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Showing papers by "Masayuki Nishiguchi published in 2001"


Patent
23 Aug 2001
TL;DR: In this article, an apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis.
Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit. A coefficient quantization section quantizes the coefficients from the orthogonal transform circuit section according to the number of allocated bits from the bit allocation calculation section.

45 citations


Patent
26 Apr 2001
TL;DR: In this article, a signal processing device and signal processing method can accurately detect CMs out of broadcast signals and also be used for storing, accessing, retrieving and viewing/listening to CMs.
Abstract: A signal processing device and a signal processing method can accurately detect CMs out of broadcast signals and also be used for storing, accessing, retrieving and viewing/listening to CMs. CM detecting section ( 202 ) detects CM sections out of a broadcast signal. CM extracting section ( 201 ) extracts the signal part for the CM section from the broadcast signal on the basis of the CM section detection signal ( 202 a ) of the CM detecting section ( 202 ) and CM characteristics extracting section ( 203 ) extracts the characteristic value of the CM. Then, CM recording section ( 205 ) records the signal of the CM section and the characteristic value. CM index generating section ( 206 ) generates CM index information by using the signal of the CM section and the characteristic value and characteristics comparing section ( 204 ) determines agreement/disagreement of CMs. CM viewing section ( 208 ) displays CM retrieval information and reproduce the CM signal from the CM recording scanning ( 205 ) in response to the instruction of the user ( 209 ). As a result, the user ( 209 ) can view, listen to and retrieve the desired CM.

28 citations


Patent
21 Aug 2001
TL;DR: In this article, a search processor determines whether a search query has been received and then deletes a content having the degree of similarity equal to or smaller than a predetermined threshold, based on fuzzy information.
Abstract: A search processor determines whether a search query has been received. When it is determined that the search query has been received, the search processor acquires the search query, calculates the degree of similarity of the search query, and deletes a content having the degree of similarity equal to or smaller than a predetermined threshold. When no search query has been received, the search processor determines whether the number of contents in a candidate list is equal to or larger than a predetermined number. When it is determined that the number of contents is equal to or larger than the predetermined number, the search processor issues an additional question. A content is thus searched in an interactive fashion based on fuzzy information.

25 citations


Patent
26 Jul 2001
TL;DR: In this paper, an information providing system includes a terminal device and an identification device which are linked with each other via a network, and when a music signal is transmitted from the terminal device to the identification device via the network, an IDC extracting unit extracts an identification code corresponding to a category which is inserted as a digital watermark in the music signal.
Abstract: An information providing system includes a terminal device and an identification device which are linked with each other via a network. When a music signal is transmitted from the terminal device to the identification device via the network, an identification code extracting unit extracts an identification code corresponding to a category which is inserted as a digital watermark in the music signal. A feature extracting unit extracts features of the music signal. A database search unit compares the features of information which belongs to the category corresponding to the identification code extracted by the identification code extracting unit with the features extracted by the feature extracting unit. If a match is found, the music title stored in the database search unit is read according to the features, and is then output to the terminal device.

21 citations


Patent
25 Jan 2001
TL;DR: In this paper, an apparatus and method for performing transform encoding, in which time domain samples can overlap one another by any desired percentage and can be added so that signals may be reproduced completely.
Abstract: An apparatus and method for performing transform encoding, in which time domain samples can overlap one another by any desired percentage and can be added so that signals may be reproduced completely. In the apparatus and method, the linear/nonlinear prediction analysis section 3 receives an audio signal from the input terminal 2 and effectuates linear or nonlinear prediction on the audio signal, generating a prediction residual. The constancy inferring section 7 infers the constancy of the audio signal. The block-length determining section 8 determines the length of an MDCT block from the constancy of the input signal, which the section 7 has inferred. The MDCT section 5 receives M time domain samples supplied from the buffer 4 and having the prediction residual. The MDCT section 5 applies the block length determined by the section 8 , performing MDCT transform on the time domain samples, thus generating MDCT coefficients. The quantization section 6 quantizes the MDCT coefficients.

13 citations


Patent
31 Oct 2001
TL;DR: In this paper, a TV-broadcast signal is supplied to a demodulator, which demodulates the audio signal and video signal, both contained in the TV broadcast signal, at intervals of 15 seconds, 30 seconds and 60 seconds.
Abstract: A tuner 1 receives a TV-broadcast signal from an antenna 6. The TV-broadcast signal is supplied to a demodulator 2. The demodulator 2 demodulates the audio signal and video signal, both contained in the TV-broadcast signal. A CM-detecting section 4 is provided. In the section 4, characteristic patterns are generated at intervals of 15 seconds, 30 seconds and 60 seconds, from the audio and video signals. Then, CM candidates are detected in accordance with the characteristic patterns. Characteristic data representing the characteristics of a CM is extracted from each CM candidate or the audio and video signals preceding and following the CM candidate. The section 4 detects a CM from the characteristic data thus extracted.

12 citations


Patent
26 Jan 2001
TL;DR: In this article, the audio signals or video signals of a broadcasting program are made into frames (short time frame, medium time frame and long time frame) in respectively hierarchically different time zones and the audio signal or video signal in the frames are respectively processed.
Abstract: PROBLEM TO BE SOLVED: To identify various kinds of broadcasting programs while exactly and surely sorting contents or categories. SOLUTION: In a short-time signal processing part 3, a medium-time signal processing part 4 and a long-time signal processing part 5, the audio signals or video signals of a broadcasting program are made into frames (short-time frame, medium-time frame and long-time frame) in respectively hierarchically different time zones and the audio signals or video signals in the frames are respectively processed. A stream processing part 6 discriminates the category of the broadcasting program from a program category sorting vector yr and a hierarchy update coefficient ξ obtained from the long-time signal processing part 5 on the final stage.

10 citations


Patent
22 Jun 2001
TL;DR: In this paper, a consumer terminal detects these CMs from the aired signals and connects to an access site information furnishing device based on the airing time and the airing channel of the detected CMs, for e.g., an article of commerce owning the detailed information pertinent to the CM.
Abstract: It is targeted that CMs by e.g., television broadcast be detected automatically, an access site to detailed contents of a specified one of the CMs be acquired automatically, and connection be made automatically to the access site to enable accessing to the detailed information. A broadcasting station 302 sends out airing signals containing CMs. A consumer terminal ( 303 ) detects these CMs from the aired signals and connects to an access site information furnishing device ( 305 ) based on the airing time and the airing channel of the detected CM to acquire the access site information of the distribution terminal ( 301 ) for e.g., an article of commerce owning the detailed information pertinent to the CM. Based on the access site information, the consumer terminal ( 303 ) accesses the distribution terminal ( 301 ) to acquire the detailed information pertinent to the CM.

10 citations


Patent
07 Mar 2001
TL;DR: In this paper, an audio signal processing apparatus and method using pitch information to change a length of predictive residual signals while maintaining continuity and thereby enabling conversion of a reproduction speed without changing a pitch and enabling a conversion of speed by a small amount of calculation, comprising shortening or extending residual signals on a time axis.
Abstract: An audio signal processing apparatus and method using pitch information to change a length of predictive residual signals while maintaining continuity and thereby enabling conversion of a reproduction speed without changing a pitch and enabling a conversion of speed by a small amount of calculation, comprising shortening or extending residual signals on a time axis while maintaining pitch information, cutting out signals and connecting of different pitch sections in the respective frames based on resemblance of signals at the time of shortening, and extending predictive residual signals in respective frames by extrapolation at the time of extension. An audio signal compressed or expanded on the time axis can be reproduced without changing the pitch by synthesizing an audio signal by an LPC synthesis filter based on the generated new predictive residual signals.

9 citations


Patent
15 Jun 2001
TL;DR: In this paper, the mean value of quantization precision information is calculated for every number of unit quantization units by an outline extracting part 20 so that the mean values can be turned into outline information, and the residual signal is variable length-encoded by a residual signal encoding part 23.
Abstract: PROBLEM TO BE SOLVED: To efficiently compress quantization precision information and normalization information SOLUTION: In a quantization precision information encoding part, the mean value of quantization precision information is calculated for every number of unit quantization unit by an outline extracting part 20 so that the mean value can be turned into outline information The outline information is vector quantized by an outline encoding part 21, and the residual of the quantization precision information and the quantized outline vector is calculated by a residual signal calculating part 22 The residual signal is variable length-encoded by a residual signal encoding part 23, and the encoded residual signal and the index of the vector quantized codebook are outputted COPYRIGHT: (C)2003,JPO

7 citations


Patent
30 May 2001
TL;DR: In this paper, a wide-band LPC combining section was proposed to improve the accuracy of an excitation source for a band-spreading apparatus and to generate a wideband signal having no gaps.
Abstract: In order to improve the accuracy of an excitation source for a band-spreading apparatus and to generate a wide-band signal having no gaps, an a band-widening section generates a prediction coefficient alpha W of a wide-band speech signal from a prediction coefficient alpha N of a narrow-band speech signal. An oversampling apparatus oversamples a narrow-band speech signal sndN. An interpolation section generates an adaptive signal excPW of a wide-band speech signal from an adaptive signal excPN of the narrow-band speech signal. A zero-filling section generates a noise signal of a wide-band speech signal from a noise signal excNN of the narrow-band speech signal. A noise addition section adds a noise signal which is a gap of the wide-band speech signal and generates a noise signal excNW. An adder generates an excitation source excW for the wide-band speech signal from the adaptive signal excPW and the noise signal excNW of the wide-band speech signal. A wide-band LPC combining section generates a wide-band speech signal. A band suppression section suppresses a frequency band contained in the narrow-band speech signal within the wide-band speech signal. An adder outputs a wide-band speech signal sndW from the wide-band speech signal and the oversampled narrow-band speech signal.

Patent
16 May 2001
TL;DR: In this article, a signal processing apparatus comprises a time block splitting section 3 for splitting an audio signal into blocks that are typically 1 second long, a feature extracting section 4 for extracting a characteristic quantity of 18 degrees on the signal attribute from the audio signal in each block and a vector quantizing section 5 for carrying out an operation of categorical classification for the audio signals of each block by means of a VQ code book 8 and a characteristic vector formed from the characteristic quantity.
Abstract: The input signal can be quickly and accurately classified and a descriptor can be generated according to the result of classification. Then, the input signal can be retrieved on the basis of the result of classification or the descriptor. A signal processing apparatus comprises a time block splitting section 3 for splitting an audio signal into blocks that are typically 1 second long, a feature extracting section 4 for extracting a characteristic quantity of 18 degrees on the signal attribute from the audio signal in each block and a vector quantizing section 5 for carrying out an operation of categorical classification for the audio signal of each block by means of a vector quantization technique that uses a VQ code book 8 and a characteristic vector formed from the characteristic quantity of 18 degrees. The vector quantizing section 5 outputs a classification label obtained as a result of the categorical classification and a descriptor indicating the reliability of the label. If a signal retrieving operation is conducted in the downstream, the result of the classification or the descriptor is used for the signal retrieval.

Patent
10 Jan 2001
TL;DR: In this article, a method for detecting the noise domain is provided in which the value employed for finding the threshold value Th 1 for noise domain discrimination is calculated using the RMS value of the current frame or the value th of the previous frame multiplied by the coefficient a, whichever is smaller.
Abstract: A method for detecting the noise domain is provided in which the value th employed for finding the threshold value Th1 for noise domain discrimination is calculated using the RMS value of the current frame or the value th of the previous frame multiplied by the coefficient a, whichever is smaller, and the coefficient a is changed over depending on the RMS value of the current frame. Noise domain discrimination by an optimum threshold value responsive to the input signal may be achieved without producing mistaken judgment even on the occasion of noise level fluctuations.

Patent
29 Jun 2001
TL;DR: In this article, an audio signal processing method by which the mismatching state of audio signals such as noise, discontinuity, sound discontinu ity, etc. can be restored is proposed.
Abstract: PROBLEM TO BE SOLVED: To provide an audio signal processing method by which the mismatching state of audio signals such as noise, discontinuity, sound discontinu ity, etc. can be restored. SOLUTION: In the audio signal processing method, the mismatching state of the audio signals is detected. When the mismatching state is detected, the audio signals in a mismatching section are removed and the removed audio signals are inferred by referring to the waveforms of the audio signals before and after the removed section. Then restore signals which restore the signals in the removed section are generated on the basis of the inferred results, inserted into the removed section, and connected to the audio signals before and after the removed section.

Patent
24 Apr 2001
TL;DR: In this article, a TV-broadcast signal is supplied to a demodulator, which demodulates the audio signal and video signal, both contained in the TV broadcast signal, at intervals of 15 seconds, 30 seconds and 60 seconds.
Abstract: A tuner 1 receives a TV-broadcast signal from an antenna 6. The TV-broadcast signal is supplied to a demodulator 2. The demodulator 2 demodulates the audio signal and video signal, both contained in the TV-broadcast signal. A CM-detecting section 4 is provided. In the section 4, characteristic patterns are generated at intervals of 15 seconds, 30 seconds and 60 seconds, from the audio and video signals. Then, CM candidates are detected in accordance with the characteristic patterns. Characteristic data representing the characteristics of a CM is extracted from each CM candidate or the audio and video signals preceding and following the CM candidate. The section 4 detects a CM from the characteristic data thus extracted.

Patent
25 Jan 2001
TL;DR: In this article, an M-point DFT is applied to the data fetched at step S 2 above, followed by an FFT with N/M (=2 n ) points within the range of 0≦k
Abstract: First, at step S 1 , i=0 is set. At step S 2 , data comprising M samples is fetched. At step S 3 , an M-point DFT is applied to the data fetched at step S 2 above. At step S 4 , an obtained y(k) is multiplied by a twist coefficient w(i, k). The result is placed in y(k). At step S 5 , the value in y(k) is overwritten to an array x which contains original data. The above processing is repeated N/M times through steps S 6 and S 7 until all the input data is processed. At step S 9 , an FFT with N/M (=2 n ) points is performed within the range of 0≦k

Patent
22 Jun 2001
TL;DR: In this paper, a consumer terminal detects these CMs from the aired signals and connects to an access site information furnishing device based on the airing time and the airing channel of the detected CMs, for e.g., an article of commerce owning the detailed information pertinent to the CM.
Abstract: It is targeted that CMs by e.g., television broadcast be detected automatically, an access site to detailed contents of a specified one of the CMs be acquired automatically, and connection be made automatically to the access site to enable accessing to the detailed information. A broadcasting station 302 sends out airing signals containing CMs. A consumer terminal ( 303 ) detects these CMs from the aired signals and connects to an access site information furnishing device ( 305 ) based on the airing time and the airing channel of the detected CM to acquire the access site information of the distribution terminal ( 301 ) for e.g., an article of commerce owning the detailed information pertinent to the CM. Based on the access site information, the consumer terminal ( 303 ) accesses the distribution terminal ( 301 ) to acquire the detailed information pertinent to the CM.

Patent
17 Jul 2001
TL;DR: In this article, the amplitude of each sub-band signal is smoothed and sampled corresponding to time for smoothing, on the basis of a value F (n, k) of the smoothed sub- band amplitude and a value f (n-1, k), the spectrum change amount D is calculated.
Abstract: PROBLEM TO BE SOLVED: To make appropriately detectable a scene change. SOLUTION: In a step S2, the amplitude of each of sub-band signals decomposed into spectrums in a step S1 is detected. In a step S3, the amplitude of each of sub-band signal is smoothed and sampled corresponding to time for smoothing. In a step S4, on the basis of a value F (n, k) of the smoothed sub- band amplitude and a value F (n-1, k) of the preceding sampled smoothed sub- band amplitude, a spectrum change amount D is calculated. In a step S5, it is decided whether the calculated spectrum change amount D is greater than a prescribed threshold or not and in a step S6, processing based on the decided result is executed. COPYRIGHT: (C)2003,JPO

Patent
16 Feb 2001
TL;DR: In this paper, the authors propose a method to make rapidly reproducible transmitted data after transmission of audio data is requested, where audio data are composed of music score data and object data which are reproduced at a prescribed time based on the music score.
Abstract: PROBLEM TO BE SOLVED: To make rapidly reproducible transmitted data after transmission of audio data is requested. SOLUTION: Audio data are composed of music score data and object data which are reproduced at a prescribed time based on the music score data. If the reproducing time of one music is sixty seconds, the music score data are divided into one thirty, namely the music score data for every two seconds, and the object data to be reproduced are transmitted for every two seconds corresponding to the music score data.

Patent
29 Jun 2001
TL;DR: In this article, the mismatching state in an audio signal is detected and a recovery signal is inserted to the eliminated period to connect the recovery signal to the audio signal before and after the mismatched period.
Abstract: PROBLEM TO BE SOLVED: To provide an audio signal processing method by which mismatching state such as audio interruption does not take place even in high-speed reproduction. SOLUTION: The mismatching state in an audio signal is detected. When the mismatching state is detected, an audio signal for the mismatching period is eliminated and a waveform of the audio signal before and after the eliminated period is referenced to estimate the eliminated audio signal, a recovery signal recovering the signal for the eliminated period is generated based on the estimated result and the recovery signal is inserted to the eliminated period to connect the recovery signal to the audio signal before and after the eliminated period.

Patent
25 Jul 2001
TL;DR: In this article, a vector quantization is used to quantize the LSP information into LPC cepstrum, which is the characteristic amount of speech in coded speech data.
Abstract: PROBLEM TO BE SOLVED: To effectively conduct the recognition processing and the retrieval processing of speakers in a coded speech data with a small amount of calculation and memory capacity. SOLUTION: In an information retrieval device 30, a LSP decoding part 22 extracts only LSP information from coded speech data that are read in each block and decoded. A LPC converting part 23 converts the LSP information into LPC information and a cepstrum converting part 24 converts the LPC information into LPC cepstrum that is characteristic amount of speech. A vector quantizing part 25 gives the LPC cepstrum vector quantization and a speaker recognizing part 26 recognizes the speaker based on a result of the vector quantization. The recognized speaker and a retrieval condition are compared in a condition comparing part 32, whereby a retrieval result is outputted.

Patent
03 Jan 2001
TL;DR: In this paper, a method for detecting the noise domain is provided in which the value employed for finding the threshold value Th 1 for noise domain discrimination is calculated using the RMS value of the current frame or the value th of the previous frame multiplied by the coefficient a, whichever is smaller.
Abstract: A method for detecting the noise domain is provided in which the value th employed for finding the threshold value Th1 for noise domain discrimination is calculated using the RMS value of the current frame or the value th of the previous frame multiplied by the coefficient a, whichever is smaller, and the coefficient a is changed over depending on the RMS value of the current frame. Noise domain discrimination by an optimum threshold value responsive to the input signal may be achieved without producing mistaken judgment even on the occasion of noise level fluctuations.