Abstract: In an audio signal expanding/compressing apparatus adapted to expand or compress, in a time domain, a plurality of channels of audio signals by using similar waveforms, a similar-waveform length detection unit calculates similarity of the audio signal between two successive intervals for each channel, and detects a similar-waveform length of the two intervals on the basis of the similarity of each channel.

Abstract: An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.

Abstract: A speaker of encoded speech data recorded in a semiconductor storage device in an IC recorder is to be retrieved easily. An information receiving unit 10 in a speaker retrieval apparatus 1 reads out the encoded speech data recorded in a semiconductor storage device 107 in an IC recorder 100. A speech decoding unit 12 decodes the encoded speech data. A speaker frequency detection unit 13 discriminates the speaker based on a feature of the speech waveform decoded to find the frequency of conversation (frequency of occurrence) of the speaker in a preset time interval. A speaker frequency graph displaying unit 14 displays the speaker frequency on a picture as a two-dimensional graph having time and the frequency as two axes.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: The present invention relates to a data processing apparatus, a method and apparatus for encoding, a method and apparatus for decoding, and a program, that allow a reduction in an algorithm delay. An interpolator 51 produces interpolated PCM data by performing R-times oversampling on original PCM data. A frame encoder 54 fetches a predetermined number of samples of the oversampled data as one frame, encodes the oversampled data on a frame-by-frame basis, and outputs resultant encoded data. A frame decoder 55 decodes the encoded data on a frame-by-frame basis at a rate R times higher than a predetermined normal rate. A decimator 56 decimates data obtained as a result of the decoding such that the number of samples is reduced to 1/R of the number of sampled included in the original data. The present invention is applicable, for example, to an IP telephone system.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: A process of identifying a speaker in coded speech data and a process of searching for the speaker are efficiently performed with fewer computations and with a smaller storage capacity. In an information search apparatus, an LSP decoding section extracts and decodes only LSP information from coded speech data which is read for each block. An LPC conversion section converts the LSP information into LPC information. A Cepstrum conversion section converts the obtained LPC information into an LPC Cepstrum which represents features of speech. A vector quantization section performs vector quantization on the LPC Cepstrum. A speaker identification section identifies a speaker on the basis of the result of the vector quantization. Furthermore, the identified speaker is compared with a search condition in a condition comparison section, and based on the result, the search result is output.

Abstract: An audio signal expansion and compression method for expanding and compressing an audio signal in a time domain, includes the steps of setting an initial value of a signal comparison length of a first comparison interval and a second comparison interval, used for detection of two similar waveforms in the audio signal, equal to or larger than a minimum waveform detection length, determining an interval length of the two similar waveforms while changing a shift amount of the first comparison interval and the second comparison interval so that the shift amount does not exceed the signal comparison length, and expanding or compressing the audio signal in the time domain on the basis of the interval length of the two similar waveforms.

Abstract: An audio-signal time-axis expansion/compression method for subjecting an audio signal to time-axis expansion/compression at a time domain includes the steps of: cross-fade-signal generating wherein a first period and a second period which are similar within the audio signal are employed to generate the cross-fade signal of the first period signal and the second period signal; correction-signal generating wherein the difference signal between the first period signal and the second period signal is subjected to time-axis reversal, and is multiplied with a window function to generate a correction signal; and connection-waveform generating wherein the cross-fade signal and the correction signal are added to generate a connection waveform for subjecting the audio signal to time-axis expansion/compression at the time domain.

Abstract: In a similarity vector detecting apparatus (2), vector transform units (20), (21) implement transform by sequential matrix, Discrete Cosine Transform, Discrete Fourier Transform, Walsh-Hadamard Transform, or Karhunen-Lueve Transform to registered vector g and input vector f. A hierarchical distance calculating unit (23) performs, in a hierarchical manner, distance calculation between two vectors in order from vector component having high significance, i.e., component having large dispersion or eigen value in the above-described transform operations, or from low frequency component. Further, in the case where it is judged at a threshold value judgment unit (24) that integrated value of distances calculated up to a certain hierarchy is above threshold value S of distance, only output indicating that the integrated value is above the threshold value S is provided to truncate distance calculation.

Abstract: An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.

Abstract: An audio signal processing method for repairing an anomalous state such as noise, a discontinuity, and a break of sound, comprising detecting the anomalous state of an audio signal, deleting the audio signal in the anomalous segment, deducing the correct audio signal by referring to the waveform of the audio signal before and after the deleted segment, generating a repair signal for repairing the signal in the deleted segment based on the deduced result, inserting the repair signal into the deleted segment, and connecting it to the audio signal before and after the deleted segment.

Abstract: A highlight portion is detected to a high accuracy from acoustic signals in say an event, and an index is added to the highlight portion. In an acoustic signal processing apparatus 10, a candidate domain extraction unit 13 retains a domain, a length of which with short-term amplitudes as calculated by an amplitude calculating unit 11 not being less than an amplitude threshold value is not less than a time threshold value, as a candidate domain. A feature extraction unit 14 extracts sound quality featuring quantities, relevant to the sound quality, from the acoustic signals, to quantify the sound quality peculiar to a climax. A candidate domain evaluating unit 15 calculates a score value, indicating the degree of the climax, using featuring quantities relevant to the amplitude or the sound quality for each candidate domain, in order to detect a true highlight domain, based on the so calculated score value.

Abstract: In a quantization step information encoding unit, an average value of the quantization step information is found in an approximate shape extraction unit (20), first of all, from one set of a given number of unitary quantization units to another. In an approximate shape encoding unit (21), the approximate shape information is vector-quantized. In a residual signal computing unit (22), the residual signals between the quantization step information and the quantized approximate shape vector are computed. In a residual signal encoding unit (23), the residual signals are variable length encoded, and the so encoded residual signals and the vector quantized approximate shape information are output.

Abstract: A video processing apparatus includes: face-area detection means for detecting a face area included in a frame forming video data; trace-generation means for generating a frame identification corresponding to a start and an end of a trace including, as a unit, a set of frames from an appearance of the face area to a disappearance on the basis of the detection; representative face-area information generation means for selecting a representative face area from the face area included in frames forming the trace and generating representative face-area information representing contents of the representative face area; and video-data appended information generation means for generating video-data appended information relating the frame identification corresponding to a start and an end of the trace to the representative face-area information for the video data.

Abstract: The present invention relates to a data processing apparatus, a method and apparatus for encoding, a method and apparatus for decoding, and a program, that allow a reduction in an algorithm delay. An interpolator 51 produces interpolated PCM data by performing R-times oversampling on original PCM data. A frame encoder 54 fetches a predetermined number of samples of the oversampled data as one frame, encodes the oversampled data on a frame-by-frame basis, and outputs resultant encoded data. A frame decoder 55 decodes the encoded data on a frame-by-frame basis at a rate R times higher than a predetermined normal rate. A decimator 56 decimates data obtained as a result of the decoding such that the number of samples is reduced to 1/R of the number of sampled included in the original data. The present invention is applicable, for example, to an IP telephone system.

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.

Abstract: A method, an apparatus and a program for detecting similarity between two time-domain signals, at a high speed, and a recording medium on which is recorded such program. The process for detecting the similarity is split into a stage for calculating the similarity between feature vectors at each discrete time t and a stage for counting the number of similar vectors in the entire time series. In the calculating stage, the distances between the components for the same frame numbers of the feature vector time series gt, ft are hierarchically integrated. If the ultimate integrated value is lower than the distance threshold value S, the feature vector time series gt, ft are determined to be similar. If the integrated value exceeds the threshold value part way in the course of the hierarchical integration, the feature vector time series gt, ft are determined to be dissimilar and the integrating calculations for the remaining components are discontinued.

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.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.