speech frequency 中文意思是什麼
speech frequency
解釋
話音頻率-
Following this analytical framework, we conducted an investigation into the explicit and conventionalized legal speech acts as used in english legislative discourses in terms of their functions, realization, distribution and frequency. the general pragmatic principles of forensic communication and the features of legal discourses are also discussed in this paper. directed by professor mao junchun
文章首先簡要地討論了法律語言的一般語用原則、特點和與法律言語行為有關的理論，然後分析了法律言語行為的功能、分類，最後分析了法律言語行為在英語立法語篇中的實施情況以及各類言語行為的分類和分佈。 -
The results of simulation indicated speech signal processed by the optimum algorithm presents obvious periodicity in time domain, and effect of the formant is removed or restrained effectively in frequency domain
處理后的語音信號在時域上表現出明顯的周期性特徵，同時在頻域上也觀察到聲道的共振峰結構影響得到消除或有效的抑制。 -
In various speech character parameters, formant frequency, bandwidth and pitch frequency are chosen as voice character parameters. the reasons are as follows : hearing apperceive experiments indicates that formant frequency can stand for a majority of voice information, while average pitch frequency can explain 55 % ability of speaker verification
數據結果與多項式回歸和線性多變量回歸相比，支持向量回歸既提高了泛化性能又避免了頻譜不連續性，從而使轉換語音與目標語音的頻譜距離失真分別減少了33 . 29 %和35 . 24 % 。 -
Firstly, we study the construction of emotion - speech template database, and analyze the common features such as pitch, energy and formant. after choosing the useful features by using fuzzy entropy effectiveness analysis, we get better performance with the application of neural network. in addition, we propose some more efficient features such as speech rate, pitch slope, mel - frequency cepstral coefficients and its transient parameters, and design a processing model based on vector quantization for cepstral features to fusing different features
本文首先介紹了情感語音數據庫的建立情況，然後研究了基音頻率、振幅能量和共振峰等目前常用的情感特徵在語音情感識別中的作用，並且通過一種基於模糊熵的特徵有效性分析方法進行了有效特徵的篩選，應用人工神經網路建立了初步的語音情感識別模型，經過實驗發現特徵篩選后系統的識別效果有著一定程度的提高。 -
At the same time according to the low recognition rate in speech recognition system, the author used the method of fundamental frequency analysis to build male / female recognition model respectively
同時針對語音識別系統中識別率不高的問題，採用基音頻率分析的方法分別建造男女聲識別模型。 -
Because the speech signal is periodicity at sonant which vocal cords surge in low frequency and similarity to white noises at surd, the pitch can be detected in traditional way through the correlation operation without the speech produce model
在人類語音的濁音段，聲帶發生較低頻率的振蕩，語音信號呈明顯的準周期性，而在清音段，語音信號則類似於白噪聲。 -
It reduces the “ music noise ” using the human auditory characteristics, and enhances the hearing quality using the speech spectrum distribution characteristics in the time - frequency dimension
主要以譜減法為基礎，結合人耳的聽覺特性從而減少殘留「音樂噪聲」的影響;結合語音的語譜在時-頻域分佈特性從而提高增強后語音的聽覺質量。 -
According to the characteristics of the human pronunciation and the speech spectrum distribution in the time - frequency dimension, the paper finds out that there is a shortcoming of the speech enhancement system which is based on the masking properties of human auditory
根據人的發音特點，通過分析語音的語譜在時-頻域的分佈，發現把聽覺掩蔽效應應用於語音增強時存在不足之處。 -
In order to reduce the musical residual noise and the background noise, a speech enhancement method based on masking properties of the human auditory system is described. this method uses bark wavelet packet transform to simulate the frequency feature of human auditory model to get the threshold
本文以最大限度減少殘留噪聲和背景噪聲為目的，採用bark子波分析的方法模擬人耳基底膜的頻率分析特性來進行語音增強，重點進行模擬人耳聽覺掩蔽效應來確定除噪閾值的研究。 -
As for the feature of mandarin digit speech, the existing arithmetic is cited to design the software system, and the design process is described in the part. here, the shore - time ^ relative efp ( energy - frequency - product ) is used to make the capsheaf of chinese speech signal, and the short - time relative efq ( energy - frequency - quotient ) is used to separate its syllable and consonant - vowel segment, and it improves the correct rate
本文採用的漢語語音的端點信號的檢測和清濁音信號切分方法是：短時相對能頻積的方法對漢語語音信號的端點進行檢測；短時相對能頻比的方法對語音信號的清濁音進行切分，提高漢語語音信號切分的成功率。 -
Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method
其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求，並在此基礎上加入g . 729b的靜音檢測模塊，以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法，通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計，信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號，信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計，除了類似雙音多頻的信號發生及頻率檢測設計外，還需要檢測信號持續時間，作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。 -
Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates
但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。 -
This paper raised a new way for overlapping speech segregation based on sound localization cues. in this paper, we first divide the speech stream into some time - frequency regions and calculate the itd and iid of each region. then the notion of a " time - frequency " binary mask is given, which selects the target if it is stronger than the interference in a local time - frequency region
然後求取每個小片段上的itd （到達雙耳時間上的差異性）和iid （到達雙耳強度上的差異性）值，經過實驗證明某個片段上的itd值和iid值與該片段上的信噪能量比是單調遞增關系，因此通過和域值的比較，得出掩蔽系數，來判斷每個小片段具體是屬于哪個聲源。 -
Speech communication is one of the most used modes in the digital trunking communication system. excellent algorithm of speech coding can save the bandwidth resource, improve the utilization of frequency, so it has important value for investigation
語音通信是數字集群通信系統中最常用的通信方式之一，優良的語音編解碼演算法能夠更加有效地節省帶寬資源，提高頻率利用率，因此具有重要的研究價值。 -
Speech coding systems are generally based on narrowband speech at present, but its frequency is restricted in 200hz ~ 3400hz and its sample rate is 8khz. with the development of the wideband speech, its bandwidth which is from 50hz to 7khz causes the quality of speech communication to approach in the feeling of face - to - face conversation, and makes the speech natural, expressive and comfortable. hence, it ’ s quite significant in researching on wideband speech coding system. in recent 20 years, the dsp and its software development kit has improved greatly, but the price has fallen sharply, thus it has more and more widespread applications now
隨著當今世界的飛速發展，寬帶語音越來越受到人們的青睞，因為它的50hz 7khz的帶寬使得語音通訊質量接近於面對面交流的感覺，大大提高了語音的自然度、表現力和舒適度。因此，開發研製基於寬帶語音的編解碼系統具有十分重要的意義。在過去的短短二十年裡， dsp處理器的性能得到很大的改善，軟體和開發工具也得到相應的發展，價格卻大幅度地下降，從而得到越來越廣泛的應用。 -
Impact will allow you to hear the high frequency speech cues that traditional amplification does not
「移頻」可以使您聽到並辨別傳統助聽器無法使您聽到的高頻語言聲。 -
Under the condition of " comparatively weak correlation between the two noises involved, coherence function is used as a frequency domain amplification factor for improving snr of the output signal to the filter and the speech enhancement effect. meanwhile, a real - time recursive algorithm is put forward in substitute for current algorithms based on short time fourier transform. the new algorithm will simplify computations and will be suited for real - time implementation together with the adaptive systems
接著針對上述nanc系統兩路輸入信號噪聲相關性弱的情況，用相干函數作頻域增益因子來提高輸出信噪比與改善語音增強效果，同時，通過一種實時迭代演算法解決了短時傅氏變換計算量大的問題，簡化了計算，便於實時處理與實際應用。 -
A method of pitch mark determination for a speech, includes : acquiring a fundamental frequency point and fundamental frequency passband signals by using an adaptable filter ; detecting a number of passing zero positions of the fundamental frequency passband signals ; and generating at least a set of pitch marks from a number of passing zero positions
一種決定語音音高標記的方法，系藉以找出一語音之一組音高標記，此決定語音音高標記的方法系利用一可適性濾波器取得一基頻點與一基頻帶通訊號；求取基頻帶通訊號之復數個過零點位置；然後經由復數個過零點位置產生至少一組音高標記。 -
Speech detection using matching pursuits algorithm in time frequency domain
將頻域的線譜對 -
It was composed of three main modules, they were word segmentation module based on the maximum word - length matching algorithm, part of speech tagging module based on statistical method of training of relative frequency, and the syntax parsing module based on the improved chart analysis algorithm
該系統實現了基於最大詞長匹配演算法的分詞模塊、基於統計方法的詞性標注模塊和基於改進的線圖分析演算法的句法分析模塊。
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