model based on observed sound quality trend. A system monitors sound quality of voice-call communications coming from various devices and detects based on the monitoring that the sound quality of voice-call communications coming from devices of a particular model tends to be threshold low even when the devices have threshold high quality wireless coverage. In response the system then imposes a policy for devices of that model to restrict the voice-filter bandwidth that they applysuch as by limiting the devices to use a voice-filter bandwidth that is no wider than a designated bandwidth.

Systems and methods for adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using suprathreshold and/or threshold hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone.

A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone.

Disclosed is a method for managing acoustic feedback in real-time audio communications in a communications system, the method comprising determining, by means of a detection module, whether a first communication device is in loudspeaker mode, whether the first communication device is in real-time audio communications with a second communication, and whether the first communication device and the second communication device are in a same acoustic space. Upon determining that this is the case a request signal for requesting one or more measures against acoustic feedback is provided to a mitigation module. Further disclosed are a device and a system configured to perform the method, a non-transitory computer-readable medium, an encoder and a decoder.