The present invention provides a processing method based on an OFDM-TDMA two-way service and a communications device. The method includes assigning timeslots in a TDMA frame structure to multiple users in a TDMA manner, so that each user has a different timeslot, and each timeslot of the users is simultaneously used for uplink and downlink communication. The methods further includes simultaneously sending a local-end OFDM signal and receiving a remote-end OFDM signal in each timeslot of the users, and cancelling, in an OSD manner, interference caused by an echo OFDM signal, which is produced according to the local-end OFDM signal, to the remote-end OFDM signal, where a spectrum of the echo OFDM signal and a spectrum of the remote-end OFDM signal overlap.

The present invention provides a processing method based on an OFDM-TDMA two-way service and a communications device. The method includes assigning timeslots in a TDMA frame structure to multiple users in a TDMA manner, so that each user has a different timeslot, and each timeslot of the users is simultaneously used for uplink and downlink communication. The methods further includes simultaneously sending a local-end OFDM signal and receiving a remote-end OFDM signal in each timeslot of the users, and cancelling, in an OSD manner, interference caused by an echo OFDM signal, which is produced according to the local-end OFDM signal, to the remote-end OFDM signal, where a spectrum of the echo OFDM signal and a spectrum of the remote-end OFDM signal overlap.

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

A signal processing apparatus for dereverberating a number of input audio signals, where the signal processing apparatus includes a processor configured to transform the number of input audio signals into a transformed domain to obtain input transformed coefficients, the input transformed coefficients being arranged to form an input transformed coefficient matrix, determine filter coefficients upon the basis of eigenvalues of a signal space, the filter coefficients being arranged to form a filter coefficient matrix, convolve input transformed coefficients of the input transformed coefficient matrix by filter coefficients of the filter coefficient matrix to obtain output transformed coefficients, and the output transformed coefficients being arranged to form an output transformed coefficient matrix.

A method of echo path delay destination and echo cancellation is described in this disclosure. The method includes: obtaining a reference signal, a microphone signal, and a trained multi-task deep neural network, wherein the multi-task deep neural network comprises a first neural network and a second neural network; generating, using the first neural network of the multi-task deep neural network, an estimated echo path delay based on the reference signal and the microphone signal; updating the reference signal based on the estimated echo path delay; and generating, using the second neural network of the multi-task deep neural network, an enhanced microphone signal based on the microphone signal and the updated reference signal.

An echo cancellation system that synchronizes output audio data with input audio data in a heterogeneous system. The system may increment a counter as outgoing audio frames are sent to a digital-to-analog converter in a speaker. As incoming audio frames are received by an analog-to-digital converter in a microphone, the system may copy contents of the counter into the incoming audio frames. Based on the contents of the counter, the incoming audio frames may be associated with corresponding outgoing audio frames. After synchronizing the incoming audio frames and the outgoing audio frames, the system may perform Acoustic Echo Cancellation by removing the outgoing audio frames from the incoming audio frames.

At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated to remove reverberation therefrom, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.

Method and system for use with audible signals that analyzes the signals into time-frequency frames over first and second time periods. Estimates of the noise and/or reverberation are derived from the frames. Gains are derived from the estimates and raised to a power to create modified gains. The modified gains are applied to the frames in the appropriate time periods. Modified audible signals are output after being processed by the modified gains.

Example methods and systems use multiple sensors to determine whether a speaker is speaking. Audio data in an audio-channel speech band detected by a microphone can be received. Vibration data in a vibration-channel speech band representative of vibrations detected by a sensor other than the microphone can be received. The microphone and the sensor can be associated with a head-mountable device (HMD). It is determined whether the audio data is causally related to the vibration data. If the audio data and the vibration data are causally related, an indication can be generated that the audio data contains HMD-wearer speech. Causally related audio and vibration data can be used to increase accuracy of text transcription of the HMD-wearer speech. If the audio data and the vibration data are not causally related, an indication can be generated that the audio data does not contain HMD-wearer speech.