Abstract: A method for decoding audio frames includes producing a first frame of coded audio samples, producing at least a portion of a second frame of coded audio samples, generating audio gap filler samples based on parameters representative of a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples, and forming a sequence including the audio gap filler samples and the portion of the second frame of coded audio samples.

Abstract: A method for encoding audio frames by producing a first frame of coded audio samples by coding a first audio frame in a sequence of frames, producing at least a portion of a second frame of coded audio samples by coding at least a portion of a second audio frame in the sequence of frames, and producing parameters for generating audio gap filler samples, wherein the parameters are representative of either a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples.

Abstract: An encoder/decoder architecture including an arithmetic encoder that encodes the MSB portions of a Factorial Pulse Coder output, and that encodes an output of a first-level source encoder, e.g., MDCT. Sub-parts (e.g., frequency bands) of portions (e.g., frames) of the signal are sorted in increasing order based on a measure related to signal energy (e.g., signal energy itself). In a system that overlays Arithmetic Encoding on Factorial Pulse coding, the result is bits re-allocated to bands with higher signal energy content, yielding higher signal quality and higher bit utilization efficiency.

Abstract: Hybrid range coding/combinatorial coding (FPC) encoders and decoders are provided. Encoding and decoding can be dynamically switched between range coding and combinatorial according to the ratio of ones to the ratio of bits in a partial remaining sequence in order to reduce the computational complexity of encoding and decoding.

Abstract: A method for processing an audio signal including classifying an input frame as either a speech frame or a generic audio frame, producing an encoded bitstream and a corresponding processed frame based on the input frame, producing an enhancement layer encoded bitstream based on a difference between the input frame and the processed frame, and multiplexing the enhancement layer encoded bitstream, a codeword, and either a speech encoded bitstream or a generic audio encoded bitstream into a combined bitstream based on whether the codeword indicates that the input frame is classified as a speech frame or as a generic audio frame, wherein the encoded bitstream is either a speech encoded bitstream or a generic audio encoded bitstream.

Abstract: An encoder/decoder architecture including an arithmetic encoder that encodes the MSB portions of a Factorial Pulse Coder output, and that encodes an output of a first-level source encoder, e.g., MDCT. Sub-parts (e.g., frequency bands) of portions (e.g., frames) of the signal are sorted in increasing order based on a measure related to signal energy (e.g., signal energy itself). In a system that overlays Arithmetic Encoding on Factorial Pulse coding, the result is bits re-allocated to bands with higher signal energy content, yielding higher signal quality and higher bit utilization efficiency.

Abstract: A encoder/decoder architecture (200, 300, 700) that uses an arithmetic encoder (220) to encode the MSB portions of the output of a Factorial Pulse Coder (212), that encodes the output of a first-level source encoder (210), e.g., MDCT. Sub-parts (e.g., frequency bands) of portions (e.g., frames) of the signal are suitably sorted in increasing order based on a measure related to signal energy (e.g., signal energy itself). Doing this in a system (100) that overlays Arithmetic Encoding on Factorial Pulse coding results in bits being re-allocated to bands with higher signal energy content, ultimately yielding higher signal quality and higher bit utilization efficiency.

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: A method and apparatus for prediction in a speech-coding system extends a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, to a multi-tap LTP filter. From another perspective, a conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Such a multi-tap LTP filter offers a number of advantages over the prior-art. 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 (?i's) of the multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component. Consequently their main function is to maximize the prediction gain of the LTP filter via modeling the degree of periodicity that is present and by imposing spectral shaping.

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: 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: 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: 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.