Panasonic Intellectual Property Corporation of America

Panasonic Corporation

Panasonic Intellectual Property Management Co., Ltd.

Haishan Zhong

Kok Seng Chong

Takeshi Norimatsu

Tomokazu Ishikawa

Huan Zhou

Koji Yoshida

Zongxian Liu

Dan Zhao

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

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

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

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

Provided is an encoding device () including: a pitch contour analysis unit () which detects information, a dynamic time-warping unit () which generates, based on the information, pitch change ratios (Tw_ratio in FIG.) within a range () including a range () of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit () which codes the generated pitch parameters (); a time-warping unit () which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal () obtained by the shifting.

Audio encoding device, decoding device, method, circuit, and program

An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients. A selection unit selects an optimum reference signal from among the reference signal candidates by checking the relationship between the respective reference signal candidates and the frequency coefficient in each sub-band portion of the side residual signal and outputs, to an ICP parameter quantization unit, a reference signal ID indicating the selected reference signal and an ICP coefficient corresponding to the reference signal.

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Singapore, Singapore

Haishan Zhong