This paper introduces a novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates. The method is enabled only in stable voiced speech frames, and preserves the original time scale at the end of each frame. This feature helps to avoid artifacts and simplifies an encoder implementation. The signal modification includes a classification algorithm as an integral part. The classification algorithm detects the frames most suitable for signal modification and low bit rate coding, and can be employed in a rate selection module of variable bit rate (VBR) codecs. In this paper, the signal modification method is applied in an experimental VBR wide-band speech codec derived from the 3GPP adaptive multirate wideband (AMR-WB) standard (ITU-T Recommendation G.722.2). The codec fulfills the system requirements of IS-95/CDMA2000 Rate Set II, operating at source coding bit rates 12.65, 6.2, and 1.0 kb/s. The signal modification is used in the 6.2 kb/s mode dedicated for voiced speech frames. Listening test results demonstrate the good performance of the proposed method. The signal modification method is used in the Nokia/VoiceAge codec that was declared in April 2003 as the winner of the selection phase in the 3GPP2 CDMA2000 wide-band speech codec standardization.

This paper presents a coding method for successive pitch periods of a speech signal. In the proposed method, a priori knowledge of the statistical properties of successive pitch periods is used by designing a shaped lattice structure that covers the most probable points in the pitch space. We also present briefly a pitch search algorithm for the shaped lattice. An example implementation of shaped lattice coding is given for a modified IS-641 speech coder. Based on the simulation results, the proposed scheme achieves capacity savings compared to the conventional methods.

A signal model used in waveform interpolation (WI) coding assumes implicitly that the pitch period remains constant within a waveform to be extracted. This assumption is utilized for determining waveform boundaries and for forming the discrete Fourier transform (DFT) of the extracted waveform. Therefore, the waveforms do not describe the original signal faithfully during pitch changes even if an accurate pitch estimate is available for each time instant. Due to the distortions occurring in waveform extraction, the original signal cannot be recovered. In this paper, we discuss assumptions of the signal model and present a waveform extraction algorithm which provides asymptotically perfect reconstruction with vanishing quantization error. The algorithm uses cubic B-spline interpolation as the continuous approximation of the signal. The full benefit of the waveform extraction algorithm is obtained when an accurate pitch estimate is available.