Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: A sound reproduction apparatus includes: a sound source localization estimating unit (1) that estimates whether or not a sound image is localized using input audio signals in an acoustic space when the input audio signals are reproduced by speakers (FL, FR, SL, SR) placed in standard positions; a sound source signal separating unit (2) that calculates a sound source localization signal Z(i) indicating the sound image that is localized and separates, from the input audio signals, sound source non-localization signals (FLa, FRb, SLa, SLb) which are signal components not contributing to localization of the sound image; a sound source position parameter calculating unit (3) that calculates parameters (R, ?) indicating a position of the sound source localization signal in the acoustic space; and a reproduction signal generating unit (4) that uses the sound source position is parameters indicating the position of the sound source localization signal to distribute the sound source localization signal to front speake

Abstract: An audio decoder which reproduces original signals from a bit stream including (i) a downmix signal of the original signals, and (ii) supplementary information indicating a gain ratio D and phase difference ? between the original signals. The audio decoder includes: a decoding unit extracting the downmix signal; a transformation unit transforming the extracted downmix signal into a frequency domain signal; a phase rotator determination unit determining two phase rotators having, as the phase rotation angles, angles ? and ? respectively obtained by dividing a contained angle by a diagonal of a parallelogram; a separation unit separating the frequency domain signal into two separation signals respectively indicating angles ? and ? as phase differences between the signals and the decoded downmix signal; and an inverse transformation unit inversely transforming the respective two separation signals into time domain signals so as to reproduce the two audio signals.

Abstract: The encoding apparatus includes an MDCT unit which transforms an inputted audio signal into a frequency parameter, for every predetermined time-frequency transformation frame length, and an MDCT coefficient encoding unit which encodes the frequency parameter. The encoding apparatus also includes a pitch cycle detection unit which detects a pitch cycle of the audio signal, a framing unit which frames the audio signal based on the detected pitch cycle, and a waveform modification unit which performs waveform modification on the audio signal framed based on the pitch cycle, in conformance with the time-frequency transformation frame length, and outputs the waveform-modified audio signal to the MDCT unit. A multiplex unit multiplexes the frequency parameter encoded by MDCT coefficient encoding unit and the pitch cycle, and outputs the multiplexed result as a bitstream.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) 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.

Abstract: Provided are an audio encoding device and an audio decoding device, by which optimal trade-off between code rates and sound quality can be flexibly adjusted. A variable frequency segmentation encoding unit includes: difference degree calculation units for calculating a difference degree between first and second input signals depending on a segmentation method for segmenting a frequency band into sub-bands; a selection unit for selecting one of the segmentation methods; and a difference degree and segmentation information encoding unit for encoding the selected method and the difference degree for each sub-band. A variable frequency segment decoding unit includes: a segmentation information decoding unit for decoding the segmentation information to learn the segmentation method; a switching unit for outputting a difference degree code corresponding to the segmentation method; and difference degree decoding units for decoding the difference degree code to the difference degree for each sub-band.

Abstract: An audio encoder, which is capable of encoding multiple-channel signals so that only a downmixed signal is decoded and of further generating specific auxiliary information necessary for dividing the downmixed signal, is provided. An audio encoder (10), which compresses and encodes audio signals of N-channels (N>1), includes a downmixed signal encoding unit (11) which encodes the downmixed signal obtained by downmixing the audio signals, and an auxiliary information generation unit (12a) which generates auxiliary information necessary for decoding the downmixed signal encoded by the downmixed signal encoding unit (11) into N-channel audio signals.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: In the conventional art inventions for coding multi-channel audio signals, three of the major processes involved are: generation of a reverberation signal using an all-pass filter; segmentation of a signal in the time and frequency domains for the purpose of level adjustment; and mixing of a coded binaural signal with an original signal coded up to a fixed crossover frequency. These processes pose the problems mentioned in the present invention. The present invention proposes the following three embodiments: to control the extent of reverberations by dynamically adjusting all-pass filter coefficients with the inter-channel coherence cues; to segment a signal in the time domain finely in the lower frequency region and coarsely in the higher frequency region; and to control a crossover frequency used for mixing based on a bit rate, and if the original signal is coarsely quantized, to mix a downmix signal with an original signal in proportions determined by an inter-channel coherence cue.

Abstract: A hybrid integrated circuit device includes: an insulating substrate (1) having a lower surface formed with wiring patterns including ends arranged along ends of the lower surface at a predetermined pitch (P); electronic components (3) mounted on the surfaces of the insulating substrate to be connected to the wiring patterns; a pair of insulating legs (2) arranged at the ends of the lower surface of the substrate (1), each insulating leg extending in parallel to the lower surface of the substrate (1); and a plurality of terminal electrodes (5) formed on each leg at the pitch and extending perpendicularly to the substrate, where the plurality of terminal electrodes are connected to the wiring patterns on the lower surface of the substrate (1). Each leg has a surface bonded to the substrate and formed with electrode films connected to the terminal electrodes.

Abstract: To reduce the amount of transmitted information and further reduce the processing amount at a decoding apparatus.

Abstract: An acoustic signal encoding device for down-mixing at different ratios to encode a multichannel signal with a small number of channels, and an acoustic signal decoding device for decoding the signal encoded by the acoustic signal encoding device. In these devices, weighting means (103) in the acoustic signal encoding device (100) weights input signals of two channels individually according to a down-mixing coefficient thereby to calculate the level difference of the signals of two channels weighted by a level difference calculation unit (104). A separating unit (202) in the acoustic signal decoding device (200) separates the down-mixed signals into signals of two channels with the level difference information weighted.

Abstract: The invention provides a downsized processing machine, equipped with a machine tool and a multijoint robot. The processing machine 100 comprises a machine tool 10, a workpiece pallet 70 and a gantry rail 120 disposed above the machine tool 10. A multijoint robot 130 traveling on the gantry rail 120 is equipped with a dual-partitioning electron system speed monitoring function. Through use of this function, the range of movement D2 of the multijoint robot 130 is controlled to fit within the inner side of a front cover 11 of the machine tool 10, according to which the safety barrier 180 can be downsized. This arrangement enables the operator to open or close a door 12 of the machine tool 10 and to manipulate a control panel 16 even when the multijoint robot is in operation, and the footprint of the machine can be minimized.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: A protection circuit module for rechargeable batteries includes a substrate, electronic components mounted on the obverse surface of the substrate for forming a protection circuit for rechargeable batteries, a housing mounted on the obverse surface of the substrate and including a top wall partially defining an enclosure for accommodating the electronic components, and a resin sealer loaded in the enclosure of the housing for sealing the electronic components. The top wall of the housing includes an opening for loading the resin sealer into the housing.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) 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.

Abstract: The present invention provides a tool holder interchangeably attached to a machine tool such as a combined machining lathe. A combined machining lathe 1 comprises main spindles 20 and 30 for gripping a workpiece, and a machining main spindle head 50 which rotates around a B axis. There are provided stockers S1 and S2 inside the lathe, and tool holders 200 and 300 are respectively stored in the stockers S1 and S2. The tool holder 200 comprises a taper shank portion 230 of a main shaft in a housing 210 and four pull studs PS2 around the taper shank portion 230 of a main shaft, and is gripped by collet chucks of the machining main spindle head 50. Rotation of the taper shank portion 230 of a main shaft is transmitted to a shaft 250 via a gear mechanism, to drive a drill tool 252 to drill a deep hole.

Abstract: An encoding and decoding apparatus quantizes LSP (Line Spectrum Pair) parameters, which are characteristics parameters of spectrum information included in a voice signal, with high accuracy and stability. This encoding and decoding apparatus comprises a first quantizer in which quantization is performed independently in the unit of one frame, and a second quantizer which uses a correlation between adjacent frames. An error comparator compares quantization errors produced by the first quantizer and quantization errors produced by the second quantizer to select the one quantizer whose quantization errors are less than that of the other quantizer. The first quantizer produces a highly accurate quantization with stability regardless a condition of an input voice signal, while the second quantizer produces a highly accurate quantization when input voice signal stays quasi-stationary.