Panasonic Corporation

Panasonic Intellectual Property Corporation of America

Nec Corporation

Matsushita Electric Industrial Co., Ltd.

Panasonic Intellectual Property Management Co., Ltd.

Panasonic Holdings Corporation

Iii Holdings 12, LLC

Kok Seng Chong

Takeshi Norimatsu

Tomokazu Ishikawa

Huan Zhou

Haishan Zhong

Sua Hong Neo

Zongxian Liu

Naoya Tanaka

Mineo Tsushima

Masahiro Oshikiri

Akihisa Kawamura

Shuji Miyasaka

Masahiro Serizawa

Toshiyuki Nomura

Osamu Shimada

Yuichiro Takamizawa

Kojiro Ono

Kim Hann Kuah

Koji Yoshida

Yoshiaki Takagi

Jiong Zhou

Zhong Hai Shan

Yong Hwee Sim

Zong Xian Liu

Chun Woei Teo

Anh Tuan Tran

Dan Zhao

Yosiaki Takagi

Lih Horng Tee

Shu Feng Fan

Kok Keong Teo

Takeshi Narimatsu

To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

Bandwidth extension method, bandwidth extension apparatus, program, integrated circuit, and audio decoding apparatus

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.

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

A sound signal hybrid encoder includes: a signal analysis unit which determines a scheme for encoding a frame included in a sound signal; an LFD encoder which encodes a frame to generate an LFD frame; an LP encoder which encodes a frame to generate an LP frame; a switching unit which switches between the encoders according to a result of the determination by the signal analysis unit; and an AC signal generation unit which generates an AC signal according to a scheme selected from among schemes, outputs the generated AC signal, and also outputs an AC flag indicating the selected scheme.

Sound signal hybrid encoder, sound signal hybrid decoder, sound signal encoding method, and sound signal decoding method

A bandwidth extension parameter generation device includes: a derivation unit which derives a high-band signal representing a high-band portion of an input sound signal; and a calculation unit which calculates a tone parameter and a floor parameter, the tone parameter indicating a magnitude of energy of a tone component of the high-band signal, the floor parameter indicating a magnitude of energy of a floor component obtained by subtracting the tone component from the high-band signal.

Bandwidth extension parameter generation device, encoding apparatus, decoding apparatus, bandwidth extension parameter generation method, encoding method, and decoding method

A new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals are provided. Currently, very low bitrate audio coding methods for speech and audio signals are proposed. These audio coding methods cause very long delays. Generally, in coding an audio signal, an algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for speech and audio input signals with very low bitrate, a combination of a low delay filter bank like AAC-ELD and a CELP coding method is provided.

Hybrid audio encoder and hybrid audio decoder which perform coding or decoding while switching between different codecs

An audio rendering device which uses multichannel speakers includes: a first delay computation unit which computes, based on rendering information about the multichannel speakers, a first delay corresponding to a primary wavefront which propagates in a predetermined traveling direction from a sound source that is each of speakers included in the multichannel speakers; a second delay computation unit which computes, based on the rendering information about the multichannel speakers, a second delay corresponding to a secondary wavefront which is generated by the primary wavefront and has a scattering wavefront; an addition unit which adds up the first delay and the second delay to compute a total delay; and a delay filter which applies the total delay to an input audio signal to generate a multichannel audio signal for use in rendering with the multichannel speakers, and outputs the multichannel audio signal to the multichannel speakers.

Audio rendering device and audio rendering method

A coding device includes: a pitch contour detection unit which detects a pitch contour of an input audio signal; a dynamic time warping unit which determines the number of pitch nodes based on the pitch contour and generates a first time warping parameter including information indicating the determined number of pitch nodes, a pitch change position, and a pitch change ratio; a first encoder which codes the first time warping parameter; a time warping unit which corrects pitch, using the information obtained from the first time warping parameter, to approximate the pitches of the number of pitch nodes to a predetermined reference value; a second encoder which codes the input audio signal at the corrected pitch; and a multiplexer which multiplexes the coded time warping parameter and the coded audio signal to generate a bitstream.

Coding device, decoding device, coding method, and decoding method for audio signals

A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit () that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit () that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit () that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit () includes: an object classifying unit () that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit () that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.

Coding apparatus and decoding apparatus

To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank () which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit () which adjusts the QMF coefficients based on a predetermined adjustment factor.

Audio signal processing apparatus, audio coding apparatus, and audio decoding apparatus

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Singapore, Singapore

Kok Seng Chong