42 results about "Spectral flatness" patented technology

In accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream of the audio bitstream to get low band coefficients in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients. The method further includes processing the high band coefficients to form processed high band coefficients. Processing includes modifying an energy envelope of the high band coefficients by multiplying modification gains to flatten or smooth the high band coefficients, and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients. The low band coefficients and the processed high band coefficients are then inverse-transformed to the time domain to obtain a time domain output signal.

A voice activity detector that detects talkspurts in a given signal at a high accuracy, so as to improve the quality of voice communication. A frequency spectrum calculator calculates frequency spectrum of a given input signal. A flatness evaluator evaluates the flatness of this power spectrum by, for example, calculating the average of power spectral components and then adding up the differences between those components and the average. The resultant sum of differences, in this case, is used as a flatness factor of the spectrum. A voice / noise discriminator determines whether the input signal contains a talkspurt or not, by comparing the flatness factor of the frequency spectrum with a predetermined threshold.

An audio signal is divided among exponentially related subband filters. The spectral flatness measure in each subband signal is determined and the measures are weighted and combined. The sum is compared with a threshold to determine the presence of music or noise. If music is detected, the noise estimation process in the noise reduction circuitry is turned off to avoid distorting the signal. If music is detected, residual echo suppression circuitry is also turned off to avoid inserting comfort noise.

The invention discloses a method for classifying digital audio automatically. The method particularly comprises the following steps: pretreating an audio signal: pre-emphasis treatment, framing treatment and windowing treatment; extracting an audio characteristics comprising frequency spectrum mass center, spectrum spreading, frequency spectrum flatness, frequency spectrum transition parameters, short time energy, fundamental frequency and Mel frequency cepstrum coefficient (MFCC) and MFCC first order difference; indexing and classifying according to the audio characteristic, and finding out the similar audio. According to the invention, the problem of error caused by indexing audio by adopting a single audio characteristic in the prior art is solved; moreover, the method is simple in computation process, is easy to apply practically, and is high in indexing efficiency.

An automated ship detection technique includes accessing data associated with an image of a portion of Earth. The data includes reflectance values. A first portion of pixels within the image are masked with a cloud and land mask based on spectral flatness of the reflectance values associated with the pixels. A given pixel selected from the first portion of pixels is unmasked when a threshold number of localized pixels surrounding the given pixel are not masked by the cloud and land mask. A spatial variability image is generated based on spatial derivatives of the reflectance values of the pixels which remain unmasked by the cloud and land mask. The spatial variability image is thresholded to identify one or more regions within the image as possible ship detection regions.

An apparatus, method, and medium for detecting a voiced sound and an unvoiced sound. The apparatus includes a blocking unit for dividing an input signal into block units; a parameter calculator for calculating a first parameter to determine the voiced sound and a second parameter to determine the unvoiced sound by using a slope and spectral flatness measure (SFM) of a mel-scaled filter bank spectrum of an input signal existing in a block; and a determiner for determining a voiced sound zone and an unvoiced sound zone in the block by comparing the first and second parameters to predetermined threshold values.

A method for digital encoding of an audio stream in which the psychoacoustic modeling bases its computations upon an MDCT for the intensity and a spectral flatness measurement that replaces the phase data for the unpredictability measurement. This dramatically reduces computational overhead while also providing an improvement in objectively measured quality of the encoder output. This also allows for determination of tonal attacks to compute masking effects.

The embodiment of the invention discloses a band spreading method and device, electronic equipment and a computer readable storage medium. The method comprises the steps of performing time-frequency transformation on a narrow-band signal to be processed to acquire a corresponding initial low frequency spectrum; based on the initial low frequency spectrum, acquiring correlation parameters of a highfrequency part and a low frequency part of a target wideband spectrum via a neural network model, wherein the correlation parameters comprise at least one of a high frequency spectrum envelop and relative flatness information, and the relative flatness information represents correlation between the spectrum flatness of the high frequency part and the spectrum flatness of the low frequency part ofthe target wideband spectrum; based on the correlation parameters and the initial low frequency spectrum, acquiring an initial high frequency spectrum; and acquiring a broadband signal after band spreading according to a target low frequency spectrum and a target high frequency spectrum, wherein the target low frequency spectrum is the initial low frequency spectrum or the spectrum acquired afterfiltering treatment is performed on the initial low frequency spectrum, and the target high frequency spectrum is the initial high frequency spectrum or the spectrum acquired after filtering treatment is performed on the initial high frequency spectrum.

An apparatus, method, and medium for detecting a voiced sound and an unvoiced sound. The apparatus includes a blocking unit for dividing an input signal into block units; a parameter calculator for calculating a first parameter to determine the voiced sound and a second parameter to determine the unvoiced sound by using a slope and spectral flatness measure (SFM) of a mel-scaled filter bank spectrum of an input signal existing in a block; and a determiner for determining a voiced sound zone and an unvoiced sound zone in the block by comparing the first and second parameters to predetermined threshold values.

The embodiments of the invention disclose an audio popping detection method and device and a storage medium. A to-be-detected audio signal can be obtained when the audio signal is subjected to popping detection, the audio signal is divided into a plurality of frame signals, then the short-time energy difference between every two adjacent frame signals is calculated, then frame signals satisfying the preset condition interval are obtained according to the short-term energy differences, sudden change audio signals are obtained, then the frequency spectrum flatness of the sudden change audio signals is calculated, and if the frequency spectrum flatness is greater than a preset flatness value, the popping of the audio signals is determined. According to the scheme, whether the audio signals have the popping or not can be accurately detected.

The present invention discloses a perceptual weighing filtering method. The method mainly comprises treating the input voice or audio signals through spectrum inclined filtering, processing the output voice or audio signals treated through perceptual weighing filtering, selecting the corresponding weighing factor value according to the spectral flatness of the input signals, and outputting the voice or audio signals processed through the traditional perceptual weighing filtering. Moreover, part of the coefficients of the transmission function acquired in the processing of perceptual weighing filtering are directly used as the coefficients of the transmission function acquired in the processing of spectrum inclined filtering. Simultaneously, the present invention also provides a perceptual weighing filter. The perceptual weighing filter can well simulate the formant structure of the frequency spectrum of the input wideband voice or audio signals without additional encoding bit rate and reducing the computation complexity, make full use of he masking effects of human acoustic feelings, and achieve the purposes of improving the encoding efficiency and enhancing the subjective hearing effects.

A method is disclosed of compensating the output of an ADC for non-linearity in the response of the ADC. The method comprises converting an analog input signal to uncorrected digital ADC output samples, applying a vector of correction variables to each of a block of uncorrected ADC output samples to provide a block of corrected ADC samples, and iteratively minimizing a measure of the spectral flatness of the block of corrected ADC samples with response to the vector of correction variables.

A method for selecting an antenna direction setting for optimum signal reception prior to channel equalization provides a set of metrics, referred to as channel quality metrics (CQM), that characterize the quality of the received signal for a given antenna setting and a generic algorithm that uses these metrics to select the antenna setting for an optimum reception. This invention utilizes five main CQMs: a Signal Strength Metric (SSM), a minimum mean squared error of a decision feedback equalizer (MMSE (DFE)) channel quality metric, a MMSE for a linear equalizer (MMSE(LE)) channel quality metric, a Spectral Flatness Metric (SFM) and an interference degradation metric (IDM).

The invention discloses an audio detecting method and device, electronic equipment and a storage medium, and belongs to the technical field of digital audio processing. On the basis of an energy valueof an audio frame, a zero crossing rate, a weight frequency spectrum value and a frequency spectrum flatness in an audio file, linear related coefficients between the audio frame and the previous audio frame and other characteristic parameters, whether or not burst white noise exists in the audio frame is detected, when it is determined that the energy value of the audio frame is larger than a first threshold value, the zero crossing rate is larger than a second threshold value, the weight frequency spectrum value is larger than a third threshold value, the frequency spectrum flatness is larger than a fourth threshold value, and the linear related coefficients between the audio frame and the previous audio frame are smaller than a fifth threshold value, it can be determined that burst white noise existing in the audio frame is detected, on the basis of the audio detecting method, whether or not an audio file of any type has burst white noise can be detected, on the basis of multiple characteristic parameters, the accuracy of the detecting result is higher, and the robustness of the detecting method is higher.

The invention discloses an infrared detector structure for improving response spectral flatness. The infrared detector structure for improving the spectral flatness comprises metal electrodes, a substrate, a photosensitive area and other media. The metal electrodes are located on the left side and the right side of the rectangular photosensitive area. The other media are located on the upper side and the lower side of the rectangular photosensitive area. The other media are the same as the photosensitive area in material and are electrically isolated from the metal electrodes and the photosensitive area. The infrared detector structure is simple and effective, non-linear response spectral shapes are eliminated, and the spectral flatness is improved.

Methods, systems, and devices are described for wireless communications. A wireless station includes a transmitter to generate a wideband contiguous waveform in a band. The transmitter transmits the waveform that conforms to spectral masking attributes and spectral flatness attributes. The wireless station, or another device supporting spectrum analysis functions, detects a wideband contiguous waveform and performs spectrum analysis of the waveform. The wireless station includes a resolution bandwidth of 25 KHz and a video bandwidth of 7.5 KHz.

The invention discloses an audio feature extraction method and a device thereof, a training method and electronic equipment, and the method comprises the steps: obtaining a to-be-extracted audio according to a preset window length, and dividing the to-be-extracted audio into M audio frames according to a preset frame length; calculating a frequency spectrum corresponding to each audio frame; according to the N frequency domain points of each frequency spectrum, obtaining a fitting slope and a fitting intercept corresponding to each frequency spectrum based on a linear fitting algorithm; calculating the frequency spectrum flatness of each frequency spectrum according to the frequency spectrum and a preset calculation formula; dividing each frequency spectrum into m frequency spectrum bands,calculating to obtain a logarithm frequency spectrum corresponding to each frequency spectrum band, and further calculating to obtain a frequency spectrum contrast ratio of each frequency spectrum; and according to the fitting slope, the fitting intercept, the spectrum flatness and the spectrum contrast of the audio frame, obtaining the characteristic quantity of the audio frame, and further extracting the audio characteristics of the audio to-be-extracted. When the extracted audio features are used for infant cry and other detection scenes, the accuracy of audio detection can be improved.

The invention discloses a noise removal method in an intelligent calling system based on machine learning, which comprises the following steps: slicing, normalizing and framing telephone signals; extracting MFCC features from the framed slice signals and carrying out averaging processing on the MFCC features; inputting the averaged MFCC features into a machine learning classifier for model training, wherein the trained classification model is used as a noise classification model; slicing the newly added telephone signal; carrying out normalization and framing preprocessing on the slice signals; carrying out frequency spectrum flatness preliminary screening on the framed slice signals; extracting MFCC features and then averaging the MFCC features; and inputting the MFCC features of each segment of signal after average processing into a noise classification model for identification. The beneficial effects of the method are that the method achieves the recognition of the voice or noise ofa signal through the classification model based on machine learning, can remove a large number of noise signals in a telephone signal, and reduces the error rate that the signal is transmitted to anASR and translated into characters.

The invention described herein provides a method and apparatus that allows direct digital to IF / RF conversion using pulse-shaping. The method facilitates obtaining a flat or near flat output spectrum after digital to analog conversion with a minimal loss in signal energy. As opposed to pulse-shortening, where the DAC output pulse and consequently the energy per sample are reduced to a fraction a<1 of the maximum, the pulse-shaping does not shorten the DAC pulse. For each sample, pulse-shaping first stores the energy delivered by DAC and then releases the stored energy to the output during a short period of time aT. This way little signal energy is lost even for very small values of a. With pulse-shaping, the duration aT of the output pulse contributes to the spectral flatness in a way similar to that pulse-shortening, but has the additional benefit that the shape of the output pulse contributes significantly to a flat spectrum. Two embodiments of the pulse-shaping method are described in the context of two types of DAC that are used: one with current output and the other with voltage output. Then, two examples of pulse-shaping implementation are given for a single-ended current-output DAC and for a differential current-output DAC.

The present invention provides multi-band ultra-wide band (UWB) communication methods and systems capable of adaptively and scalably supporting different applications with different requirements, as well as different desired or ideal properties relating to the communications. A method is provided for transmitting information using multi-band ultra-wide band transmission, including transmitting a signal over each of multiple frequency sub-bands, and allowing variation of at least one transmission parameter to facilitate trade-off between at least two of power consumption, energy collection, bit rate, performance, range, resistance to multiple access interference, and resistance to multipath interference and spectral flatness.

The invention discloses an ultrahigh-spectral transmittance etalon, which comprises a fixed part and light filters, wherein the light filters are of a sector structure, are alternately arranged on thefixed part and are used for filtering or shielding continuous light emitted from a light source. The upper limit of measurement of the transmittance is improved or the lower limit of measurement of the transmittance is reduced through alternately arranging at least one light filter with fixed transmittance on the fixed part and setting the sector area / central angle ratios of the light filters, the measurement range of the spectral transmittance etalon is greatly improved, meanwhile, the measurement uncertainty on the transmittance is greatly reduced, the spectral flatness of a transmittance curve is improved and the measurement uniformity on the transmittance is improved.

The present invention discloses a novel optical fiber temperature measuring system for an electric power system. The novel optical fiber temperature measuring system comprises a light source module and a coupler which are connected with each other by a single-mode optical fiber; the coupler is connected with a tunable filter module, an etalon module and a sensor probe; the light source module does not need a spectral flatness characteristic which is necessary for a conventional optical fiber temperature measuring system; the light suppression ratio of the tunable filter module is 45 dB-60 dB, and therefore, a light spectrum corresponding to a tested probe can be selected from uneven spectra, crosstalk imposed to a low-power spectral region by a high-power spectral region caused by non-flat spectra can be decreased; and the output end of the light source module is also connected with a logarithmic amplifier module, and therefore, the influence of the non-flat spectra can be further decreased.