Nec Corporation

Matsushita Electric Industrial Co., Ltd.

Panasonic Corporation

Sony Corporation

Yuichiro Takamizawa

Toshiyuki Nomura

Takeshi Norimatsu

Naoya Tanaka

Masahiro Serizawa

Mineo Tsushima

Kok Seng Chong

Osamu Shimada

Sua Hong Neo

Masahiro Iwadare

Kim Hann Kuah

Satoshi Kanda

Naoki Fukino

Taichi Yuki

The information processing device of the present invention includes: a display portion that displays objects that respectively performed a function assigned in advance by being selected; an operation portion that inputs operation information to select one of the objects; and a control portion that displays supplementary information linked to the object on the display portion, based on the operation information. The control portion, based on predetermined conditions, performs a first process to display first supplementary information on the display portion. If operation information is not newly input during a prescribed period of time after the first process is performed, the control portion performs a second process to acquire second supplementary information and display the second supplementary information on the display portion.

Information processing device and information providing method

A bit-rate converting apparatus and a method thereof, in which bit-rate conversion is executed by low computational complexity, are provided. The bit-rate conversion is executed in a frequency domain, and psycho-acoustic analysis is not needed by using information included in an inputted bit-stream before the bit-rate conversion is applied. With this, the computational complexity is lowered. And in order that many equal values are not contained in a frequency domain signal, which is inputted to a quantizing means, a quantized value before inverse quantizating is applied is modified, or an inverse quantized value after the inverse quantizing was applied is modified. With this, fine control for the bit-rate is made to be easy.

Bit-rate converting apparatus and method thereof

An energy corrector () for correcting a target energy for high-frequency components and a corrective coefficient calculator () for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Audio decoding apparatus and decoding method and program

In an adaptive transform coding system and/or and adaptive transform decoding system, coding efficiency in the case where a small number of quantized values having large absolute value are present, is improved. The adaptive transform coding system codes the small number of quantized values having large absolute values and other quantized values are coded separately. More particularly, the adaptive transform coding system includes a selector () discriminating the small number of quantized value having large absolute value from other quantized value, a pulse coding means for coding the small number of quantized values having large absolute values () and the pulse decoding means () for decoding the same, a coding means () for coding the quantized value other than those having large absolute values and a decoding means () decoding the same, and a synthesis means () for synthesizing the small number of quantized values having large absolute value and other quantized values.

Adaptive transform coding system, adaptive transform decoding system and adaptive transform coding/decoding system

Audio decoding device, decoding method, and program

A hierarchical lossless encoding and decoding technology for digital signals such as of music data, audio data, or the like. A lossless reproduced signal is made identical to an input signal even when the processing accuracy in an encoding apparatus and the processing accuracy in a decoding apparatus are different from each other. the encoding apparatus transmits low-bit-rate encoded data produced by encoding an input signal, lossless encoded data produced by effecting lossless encoding on a differential signal between a low-bit-rate decoded signal decoded from the low-bit-rate encoded data and the input signal, and corrective information extracted from the low-bit-rate decoded signal to respective input terminals of the decoding apparatus. A low-bit-rate decoder decodes the low-bit-rate encoded data. A lossless decoder decodes the differential signal. A corrector corrects the low-bit-rate decoded signal based on the corrective information. An adder adds the differential signal and the corrected low-bit-rate decoded signal into a lossless reproduced signal.

Hierarchical lossless encoding/decoding method, hierarchical lossless encoding method, hierarchical lossless decoding method, its apparatus and program

A mapping transform unit subjects input audio signals to a mapping transform and generates frequency region signals that take frequency as a variable; a code amount designation unit supplies a preset coding bit rate as a code amount output; a frequency region signal compression encoder, based on the code amount, subjects input frequency region signals to a compression encoding process and generates a bitstream; and a bandwidth-limiting unit executes a bandwidth-limiting processing in which a part of the frequency zone covered by frequency region signals is allotted to an attenuation frequency zone, and in which the value of the frequency region signal is multiplied by an attenuation coefficient having a value less than 1 in the attenuation frequency zone to attenuate the frequency region signal in the attenuation frequency zone, and supplies the frequency region signals that have undergone the bandwidth-limiting processing to the frequency region signal compression encoder.

Audio encoder utilizing bandwidth-limiting processing based on code amount characteristics

A random number generating apparatus is equipped with a random number memory section in which a plurality of sets of random numbers, whose total energy R, has a fixed value are stored. N random numbers are generated on the basis of more than one set of random numbers read from the random number memory section, and further multiplied by a square root of (T&#xd7;M&#xf7;R&#xf7;N) to obtain a random number sequence having a predetermined energy. A normalization coefficient S can be obtained with a simple process without any usage of the calculation of division and square root multiplication, thereby enabling a small-scale apparatus to be realized. Accordingly, the random number generating apparatus has a decreased scale in the circuit arrangement capable of outputting the random numbers having a predetermined energy.

Random number generating apparatus, random number generating method, program for generating random numbers, audio decoder and audio decoding method

A coded voice signal format converting apparatus is provided which is capable of converting a signal format of a coded voice signal by computations in reduced amounts. In the coded voice signal format converting apparatus, in a second coding device is employed a quantizing accuracy information converting section to which a first quantizing accuracy information output from a quantizing accuracy information decoding section in a first decoding device is input. Second mapping signal is quantized by a mapped signal coding section to produce a coded voice signal and the first quantizing accuracy information is converted so that it can be used by mapped signal coding section to determine a second quantizing accuracy information.

Coded voice signal format converting apparatus

An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.

Audio decoding apparatus and method for band expansion with aliasing adjustment

A wideband, high quality audio signal is decoded with few calculations at a low bitrate. Unwanted spectrum components accompanying sinusoidal signal injection by a synthesis subband filter built with real-value operations are suppressed by inserting a suppression signal to subbands adjacent to the subband to which the sine wave is injected. This makes it possible to inject a desired sinusoid with few calculations.

Audio decoding apparatus and method for band expansion with aliasing suppression

To provide an audio signal quantization device in which an amount of arithmetical operation can be reduced. A 0.75-power computing apparatus includes: an inverse number computing unit; a first &#x2212;0.5-power computing unit; a multiplication unit; a second &#x2212;0.5-power computing unit; and a program storage medium that is connected to an outside, whereby a program stored in the program storage medium is executed to compute the 0.75-power in a quantization computing expression. The inverse number computing unit, the first &#x2212;0.5-power computing unit, the multiplication unit, and the second &#x2212;0.5-power computing unit operate in such a manner that a multiplication between an infinity and zero which leads its operation result to an indefinite value is inhibited.

0.75-power computing apparatus and method and program for use therewith

In a data compression device having a plurality of Huffman coders given identification (ID) values, respectively, to produce Huffman codes together with the corresponding ID codes obtained from the ID values, a re-coding portion is included to change a sequence of the ID values so that a reduction is accomplished about a code amount of the Huffman codes and the ID codes obtained from the ID value sequence and is operable to re-code an input data sequence again by Huffman coders indicated by the changed ID values. Such re-coded Huffman codes and ID codes based on the changed ID values are produced as output data signals.

Data compression, control program for controlling the data compression

In an adaptive transform coding system and/or and adaptive transform decoding system, coding efficiency in the case where a small number of quantized values having large absolute value are present, is improved. The adaptive transform coding system codes the small number of quantized values having large absolute values and other quantized values are coded separately. More particularly, the adaptive transform coding system includes a selector ( ) discriminating the small number of quantized value having large absolute value from other quantized value, a pulse coding means for coding the small number of quantized values having large absolute values ( ) and the pulse decoding means ( ) for decoding the same, a coding means ( ) for coding the quantized value other than those having large absolute values and a decoding means ( ) decoding the same, and a synthesis means ( ) for synthesizing the small number of quantized values having large absolute value and other quantized values.

In a coded speech decoding system, an n-channel time domain speech signal is converted to a frequency domain speech signal. A predetermined weighting adding process is executed on the frequency domain speech signal for each of a plurality of different transfer functions. The frequency domain speech signal obtained through the weighting adding process is converted to an m-channel (m<n) time domain speech signal. A predetermined windowing processing is executed on the time domain speech signal.

Growing community of inventors

Tokyo, Japan

Yuichiro Takamizawa