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A scalable speech coding scheme using compressive sensing and orthogonal mapping based vector quantization

M. S. Arun Sankar, P. S. Sathidevi · 2019 · 10 citationsRead the paper

A novel scalable speech coding scheme based on Compressive Sensing (CS), which can operate at bit rates from 3.275 to 7.275 kbps is designed and implemented in this paper. The CS based speech coding offers the benefit of combined compression and encryption with inherent de-noising and bit rate scalability. The non-stationary nature of speech signal causes the recovery process from CS measurements very complex due to the variation in sparsifying bases. In this work, the complexity of the recovery process is reduced by assigning a suitable basis to each frame of the speech signal based on its statistical properties. As the quality of the reconstructed speech depends on the sensing matrix used at the transmitter, a variant of Binary Permuted Block Diagonal (BPBD) matrix is also proposed here which offers a better performance than that of the commonly used Gaussian random matrix. To improve the coding efficiency, formant filter coefficients are quantized using the conventional Vector Quantization (VQ) and an orthogonal mapping based VQ is developed for the quantization of CS measurements. The proposed coding scheme offers the listening quality for reconstructed speech similar to that of Adaptive Multi rate - Narrowband (AMR-NB) codec at 6.7 kbps and Enhanced Voice Services (EVS) at 7.2 kbps. A separate de-noising block is not required in the proposed coding scheme due to the inherent de-noising property of CS. Scalability in bit rate is achieved in the proposed method by varying the number of random measurements and the number of levels for orthogonal mapping in the VQ stage of measurements.

What the full text shows

Working prototype
  • The proposed coding scheme achieves listening quality similar to AMR-NB at 6.7 kbps and EVS at 7.2 kbps. — MOS score equivalent to 6.7 kbps AMR-NB and 7.2 kbps EVS (At bit rates of 5.275 kbps and 7.275 kbps respectively)
  • The system demonstrates inherent noise reduction capabilities without a separate de-noising block. — 2-5 dB SNR improvement (Noisy inputs with SNR ranging from 0 to 10 dB)
  • The proposed sparse binary sensing matrix reduces computational complexity compared to Gaussian random matrices. — Elimination of multiplications (Transmitter side sensing process)
  • The coding scheme supports scalable bit rates. — 3.275 to 7.275 kbps (Varying number of random measurements and orthogonal mapping levels)
Setup
50 random speech samples from male and female speakers of the TIMIT B database, band-limited to 4 kHz; frame length of 20 ms (160 samples).
How
The authors implemented a speech codec using Compressive Sensing (CS) with a modified Binary Permuted Block Diagonal (BPBD) sensing matrix. They used a multi-feature categorization (ZCR and average power) to assign specific hybrid bases (DCT, Daubechies wavelets, or LPC) to voiced, unvoiced, and inactive frames, and employed orthogonal mapping-based Vector Quantization (VQ) for measurements.
Limitations
Performance depends on the reduction factor (s), with quality increasing as s increases from 0.3 to 0.7.

Summarized by Gemma 4 from the open-access full text · 8 Oct 2026. Read it