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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度硬件架构

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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音频标准的主要目的之一是为适用于低比特率无线通信的音频编码提供一个技术框架。 MPEG4音频编码器的所有变体都使用线性预测(LP)过滤系统。 LP滤波器系数由于其出色的量化特性而基于线频谱频率(LSF)参数进行计算。这项工作提出了一种硬件体系结构,该体系结构将LSF参数转换为LP系数和LP合成滤波器。 LSF转换模块基于简化的算法,该算法减少了许多算术运算,因此导致门数低的模块。该架构是在Xilinx Virtex5 FPGA平台上实现的,用于性能评估和验证。根据每个子模块占用的时钟周期数来报告模块速度,以显示其适用于实时应用。使用各种并行度报告综合结果,以探索门数与时间延迟之间的权衡。定点实现的方式应保持LSF参数和输入数据的原始精度,以便减轻紧缩的比特率预算的严重影响。

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