The present invention is a system that detects the location and head orientation of a live performer on stage in front of an audience and adjusts the individual elements of the mix either by changing them at the listener's position or by feeding the location and orientation information back to the monitor mixer or a combination of both. The adjustments would include left/right panning, relative levels, equalization, transient response, reverberation levels, panning, and time delay, as well as other possible modifications to the signal so that the performer senses that he/she is actually listening to the various instruments, vocals, acoustic space and audience, in their real or virtual locations.

A computing device may maintain a database of representative spectral characteristics. The computing device may also receive particular spectral data associated with a particular playback environment corresponding to the particular playback device. Based on the particular spectral data, the computing device may identify one of the representative spectral characteristics from the database that substantially matches the particular spectral data, and then identify, in the database, an audio processing algorithm based on a) the identified representative spectral characteristic and b) at least one characteristic of the particular playback device. The computing device may then transmit, to the particular playback device, data indicating the identified audio processing algorithm.

Utterances spoken or sung by a first person can be received, in real time, from a mobile communication device. A location of the mobile communication device can be determined to be in an area designated as a quiet zone. A key indicator that indicates at least one characteristic of the detected utterances spoken or sung by the first person can be generated. Based, at least in part, on the key indicator, a determination can be made that the first person is speaking or singing too loudly in the area designated as the quiet zone. Responsive to determining that the first person is speaking or singing too loudly in the area designated as the quiet zone, feedback indicating that the first person is speaking or singing too loudly in the area designated as the quiet zone can be communicated to the mobile communication device.

A computing device may maintain a database of representative spectral characteristics. The computing device may also receive particular spectral data associated with a particular playback environment corresponding to the particular playback device. Based on the particular spectral data, the computing device may identify one of the representative spectral characteristics from the database that substantially matches the particular spectral data, and then identify, in the database, an audio processing algorithm based on a) the identified representative spectral characteristic and b) at least one characteristic of the particular playback device. The computing device may then transmit, to the particular playback device, data indicating the identified audio processing algorithm.

A method 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 acoustic input signals by the microphone system, processing said acoustic input signals by using an algorithm for focusing and steering a selected target sound signal towards a desired direction, and transmitting an output signal based on said processing.

Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).

A method for setting a filter frequency response includes: dividing a target frequency band into multiple first partial bands at a frequency corresponding to an intersection between a first line representative of a target spectrum envelope and a second line representative of a first spectrum envelope derived from smoothing a measured frequency spectrum; and setting a frequency response of a filter corresponding to a first partial band specified by an operator from among the multiple first partial bands, in accordance with a setting operation for setting an adjustment amount performed by the operator with respect to the specified first partial band.

VoIP telephone systems and methods are provided for use in a noisy listening environment. The systems and methods provide high quality voice broadcast into the noisy listening environment through independent voice channels which makes it easier for the human brain to discern broadcast words from the background noise.

Provided are a de-reverberation control method and apparatus for a device equipped with a microphone. The method includes: reverberation parameters which indicate, at respective moments, reverberation levels of a room environment where the device is located are acquired from an audio signal played by the device; and a de-reverberation mode adopted by the device is dynamically adjusted according to the reverberation levels indicated by the reverberation parameters at different moments and preset correspondences between reverberation levels and de-reverberation modes. By adopting a dynamic de-reverberation mode, the method and the apparatus disclosed herein significantly improve the rate of the recognition of a device for the voice of the user.

An exemplary microphone capture simulation system accesses a captured set of audio signals captured by a plurality of directional microphones disposed at a plurality of locations on a perimeter of a capture zone of a real-world scene. The system identifies a location within the capture zone that corresponds to a virtual location at which a user is virtually located within a virtual reality space that is based on the capture zone. Based on the captured set of audio signals and the identified location, the system generates a simulated set of audio signals representative of a simulation of a full-sphere multi-capsule microphone capture at the identified location. The system processes the simulated set of audio signals to form a renderable set of audio signals configured to be rendered to simulate full-sphere sound for the virtual location while the user is virtually located at the virtual location.