預測減法編碼 的英文怎麼說

中文拼音 [jiǎnbiān]
預測減法編碼 英文
prediction subtraction coding
  • : Ⅰ副詞(預先; 事先) in advance; beforehand Ⅱ動詞(參與) take part in
  • : 動詞1. (測量) survey; fathom; measure 2. (測度; 推測) conjecture; infer
  • : Ⅰ名詞1 (由國家制定或認可的行為規則的總稱) law 2 (方法; 方式) way; method; mode; means 3 (標...
  • : Ⅰ動詞1 (編織) weave; plait; braid 2 (組織; 排列) make a list; arrange in a list; organize; gr...
  • : Ⅰ名詞(表示數目的符號或用具) a sign or object indicating number; code Ⅱ量詞1 (指一件事或一類的...
  • 預測 : calculate; forecast; prognosis; divine; forecasting; foreshadowing; predetermination
  • 編碼 : encoded; code; coded; encrypt; codogram; coding編碼表 encode table; 編碼程序 builder; 編碼尺 code...
  1. This thesis tries to update the cmdsr system to achieve the characters below : real - time, better robust, higher recognition rate, non - special - man. considering the disadvantages of traditional improved spectrum subtraction speech enhancement, this thesis proposes the theory of fuzzy spectrum subtraction based on the fuzzy theory and improved spectrum subtraction speech enhancement ; as for the difficulties of detecting the endpoint of speech signal, the thesis gives the table of initial and the improved parameters, with which we can confirm the endpoints of mandarin digit speech ; the thesis puts forward two - level digit real - time speech recognition system, the first level is based on discrete hidden markov model which is linear predictive coding cepstrum ( lpcc ) and difference linear predictive coding cepstrum ( dlpcc ), the second level is based on formant parameters ; as for the realization of hardware, the thesis depicts the realization of every part of cmdsr based on the tms320vc5402 in detail ; as for the development of software, the thesis gives the software design flow chart of cmdsr, simulates the basic theory with matlab language and gives the simulation results

    針對傳統的「改進譜相語音增強」參數設定單一、環境適應能力差的缺點,提出了一種利用模糊理論和「改進的譜相」結合的「模糊譜相語音增強」 ;針對語音信號端點檢困難的特點,通過matlab模擬試驗,給出了能夠準確確定數語音端點的初始和改進參數表;提出了利用基於線性倒譜參數和差分線性倒譜參數相結合的離散隱含馬爾可夫模型進行第一級識別、利用共振峰參數進行第二級識別的兩級漢語數語音識別系統,在保證系統實時性的同時,實現連接漢語數語音識別系統識別率的提高;在硬體實現上,詳細闡述了基於tms320vc5402的連接漢語數語音識別系統各部分硬體設計;在軟體開發上,給出了連接漢語數語音識別的軟體設計各部分的流程圖,並對各部分進行了matlab模擬,並給出了模擬結果。
  2. To extent the use of this 1c, some circuit blocks are added. in the design, digital circuits are used to process the signal and control the precision of coding. and some circuits are used in different time by several functions to reduce the number of transistors used and the dissipation

    電路設計中,為了便於信號處理以及控制精度,採用數字的方進行量階和的計算;同時,為了小電路規模,採用了時分復用的概念,用同一部分數字電路實現量階調整和的生成,增加了精度,小了電路的規模和功耗。
  3. On the basis of this introduction, some part of algorithm is improved. it includes : a new bit - allocation algorithm based on linear predication, that is to predicate the initial value used in bit - allocation procedure by the initial value of previous frame according to the correlation of the previous frame and the current frame. so the iterated loop number is reduced and the complexity of audio coding is reduced

    其中的一些主要改進有:提出一種基於線性的比特分配演算,即利用幀與幀之間存在的相關性,根據前幾幀的比特分配信息初值出當前幀的比特分配信息初值,通過合理設定比特分配信息初值,使得比特分配的迭代次數少,從而節省了音頻的運算量。
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