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Model-based multirate representation of speech signals and its application to recovery of missing speech packets

机译:基于模型的语音信号多速率表示及其在丢失语音包恢复中的应用

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When the samples of a critically sampled speech signal are lost, objectionable aliasing occurs and perfect recovery of the original speech becomes impossible. In this work, a multirate state-space representation of the autoregressive (AR) speech process is derived to describe the generation of regularly missing-sample speech sequences. Next, a new sample-interpolation algorithm based on the multirate Kalman reconstruction filter is proposed to reduce speech quality degradation caused by packet losses. This method is used together with packet interleaving configuration, thereby simplifying the recovery of missing packets to the interpolation of missing samples. Subjective tests indicate that the proposed Kalman-based sample-interpolation algorithm performs better than the conventional odd-even sample-interpolation procedure for mitigating the effects of random packet losses in 64 kb/s PCM codes. The tolerable packet loss rate P/sub L/, which is strictly input-speech-dependent, can be as high as 10-20% with Kalman interpolation. These observations are based on computer simulations in terms of signal-to-noise ratio (SNR) values, waveform reconstruction plots, error spectral shapes, and summaries of informal listening tests.
机译:当严重采样的语音信号的采样丢失时,会产生令人讨厌的混叠现象,并且无法完美恢复原始语音。在这项工作中,自回归(AR)语音过程的多速率状态空间表示被导出以描述规则缺失样本语音序列的生成。接下来,提出了一种新的基于多速率卡尔曼重构滤波器的样本插值算法,以减少由于丢包引起的语音质量下降。此方法与数据包交织配置一起使用,从而简化了丢失数据包对丢失样本的插值的恢复。主观测试表明,所提出的基于Kalman的样本插值算法在减轻64 kb / s PCM码中随机数据包丢失的影响方面比常规的奇偶样本插值程序性能更好。严格取决于输入语音的容忍丢包率P / sub L /,使用卡尔曼插值法可以高达10-20%。这些观察是基于计算机模拟的信噪比(SNR)值,波形重构图,误差频谱形状以及非正式听觉测试的摘要。

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