摘要:
A system for improved audio in a headset comprising a first headset microphone generating a first signal. A second headset microphone generating a second signal. A multiplexer coupled to the first headset microphone and the second headset microphone for multiplexing the first signal and the second signal. A power extractor for extracting power for use by one or more of the multiplexer, the first headset microphone and the second headset microphone. A demultiplexer for extracting the first signal and the second signal. A signal processor for generating a noise reduced microphone signal. An audio subsystem for receiving the noise reduced microphone signal and for generating speaker signals for a first headphone speaker and a second headphone speaker.
摘要:
Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.
摘要:
An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.
摘要:
A system and apparatus for constructing a displacement model across a frequency range for a loudspeaker is disclosed. The resultant displacement model is centered around the distortion point. Once a distortion model is constructed it can be incorporated into an audio driver to prevent distortion by incorporating the model and a distortion compensation unit with a conventional audio driver. Various topologies can be used to incorporate a distortion model and distortion compensation unit into an audio driver. Furthermore, a wide variety of distortion compensation techniques can be employed to avoid distortion in such an audio driver.
摘要:
Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.
摘要:
An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.
摘要:
Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.
摘要:
Acoustic echoes in communications systems are distracting and undesirable. Acoustic echoes occur in communications systems where sound produced by a speaker is picked up by a microphone in a communications system. In a stereo playback environment, echo cancellation techniques become more complicated. Echo cancellation can be performed by performing echo cancellation on a center signal, which is the sum of a left channel signal and the right channel signal, or left signal and a difference signal, which is the difference of the right channel signal and the left channel signal. The adaptation rates of the two echo cancellers meet certain constraints to prevent degeneracies in the echo cancellation system.
摘要:
A system for processing audio data comprising an amplifier configured to receive an audio signal and to perform nonlinear processing on the audio signal. An encoder coupled to the amplifier, the encoder configured to receive the nonlinearly processed audio signal and to encode the nonlinearly processed audio signal into a data transmission format. A transmitter configured to receive and transmit the encoded nonlinearly processed audio signal. A receiver configured to receive the transmitted encoded nonlinearly processed audio signal and to decode the encoded nonlinearly processed audio signal. A digital voice processor configured to receive the nonlinearly processed audio signal and to use the nonlinearly processed audio signal for echo estimation and to subsequently subtract the estimated echo signal from a microphone signal.
摘要:
A system for controlling distortion comprising a total harmonic distortion (THD) modeling system configured to apply a chirp signal to a system and to identify one or more frequency bands at which distortion is present, and to apply a ramping signal to identify for each of the one or more frequency bands an input signal level at which distortion is initiated, a signal processing system configured to receive an input signal, to determine whether frequency components are present in the input signal that are associated with the one or more frequency bands at which distortion is present, and to limit the amplitude of the input signal at the one or more frequency bands, such as by applying dynamic range compensation.