Abstract: A digital information encoder including a divider configured to divide a block of information into a plurality of sub-parts, an initial bit allocator configured to perform an initial allocation of bits to a KTH sub-part of said plurality of sub-parts, a processor configured to compute an estimated number of bits for encoding said KTH sub-part, and a bit allocation adjuster configured to obtain an adjusted bit allocation for said KTH sub-part by adjusting said initial allocation of bits to said KTH sub-part based, at least in part, on said estimated number of bits, wherein the encoder encodes said KTH sub-part using said adjusted bit allocation for said KTH sub-part.

Abstract: A method and apparatus are disclosed for generating a coded audio signal based on a multiple channel audio input 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.

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 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: 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: 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: 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 method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

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 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 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: 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: 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 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: 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 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: 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: 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.