語音頻率 的英文怎麼說
中文拼音 [yǔyīnbīnlǜ]
語音頻率
英文
speech frequency-
Moreover in speech enhancement, especially in reducing the pulse noise, morphological algorithm has its unique advantage. particularly morphological filter may maintain the preferable accurate of the speech signal in speech waveform, and which produces little impairment to the formant of speech. so the spectrum structure of the speech is retained well, and the quality of the speech will not be reduced
特別是，在時域波形分析中，形態學濾波增強較小波去噪更好地保持語音信號的細節；在頻域分析中，形態學濾波對語音信號的基音頻率、頻譜斜率、共振峰等語音特徵的影響很小，因而能夠較好的保留語音信號的頻譜結構，使語音品質不致降低。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。Furthermore, the network technique only support the voice service cannot meet the communication requirements of people. people hope to obtain multimedia service such as packet data, video and picture phone etc besides voice service. all these need operators and researchers to look after the mobile communication scheme which optimizes spectrum efficiency and expands system capacity to accommodate more users in major metropolitan markets
此外，僅支持語音業務的網路技術已不能滿足人們對信息交流的需求，人們希望能隨時隨地獲取除語音之外的數據、視頻和圖像等多媒體業務，這些因素都促使運營商和研究者尋求頻譜利用率更高、通信容量更大的移動通信解決方案。He has also shown how electronic technology can be harnessed to map out and account for phonological developments in china through the example of retroflex initials. mitsuaki endo follows the admirable tradition of toiling with patience and perseverence to provide a foundation that will facilitate research by others. he has compiled frequency data on initials, medials ( glides ), nucleus vowels, rhyme endings and tones in the chinese rhyme book
遠藤光曉經過長期艱苦的努力，對《廣韻》中所出現的音位（聲母、介音、主要母音和聲調）進行了統計，並求出了各音位及各種語音類別的頻率，為今後從計算與計量的角度研究從中古到現代各方言的語音演變過程及原因提供了量化的依據。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。Algorithm for pitch detection of chinese tri - syllabic words based on wavelet transform
基於小波變換的漢語三字詞語音基音頻率提取In this paper, on the basis of absorption of achievements of the research on auditory physiology, an auditory model simulationg the peripheral auditory system and part of the central auditory system is set up. the model is made of the fitlters presenting the characteristics of the basilar membrane for analyzing the voice signals, the half wave rectification modeling the inner hair cells and energy transfer of nerve fiber
在吸收聽覺生理學研究成果基礎上，建立了一個模擬外圍聽覺系統和部分中樞聖經系統功能的聽覺模型。模型由表徵基底膜的頻率分析的帶通濾波器組、內毛細胞的半波整流特性和神經纖維的能量轉換特性組成，該模型可以作為前端處理來提取語音信號的自相關圖譜。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。Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. the result is not degraded too much while the complexity is reduced. at the end of the paper the development prospect of cs - acelp and speech coding are described
對lsp參數量化中的第一級碼書的128個碼字的使用頻率進行了統計試驗，選用了128個碼字中使用頻率高的112個碼字作為新碼書，語音質量基本不變但降低了碼書搜索的復雜度。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。The project uses for reference the algorithm thought of sbc ( subband coding ) to measure off the audio to the corresponding frequency width and encode it by the different sensitivity of human hearing, which results in the lower coding rate and bearable voice quality. the algorithm processing low bit - rate audio is designed to be self - adaptive by the situation of network. the component developped by that algorithm and project has already been used in the realtime interactive educational system
該方案借鑒sbc （ subbandcoding ）子帶編碼演算法思想，將音頻按對人聽覺敏感程度不同劃分為相應的頻帶並進行相應的編碼，從而得到較低的編碼率和較好的語音質量，設計了可根據網路狀況進行自適應的低帶寬音頻處理演算法。In this paper, the author provides frequency counts of the phonemes ( e. g. initials, medials, principal vowels, endings and tones ) and sound categories in the guangyun
本文對《廣韻》中所出現的音位（即聲母、介音、主要母音、韻尾和聲調）進行統計，求出各音位及各種語音類別的頻率。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。Then we consider the applications of music structure to audio - based mir and music summarization based on the labeled music structure information. first we extract the pitch class profile ( pcp ) feature vector through the analysis of music representation
首先通過分析音樂的表達方式提取了pcp特徵，這是一種基於幀的特徵，它較好的結合了聲學層的頻率和音樂語義層的十二平均律信息。In this thesis, first we analyzed and designed a traditional continued speech recognition system, which based on hmm and mfcc speech features. then we researched some noise robust technologies based on that system
本論文首先分析並實現了一個以mel頻率倒譜系數（ mfcc ）作為語音特徵，基於隱馬爾可夫模型（ hmm ） ，針對連續數字串識別任務的基本連續語音識別系統。The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "
系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。After having the keynote cycle, it separately processes harmonic waves of speech band ; whereafter takes v / u judgement / verdict and amplitude estimate to each band
得到基音周期后，對語音幀頻帶按基音頻率的諧波進行分帶處理，並對每個帶進行v / u判決和幅度估計。分享友人