Panasonic Intellectual Property Corporation of America

Panasonic Corporation

Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.

Panasonic Holdings Corporation

Iii Holdings 12, LLC

Zongxian Liu

Masahiro Oshikiri

Srikanth Nagisetty

Kok Seng Chong

Naoya Tanaka

Koji Yoshida

Hiroyuki Ehara

Haishan Zhong

Provided are a voice audio encoding device, voice audio decoding device, voice audio encoding method, and voice audio decoding method that efficiently perform bit distribution and improve sound quality. Dominant frequency band identification unit identifies a dominant frequency band having a norm factor value that is the maximum value within the spectrum of an input voice audio signal. Dominant group determination units and non-dominant group determination unit group all sub-bands into a dominant group that contains the dominant frequency band and a non-dominant group that contains no dominant frequency band. Group bit distribution unit distributes bits to each group on the basis of the energy and norm variance of each group. Sub-band bit distribution unit redistributes the bits that have been distributed to each group to each sub-band in accordance with the ratio of the norm to the energy of the groups.

Voice audio encoding device, voice audio decoding device, voice audio encoding method, and voice audio decoding method

An apparatus for decoding a surround audio signal includes a Bitstream De-multiplexer for unpacking a bitstream into spatial parameters and core parameters, a set of core decoders for decoding the core parameters into a set of core signals, a matrix derivation unit for deriving the rendering matrix from the spatial parameters and playback speaker layout information, and a renderer for rendering of the decoded core signals to playback signals using the rendering matrix.

Apparatus and method for surround audio signal processing

An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes the encoded information, including encoding parameters that are used for decoding a low frequency spectrum and index information that identifies a most, correlated portion from a low frequency spectrum for one or more high frequency subbands. The processor also replicates a high frequency subband spectrum based on the index information using a synthesized low frequency spectrum, the synthesized low frequency spectrum being obtained by decoding the encoding parameters. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

Device and method for bandwidth extension for audio signals

An audio/speech encoding method is provided that includes transforming a time domain input signal to a frequency spectrum, and dividing the frequency spectrum to a plural of bands. The method also includes calculating a level of energies for each band, quantizing the energies for the each band, and calculating differential indices. The method additionally includes modifying a range of the differential indices for the Nth band when N is an integer of 2 or more, and replacing the differential index with the modified differential index, and not modifying a range of the differential indices for the Nth band when N is an integer of 1. The method further includes encoding the differential indices using a Huffman table selected based on a minimum value and a maximum value of the differential indices, and transmitting the encoded differential indices and a flag signal for indicating the selected Huffman table.

Audio/speech encoding apparatus and method, and audio/speech decoding apparatus and method

An audio/speech encoding apparatus/method and an audio/speech decoding apparatus/method are provided. The audio/speech encoding apparatus includes a memory that stores instructions, and a processor that performs operations. The operations include transforming a time domain input audio/speech signal to a frequency spectrum, dividing the frequency spectrum to a plural of bands, calculating norm factors, and quantizing the norm factors. The operations also include calculating differential indices between an Nth band index and an (N&#x2212;1)th band index, and modifying a range of the differential indices for the Nth band when N is 2 or more. The operations further include replacing the differential index with the modified differential index, and not modifying a range of the differential indices for the Nth band when N is 1. The apparatus encodes the differential indices using a selected Huffman table, and transmits the encoded differential indices and a flag signal over a communication network.

An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes the encoded information, including encoding parameters that are used for decoding a low frequency spectrum and index information that identifies a most correlated portion from a low frequency spectrum for one or more high frequency subbands. The processor also replicates a high frequency subband spectrum based on the index information using a synthesized low frequency spectrum, the synthesized low frequency spectrum being obtained by decoding the encoding parameters. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

An apparatus for decoding surround audio signal, includes a Bitstream De-multiplexer for unpacking a bitstream into spatial parameters and core parameters, a set of Core Decoder for decoding the core parameters into a set of core signal, a matrix derivation unit for deriving the rendering matrix from the spatial parameters and playback speaker layout information, a renderer for rendering of the decoded core signal to playback signals using the rendering matrix.

In this invention, the design of the Huffman table can be done offline with a large input sequence database. The range of the quantization indices (or differential indices) for Huffman coding is identified. For each value of range, all the input signal which have the same range will be gathered and the probability distribution of each value of the quantization indices (or differential indices) within the range is calculated. For each value of range, one Huffman table is designed according to the probability. And in order to improve the bits efficiency of the Huffman coding, apparatus and methods to reduce the range of the quantization indices (or differential indices) are also introduced.

Device and method for execution of huffman coding

An acoustic signal coding apparatus includes a subband classifier that classifies subbands obtained by dividing a frequency-domain spectrum into a plurality of perceptually important first-category subbands and the other subbands referred to as second-category subbands according to at least one of measures in terms of energy and peak property, a subband peak-algebraic vector quantization (SBP-AVQ) vector generator that generates an SBP-AVQ vector by collecting a maximum peak from each first-category subband, outputs the generated SBP-AVQ vector, and outputs peak position information indicating the positions of the maximum peaks, a bit distributor that distributes bits for AVQ coding to the SBP-AVQ vector and the second-category subband vector, and an AVQ coder that performs AVQ coding on the SBP-AVQ vector and the second-category subband vector.

Acoustic signal coding apparatus, acoustic signal decoding apparatus, terminal apparatus, base station apparatus, acoustic signal coding method, and acoustic signal decoding method

An audio encoding apparatus capable of reducing the bit rate even if a codebook having a larger codebook number is selected in a split multi-rate lattice vector quantization is provided. Sub-vector determining unit () determines, in the spectrum of an input signal having been divided into a predetermined number of sub-vectors, a sub-vector using the largest number of bits. Positional information encoding unit () encodes the positional information of the determined sub-vector. Codebook indication value estimating unit () estimates a number of used bits for a codebook indication value of the largest number of used bits by use of the (N&#x2212;1) other codebook indication values, and generates a number-of-used-bits estimation value. Difference calculating unit () calculates a difference by subtracting the number-of-used-bits estimation value from the actual value of the codebook indication value of the largest number of used bits. Difference encoding unit () encodes the difference information.

Audio encoding apparatus, audio decoding apparatus, audio encoding method, and audio decoding method

An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes low-band encoding parameters, index information, and scale factor information from the encoded information. The processor also decodes the low-band encoding parameters to obtain a synthesized low frequency spectrum, replicates a high frequency subband spectrum based on the index information using the synthesized low frequency spectrum, and adjusts an amplitude of the replicated high frequency subband spectrum using the scale factor information. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency spectrum, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

The purpose of the present invention is to more efficiently extend, using a low bit rate, the bandwidth of input signals having a harmonics structure, in order to obtain better audio quality. The present invention is installed in a device that extends bandwidth for audio signal encoding and decoding. This novel bandwidth extension encoding identifies a low-frequency spectrum component having the highest correlation to a high-frequency bandwidth signal among input signals, duplicates a high-frequency spectrum by energy adjustment of said component, and maintains the harmonic relationship between the low-frequency spectrum and the duplicated high-frequency spectrum by adjusting the spectral peak position of the duplicated high-frequency spectrum, on the basis of a harmonic frequency estimated from a composite low-frequency spectrum.

This invention introduces audio/speech encoding apparatus audio/speech decoding apparatus, audio/speech encoding method and audio/speech decoding method to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, the position of the sub-vector whose codebook indication consumes the most bits is firstly located, and then the value of the codebook is estimated based on the total number of bits available and the bits usage information for other sub-vectors. The difference value is calculated between the actual value and estimated value. Finally, instead of transmitting the codebook indication which consumes the most bits, the position of the sub-vector whose codebook indication consumes the most bits and the difference value between the actual value and the estimated value are transmitted. By applying of the invented method, bits can be saved by the codebook indications.

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Singapore, Singapore

Zongxian Liu