A spatial audio scene consists of discrete audio sources and ambience. The 3D audio image is observed due to the directional sounds, but even more important is the reverberation and so called room effect caused by the properties of the space, the sources and listener location. It is obvious that a human being is able to capture the 3D image using the signal from left and right ear. Hence, two audio channels are sufficient to represent the spatial audio image for the listener. An efficient transmission and representation of a spatial audio image using two channels requires a specific coding and rendering algorithm for the audio content. In this paper we present novel mechanisms to efficiently parameterize, quantize and represent the spatial audio signal.

This communication presents the recently standardized superwideband (SWB) extensions of ITU-T G.718 and G.729.1. These extensions were standardized as G.718 annex B and G.729.1 annex E. The SWB functionality is implemented using embedded scalable layers on top of the wideband (WB) core codecs, and it extends the bit rate of the codecs to 48 and 64 kbit/s for the G.718 and G.729.1, respectively. The main technology is a two-mode SWB coding method of the high frequencies. In addition, the G.729.1 SWB extension enhances the lower frequency range. The codec performance is illustrated with some listening test results extracted from the ITU-T Characterization phase.

Recent trends in speech and audio codec standardization include scalability and extending the signal bandwidth beyond wideband (WB) to superwideband (SWB). In this paper we introduce a SWB extension for the ITU-T G.718 WB codec. In the SWB extension the high frequency content is generated utilizing the quantized MDCT domain coefficients of the WB core, which enables low additional delay. The proposed implementation is scalable with 4 kbps layers. In the first layer two different coding modes are used depending on the input signal type. The proposed SWB extension is evaluated with listening tests and complexity analysis.

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.

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.