73 results about "Harmonic structure" patented technology

A common narrow-band speech signal is expanded into a wide-band speech signal. The expanded signal gives the impression of a wide-band speech signal regardless of what type of vocoder is used in a receiver. The robust techniques suggested herein are based on speech acoustics and fundamentals of human hearing. That is the techniques extend the harmonic structure of the speech signal during voiced speech segments and introduce a linearly estimated amount of speech energy in the wide frequency-band. During unvoiced speech segments, a fricated noise may be introduced in the upper frequency-band.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

There is provided a harmonic structure acoustic signal detection device not depending on the level fluctuation of the input signal, having an excellent real time property and noise resistance. The device includes: an FFT unit (200) which performs FFT on an input signal and calculates a power spectrum component for each frame; a harmonic structure extraction unit (201) which leaves only a harmonic structure from the power spectrum component; a voiced feature evaluation unit (210) which evaluates correlation between the frames of harmonic structures extracted by the harmonic structure extraction unit (201), thereby evaluates whether or not the segment is a vowel segment, and extracts the voiced segment; and a speech segment determination unit (205) which determines a speech segment according to the continuity and durability of the output of the voiced feature evaluation unit (210).

A decoding device (100) is a decoding device that generates frequency spectral data from an inputted encoded audio data stream, and includes: a core decoding unit (102) for decoding the inputted encoded data stream and generating lower frequency spectral data representing an audio signal; and an extended decoding unit (104) for generating, based on the lower frequency spectral data, extended frequency spectral data indicating a harmonic structure, which is same as an extension along the frequency axis of the harmonic structure indicated by the lower frequency spectral data, in a frequency region which is not represented by the encoded data stream.

A voice activity detection method in a low SNR environment. The voice activity detection is performed by extracting a long-term spectrum variation component and a harmonic structure as feature vectors from a speech signal and increasing difference in feature vectors between speech and non-speech (i) using the long-term spectrum variation component feature or (ii) using a long-term spectrum variation component extraction and a harmonic structure feature extraction. A correct rate and an accuracy rate of the voice activity detection is improved over conventional methods by using a long-term spectrum variation component having a window length over an average phoneme duration of an utterance in the speech signal. The voice activity detection system and method provides speech processing, automatic speech recognition, and speech output capable of very accurate voice activity detection.

The invention discloses a voice enhancement system and method for a cellphone microphone, and the method comprises the steps: firstly carrying out the time domain noise reduction processing of noise voice signals of a main microphone and an auxiliary microphone; secondly filtering out the linear related noises between the main microphone and the auxiliary microphone; thirdly converting the signals after time domain noise reduction to a frequency domain for frequency domain noise reduction, thereby further filtering the linear uncorrelated noises between the main microphone and the auxiliary microphone. The method determines a voice frequency point and a noise frequency point in the frequency domain through employing the reference sound of voice and a harmonic structure, adjusts the parameters of a frequency domain wiener filter for the voice and noise frequency points, retains the voice frequency point while enabling the wiener filter to precisely filter the noise frequency point, achieves the reduction of voice distortion, and improves the call quality of a cellphone.

A method and system for analyzing patterns in the relationships of notes of an input piece of music. The method comprises generating a set of the most frequently occurring note pitches in ascending pitch order that matches an interval pattern, and detecting out-of-key pitches that lie outside of this interval pattern. One or more potential key sequence bifurcations are identified which represent a list of possible key sequences according to forwards and backwards analysis. By finding patterns of repetition in the chordal sequences that may be generated according to these key sequence bifurcations, a key sequence that allows the most frequently recurring chord sequences may be chosen. Chord sequences may be analyzed by using ghost chords, temporary harmonic structures that are created, updated and finalized over time according to a combination of essential and inessential note fragments. The method further comprises identifying non-harmony pitches according to the analyzed chord sequence.

There is disclosed a noise suppression device capable of improving the noise suppression accuracy while reducing the audio distortion. In this device, a suppression unit suppresses a noise component from the audio power spectrum by using the detection result of the audio-existing band and the noise band in the audio power spectrum including the noise component. A pitch harmonic structure extracting unit (105) extracts a pitch harmonic power spectrum from the audio power spectrum. An audio-existence judgment unit (106) judges whether the audio power spectrum has audio existence according to the extracted pitch harmonic power spectrum. A pitch harmonic structure repair unit (108) repairs the extracted pitch harmonic power spectrum. A per-band audio / noise correction unit (109) corrects the detection result according to the pitch harmonic power spectrum selected according to the result of judgment by the audio-existence judgment unit (106) among the repaired pitch harmonic power spectrum and the extracted pitch harmonic power spectrum.

The invention discloses a nuclear magnetic resonance underground water detection signal noise eliminating method. The method comprises the steps of: recoding response data entry three groups, each of these three sets of data Fourier transform nuclear magnetic resonance to determine the frequency of the data center in each neighborhood contains frequency harmonics, and frequency harmonic structure and the same frequency, and magnetic resonance corresponding data with sine function of the length of the cosine function, and composition and magnetic resonance response data observed signal using independent component analysis for each group observed signals to separate the mixed signal obtained solution for data reconstruction to eliminate frequency harmonic interference, the three groups of removing harmonic frequency data as observational signals, and then use algorithm processing, weaken the remaining random noise. The invention is to eliminate harmonic frequency noise at the same time, the signal will not destroy any details, without laying a reference coil, simple operation, suppress random noise does not require a lot of data, reducing the processing time.

Non-linear asymmetric amplifiers incorporate different non-linear characteristics in single ended or differential configurations to avoid cancellation of non-linear characteristics and to produce the harmonic structures of vacuum tubes. The operating currents of the asymmetrical devices are different to create a bias shift in the coupling capacitance between the asymmetrical devices.

The invention discloses a color optical filter insensitive to an incident angle and a manufacture method thereof. The color optical filter comprises a substrate; a metal film, a medium film and a metal film are successively arranged on the substrate; the three layers of films constitute a harmonic structure; the material of the medium film is made of the silicon-enriched silicon oxide SiOx or silicon-enriched silicon nitride SiNy, wherein the value ranges of the x and the y are as follow:0<x&1t,0<y*&1t;and 4 / 3, The manufacture method for the color optical filter insensitive to the incident angle is simple in preparation, low in cost, and applicable to mass production and batch production. The invention is hopefully and widely applied in the fields of liquid crystal display, color printing, sensor detection, anti-counteracting, etc.

The invention discloses a double-talk detection and echo cancellation method based on zero-crossing rate, comprising the following steps: S1, zero-crossing rate calculation and double-talk monitoring: different echo cancellation strategies are adopted for double-talk and single-talk scenarios; S2, echo filter estimation and echo cancellation; there is difference between an echo signal collected by a microphone and a far-end signal because of room impulse response, and the echo component is eliminated from the signal collected by the microphone; and S3, recovery of a target speech harmonic structure. A harmonic structure analysis method is used to compensate for the missing harmonic components of near-end speech to further inhibit speech distortion. Compared with the traditional echo cancellation technology, double-end monitoring is realized based on the zero-crossing rate, and elimination of target speech is avoided. Based on the criterion of frequency-domain minimum mean square error, an echo cancellation filter converges rapidly. By adopting a frequency-domain parallel processing framework, the complexity is low. A speech distortion inhibition module is added, and the distortion degree of target speech is reduced.

A signal portion is extracted per frame having a specific duration from an input signal, thus generating a per-frame input signal. The per-frame input signal in the time domain is converted into a per-frame input signal in the frequency domain, thereby generating a spectral pattern of spectra. Peak spectra having peaks are detected in the spectral pattern. A harmonic spectrum is determined, in the peak spectra, having a harmonic structure showing a relationship between a fundamental pitch and a harmonic overtone.

The invention relates to a voice activity detection system based on the fundamental frequency and a calculation method thereof. All possible positions where a fundamental frequency appears are acquired through a fundamental frequency extraction algorithm, and the fundamental frequency is determined by cost. However, because the fundamental frequency may be interfered by low-frequency noise, whether a point is voice is judged by aid of the fact that a position where a fundamental frequency appears has a harmonic structure. Meanwhile, the speed of voice activity detection is increased and the detection accuracy is improved according to adaption of background energy. According to the voice activity detection system based on the fundamental frequency and the calculation method thereof of the invention, the voice activity detection system is enabled to have high robustness under the condition of low signal-to-noise ratio. When noise is difficult to distinguish in a time domain, the method enables noise to be correctly distinguished in a frequency domain according to significantly different characteristics of spectral distribution of noise signals and voice signals from time-domain distribution. The method can be widely applied to the field of voice signal processing.

A voice detecting apparatus includes a first determining unit to determine that human voice has been input if a signal component having a harmonic structure is detected from an input voice signal; a second determining unit to determine that human voice has been input if a frequency center-of-gravity of the input voice signal is within a predetermined range; a noise level storing unit to store a noise level; a third determining unit to determine that human voice has been input if the ratio of the power of the input voice signal to the noise level is above a predetermined threshold; a final determining unit configured to finally determine whether human voice has been input based on determination results of the first to third determining units; and a noise level updating unit configured to update the noise level if the final determining unit determines that human voice has not been input.

An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoding section (202) encodes an input signal and generates first encoded information, a first layer decoding section (203) decodes the first encoded information and generates a first decoded signal, a characteristic judging section (206) analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoding section (207) changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information .

A noise suppression device includes: a power spectrum calculator converting an input signal of time domain into power spectra of frequency domain; a voice / noise determination unit determining whether the power spectra indicate voice or noise; a noise spectrum estimation unit estimating noise spectra of the power spectra; a period component estimation unit analyzing a harmonic structure constituting the power spectra and estimating periodical information about the power spectra; a weighting coefficient calculator calculating a weighting coefficient for weighting the power spectra; a suppression coefficient calculator calculating a suppression coefficient for suppressing noise included in the power spectra; a spectrum suppression unit suppressing amplitude of the power spectra in accordance with the suppression coefficient; and an inverse Fourier transformer converting the power spectra output by the spectrum suppression unit into a signal of time domain to generate a noise-suppressed signal.

The invention discloses a polarization switching surface based on a sub-wavelength harmonic structure. The polarization switching surface is mainly obtained in a mode that a sub-wavelength harmonic unit is subjected to a horizontal cycle array and a longitudinal cycle array, the sub-wavelength harmonic unit is of a square structure and comprises a top-layer metal copper sheet, an upper medium plate, a middle grounded metal copper sheet, a lower medium plate and a bottom-layer metal copper sheet, and the middle grounded metal copper sheet is located between the upper medium plate and the lower medium plate; penetrating through holes are formed in the five layers of materials of the sub-wavelength harmonic unit, the top-layer metal copper sheet and the bottom-layer metal copper sheet are oppositely arranged, and the polarization rotation directions of the top-layer metal copper sheet and the bottom-layer metal copper sheet are contrary to each other. The polarization switching surface is low in cost, simple in structure, thin, light, good in conversion performance and capable of flexibly controlling the polarization state of electromagnetic waves and conducting selective electromagnetic wave transmittance and reflectance and is widely applied to the fields of imaging, antennas, communication systems, electromagnetic confrontation and governance, military invisibility and the like.

An apparatus and method for extracting a predetermined non-harmonic structured spectral component contained in an audio signal. Then, the extracted predetermined spectral component is increased or decreased. In this process, the spectrum of the audio signal is calculated by frequency analysis, so that a spectrum component corresponding to the predetermined non-harmonic structured spectral component is extracted and then increased or decreased. The extraction of the predetermined non-harmonic structured spectral component is performed with reference to a spectral component of a template stored in advance. In this process, the spectral component of the template is adapted in such a manner that the difference between the extracted spectral component and the spectral component of the template goes below or at a predetermined value. This allows the audio-signal contained predetermined non-harmonic structured spectral component to be independently increased or decreased without an influence on other spectral components.

The invention provides a heterogeneous microphone automatic gain calibration method and system. The method comprises the following steps: acquiring a microphone signal, performing channel noise estimation on the signal, calculating a noise suppression control factor and correcting the signal to obtain a first target signal; performing reverberation estimation on the first target signal, calculating a reverberation control factor and correcting the first target signal to obtain a second target signal; and performing gain control on the second target signal to obtain a third target signal. The method and system can effectively suppress the channel noise without causing excessive distortion to the target signal, and repair the harmonic structure to avoid signal distortion, may adaptively estimate the reverberation time, ensure the accuracy of the reverberation suppression, accurately restore frequency band signals by calculating the amplitude gain in each frequency band, and avoids a defect that same-proportion expansion is achieved but the signal resolution in each band is not improved.

In a pitch estimation apparatus, a function estimation part estimates a fundamental frequency probability density function of an audio signal by repeating a weight calculation process and an estimated shape specification process. The weight calculation process calculates a weight of each tone model of each fundamental frequency based on an estimated shape of each tone model of each fundamental frequency. The estimated shape indicates a degree of dominancy of a corresponding tone model in a total harmonic structure of the audio signal. The estimated shape specification process specifies each estimated shape of each tone model based on an amplitude spectrum of the audio signal, the harmonic structure of each tone model and the weight of each tone model. A similarity analysis part calculates a similarity index value indicating a degree of similarity between each tone model and corresponding estimated shape. A weight correction part reduces a weight of a tone model of a certain fundamental frequency having the similarity index value indicating that the tone model and the corresponding estimated shape are not similar to each other.

An improved electro-larynx includes a linear transducer and / or an improved waveform generator. The improved electro-larynx sets up a sound wave within the pharynx of the user which closely approximates a normal glottal excitation. The linear transducer preserves the harmonic structure of a glottal source wave generated by the waveform generator and translates it into a vibration. The transducer includes an armature assembly, suspension assembly, and coupler disk coupled together to move in concert. The armature assembly vibrates as a function of the desired and input glottal source wave, which in turn causes an immediate and corresponding vibration of the coupler disk. The suspension assembly constrains armature movement to one dimension and provides additional compliance. The coupler disk includes a substantially flat surface suitable for engaging the surface of a user's throat and vibrates as a linear function of the input glottal source wave. The improved waveform generator produces a relatively good approximation of an actual glottal source waveform by preferably deriving it from actual voice data and having the effects of the modulation of the vocal tract removed. As a result, the harmonic structure of the glottal source waveform has overtones which drift in frequency, similar to normal glottal excitations. The waveform generator also allows user adjustment of the pitch and amplitude of the glottal source wave and smoothes out any distortions caused by the process of obtaining the glottal data used to generate the glottal source wave. The waveform generator bolsters the frequency response at the high end of the spectrum to compensate for any roll-off, yielding a frequency response spectrum of about 20–5 Khz. The responsiveness of the linear transducer allows the glottal source wave's pitch, amplitude, and harmonic structure to be communicated through the coupler disk and realistic glottal source waves to be transduced into the user's pharynx, resulting in the production of substantially normal speech.

The invention relates to a method and a device for detecting respiration signals. The method comprises the following steps: original signals are collected; filter processing is conducted on the original signals, and filtering parameters adopted in filter processing are confirmed according to a harmonic structure of slow time domain respiration signals by analyzing in advance; a periodogram of slow time domain signals of the original signals is calculated after the filter processing; noise power of the slow time domain signals of the original signals is estimated after the filter processing; a harmonic graph is calculated according to the harmonic structure of the slow time domain respiration signals, the periodogram of the slow time domain signals and the noise power of the slow time domain signals; and whether the harmonic graph is larger than a preset threshold or not is judged, if the harmonic graph is larger than the preset threshold, the respiration signals exist, and if the harmonic graph is not larger than the preset threshold, the respiration signals do not exist. According to the method and the device for detecting the respiration signals, the filtering parameters are confirmed by adopting the harmonic structure of the respiration signals so that the filter processing is conducted, environmental noise can be filtered well, the respiration signals can be reserved as much as possible, and detecting performance is improved.

The invention provides a honeycomb-shaped disc-shaped vibrating gyro which comprises a harmonic oscillator and electrodes arranged in or / and outside the harmonic oscillator. The harmonic oscillator is in a disc shape which is overall in axisymmetry. The center of the disc-shaped harmonic oscillator is an anchoring point used for fixing the whole harmonic oscillator, and a circle of round harmonic rings are arranged on the outer edge of the disc-shaped harmonic oscillator. The structures for connecting the harmonic rings and the anchoring point are honeycomb-shaped frames which are distributed periodically in the circumferential direction and are in a honeycomb topology shape, a plurality of rings of hexagonal holes distributed periodically in the circumferential direction are formed in the honeycomb-shaped frames, and the whole honeycomb-shaped frames and a circular ring form a harmonic structure of the harmonic oscillator. The electrodes used for conducting driving, detecting and controlling can be arranged inside or outside the harmonic oscillator or arranged inside and outside the harmonic oscillator at the same time. The honeycomb-shaped disc-shaped vibrating gyro has the advantages that large harmonic mass is achieved, the number of driving electrodes is large, the detection capacitance area is large, and the number of control electrodes is large.

A pitch search method and device for digitally encoding a wideband signal, in particular but not exclusively a speech signal, in view of transmitting, or storing, and synthesizing this wideband sound signal. The new method and device which achieve efficient modeling of the harmonic structure of the speech spectrum uses several forms of low pass filters applied to a pitch codevector, the one yielding higher prediction gain (i.e. the lowest pitch prediction error) is selected and the associated pitch codebook parameters are forwarded.

A method and apparatus provide the ability to code signals in a layered manner for compression and networking applications. The solution involves a relaxed hierarchical structure of layers, wherein only an optimal subset of information from lower quality levels is transmitted to higher level decoders. This framework is complemented with a design method to optimize system parameters. Specialization may include techniques for employing irregular quantizers and / or estimation theoretic optimal parameter selection and / or content specific optimization (e.g., exploiting harmonic structure in audio, adaptive transform coding and enhanced motion compensated prediction for video) and / or optimization of the structure of the layers, where the potential of exploiting all the common information is realized to improve overall system performance for that application. One specific technique provides improved compression of signals with multiple quality levels with or without prediction. Another specific technique provides improved compression of signals with multiple heterogeneous quality-levels with or without prediction.