Abstract: In a packet receiving method and device which convert a voice packet received into a voice, a receiving packet buffer temporarily stores a voice packet received; a plurality of parameter information monitors respectively determine different buffer adjustment values for determining a buffering amount of the receiving packet buffer based on one or more pieces of parameter information obtained from the voice packet temporarily stored; a buffer adjustment value determiner determines a receiving buffer adjustment value from the plural buffer adjustment values; and a buffer controller controls the buffering amount based on the receiving buffer adjustment value.

Abstract: A decoding apparatus includes a unit decoding and inversely quantizing coded data to obtain frequency domain audio signal data, a unit computing from the coded data one of the number of scale bits composed of the number of bits corresponding to the scale value of the coded data and the number of spectrum bits composed of the number of bits corresponding to the spectrum value of the coded data, a unit estimating a quantization error of the frequency domain audio signal data based on one of the number of scale bits and the number of spectrum bits of the coded data, a unit computing a correction amount based on the estimated quantization error and correct the frequency domain audio signal data obtained by the frequency domain data obtaining unit based on the computed correction amount, and a unit converting the corrected frequency domain audio signal data into the audio signal.

Abstract: An audio encoding apparatus that encodes audio signals of a plurality of channels, includes an adaptive bit allocation control unit that adaptively controls a number of encoding bits assigned to the audio signal of each channel in accordance with perceptual entropy of the audio signal of each of the channels, a fixed bit allocation control unit that fixedly controls the number of encoding bits assigned to the audio signal of each of the channels in predetermined allocations, and a channel encoding unit that encodes the audio signal of each of the channels based on the number of adaptive allocation bits assigned by the adaptive bit allocation control unit and the number of fixed allocation bits assigned by the fixed bit allocation control unit.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.

Abstract: An encoding apparatus converts an input signal into a frequency-domain spectrum signal, divides the converted spectrum signal into an arbitrary number of segments with respect to a time axis and a frequency axis, calculates a spectrum power of each segment and a feature parameter that represents a feature of the corresponding spectrum power, calculates a masking threshold using the calculated spectrum power of each segment, detects a segment having a spectrum power equal to or less than the calculated masking threshold, corrects the spectrum power of the detected segment, and encodes both the spectrum power of the corrected segment and the calculated parameter.

Abstract: A decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: A voice communication system having, on a transmission side, a data embedding apparatus provided with an embedding allowability judgment unit (41) calculating an analysis parameter with respect to an input audio signal and judging based on the analysis parameter whether there is a part of the input audio signal allowing embedding of data and an embedding unit (42) outputting an audio signal having the data embedded in the allowable part when the result of judgment of the embedding allowability judgment unit is data can be embedded and outputting the audio signal as is when the result of judgment of the embedding allowability judgment unit is data cannot be embedded and having, on the receiving side, a data extraction apparatus provided with a data extraction apparatus extracting data by a reverse operation is provided, whereby data can be embedded in voice signals without causing an unallowable change in audio quality and a drop in amount of embedded data due to embedding data in parts unsuitable for embeddin

Abstract: In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.

Abstract: A code separation/decoding unit restores a vocal tract characteristic sp1 and a vocal source signal r1. A vocal tract characteristic modification unit modifies the vocal tract characteristic sp1 and outputs the modified vocal tract characteristic sp2. In this method, an emphasized vocal tract characteristic sp2 is generated to output by applying formant emphasis, using amplification ratios calculated based on estimated formants, directly to the vocal tract characteristic sp1 for instance. A signal synthesis unit synthesizes the modified vocal tract characteristic sp2 and the vocal source signal r1 to generate and output an output voice, s.

Abstract: A sensing apparatus which controls the detection timing of a sensor by a prescribed guidance signal includes a guidance signal analyzing unit for analyzing the prescribed guidance signal, and a sensing control unit for determining, based on the result of the detection, an effective detection period during which an output of the sensor is effective.

Abstract: It is so arranged that a voice code can be converted even between voice encoding schemes having different subframe lengths. A voice code conversion apparatus demultiplexes a plurality of code components (Lsp1, Lag1, Gain1, Cb1), which are necessary to reconstruct a voice signal, from voice code in a first voice encoding scheme, dequantizes the codes of each of the components and converts the dequantized values of code components other than an algebraic code component to code components (Lsp2, Lag2, Gp2) of a voice code in a second voice encoding scheme. Further, the voice code conversion apparatus reproduces voice from the dequantized values, dequantizes codes that have been converted to codes in the second voice encoding scheme, generates a target signal using the dequantized values and reproduced voice, inputs the target signal to an algebraic code converter and obtains an algebraic code (Cb2) in the second voice encoding scheme.

Abstract: A disclosed encoding device converts an audio signal into frequency spectra, determines allowable error powers with respect to bands divided by the frequency of the audio signal by a predetermined with, detects a tonal frequency spectrum from the frequency spectra, and detects a band containing the frequency spectrum. Using the detection result and the allowable error powers, the encoding device performs correction such that allowable error powers determined by a power determining unit with respect to bands adjacent to the band detected by a detecting unit become smaller than the powers of the frequency spectra with respect to the adjacent bands, and quantizes each of frequency spectra having greater powers than the corrected allowable error powers.

Abstract: According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component.

Abstract: An exhaust manifold made of heat-resistant cast steel comprising pluralities of flanges each having a hole connected to each exhaust port of a cylinder head of an engine with bolts, pluralities of ports connected to the flanges, and a convergence portion in which the ports are converging, the thickness of the flanges being 80-150% of that of the ports.

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: When creating SBR data in a the low-resolution mode, an encoding device divides a high-frequency component of input audio data being encoded by SBR method into a high-frequency band and a low-frequency band, and calculates an average high-frequency power value that indicates the average value of the power in the high-frequency band of the audio data, as well as an average low-frequency power value that indicates the average value of the power in the low-frequency band of the audio data. The encoding device then compares the average high-frequency power value and the average low-frequency power value, selecting the smaller of the two. The encoding device then corrects the power of the high-frequency component of the signal being encoded by the SBR method so that it equals the selected average power value.

Abstract: An articulated robot includes a first arm which is horizontally angularly movable, a sectorial support plate coaxial with a center about which the first arm is angularly movable, an auxiliary arm parallel to the first arm, and a joint member connected to respective distal ends of the first arm and the auxiliary arm. An arcuate rail is mounted on the support plate. The first arm, the auxiliary arm, and the joint member make up a parallel link mechanism. The arcuate rail engages an engaging assembly mounted on an upper surface of the first arm. A second arm is angularly movably connected to the joint member, and a third arm is angularly movably connected to the distal end of the second arm. An end effector for attracting a workpiece is connected to the distal end of the third arm.

Abstract: To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.