Real-Time Implementation of Wideband Adaptive Multi Rate (AMR-WB) Speech Codec Using TMS32OC6201 


Vol. 29,  No. 9, pp. 1337-1344, Sep.  2004


PDF
  Abstract

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

  Statistics
Cumulative Counts from November, 2022
Multiple requests among the same browser session are counted as one view. If you mouse over a chart, the values of data points will be shown.


  Cite this article

[IEEE Style]

S. Lee and K. Bae, "Real-Time Implementation of Wideband Adaptive Multi Rate (AMR-WB) Speech Codec Using TMS32OC6201," The Journal of Korean Institute of Communications and Information Sciences, vol. 29, no. 9, pp. 1337-1344, 2004. DOI: .

[ACM Style]

Seung-won Lee and Keun-sung Bae. 2004. Real-Time Implementation of Wideband Adaptive Multi Rate (AMR-WB) Speech Codec Using TMS32OC6201. The Journal of Korean Institute of Communications and Information Sciences, 29, 9, (2004), 1337-1344. DOI: .

[KICS Style]

Seung-won Lee and Keun-sung Bae, "Real-Time Implementation of Wideband Adaptive Multi Rate (AMR-WB) Speech Codec Using TMS32OC6201," The Journal of Korean Institute of Communications and Information Sciences, vol. 29, no. 9, pp. 1337-1344, 9. 2004.