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A low-complexity, hardware architecture for a parametric, real-time, LSF speech decoder

机译:用于参数,实时,LSF语音解码器的低复杂性,硬件架构

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One of the main objectives of the widely used MPEG4 audio standard was to provide a technical framework for audio coding suited for low bit rate wireless communication. All variants of the MPEG4 audio coders use a Linear Prediction (LP) filtering system. The LP filter coefficients are calculated based on the Line Spectral Frequencies (LSF) parameters due to their excellent quantization characteristics. This work presents a hardware architecture that implements the conversion of the LSF parameters into the LP coefficients and the LP synthesis filter. The LSF conversion module is based on a simplified algorithm that cuts down on many arithmetic operations and thus results in a low gate count module. The architecture was implemented on Xilinx Virtex5 FPGA platform for performance measurement and verification. Module speeds are reported in terms of the number of clock cycles taken by each of their sub-modules to show suitability for real-time applications. Synthesis results are reported using various degrees of parallelism to explore the trade-off between gate count and time delay. Fixed-point implementation was carried out in a way that kept the original precision of the LSF parameters and input data in order to mitigate the severe impact of the tight bit rate budget.
机译:广泛使用的MPEG4音频标准的主要目标之一是为适用于低比特率无线通信的音频编码提供技术框架。 MPEG4音频编码器的所有变体使用线性预测(LP)过滤系统。由于它们出色的量化特性,基于线谱频率(LSF)参数来计算LP滤波器系数。这项工作介绍了一种硬件架构,它将LSF参数转换为LP系数和LP合成滤波器。 LSF转换模块基于简化算法,该算法减少了许多算术运算,从而导致低栅极计数模块。该架构在Xilinx Virtex5 FPGA平台上实现,用于性能测量和验证。模块速度以每个子模块所拍摄的时钟周期数来报告,以显示适合实时应用程序的适用性。据报道,使用各种程度的平行度来探讨栅极计数和时间延迟之间的权衡的合成结果。固定点实现以一种方式执行LSF参数和输入数据的原始精度,以减轻紧密比特率预算的严重影响。

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