The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Audio encoder and decoder

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

Active noise cancellation is combined with spectrum modification of a reproduced audio signal to enhance intelligibility.

Systems, methods, apparatus, and computer-readable media for automatic control of active noise cancellation

Systems, methods, and apparatus for spectral contrast enhancement of speech signals, based on information from a noise reference that is derived by a spatially selective processing filter from a multichannel sensed audio signal, are disclosed.

Systems, methods, apparatus, and computer program products for spectral contrast enhancement

Techniques described herein include the use of equalization techniques to improve intelligibility of a reproduced audio signal (e.g., a far-end speech signal).

Systems, methods, apparatus and computer program products for enhanced intelligibility

There is provided a speech post-processor (250) for enhancing a speech signal (320) divided into a plurality of sub-bands (330) in frequency domain. The speech post-processor comprises an envelope modification factor generator (260) configured to use frequency domain coefficients representative of an envelope derived from the plurality of sub-bands to generate an envelope modification factor for the envelope derived from the plurality of sub-bands, where the envelope modification factor is generated using FAC = α ENV / Max + (1-α), where FAC is the envelope modification factor, ENV is the envelope, Max is the maximum envelope, and α is a value between 0 and 1, where α is a different constant value for each speech coding rate. The speech post-processor further comprises an envelope modifier (265) configured to modify the envelope derived from the plurality of sub-bands by the envelope modification factor corresponding to each of the plurality of sub-bands.

Yang Gao

Papers

Speech post-processing using mdct coefficients