Abstract: A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

Abstract: An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames.

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 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: 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 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: Disclosed are systems and methods for the efficient conversion of linear predictive coefficients. This method is usable, for example, in the conversion of full band linear predictive coding (“LPC”) coefficients to sub-band LPCs of a sub-band speech codec. The sub-bands may or may not be down-sampled. In an embodiment, the LPC coefficients of the sub-bands are obtained from the correlation coefficients, which are in turn obtained by filtering the auto-regressive extended auto-correlation coefficients of the full band LPCs. The method also allows the generation of an LPC approximation of a pole-zero weighted synthesis filter.

Abstract: A method and apparatus for low complexity combinatorial coding and decoding of signals is described herein. During operation, an encoder and a decoder will utilize a first function in determining a codeword or vector when the size of the function is small. The encoder and the decoder will also utilize a second function in determining the codeword or vector when the size of the function is large.

Abstract: A method and apparatus are for performing one of encoding and decoding a code word that is used to communicate a portion of a signal. For encoding, at least a portion of a code word is encoded from a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. For decoding, at least a portion of a code word is decoded to a signal based value using an approximation of a combinatorial function, wherein the signal based value represents one or more aspects of a signal. The approximation of the combinatorial function is based on a linear combination of a set of basis functions.

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 (1100) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (1110) a weighted target vector from an input signal. The method can include processing (1120) the weighted target vector through an inverse weighting function to create a residual domain target vector. The method can include performing (1130) a first search process on the residual domain target vector to obtain an initial fixed codebook code-vector. The method can include performing (1140) a second search process over a subset of possible codebook code-vectors for a low weighted-domain error to produce a final fixed codebook code-vector. The subset of possible codebook code-vectors can be based on the initial fixed codebook code-vector. The method can include generating (1150) a codeword representative of the final fixed codebook code-vector. The codeword can be for use by a decoder to generate an approximation of the input signal.

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 method and apparatus provides for frame loss recovery following a loss of a frame in an audio codec. The lost frame is identified. Estimated linear predictive coefficients of a previous transform frame are generated based on a decoded audio of the previous transform frame. An estimated residual of the previous transform frame is generated based on the estimated linear predicative coefficients and the decoded audio. A pitch delay is determined from frame error recovery parameters received with the previous transform frame. An extended residual is generated based on the pitch delay and the estimated residual. A first synthesized signal is generated based on the extended residual and the linear predicative coefficients. A decoded audio output of at least the lost frame is generated based on the first synthesized signal. The frame error recovery parameters are generated by an encoder.

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: A method and apparatus provide for audio frame recovery by identifying a sequence of lost frames of coded audio data as being lost or corrupted; identifying a first frame of coded audio data which immediately preceded the sequence of lost frames, as having been encoded using a time domain coding method; identifying a second frame of coded audio data, which immediately followed the sequence of lost frames of coded audio data, as having been encoded using a transform domain coding method; obtaining a pitch delay; generating a second decoded audio portion of the second frame based on the second frame; generating a first decoded audio portion of the second frame based on the pitch delay and decoded audio samples; and generating a decoded audio output of the second frame based on a sequential combination of the first and second decoded audio portions.

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: A method (300) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (310) a target vector from a received input signal. The method can include constructing (320) a plurality of inverse weighting functions based on the target vector. The method can include evaluating (330) an error value associated with each of the plurality of inverse weighting functions to produce a fixed codebook code-vector. The method can include generating (340) a codeword representative of the fixed codebook code-vector, where the codeword can be used by a decoder to generate an approximation of the input signal.

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 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: An audio signal processing system includes parallel speech and generic audio signal processing paths. One path includes a linear predictive coder and a resampling filter having a non-linear phase characteristic. A phase compensation filter is disposed along the one of the processing paths to compensate for the non-linearity of the resampling filter thereby enabling relatively seamless switching between the coders resulting in a reduction of audio artifacts that would otherwise result from the non-linear phase characteristic of the resampling filter during playback.