語音頻率分量 的英文怎麼說
中文拼音 [yǔyīnbīnlǜfēnliáng]
語音頻率分量
英文
speech frequency component- 語 : 語動詞[書面語] (告訴) tell; inform
- 音 : 名詞1. (聲音) sound 2. (消息) news; tidings 3. [物理學] (音質) tone 4. (姓氏) a surname
- 頻 : Ⅰ形容詞(次數多) frequent Ⅱ副詞(屢次) frequently; repeatedly Ⅲ名詞1 [物理學] (物體每秒鐘振動...
- 率 : 率名詞(比值) rate; ratio; proportion
- 分 : 分Ⅰ名詞1. (成分) component 2. (職責和權利的限度) what is within one's duty or rights Ⅱ同 「份」Ⅲ動詞[書面語] (料想) judge
- 量 : 量動1. (度量) measure 2. (估量) estimate; size up
- 語音 : speech sounds; pronunciation; voice
- 頻率 : frequency; rate
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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。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。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。Firstly, it introduces the development of speech coding, along with the significance of the low bit rate speech coding. it also compares the model of traditional dualistic excitation lpc vocoder and the multi - band excitation vocoder, and lucubrates the analytical method of frequency domain and time domain in the parameter extraction of multi - band excitation vocoding. secondly, based on the parameter extraction operation of keynote cycle, it adopts time domain in rough estimate operation of keynote and frequency domain in fine estimate operation of keynote, in according to the immediacy required in practice, to minish operation amount
本文闡述了一種基於fpga的多帶激勵語音編碼器的研究與設計,首先介紹語音編碼研究的發展狀況以及低速率語音編碼研究的意義,接著對比分析了傳統二元激勵lpc聲碼器模型和多帶激勵編碼器模型,並深入研究了多帶激勵語音編碼參數提取的頻域和時域分析法,然後根據實際應用的實時性要求,為了減小運算量,在基音周期參數的提取的演算法實現上,本文採用在時域進行基音粗估運算,在頻域進行基音精細估計運算。Code - division multiple - access ( cdma ) technology has become the main technology studied in the personal communication service ( pcs ) recently because it has lots of benefits, such as large system content, high voice quality, easy frequency scheme, low cost of the net making and so on
碼分多址技術因其具有系統容量大、語音質量高、頻率規劃簡單、建網成本較低等眾多優點,已成為近年來個人通信業務中研究的主要技術。For those multi - component signals like speech signal, such interaction is a common situation. secondly, a new common ridge method is discussed ; a common ridge across the complex ridge image is used instead of several interacted single ridges
因此對于語音等含有多個分量的信號,頻率相鄰的分量分別形成幾個帶狀區域,每個區域內脊互相干擾,無法分辨的情況是普遍的。System scheme of speech coding plus spread spectrum communication was presented based on a full analysis of noise characteristic, attenuation characteristic and impedance characteristic of low - voltage power line. spread spectrum carrier ( abbreviated as ssc ) technology is adopted to overcome problems existing in signal transmission over power line. high quality, low rate mbe compression algorithm was used to complete speech encoding and decoding
在對低壓電力線路的噪聲特性、衰減特性和阻抗特性三個方面充分分析的基礎上,本文提出一種語音編碼+擴頻傳輸的系統總體方案,採用擴頻載波( spreadspectrumcarrier ,縮寫為ssc )技術克服電力線傳輸信號存在的問題,採用語音合成質量高並具有較低碼率的mbe壓縮演算法完成語音信號的編解碼。Based on the research of videoconference systems of h. 323 protocol over ip networks and the author ' s experiences of implementing h. 323 videoconference systems in remote education area, in this thesis the main factors that affect videoconference quality are analyzed, and a dynamic bit rate allocation model is proposed and partly implemented. this model is designed to dynamically allocate bit rate for multi - media data flow ( including audio data and video data ) in fixed bandwidth network environment. when continuous multi - media packet losses are detected in ip based h. 323 videoconference system, the bit rate of video data is adjusted meanwhile the bit rate of audio data remains unchanged, and the bit rate allocation of multi - media data ( including audio data and video data ) is optimized as a whole effect
本文結合作者在h . 323視頻會議系統應用於遠程教育的經驗,通過對現有的基於ip網路的h . 323協議視頻會議系統的研究,分析了影響視頻會議質量的原因,提出並部分實現了在固定帶寬的網路環境下,基於ip的h . 323視頻會議的多媒體數據包發生丟包、抖動或延時時,保持音頻數據位速率不變,通過對視頻數據的位速率的進行調整,最終實現旨在提高視頻會議語音質量的多媒體數據位速率動態調整的模型。分享友人