Abstract: Hybrid audio beamforming systems and methods with narrower beams and improved directivity are provided. The hybrid audio beamforming system includes a time domain beamformer for processing upper frequency band signals of an audio signal using a time domain beamforming technique, and a frequency domain beamformer for processing groups of lower frequency band signals of the audio signal using frequency domain beamforming techniques.

Abstract: Audio beamforming systems and methods that enable more precise control of lobes and nulls of an array microphone are provided. Optimized beamformer coefficients can be generated to result in beamformed signals associated with one or more lobes steered towards one or more desired sound locations and one or more nulls steered towards one or more undesired sound location. The performance of acoustic echo cancellation can be improved and enhanced.

Abstract: Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Abstract: A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

Abstract: An audio signal processor includes a digital filter block configured to receive an audio signal and output a first filtered audio signal, and a phase linearization block configured to receive the first filtered audio signal and output a second filtered audio signal with a more linear phase.

Abstract: Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Abstract: Hybrid audio beamforming systems and methods with narrower beams and improved directivity are provided. The hybrid audio beamforming system includes a time domain beamformer for processing upper frequency band signals of an audio signal using a time domain beamforming technique, and a frequency domain beamformer for processing groups of lower frequency band signals of the audio signal using frequency domain beamforming techniques.

Abstract: Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Abstract: A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

Abstract: An audio signal processor includes a digital filter block configured to receive an audio signal and output a first filtered audio signal, and a phase linearization block configured to receive the first filtered audio signal and output a second filtered audio signal with a more linear phase.

Abstract: A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

Abstract: Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Abstract: A method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted to the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-state signal. The zero-state response of the digital filtering system is added to the zero-input response of the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.

Abstract: A method for long impulse response digital filtering of an input data stream, by use of a digital filtering system. Where the input data stream is divided into zero-input signals and zero-state signals. One of the zero-input signals and a corresponding impulse response of the digital filtering system is converted lo the frequency domain to determine a respective zero-input response of the digital filtering system. One of the zero-state signals is convolved with a corresponding impulse response of the digital filtering system to determine a respective zero-state response of the digital filtering system, wherein at least part of the zero-input signal precedes the zero-stale signal. The zero-state response of the digital filtering system is added to the zero-input response or the digital filtering system to determine the response of the digital filtering system. Apparatus for effecting this method is also disclosed.

Abstract: A system and method for determining DTMF coefficients stores a plurality of reference templates, at least some of said reference templates representative of likelihood ratios. A plurality of LPC coefficients is received. The LPC coefficients are representative of an input signal. The system and method determines a plurality of current likelihood ratios based, at least in part, upon the LPC coefficients. The system and method also determines an initial tone based, at least in part, on the minimum of the current likelihood ratios and the minimum of the reference likelihood ratios. A plurality of LSF coefficients is received. The LSF coefficients are determined, at least in part, upon the input signal. The system and method verifies the validity of the initial tone based, at least in part, upon the LSF coefficients.

Abstract: A system for creating a comfort noise signal, said system includes a clipping circuit, a comfort noise generator, and a switch. The clipping circuit receives a near-end signal, a far-end signal and a residual signal. The clipping circuit determines the power of the far-end signal, the power of the near-end signal and the power of the residual signal. The clipping circuit determines the ratio of the power of the near-end signal to the residual signal and whether the ratio is less than a predetermined threshold. A comfort noise determination block is coupled to the residual signal and selectively forms a comfort noise signal. A switch is coupled to the comfort noise determination block and the clipping circuit. The switch is activated to transmit the comfort noise signal whenever the ratio of the power of the near-end signal to the residual signal is less than a predetermined threshold.

Abstract: A method and apparatus for reducing the computational load of a dual-rate encoding system having a multi-pulse maximum likelihood quantization process configured to transmit at a first transmission rate and to search subframes of excitation signals according to a reduced number of gain scale factors; and an algebraic code-excited linear prediction block configured to perform a first correlation threshold test for entry into an embedded signal processing loop and a second correlation threshold test for entry into a previous signal processing loop in which the embedded signal processing loop is embedded to reduce the number of times the previous signal processing loop and the embedded signal processing loop are entered, thereby reducing the computational load of the system.

Abstract: A system and method for DTMF tone detection receives an input signal. The input signal includes a DTMF tone. The DTMF tone has a first frequency and a second frequency. An initial first frequency and an initial second frequency of said DTMF tone are selected by calculating a plurality of cost functions. The initial first frequency is confirmed to be the first frequency and the initial second frequency is confirmed to be the second frequency using re-computed values of the plurality of cost functions.

Abstract: A method and device to synchronize sampled digital data transferred from an input section to an output section prevents data overrun or underrun due to timing differences of timing signals of the input and output section. The timing synchronization device has an input sampled data counter to determine a number of samples in a frame time of the input sampled data. The timing synchronization device further has an interpolator to estimate data sample values for each sample of the input sampled data to coincide with each sample of the output sampled data if the number of samples in said input sampled data is less than an expected number of samples in said output sampled data.

Abstract: In this invention digital audio data transmitted by wireless means is error corrected and concealed to remove and hide noise created errors ranging from random to burst noise. The data is interleaved into even and odd sub-frames to combat burst mode noise, and ECC is created for the MSB of the data and for command and control bytes using a Reed Solomon encoder before transmission. The LSB are not encoded for reasons of bandwidth because experiments have show the LSB have little effect on audible noise even at a bit error rate of 3.0E-3. The transmitted data is decoded using Reed Solomon decoder and error corrected. The digital audio data is then processed through a concealment procedure that hides the remaining MSB errors by using extrapolation, soft muting and muting depending on the state of audio data preceding and following the current sub-frame of the digital audio data.