A sound processing device program that enables a sound signal adapted to an intended use to be output is provided. The sound processing device includes a signal processing part that processes a sound signal picked up by a microphone, and generates a recording sound signal to be recorded in a recording device, and generates an amplification sound signal different from the recording sound signal to be output from a speaker. The sound processing device can be applied to, for example, a sound amplification system that performs off-microphone sound amplification.

Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

A computer-implemented method and system for performing testing of audio equipment in a conference room, the method executed by one or more processors, comprising: (a) commissioning the conference room with a set of audio video equipment, the set of audio equipment comprising one or more loudspeakers, one or more microphones, and audio signal processing equipment that includes at least an acoustic echo cancellation function; (b) determining an initial audio performance level in the conference room, and storing the initial audio performance level (IAPL); (c) determining that sound quality testing of the audio equipment in the conference room should be performed; (d) disabling the acoustic echo cancellation function in the audio equipment of the conference room such that an output from each of the one or more loudspeakers is not removed from a respective microphone output signal; (e) generating an electrical stimulus test signal and transmitting it to the one or more loudspeakers in the audio equipment of the conference room; (f) receiving an acoustic audio stimulus test signal generated by each of the one or more loudspeakers from each of the one or more microphones, and analyzing each of the received acoustic audio stimulus test signals to generate a current audio performance level (CAPL); (g) comparing the CAPL to the IAPL; and (h) determining if the audio equipment in the conference room passes or fails the sound quality test based on the comparison of the CAPL to the IAPL.

An auditory guidance system is installed along a predefined pathway between an initial location and a target location. The auditory guidance system comprises a plurality of nodes arranged at intervals along the predefined pathway. A first node of the plurality of nodes is a closest node to the initial location and a last node of the plurality of nodes is a closest node to the target location. Each node of the plurality of nodes comprises two directional sound generators. Each of the two directional sound generators are configured to emit sound in a predetermined direction. The two directional sound generators within each node are pointing in opposite directions along the predefined pathway.

An object is to provide a public address device that effectively prevents occurrence of howling without a drop of the gain. A public address device 100 includes: a loudspeaker 1 that generates a reproduced sound on the basis of a loudspeaker driving signal u(n); a microphone 2 that collects the reproduced sound and an input sound v(n) to generate a microphone-collected-sound signal y(n); a first filter 301 that generates, on the basis of the loudspeaker driving signal u(n), a pseudo echo signal e(n); an echo-cancelling unit 302 that obtains a difference between the microphone-collected-sound signal y(n) and the pseudo echo signal e(n) to generate an echo-cancelled signal d(n); a second filter 311 that whitens the input sound v(n) included in the loudspeaker driving signal u(n); a third filter 312 that whitens the input sound v(n) included in the microphone-collected-sound signal y(n); a first adaptive filter 313 that uses, as a reference signal, an output signal output from the second filter 311, and uses, as a desired signal, an output signal output from the third filter 311, and estimates a propagation characteristic Wo from the loudspeaker 1 to the microphone 2; a unit that repeatedly updates a filter coefficient W of the first filter 301 on the basis of a filter coefficient W identified by the first adaptive filter 313; and a frequency shifting unit 32 that performs a frequency shift on the echo-cancelled signal d(n) to generate the loudspeaker driving signal u(n).

Utterances spoken or sung by a first person can be received, in real time. The detected utterances can be compared to at least a stored sample of utterances spoken or sung by the first person. Based on the comparing, audio of the utterances spoken or sung by the first person can be isolated from a background noise. A volume of the utterances spoken or sung by a first person relative to the background noise can be determined. A key indicator that indicates the volume of the detected utterances spoken or sung by the first person relative to the background noise can be generated. Based on the key indicator, information indicating the volume of the detected utterances spoken or sung by the first person relative to the background noise can be communicated.

A method (S100) for optimizing speech pickup in a speakerphone system, wherein the speakerphone system comprises a microphone system placed in a specific configuration, wherein the method comprising receiving (S10) acoustic input signals (12) by the microphone system, processing (S20) said acoustic input signals (12) by using an algorithm (100) for focusing and steering a selected target sound signal towards a desired direction, and transmitting (S30) an output signal (13) based on said processing.

A mobile electronic device (device) operating with other devices form a group for wireless coordinated playout of audio media content. The processor performs operations that determine a common time sync shared with the other devices, and determine a timing of occurrence of a sound transient sensed relative to the common time sync. The operations receive timing reports from the other devices, where each of the timing reports indicates a timing of occurrence of the sound transient sensed at a respective one of the other devices relative to the common time sync. The sound transient sensed by the devices is generated at a preferred listening location. The operations coordinate timing of playout of audio media content by the group of devices responsive to comparison of the timing of occurrence of the sound transient sensed by the microphone of the device and the timing of occurrences indicated by the timing reports.

The invention relates to a sound diffusion system embedded in a railway vehicle and comprising: a plurality of groups of speakers distributed in the cars, each group of speakers being located in a respective diffusion zone of the railway vehicle; a control device of the speakers configured to broadcast generic sound signals via the different groups of speakers; and at least one reception device, each reception device being associated with a single group of speakers and being able to receive a control signal from a control device outside the railway vehicle, the control device being configured, upon reception of a control signal by one of the reception devices, to broadcast a specific sound signal solely via the associated group of speakers.