The standardization of the codec for Immersive Voice and Audio Services (IVAS) was completed by the 3rd Generation Partnership Project in June 2024. The IVAS codec goes beyond traditional mono voice coding by representing and reproducing the spatial characteristics of sound, creating an immersive auditory experience. It opens the space for new applications in the realm of mobile communications and user-generated live content streaming such as immersive telephony and extended reality (XR) teleconferencing. The present paper provides a brief overview of the IVAS standard framework covering key features and properties and unique perspectives that are essential for understanding the underlying development and standardization processes that have led to this new 3GPP standard.

The 3GPP IVAS codec specifies an EVS-compatible stereo downmix as one of the key functionalities. This paper describes how this novel active downmix scheme has been devised to achieve high and stable quality from stereo input to EVS encoder/decoder with no additional algorithmic delay. An example of a network configuration for a multi-party conference using the proposed EVS-compatible downmix is also provided. This paper shows several fundamental schemes, of active processes, including adaptive weight and phase-compensated weight between two channels. Subjective listening test results show that the devised scheme’s quality is better than that of a passive downmix.

International audience

We address in this paper the problem of designing high bit-rate spherical vector quantization (VQ) to achieve transparent joint coding of angular values (azimuth and elevation) representing source directions in parametric spatial audio coding. We present the design of three coding methods using an optimized 3D grid and evaluate their performance/complexity at a bitrate of 16 bits per spatial direction. A comparison with VQ based on existing 3D spherical grids is provided for benchmarking purposes.

In this paper, we present experimental results on WebRTC voice quality as a function of LTE radio coverage. Different LTE radio conditions were tested by varying the radio path loss (with fast fading) in controlled test lab conditions. Voice quality was evaluated at different speech coding bit rates in terms of POLQA score and mouth-to-ear delay. Results show that, with acknowledged mode (AM) data transmission over LTE, degraded radio coverage translate essentially into increased end-to-end delay and jitter, with virtually no packet loss up to the coverage limit. Voice quality is analyzed by studying the influence of the specific jitter buffer used in WebRTC endpoints.

With the recent standardization of the Enhanced Voice Services (EVS) codec in 3GPP, mobile operators can upgrade their voice services to offer super-wideband (SWB) audio quality (with 32 kHz sampling rate). There is however one important use case which is currently limited by existing standards: hands free communication with wireless headsets, car kits, or connected audio devices often rely on Bluetooth, and the hands free-profile (HFP) in Bluetooth is currently limited to narrowband and wideband speech. Following the approach used to extend HFP to support wideband, we study in this paper the applicability of the SBC codec to further extend HFP to SWB. An evaluation of performance is provided taking into account Bluetooth system constraints.

A new codec for Enhanced Voice Services (EVS), the successor of the current mobile HD voice codec AMR-WB, was standardized by the 3rd Generation Partnership Project (3GPP) in September 2014. The EVS codec addresses 3GPP's needs for cutting-edge technology enabling operation of 3GPP mobile communication systems in the most competitive means in terms of communication quality and efficiency. This paper provides an in-depth insight into 3GPP's rigorous and transparent processes that made it possible for the mobile industry, with its many competing players, to successfully develop and standardize a codec in an open, fair and constructive process. This paper also enables an understanding of this achievement by providing an overview of the EVS codec technology, the standard specifications, and the performance of the codec that will elevate HD voice services to the next quality level.

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.

This paper presents the two new ITU-T Recommendations G.722 Annex D and G.711.1 Annex F, which are stereo extensions of the wideband codecs ITU-T G.722 and G.711.1 and their superwideband extensions (G.722 Annex B and G.711.1 Annex D). An embedded scalable structure is used to add stereo extension layers on top of the wideband or superwideband core coding. Wideband stereo modes are supported at the bit rates of 64/80 and 96/128 kbit/s for G.722 and G.711.1 (respectively), while superwideband stereo modes are supported at 80/96/112/128 and 112/128/144/160 kbit/s. The parametric stereo coding model is based on a frequency domain downmix, wideband inter-channel differences estimation, quantization and synthesis, low complexity coherence analysis and synthesis, stereo transient detection and stereo post-processing. An overview of formal ITU-T characterization listening tests illustrates the performance of these codecs.

This paper presents the lowest bitrate mode (64 kbit/s) of the new superwideband (SWB, 50-14000 Hz) coder, recently standardized as ITU-T G.722 Annex B. This mode provides a superwideband extension of G.722 at 56 kbit/s with one 8 kbit/s enhancement layer divided in two sub-layers. The resulting bitstream is compatible with ITU-T G.722 at 64 kbit/s and can be viewed as watermarking G.722 least significant bits (LSBs). The novel technologies in this mode include G.722 enhancements (noise feedback coding, scalable quantization in G.722 higher band), as well as multimode bandwidth extension (BWE). Selected ITU-T characterization test results and additional informal test results show that the 64 kbit/s mode of G.722 Annex B gives high SWB quality with low delay and complexity.

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.

In this paper, we present a novel, frequency-domain stereo to mono downmixing, which preserves the energy of spectral components and avoids setting the left or right channel as a phase reference. Based on this downmixing technique, a parametric stereo analysis-synthesis model is described in which subband stereo parameters consist of interchannel level differences and phase differences between the mono signal and one of the stereo channels (left or right). This model is applied to the stereo extension of ITU-T G.722 at 56+8 and 64+16 kbit/s with a frame length of 5 ms. AB test results are provided to assess the quality of the proposed downmixing technique. In addition, the quality of the proposed G.722-based stereo coder is compared against reference coders (G.722.1 at 24 and 32 kbit/s dual mono and G.722 at 64 kbit/s dual mono) for clean speech, noisy speech and music.

This paper proposes a new bit plane coding method for signed integer sequences. This method consists in mapping successive bit planes onto quinary symbols (+, -, 0, 1, EoP), where the symbol ldquoEoPrdquo stands for ldquoEnd of Planerdquo, and applying arithmetic coding. Sign bits are efficiently coded in combination with the corresponding most significant bit of non-zero integers. Moreover, bit planes are scanned and coded in a non-sequential manner to exploit the correlation between successive planes. Results for conversational transform coding of wideband speech and audio signals - sampled at 16 kHz - show that the performance/complexity of the proposed bitplane coder is near equivalent to non-embedded coding (stack-run coding), while offering additional flexibility (bitstream scalability).

This paper proposes a new model-based method for transform coding of audio signals. The input signal is mapped in "perceptual" domain by linear-predictive weighting filter followed by modified discrete cosine transform (MDCT). To provide bitstream scalability, model-based bit plane coding is then applied with respect to the mean square error (MSE) criterion. We present methods to estimate the symbol probability in bit planes assuming a generalized Gaussian model for the distribution of MDCT coefficients. We compare the performance of the proposed bitstream scalable coder with stack-run coding and ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is equivalent to or slightly worse than reference coders, but presents the nice advantage of being scalable. Performance penalty due to bitstream scalability is evident at low bitrates.

This paper describes a 12.2–32 kbps scalable wideband speech and audio coder interoperable with GSM enhanced full-rate (EFR). This coder, referred to as EFR-EV, is designed using the ITU-T G.729.1 multi-stage coding structure. Specifically, EFR-EV consists of three stages: a code-excited linear prediction (CELP) stage derived from EFR, time-domain bandwidth extension (TDBWE), and time-domain aliasing cancellation (TDAC). In this paper, we show that the G.729.1 extension layers (i.e. TDBWE and TDAC) are quite generic for scalable codec design in the sense that they can be applied to EFR with limited adjustments. In addition, we propose a minor modification of the bit allocation procedure in TDAC stage, exploiting spectral masking only for higher frequency bands. The performance of EFR-EV and G.729.1 are evaluated in terms of objective/subjective quality, algorithmic delay, and complexity.

This article presents ITU-T G.722 Appendix IV which is a Packet Loss Concealment (PLC) algorithm recently standardized by ITU-T for G.722 decoding in the presence of frame erasures. This algorithm is suitable for applications that may encounter frame erasures or packet losses with a special focus on complexity constraints For example, G.722 Appendix IV is very suitable for DECT Next Generation and VoIP using low cost devices. Besides, we also describe some minor algorithmic modifications to G.722 Appendix IV that improve subjective quality. We discuss G.722-PLC performance based on formal ITU-T test results as well as additional informal experiments.

This article presents a bitrate- and bandwidth-scalable coder submitted as a candidate to the qualification phase of ITU-T embedded G.711 wideband (G.711-WB) standardization. The encoder operates on 5 ms frames and bitrate can be set on a frame basis at either 64, 80 or 96 kbit/s. The input signal, which is by default sampled at 16 kHz, is split in two bands. The low band (0-4000 Hz) is coded by a low-complexity embedded pulse code modulation (PCM) coder at 8-10 bits/sample, including noise feedback at the encoder and PCM-specific post-filtering at the decoder. The high-band (4000- 8000 Hz) is coded by a time-domain aliasing cancellation (TDAC) coder derived from ITU-T G.729.1. Frame erasures are concealed using a low-complexity split-band algorithm derived from ITU-T G.722 Appendix IV. The proposed coder passed all requirements of the G.711WB qualification phase. We summarize and discuss formal ITU-T test results.

In this paper we present a model-based method to optimize the dead- zone of uniform scalar quantization in transform audio coding. The input signal is coded in modified discrete cosine transform (MDCT) domain by uniform scalar quantization followed by context-based arithmetic coding. The optimal deadzone is derived using a non- asymptotic method, assuming that the distribution of MDCT coefficients is approximated by a generalized Gaussian model. We show that deadzone optimization improves slightly quality, especially at low bitrates.

ITU-T G.729.1 is a scalable coder recently standardized in ITU-T for wideband telephony and voice over IP (VoIP) applications. Composed of three stages, this codec provides a scalable bitstream between 8 and 32 kbit/s both in narrowband and wideband. This paper describes the first stage which is a narrowband embedded CELP coder at 8 and 12 kbit/s. The 8 kbit/s layer ensures interoperability with ITU-T G.729 standard with a reduced complexity, and with a quality better than G.729 Annex A. At 12 kbit/s, G.729.1 reaches the quality level of the 11.8 kbit/s G.729 Annex E in spite of the embedded structure. The modifications brought to the original G.729 scheme to achieve this performance are explained and formal test results provided.

In this paper we present a new model-based method to code the transform coefficients of audio signals. The histogram of transform coefficients is approximated by a generalized Gaussian model for efficient model-based bit allocation and the spectrum is coded by scalar quantization followed by arithmetic coding. An example coder operating at 16 kHz and using predictive modified discrete cosine transform (MDCT) coding is described. We compare the performance of the proposed coder with ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is better than ITU-T G.722.1 at 24 kbit/s and equivalent at 32 kbit/s.

This paper describes the scalable coder - G.729.1 - which has been recently standardized by ITU-T for wideband telephony and voice over IP (VoIP) applications. G.729.1 can operate at 12 different bit rates from 32 down to 8 kbit/s with wideband quality starting at 14 kbit/s. This coder is a bitstream interoperable extension of ITU-T G.729 based on three embedded stages: narrowband cascaded CELP coding at 8 and 12 kbit/s, time-domain bandwidth extension (TDBWE) at 14 kbit/s, and split-band MDCT coding with spherical vector quantization (VQ) and pre-echo reduction from 16 to 32 kbit/s. Side information - consisting of signal class, phase, and energy - is transmitted at 12, 14 and 16 kbit/s to improve the resilience and recovery of the decoder in case of frame erasures. The quality, delay, and complexity of G.729.1 are summarized based on ITU-T results.

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

This paper describes a 8-32 kbit/s scalable speech and audio coder submitted as a candidate for the ITU-T G729-based embedded variable bitrate (G729EV) standardization. The coder is built upon a 3-stage coding structure consisting of: narrowband cascade CELP coding at 8 and 12 kbit/s, bandwidth extension based on wideband linear-predictive coding (WB-LPC) at 14 kbit/s, and MDCT coding in a WB-LPC weighted signal domain from 14 to 32 kbit/s. ITU-T test results showed that this coder passed all the requirements of the G729EV qualification phase

We present a new method, called Voronoi extension, for the design of low-complexity multi-rate lattice vector quantization (VQ). With this technique, lattice codebooks of arbitrarily large bit rates can be generated algorithmically and the problem of lattice codebook overload can be bypassed. We describe a practical multi-rate quantization system based on Voronoi extension and derived from the lattice RE/sub 8/. This system is applied to the TCX coding model using pitch prediction, so as to extend AMR-WB speech coding at high bit rates (in particular 32 kbit/s).

We consider a special case of Voronoi coding, where a lattice /spl Lambda/ in /spl Ropf//sup n/ is shaped (or truncated) using a lattice /spl Lambda/'={(m/sub 1/x/sub 1/,...,m/sub n/x/sub n/)|(x/sub 1/,...,x/sub n/)/spl isin//spl Lambda/} for a fixed m_=(m/sub 1/,...,m/sub n/)/spl isin/(/spl Nopf//spl bsol/{0,1})/sup n/. Using this technique, the shaping boundary is near-ellipsoidal. It is shown that the resulting codes can be indexed by standard Voronoi indexing algorithms plus a conditional modification step, as far as /spl Lambda/' is a sublattice of /spl Lambda/. We derive the underlying conditions on m_ and present generic near-ellipsoidal Voronoi indexing algorithms. Examples of constraints on m_ and conditional modification are provided for the lattices A/sub 2/, D/sub n/ (n/spl ges/2) and 2D/sub n//sup +/ (n even /spl ges/4).

We present the performance of several suboptimal algebraic quantizers in 24 dimensions. The Gaussian source is encoded at 1/2 bit per sample using the binary extended Golay code C/sub 24/ and the hexacode H/sub 6/. We also propose two new suboptimal decoding algorithms for the hexacode H/sub 6/.

This paper describes a fixed-rate quantizer using algebraic techniques as well as conventional stored codebooks to represent the wideband line spectral frequencies (LSF). It is based on two-stage split vector quantization with random and lattice codebooks in the first and second stage, respectively. We investigate the use of variable bit allocation in lattice quantization in order to capture outliers and reduce the overload distortion in spectral quantization, particularly for non-speech signals. Experimental results show that a variable bit allocation in split quantization yields performances slightly better than a fixed allocation at the same rate.