Implementation of FIR Interpolation Filter on TMS320C6713 for VoIP Analysis

Jasvir Singh, H. Singh, Sarabjeet Singh
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引用次数: 5

Abstract

In voice over internet protocol (VoIP) system, the speech signal is degraded when passed through the network layers. The speech signal is processed through best effort policy based IP network, which include delay, packet loss and jitter network problems. The quality of VoIP speech signal can be improved by filtering the degraded VoIP speech signal through multirate filters. The work in this paper deals with the implementation of Interpolated finite impulse response (IFIR) filtering algorithm on degraded VoIP speech signal using TMS320C6713 DSP processor. The filter was designed using polyphase structure. The results of implementation experiment indicate the improvement of signal quality. The results are validated through the measurement of enhancement signal using perceptual evaluation of speech quality (PESQ) measurement, which shows the increase in MOS score at different packet loss rates (PLR)
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在TMS320C6713上实现用于VoIP分析的FIR插值滤波器
在网络语音协议(VoIP)系统中,语音信号在通过网络层时存在降级问题。语音信号是通过基于最大努力策略的IP网络进行处理的,该网络存在时延、丢包和抖动等问题。通过多速率滤波器对降级的VoIP语音信号进行滤波,可以提高VoIP语音信号的质量。本文的工作是利用TMS320C6713 DSP处理器对退化的VoIP语音信号实现插值有限脉冲响应滤波算法。滤波器采用多相结构设计。实现实验结果表明,信号质量得到了改善。通过语音质量感知评价(PESQ)测量增强信号对结果进行了验证,结果表明,在不同丢包率(PLR)下,MOS分数有所提高。
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