Digital processing of speech signals

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

An electronic device and a control method thereof includes an infrared module which receives an infrared control signal having a predetermined section-specific waveform from an infrared transmitter, a switching unit which turns on/off the infrared module, and a controller which controls the switching unit to alternately turn on and off the infrared module in a predetermined cycle for recognizing the predetermined section-specific waveform of the infrared control signal. Thus, electric power consumed by an infrared module in a standby mode may be reduced while receiving a signal of an infrared transmitter.

Electronic device and control method thereof

Multiple aspects of systems and methods for voice control and related features and functionality for various embodiments of media playback devices, networked microphone devices, microphone-equipped media playback devices, and speaker-equipped networked microphone devices are disclosed and described herein, including but not limited to designating and managing default networked devices, audio response playback, room-corrected voice detection, content mixing, music service selection, metadata exchange between networked playback systems and networked microphone systems, handling loss of pairing between networked devices, actions based on user identification, and other voice control of networked devices.

Voice control of a media playback system

A method, an apparatus, a system, and instructions stored in a non-transitory computer-readable medium to instruct a processing system to carry out the method. The method includes applying corrective filters directly in a portable media device to correct, e.g., equalize for the overall system comprising the portable media device and the playback system to which it is attached. Also a method of determining the corrective filters by playing back one or more calibration signals on the playback system while recording the resulting sound field on the portable media device.

Audio system equalization for portable media playback devices

The purpose of room impulse response (RIR) shortening and reshaping is usually to improve the intelligibility of the received signal by prefiltering the source signal before it is played with a loudspeaker in a closed room. In an alternative, but mathematically equivalent setting, one may aim to postfilter a recorded microphone signal to remove audible echoes. While least-squares methods have mainly been used for the design of shortening/reshaping filters for RIRs until now, we propose to use the infinity- or p-norm as optimization criteria. In our method, design errors will be uniformly distributed over the entire temporal range of the shortened/reshaped global impulse response. In addition, the psychoacoustic property of masking effects is considered during the filter design, which makes it possible to significantly reduce the filter length, compared to standard approaches, without affecting the perceived performance.

/pdf/room-impulse-response-shortening-reshaping-with-infinity-and-42isw3699r.pdf

Room Impulse Response Shortening/Reshaping With Infinity- and $p$ -Norm Optimization

An apparatus for generating an audio output signal to simulate a recording of a virtual microphone at a configurable virtual position in an environment is provided. The apparatus comprises a sound events position estimator and an information computation module (120). The sound events position estimator (110) is adapted to estimate a sound source position indicating a position of a sound source in the environment, wherein the sound events position estimator (110) is adapted to estimate the sound source position based on a first direction information provided by a first real spatial microphone being located at a first real microphone position in the environment, and based on a second direction information provided by a second real spatial microphone being located at a second real microphone position in the environment. The information computation module (120) is adapted to generate the audio output signal based on a first recorded audio input signal, based on the first real microphone position, based on the virtual position of the virtual microphone, and based on the sound source position.

Sound acquisition via the extraction of geometrical information from direction of arrival estimates

An embodiment of an apparatus (200) for computing control information for a suppression filter (210) for filtering a second audio signal to suppress an echo based on a first audio signal includes a computation means (220) having a value determination means (230) for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computation means (220) further includes a mean value determination means (250) for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computation means (220) further includes a modification means (260) for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computation means (220) ' further includes a control information computation means (270) for computing the control information for the suppression filter (210) on the basis of the at least one modified energy-related value.

Apparatus and method for computing control information for an echo suppression filter and apparatus and method for computing a delay value

An audio format transcoder (100) for transcoding an input audio signal, the input audio signal having at least two directional audio components. The audio format transcoder (100) comprising a converter (110) for converting the input audio signal into a converted signal, the converted signal having a converted signal representation and a converted signal direction of arrival. The audio format transcoder (100) further comprises a position provider (120) for providing at least two spatial positions of at least two spatial audio sources and a processor (130) for processing the converted signal representation based on the at least two spatial positions to obtain at least two separated audio source measures.

Audio format transcoder

This paper addresses the usability of channel shortening equalizers known for data transmission systems for the equalization of acoustic systems. In multicarrier systems, equalization filters are used to shorten the channel’s effective length to the size of a cyclic prefix or the guard interval. In most applications the equalizer succeeds the channel. In acoustic systems, an equalizer is placed in front of a playback loudspeaker to generate a desired impulse response for the concatenation of the equalizer, a loudspeaker, a room impulse response, and a reference microphone. In this paper, we show that shaping the desired impulse response to a shorter reverberation time is more appropriate for acoustical systems than trying to exactly truncate it to a maximum length. Investigations are carried out using a multi-loudspeaker-multi-microphone system.

/pdf/multi-channel-room-impulse-response-shaping-a-study-20empn52xc.pdf

Markus Kallinger

Papers

Room Impulse Response Shortening/Reshaping With Infinity- and $p$ -Norm Optimization

Sound acquisition via the extraction of geometrical information from direction of arrival estimates

Apparatus and method for computing control information for an echo suppression filter and apparatus and method for computing a delay value

Audio format transcoder

Multi-Channel Room Impulse Response Shaping - A Study