speech error 中文意思是什麼
speech error
解釋
言語誤差-
On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system
在此基礎上，描述了一種較為先進的大型dsp系統程序開發策略，並以此為指導思想，以美國ti公司c6000dsp開發平臺開發出了整個amr解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語音自然度不好的缺點，論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射系數參數進行誤碼消除處理。 -
Experimental results in different noises and snr indicated that this vad algorithm can divide speech segments from non - speech segments accurately and reduce voiced - unvoiced error obviously. ( 2 ) an improved dct - hn speech decomposition algorithm based on the harmonic - noise model is presented
不同噪聲、信噪比下的實驗結果表明，該演算法可以準確區分語音段與非語音段，明顯降低了基音檢測中清濁誤判現象的發生; （ 2 ）基於「諧波-噪聲」模型提出了一種改進的dct - hn語音分解演算法。 -
In this thesis, the research focuses on pitch detection techniques of the low - rate wi speech coding. aimed at the problems of voiced - unvoiced error, pitch doubling and halving, accuracy of pitch detection and pitch quantization, a series of pitch detection techniques including pre - processing, pitch detection and pitch quantization were proposed
本文就低速率wi語音編碼中的基音檢測技術進行了深入研究，針對基音檢測中的清濁誤判、基音加倍減半、基音檢測精度及基音量化問題，提出了包括基音檢測前端處理、基音檢測演算法及基音量化的一整套基音檢測技術。 -
According to the project of adaptive multi - rate speech coding ( amr ) being put forward by the third generation group of the mobile communication, this paper takes the principle of the speech arithmetic as the base, studies the technologies including the source controlled rate, voice activity detector, comfort noise and the error concealment unit in amr, discusses its the characteristic of adaptation and analyses its performances particularly. amr c codes are researched carefully through the modules being divided into and debugged under the tms320c54x provided by the ti corporation, and optimized in selecting the method of c code embedded assembler codes and simplified in the search codebook combining with the theory of speech coding, which are based on the realization about theory and practice of the optimization of amr speech coding
從自適應多碼率語音編碼演算法的c代碼出發，對它進行模塊劃分後作了系統分析，將其在vc下調試通過，進一步在ti公司提供的tms320c54x環境中進行調試，結合語音編碼理論，對演算法進行優化，採用了在c代碼中嵌入匯編和簡化自適應碼本和固定碼本搜索的方法，部分地提高了c代碼效率，為實現自適應多碼率語音編碼的優化奠定了理論和實踐基礎。 -
Error is an unavoidable speech act in the process of second language acquisition
摘要錯誤是二語習得過程中不可避免的語言行為。 -
An error - tolerant algorithm in decoding module of mandarin continuous speech recognition is examined to correct substitution, insertion and deletion errors in acoustic recognition
摘要本文研究了漢語連續語音識別音字轉換中的容錯演算法，以糾正聲學識別的替代、插入、刪除錯誤。 -
A study on noisy speech recognition linear predictive coding prediction error
一種噪聲環境下的語音識別方法線性預測誤差法的研究 -
This paper carries out a quantificational analysis upon the corpora of the english learners from primary schools to universities, especially middle schools, and makes comparisons in longitude and latitude. we find out from the comparisons that the language items puzzling the students most are : omission / addition of " ~ s " after third person singular verbs, omission / addition of " be ", error of tenses, error of articles, error of preposition collocations, agreement of subjects and verbs, error of singular / plural nouns, error of infinitives, error of part of speeches, error of possessive cases, empty of conjunctions in coordinative sentences, error of spellings ; elementary or intermediate learners ( such as freshmen ) suffer from native language transfer or simplification a lot ; advanced learners ( such as sophomores ) are affected by the overgeneralization of target language ; the error of part of speech or semantic selection ( except conceptual meaning ) runs through all phases, and it is likely to be fossilized
我們從研究中發現，困擾我國中學英語學習者的語言項目有動詞第三人稱單數s遺漏誤加、 be動詞誤加省、時態錯誤、冠詞錯誤、介詞搭配錯誤、主謂不一致、名詞單復數錯誤、不定式錯誤、詞性混用、所有格錯誤、並列句關聯詞空缺、拼寫錯誤；初、中級階段（大學一年級以前）學習者受母語遷移、目的語規則簡化影響較大；高級（大學二年級及以後）階段的學習者受目的語規則泛化影響較明顯；過渡語水平和學習者的學習階段呈正相關；英語詞性、詞義（概念意義除外）等方面的錯誤貫穿所有學習階段，且易形成「僵化」 。 -
There are difficulties in noisy speech recognition, especially low signal - to - noise rations are more difficult. this paper describes briefly six methods for speaker - dependent noisy speech recognition isolated words. they are lpc prediction error method, one - side auto - correlation sequence lpc, acoustic front end processing, canonical correlation based on compensation method, combination of features method and increase of poles method. the experimental results show that all the six techniques can improve effectively noisy speech recognition, and the best noisy speech recognition rate is above 80 % when snr 0db
它們是：線性預測誤差法，單邊自相關線性預測法，語音前端聲學處理法，正則相關分析的譜變換補償方法，特徵綜合法和同模極點增加法。實驗結果表明，這6種方法都有效地提高了噪聲環境中語音識別率，其中較好的方法在強噪聲環境中信噪比為0db的語音識別率達到80 %以上，為信噪比較低的噪聲環境中自動語音識別展現了美好前景。 -
It can take advantage of the advancement of hmm and gmm, utilize dynamic programming technique to realize the nonlinear time alignment between speech feature vectors and markov state sequences, use expectation - maximum algorithm to re - estimate the gmm parameters and finally employ levenshtein distance to calculate the word error rate between the recognized and expected results
它將隱markov模型和gaussian混合密度分佈緊密聯系，結合動態規劃演算法對時間序列和markov狀態鏈進行非線性時間對齊，並運用em演算法對gaussian混合模型的參數進行重新估計，識別出來的結果與期望結果採用levenshtein距離進行比較並得出其字誤差率。 -
This paper introduces a project of the wireless data transferring and the realization of speech encoding / decoding arithmetic based on the embedded system. in embedded system based on arm ? cpu, we accomplished the update of the system data by using the paging system, and emphatically researched how to avoid bit error. and, realizes the speech compression and decompression based on itu - t g. 729a, implement the speech synthesize of personal paging
在以arm7為處理器內核的嵌入式系統上，通過尋呼系統實現了系統數據的無線動態更新，重點解決了尋呼誤碼造成的數據錯誤等問題；以itu ? tg . 729a語音編解碼標準為基礎，通過語音壓縮與解壓演算法實現了個人尋呼的語音合成。 -
According to the distribution of chinese single - character after word segmentation in chinese text and the conception of " non - multi - character word error ", we proposed a group of rules to find errors in texts, to construct the automatic error - detection model and to implement its algorithm by combining the scattered single - character bigram models, part - of - speech bigram and trigram models
根據正確文本分詞后單字詞的出現規律以及「非多字詞錯誤」的概念，提出一組錯誤發現規則，並與針對分詞后單字散串建立的字二元、三元統計模型和詞性二元、三元統計模型相結合，建立了文本自動查錯模型與實現演算法。 -
Then, to facilitate error correction, highly accurate word alternatives are suggested by combining speech data with statistical predictive data. as a result, the desired word can be selected with a minimum number of keystrokes
然後，為方便糾錯，語音識別數據和統計預測數據結合在一起，極大地提高了選詞精度，最終以最少的擊鍵次數找到想要的單詞。 -
Meanwhile, the telephone gateway in tetra system is introduced. in further research, the principle of tetra speech coding algorithm ? algebraic codebook excitation linear prediction ( acelp ) is introduced and analysed in detail, which is a advanced codebook excitation linear prediction ( celp ). acelp algorithm replaces the excitation signals with algebraic codebook and uses some technique such as minimizing the mean square error ( mse ) and the analysis - synthesis method to obtain characteristic parameters of speech
同時，介紹tetra系統的市話網關，並在接下來的研究中詳細介紹tetra電話網關中應用到的語音編解碼演算法? ?代數碼本激勵線性預測碼（ acelp ）的基本原理，它是一種簡化了的碼本激勵線性預測碼（ celp ） ，它把激勵信號用代數碼本代替，並且運用了均方誤差最小、分析?合成等技術提取出語音的特徵參數，極大地降低了比特率，而且具有較好的重建語音質量。 -
By using the lpc coefficients of noise to predict all the speech signal, the method gets the lpc prediction error lpcpe sequence. then use it to substitute the speech sequence to detect the speech terminal extract the speech features and to recognize in a suitable way
該方法利用噪聲的lpc系數去預測語音信號，從而得到lpc預測序列，然後把它代替原語音序列來進行語音端點的檢測語音特徵的提取和在合適的匹配方式下的識別。
分享友人