A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter.

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.

The disclosure provides a method for processing audio data, an apparatus for processing audio data and an electronic device, and relates to a field of natural language processing technologies, and in particular to the fields of audio technology, digital conference and speech transliteration technologies. The method includes: receiving at least two pieces of audio data sent by at least one audio matrix, in which the audio data is collected by a microphone array and sent to the audio matrix; converting all the audio data into corresponding text data; and sending the audio data and the text data corresponding to the audio data.

A conference terminal and a feedback suppression method are provided. In the method, a transmitting sound signal is divided into sub sound signals of multiple frequency bands. Different sub sound signals correspond to different frequency bands. An interfered frequency band corresponding to the howling interference is detected according to the sub sound signals of those frequency bands. The power of the sub sound signal of the interfered frequency band increases along with time. The interfered frequency band is affected by the howling interference. An interference direction is determined according to multiple input sound signals received by the microphone array and merely pass through the interfered frequency band. A sound from the interference direction leads to the howling interference. A beam pattern of the microphone array is determined according to the interference direction. The gain of the beam pattern in the interference direction is reduced.

Systems and methods are disclosed for networked audio automixing using array microphones and an aggregator unit that participate in making a common gating decision to determine which channels to gate on and off. Through the use of such a network of array microphones having the capability to generate submix audio signals and reduced bandwidth metrics, as well as AEC processing capability, array microphone lobe selection can be enhanced while maximizing signal-to-noise ratio, increasing intelligibility, and increasing user satisfaction.

A microphone module comprises a housing, an audio bus, and a first plurality of microphones in communication with the audio bus. The microphone module further comprises a module processor in communication with the first plurality of microphones and the audio bus. The module processor is configured to detect the presence of an array processor in communication with the audio bus, detect the presence of a second microphone module in communication with the audio bus, and configure the audio bus to pass audio signals from both the first plurality of microphones and the second microphone module to the array processor.

Systems, apparatus, and methods for processing audio signals associated with conferencing devices communicatively connected in a daisy-chain configuration using local connection ports included on each device are provided. One method involving a first conferencing device comprises receiving auxiliary mixed microphone signal(s) from at least one other conferencing device via at least one local connection port, each auxiliary signal comprising a mix of microphone signals captured by the at least one other conferencing device; determining a gain adjustment value for each auxiliary mixed microphone signal based on a daisy-chain position of the at least one other conferencing device relative to the position of the first conferencing device; adjusting a gain value for each auxiliary mixed microphone signal based on the corresponding gain adjustment value; generating a loudspeaker output signal from the gain-adjusted auxiliary mixed microphone signal(s); and providing the loudspeaker signal to the loudspeaker of the first conferencing device.

A conference terminal and a feedback suppression method are provided. In the method, a transmitting sound signal is divided into sub sound signals of multiple frequency bands. Different sub sound signals correspond to different frequency bands. An interfered frequency band corresponding to the howling interference is detected according to the sub sound signals of those frequency bands. The power of the sub sound signal of the interfered frequency band increases along with time. The interfered frequency band is affected by the howling interference. An interference direction is determined according to multiple input sound signals received by the microphone array and merely pass through the interfered frequency band. A sound from the interference direction leads to the howling interference. A beam pattern of the microphone array is determined according to the interference direction. The gain of the beam pattern in the interference direction is reduced.

Systems and methods are provided for identifying and displaying whiteboard text and/or an active speaker in a video-based presentation, e.g., a video conference. Video images of an environment including a whiteboard may be captured by a video camera system. The video images may be analyzed to detect at least one text-containing area in the environment. Each text-containing area may be analyzed to determine whether it is an area of a whiteboard. When a text-containing area is identified as a whiteboard area, an area of view including the text-containing whiteboard area may be selected for display, e.g., a subset of the full frame captured by the video system. A video feed from the video camera system may be controlled to display the selected area of view at a client device, to provide a useful view of the whiteboard text and/or a speaking person located near the whiteboard text.

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.