We developed scalable narrowband and wideband speech coding schemes based on the internet low bitrate codec (iLBC). Some of these newer codecs used the Discrete Wavelet Transform (DWT) instead of the Modified Discrete Cosine Transform (MDCT). This paper explores the choice of wavelet packet transform (WPT) for an application for a new scalable speech codec for IP networks using the Discrete Wavelet Transform (DWT) to encode the core-layer coding error in the enhancement layer. The issues regarding the design and in particular the choice of wavelet for the wideband codec are discussed. Experimental simulation results show that the DWT is a promising technique to use for encoding highly non-stationary signals such as the speech coding error. The wideband codec achieved speech quality equivalent to ITU-G.718 and similar codecs and is more robust. We also show that the best choice of wavelet depends on many factors including the order and number o f levels of the wavelet tree, delay and how well it approximates the human auditory system.

This paper presents a new scalable speech codec for IP networks using the discrete wavelet transform (DWT). The scalable narrowband speech coding scheme based on the internet low bitrate codec (iLBC) was previously presented and achieved speech quality equivalent to G.718 for narrowband signals. Whereas the performance of the core layer was satisfactory, the higher speech quality by the addition of the enhancement layer which employed the modified discrete cosine transform (MDCT) was desired. We propose the utilization of the DWT instead of the MDCT to encode the core-layer coding error in the enhancement layer. The experimental simulation results show that the DWT is a promising technique to use for encoding highly non-stationary signals such as the coding error.

High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.

Rate flexibility and high robustness to packet loss are the essential features of speech codec for voice communications over Internet Protocol (IP) networks. Multi-rate internet Low Bit-rate Codec (iLBC) is one of the speech codecs that possesses both properties and was presented by our previous papers. However the speech quality is limited in clean channel conditions because of the limitations of the current frame-independent coding scheme in time domain. In this paper, the benefits of scalable structure constructed by the addition of enhancement layer to the core layer of multi-rate iLBC are presented. The experimental simulation results show that the proposed framework can improve speech quality especially at high bit rates.