The new 3GPP IVAS codec for immersive voice and audio communication services in 5G networks supports a variety of spatial audio formats beyond mono and stereo voice and audio, and even combinations of them. This paper presents the OMASA combined audio format which provides a joint coding of object-based audio and metadata-assisted spatial audio (MASA). It enables immersive communication experiences, e.g., in teleconferencing scenarios where a mobile end-device captures the environmental audio in MASA format while individual talkers’ voices are captured with their own microphones as separate audio objects. This paper describes coding methods newly developed and optimized depending on the available bitrate and the number of audio objects to efficiently deliver a 3D audio experience at a wide OMASA bitrate range from 13.2 kbps to 512 kbps. The benefits of the presented joint coding of audio formats over separate coding at the same bitrates are presented in terms of subjective evaluation and computational resources.

The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

The recently standardized codec for Enhanced Voice Services (EVS) consists of a number of modes to achieve its high coding flexibility. In this paper we focus on techniques that enable a seamless switching between two linear prediction based modes running at different sampling rates within this codec. The first one deals with an efficient conversion of the linear prediction filter coefficients. The other one is based on a constrained-memory ACELP called transition coding (TC) that significantly limits the inter-frame long-term dependency. We show that the use of TC can be successfully extended to improve quality also in coding other transitions, e.g. strong onsets of voiced speech.

We present several new methods for coding of spectrum coefficients in embedded audio codecs. In audio codecs, the spectrum is often divided in sub-bands, and quantized using a vector quantization. In many applications, the limited bit-budget allocated to the vector quantization is not sufficient to cover all the sub-bands, and the distortion of the spectrum in the unquantized sub-bands frequently causes audio quality degradation. The presented methods have been designed to improve the representation efficiency of the spectrum in these sub-bands. They consist of a preprocessing of the spectrum before the vector quantization, a correction of the encoded spectral envelope after the quantization, and a filling of the unquantized spectrum sub-bands. These methods have been used in the superwideband embedded extensions of the audio codecs G.711.1 Annex D and G.722 Annex B, recently standardized by the ITU-T.

This paper presents high quality monaural superwideband extensions to G.711.1 and G.722, recently standardized as Recommendations ITU-T G.711.1 Annex D and G.722 Annex B. The superwideband (50-14000 Hz) functionality is achieved using embedded scalable structure that adds extension layers on top of the wideband core codecs. The bit rates are extended to 96/112/128 and 64/80/96 kbit/s for G.711.1 and G.722, respectively. The main technologies include lower and higher band (0-4 kHz and 4-8 kHz) enhancements, 8-14 kHz bandwidth extension and transform coding based on algebraic vector quantization. The codecs' performance is illustrated with listening test results extracted from formal ITU-T Characterization tests.

This paper presents a new technique for the class of code-excited linear prediction speech codecs designed to reduce error propagation after lost frames. Its principle consists in replacing the interframe long-term prediction with a glottal-shape codebook in the subframe containing the first glottal impulse in a given frame. This technique, independent of previous frames, is of particular interest in voiced speech frames following transitions as these frames are the most sensitive to frame erasures. It is a basis of a structured coding scheme calledtransition coding(TC). The TC greatly improves codec performance in noisy channels while maintaining clean channel performance. It is a part of the new embedded speech and audio codec recently standardized as Recommendation G.718 by ITU-T.

CELP-based codecs typically rely on prediction to achieve their high coding efficiency. On the other hand, the prediction makes these codecs sensitive to frame erasures as errors propagate beyond the erased frame. We present a technique that significantly limits the error propagation by replacing inter-frame long-term prediction with a non-predictive glottal-shape codebook. The technique was implemented in the winning candidate of the EV-VBR baseline codec selection by ITU-T in March 2007. To maintain the performance in clean channel, this transition mode coding technique was used only in frames following voiced onsets frames, i.e. the frames most sensitive to frame errors.