Abstract: A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component.

Abstract: During operation a multiple channel audio input signal is received and coded to generate a coded audio signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. The representation of the gain value may be output for transmission and/or storage.

Abstract: A set of peaks in a reconstructed audio vector ? of a received audio signal is detected and a scaling mask ?(?) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output.

Abstract: An aspect of the present invention is drawn to an audio data processing device for use by a user to control a system and for use with a microphone, a user demographic profiles database and a content/ad database. The microphone may be operable to detect speech and to generate speech data based on the detected speech. The user demographic profiles database may be capable of having demographic data stored therein. The content/ad database may be capable of having at least one of content data and advertisement data stored therein. The audio data processing device includes a voice recognition portion, a voice analysis portion and a speech to text portion. The voice recognition portion may be operable to process user instructions based on the speech data. The voice analysis portion may be operable to determine characteristics of the user based on the speech data. The speech to text portion may be operable to determine interests of the user.

Abstract: During operation a multiple channel audio input signal is received and coded to generate a coded audio signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal. The representation of the gain value may be output for transmission and/or storage.

Abstract: A set of peaks in a reconstructed audio vector ? of a received audio signal is detected and a scaling mask ?(?) based on the detected set of peaks is generated. A gain vector g* is generated based on at least the scaling mask and an index j representative of the gain vector. The reconstructed audio signal is scaled with the gain vector to produce a scaled reconstructed audio signal. A distortion is generated based on the audio signal and the scaled reconstructed audio signal. The index of the gain vector based on the generated distortion is output.

Abstract: Apparatus (119) for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor (119) which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k?1 subsequent frames to values, such that the values of the n bits of the k?1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Abstract: A method is provided for offering for purchase supplemental content associated with a multimedia program. The method includes receiving over a content delivery system a multimedia program and supplemental content associated with the program. While the program is being rendered, an option is presented to a user to acquire the supplemental content.

Abstract: A method that may be used in variety of electronic devices for generating comfort noise includes receiving a plurality of information frames indicative of speech plus background noise, estimating one or more background noise characteristics based on the plurality of information frames, and generating a comfort noise signal based on the one or more background noise characteristics. The method may further include generating a speech signal from the plurality of information frames, and generating an output signal by switching between the comfort noise signal and the speech signal based on a voice activity detection.

Abstract: In a selective signal encoder, an input signal is first encoded using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded to produce a reconstructed signal and an error signal is generated as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared to the input signal. One of two or more enhancement layer encoders selected dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output to the channel (for transmission or storage, for example).

Abstract: To reduce the complexity of the encoding/decoding of pulse positions and/or pulse magnitudes associated with complex combinatorial computations, a method and structure for encoding and decoding of pulse position and/or pulse magnitudes requires fewer computations of these combinatorial functions. Adaptive switching between coding or encoding is performed in accordance with the estimated density of the plurality of occupied positions.

Abstract: To reduce the complexity of the encoding/decoding of pulse positions and/or pulse magnitudes associated with complex combinatorial computations, a method and structure for encoding and decoding of pulse position and/or pulse magnitudes requires fewer computations of these combinatorial functions. Approximation of such functions is acceptable as long as certain sufficient properties are maintained. Computational complexity of certain coding and decoding operations may be reduced by two orders of magnitude or more for a given signal vector input.

Abstract: During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal.

Abstract: During operation of an encoder, a signal vector (x) is received. A first multi-precision operand (??k) will be generated based on the signal vector to be encoded. A mantissa operand and an exponent operand are generated. Both the mantissa operand and the exponent operand are representative of a second multi-precision operand that is based on the signal vector to be encoded. A portion of ??k is selected to be modified based on the exponent operand. A part of ??k is modified based on the mantissa operand to produce a modified multi-precision operand (??k+1). Finally, a multi-precision codeword is generated for use in a corresponding decoder.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: A method and system for analysis-by-synthesis encoding of an information signal is provide. The encoder (400) can include the steps of generating a first synthetic signal based on a first pitch-related codebook (402), generating a second synthetic signal based on a second pitch-related codebook (404), selecting a codebook configuration parameter based on the reference signal and the first and second synthetic signals, and conveying the codebook configuration for use in reconstructing an estimate of the input signal. The encoder can include an error expression having an error bias (506) and a prediction gain having a prediction gain bias (508) for determining the codebook configuration. The encoder can employ variable length coding and combinatorial subframe coding (600) for efficiently compressing the codebook configuration parameter and codebook related parameters for one or more subframes.

Abstract: A system (100) and method (200) of combining codewords is provided. The system can include a splitter (120) for splitting a first codeword (110) into a most significant bits part MSP (112) and a least significant bits part LSP (114), a combiner (130) for combining the MSP of the first codeword with a second codeword to produce a first group (132), and a concatenator (140) for concatenating the first group with the LSP to produce a second group (134), and multiplexing the first group with the second group to produce a multiplexed codeword (150). Bit-errors in the LSP correspond to decoding errors only in a codeword associated with the LSP, and not to decoding errors in other codewords.

Abstract: A method of enhanced tandem communication is provided between at least a first portion of a network suitable for voice communications and a second portion of a network suitable for voice communications. During operation, two representations of an encoded signal are transmitted from the first portion of a network. The two representations comprise the encoded signal produced by a first codec and a parameter translation of the first encoded signal into an encoded signal compatible with a single common compressed voice codec (CCVC) format.