A novel noise robust and low bit rate speech coding algorithm
Ümit Güz, Hakan Gürkan, Binboğa Sıddık Yarman
Abstract
Ümit Güz, Hakan Gürkan, Binboğa Sıddık Yarman
Abstract
In this work, a new noise robust and variable length frame based speech modeling method is introduced. This method consists of three major steps which includes noise removal algorithm, coding and encoding algorithms, respectively. Coding and encoding parts are developed based on SYMPES (systematic procedure for predefined envelope and signature sequence sets). These sets have been developed in two types which represent voiced and unvoiced parts of the speech signals separately in order to obtain more efficient coding strategy and higher compression ratio while preserving the perceptual quality of the speech signals. As an extension of our previous works our new framework is not only consider the coding of the clean speech signals but also noisy speech signals. The new noise robust module suppresses the noise and delivers the clean speech signal to the newly designed modeling part. The modeling part promises higher compression ratios by switching to the more appropriate type of predefined sets take into account the voiced and unvoiced frames.
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In this work, a new noise robust and variable length frame based speech modeling method is introduced. This method consists of three major steps which includes noise removal algorithm, coding and encoding algorithms, respectively. Coding and encoding parts are developed based on SYMPES (systematic procedure for predefined envelope and signature sequence sets). These sets have been developed in two types which represent voiced and unvoiced parts of the speech signals separately in order to obtain more efficient coding strategy and higher compression ratio while preserving the perceptual quality of the speech signals. As an extension of our previous works our new framework is not only consider the coding of the clean speech signals but also noisy speech signals. The new noise robust module suppresses the noise and delivers the clean speech signal to the newly designed modeling part. The modeling part promises higher compression ratios by switching to the more appropriate type of predefined sets take into account the voiced and unvoiced frames.
Key concepts: Speech coding, Speech recognition, Computer science, Harmonic Vector Excitation Coding, Codec2, Linear predictive coding, Coding (social sciences), Vector sum excited linear prediction