Broadcom Corporation

Dts, Inc.

Elias Nemer

Jes Thyssen

Wilfrid Paul LeBlanc

Juin-Hwey Chen

Robert W Zopf

Ashutosh Pandey

Zoran Fejzo

Antonius A Kalker

Mohammad Reza Zad-Issa

Bengt J Borgstrom

Jeffrey Kenneth Thompson

Jacek Stachurski

Syavosh Zad-Issa

Antonius Kalker

A frequency domain long-term prediction system and method for estimating and applying an optimum long term predictor. Embodiments of the system and method include determining parameters of a single-tap predictor using a frequency-domain analysis having an optimality criteria based on spectral flatness measure. Embodiments of the system and method also include determining parameters of the long-term predictor by accounting for the performance of the vector quantizer in quantizing the various subbands. In some embodiments other encoder metrics (such as signal tonality) are used as well. Other embodiments of the system and method include determining the optimal parameters of the long-term predictor by accounting for some of the decoder operation. Other embodiments of the system and method include extending a 1-tap predictor to a k-th order predictor by convolving the 1-tap predictor with a pre-set filter and selecting from a table of such pre-set filters based on a minimum energy criteria.

System and method for long term prediction in audio codecs

Systems and methods include audio encoders having improved coding of harmonic signals. The audio encoders can be implemented as transform-based codecs with frequency coefficients quantized using spectral weights. The frequency coefficients can be quantized by use of the generated spectral weights applied to the frequency coefficients prior to the quantization or by use of the generated spectral weights in computation of error within a vector quantization that performs the quantization. Additional apparatus, systems, and methods are disclosed.

Coding of harmonic signals in transform-based audio codecs

A transform-based codec and method with energy smoothing for mitigating vector quantization errors (such as 'birdies') during the encoding process. Embodiments of the codec and method use an encoder to apply in combination an orthogonal transformation and a vector permutation to frequency transform coefficients. In some embodiments the transformation is performed first followed by the permutation and in other embodiments the order is reversed. The order used is reversed at the decoder. A smoothing parameter containing the level of energy smoothing to be applied is passed from the encoder to the decoder and used by both to compute a transform matrix and an inverse transform matrix. In some embodiments the transform matrix is a fraction Hadamard matrix that is invertible, energy preserving, controllable, and stable.

Transform-based audio codec and method with subband energy smoothing

Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

Speaker-identification-assisted speech processing systems and methods

Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device.

Speaker-identification-assisted uplink speech processing systems and methods

A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Single-microphone wind noise suppression

A sound quality metric may be determined at a near-end telephone system, the sound quality metric associated with far-end sound quality received at a far-end telephone system. A signal adjustment may be determined, based on the sound quality metric. The signal adjustment may thus be provided at an earpiece of the near-end telephone system. In this way, a user of the near-end telephone system may be alerted that the sound quality of a far-end user is unacceptably low, so that the near-end user may take corrective action at the near end to improve the far-end sound quality.

Far-end sound quality indication for telephone devices

A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.

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Irvine, CA, United States of America

Elias Nemer