An acoustic characteristic measurement device includes a speaker configured to output an acoustic signal in which a watermark signal is embedded; a microphone configured to perform sound collection with respect to the acoustic signal output from the speaker in a predetermined listening position; and an acoustic characteristic measurement unit configured to determine an acoustic characteristic of an acoustic space from the speaker to the listening position, on the basis of the watermark signal embedded in the acoustic signal which is subjected to the sound collection by the microphone.

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

A system and method for performing perceptually-transparent RTF estimation for a two-channel system with low estimation errors. The system includes two loudspeakers for the left and right channels positioned in a room and a single microphone, positioned in the vicinity of a listener in the room. The system performs spectral splitting using a filter bank and a complementary filter bank, to obtain separation in frequency between the two channels. Calibration signals being uncorrelated across the two channels are , generated using the original audio signals and the constructed filters. Then, the calibration signals are played back by the loudspeakers, instead of playback of the original audio signals. At the end, the system records the playback of the calibration signals by the microphone and simultaneously estimates the two-channel RTF using the recorded played back calibration signals.

A multichannel microphone-based reverberation time estimation method and device which use a deep neural network (DNN) are disclosed. A multichannel microphone-based reverberation time estimation method using a DNN, according to one embodiment, comprises the steps of: receiving a voice signal through a multichannel microphone; deriving a feature vector including spatial information by using the inputted voice signal; and estimating the degree of reverberation by applying the feature vector to the DNN.

A multichannel microphone-based reverberation time estimation method and device which use a deep neural network (DNN) are disclosed. A multichannel microphone-based reverberation time estimation method using a DNN, according to one embodiment, comprises the steps of: receiving a voice signal through a multichannel microphone; deriving a feature vector including spatial information by using the inputted voice signal; and estimating the degree of reverberation by applying the feature vector to the DNN.

In embodiments of the present invention improved capabilities are described for a computer-based method for modifying a sound source, including accessing, by an acoustic processing facility including at least one processor, a multi-dimensional sound profile of an acoustic environment, wherein the multi-dimensional sound profile comprises a time-based sound reflection sequence for the acoustic environment; and modifying, by the acoustic processing facility, a sound source data based on the multi-dimensional sound profile to generate a modified sound data.

In embodiments of the present invention improved capabilities are described for a computer-based method for modifying a sound source, including accessing, by an acoustic processing facility including at least one processor, a multi-dimensional sound profile of an acoustic environment, wherein the multi-dimensional sound profile comprises a time-based sound reflection sequence for the acoustic environment; and modifying, by the acoustic processing facility, a sound source data based on the multi-dimensional sound profile to generate a modified sound data.

Provided are a directional acoustic sensor and a method of detecting a distance from a sound source using the same. The directional acoustic sensor may include a plurality of resonators arranged in different directions; and a processor configured to calculate a time difference between a first signal that is received by the plurality of resonators directly from a sound source and a second signal that is received by the plurality of resonators from the sound source after being reflected on a wall surface around the sound source, and determine a distance between the sound source and the directional acoustic sensor based on the time difference.

At least one acoustic parameter of an environment is obtained by emitting a sound signal emitted, receiving a sound signal received, estimating a sampling frequency shift and of a temporal shift between the emission and reception devices on the basis of impulse responses obtained on the basis of portions of the sound signal received and of corresponding portions of the sound signal emitted, obtaining a corrected sound signal received by applying to the sound signal received a sampling frequency shift correction and a temporal shift correction on the basis of the estimating, determining the impulse response of the environment on the basis of portions of the corrected sound signal received and of corresponding portions of the sound signal emitted, and obtaining acoustic parameters of the environment on the basis of the impulse response.