A novel algorithm for low bit rate speech compression using a hybrid LP-harmonics model

N. Abu-Shikhah, Mohamed Deriche
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Abstract

We present a new LP-harmonic speech codec. At the coder speech signal is pre-processed, and an LP analysis is performed, together with pitch estimation and voicing decision. At the decoder and when the frame is voiced, the encoded parameters are used to estimate the spectrum envelope, extract and classify the harmonics as either strong or weak depending on their relative distance from multiples of the fundamental frequency. Strong harmonics parameters are then used to generate pure sinusoids. While weak harmonics are used to generate a mixed signal of a pure sinusoid and a random-like signal. For unvoiced frames, the excitation of the LP filter is generated as a white noise signal. The proposed model allows for the mixing of strong and weak periodic signals together with random signals to produce an excitation input that results in natural speech. Informal testing of the coder working at 1.82 kb/s showed that the output speech has high intelligibility, with quality comparable to that of a 4 kb/s sinusoidal codec.
基于混合lp -谐波模型的低比特率语音压缩新算法
提出了一种新的低阶谐波语音编解码器。在编码器处,对语音信号进行预处理,进行LP分析,以及基音估计和发声决策。在解码器和帧配音时,编码参数用于估计频谱包络,提取和分类的强或弱的谐波取决于它们的相对距离的倍数基频。然后使用强谐波参数生成纯正弦波。而弱谐波则用于产生纯正弦波和随机信号的混合信号。对于非浊音帧,低频滤波器的激励以白噪声信号的形式产生。所提出的模型允许将强、弱周期信号与随机信号混合在一起,以产生产生自然语音的激励输入。对工作在1.82 kb/s的编码器进行的非正式测试表明,输出语音具有很高的可理解性,其质量可与4 kb/s的正弦编解码器相比较。
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