| Summary: | 碩士 === 國立交通大學 === 工學院聲音與音樂創意科技碩士學位學程 === 99 === With the scientific and technological progress in recent years, the speech recognition interface has been essential in many 3C products, such as hands-free car kit, smart phones, GPS, and other products. This paper will propose a speech recognition system, consists of speech enhancement algorithm, speech feature extraction, and recognition method, which employs recognizes the words of command voice. For the purpose of increasing the purity of the command voice, we adopt single-channel noise reduction (NR) as our speech enhancement algorithm. The speech feature extraction applies mel-scale frequency cepstral coefficient (MFCC). We employ dynamic time warping (DTW) to do dynamic matching after feature extraction. In experiment, we use the DSP TMS320C6416 to implement this speech recognition system. We investigate the computation of floating-point MFCCs to fixed-point ones. In particular, we have to check the scaling factors during each stage of the computation of MFCC. These scaling factors are carefully chosen such that the highest precision is achieved without overflow. The results of experiment show that the recognition rate will be improved at low SNR case.
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