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.

In real-time teleconferences over mobile networks, cross-talk can significantly impact the performance of parametric stereo codecs, particularly at low bitrates. When multiple speakers overlap, stereo quality can be improved by independently encoding the left and right channels, especially when inter-channel correlation is low. This approach can be implemented using the dual-mono EVS coder, for example. The recently standardized 3GPP IVAS codec incorporates both a parametric stereo model and an independent left/right stereo model based on the dual-mono EVS codec. To address cross-talk, the 3GPP IVAS codec features a cross-talk classifier that uses a multivariate statistical model based on GCC-PHAT and other spatial cues, allowing for seamless switching between the two stereo models. Listening tests show that the IVAS stereo codec enhances performance in single-talker scenarios while maintaining quality in cross-talk segments.

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 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.

In this paper a novel technique is presented to efficiently mix traditional ACELP time domain coding with a frequency domain coding model to improve the quality of generic audio signals coded at low bitrates without additional delay. The paper discusses how to integrate parts of a traditional Algebraic Code Excited Linear Prediction (ACELP) speech codec to create a time-domain contribution which coexists with a frequency based coding model. A mechanism to determine the value of the time-domain contribution is proposed and a method is described how the frequency-domain contribution might be added without increasing the overall delay of the codec. The proposed method forms part of the recently standardised 3GPP EVS codec.

This paper presents two new post-processing techniques to address limitations of the deployed low bit rate speech codecs in case of unvoiced speech and background noise, and in case of music. Both post-processing techniques enhance the spectrum of the decoded excitation signal without increasing the codec algorithmic delay. The paper discusses how to integrate the enhancement procedure of unvoiced speech and background noise and of generic audio signals coded by low bit rate ACELP codecs. The proposed post-processing procedures are part of the AMR-WB interoperable modes of the recently standardized 3GPP EVS codec [1].

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 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.

In this paper we present a novel technique to enhance music signals encoded using a low bit rate CELP coder. The method is based on reduction of inter-tone quantization noise for decoded music signals without affecting the quality for speech signals. The proposed technique consists of two modules. The first module is used to discriminate between stable tonal sounds and other sounds and the second module is used to reduce the inter-tone quantization noise in the stable tonal segments. The inter-tone noise is reduced by means of spectral subtraction. The proposed method is a part of the newly standardised ITU-T G.718 codec.

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.

We present the G.EV-VBR winning candidate codec recently selected by Question 9 of Study Group 16 (Q9/16) of ITU-T as a baseline for the development of a scalable solution for wideband speech and audio compression at rates between 8 kb/s and 32 kb/s. The Q9/16 codec is an embedded codec comprising 5 layers where higher layer bitstreams can be discarded without affecting the decoding of the lower layers. The two lower layers are based on the CELP technology where the core layer takes advantage of signal classification based encoding. The higher layers encode the weighted error signal from lower layers using overlap-add transform coding. The codec has been designed with the primary objective of a high-performance wideband speech coding for error-prone telecommunications channels, without compromising the quality for narrowband/wideband speech or wideband music signals. The codec performance is demonstrated with selected test results.

This paper presents a new approach for accurate detection of voiced onsets in a speech signal. The knowledge of the precise location of the beginning of a voiced segment has several potential applications in speech compression (e.g. source-controlled variable bitrate coding, classification-based frame erasure concealment). The proposed onset detection algorithm is based on Teager's energy operator applied on sub-bands. The technique has been integrated in the signal classifier of the VMR-WB speech coding standard used for frame erasure (FE) concealment. A subjective listening test using MUSHRA methodology showed an improvement in frame error conditions with respect to the legacy VMR-WB classification.

Error propagation after a frame loss is an important factor in quality degradation for predictive speech coders. This is mainly due to the lack of synchronization in the adaptive codebook in the good frames following a frame erasure. This article presents a method for resynchronizing the glottal pulse after an erased frame. The method uses an extra frame delay at the decoder, and can be applied with or without additional side information. The approach has been implemented in the G.729.1 standard and results in improved decoder convergence after erased frames. Subjective tests have demonstrated that this improves perceived quality in the presence of frame erasures.

This paper presents novel techniques for source-controlled variable-rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by the Third-Generation Partnership Project 2 (3GPP2) for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third-generation wireless system. The codec utilizes efficient coding modes optimized for different classes of speech signal including generic coding based on AMR-WB for transients and onsets, voiced coding optimized for stable voiced signals, unvoiced coding optimized for unvoiced segments, and comfort noise generation for inactive segments. Several innovations enable very good performance at average bit rates below 8 kb/s for active speech coding. The article presents an overview of the codec and describes in detail some of the codec novel features: Robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Selected results from the Selection and Characterization tests of the codec illustrate its performance

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.

The description and design of the source-controlled variable-rate multimode wideband (VMR-WB) codec recently selected by the 3/sup rd/ Generation Partnership Project 2 (3GPP2) for the cdma2000/spl reg/ system in Rate-Set II are presented. The paper gives an overview of the codec and the methodologies that enable high quality wideband coding at average data rates ranging from TIA/EIA/IS-733 ADR (average data rate) to that of TIA/EIA/IS-127. The codec has three modes of operation at different average data rates and a fourth mode that is interoperable with 3GPP/AMR-WB (ITU-T G.722.2). Despite the interoperability constraint, the codec is capable of meeting the aggressive performance requirements through the use of novel techniques such as noise suppression, efficient signal classification, new coding types optimized for stable voiced and unvoiced frames, novel post-processing technique for periodicity enhancement in the lower frequency band, and improved frame erasure concealment mechanisms.

This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

We present in this paper new methods for achieving high- quality wideband speech at low rates using the ACELP algorithm. Several innovations are introduced to optimize the quality and minimize the complexity of the coder. A multi-rate wideband speech encoding algorithm based on these techniques was recently selected by 3GPP as the standard for AMR-WB, and is currently one of the candidates for the ITU- T wideband speech coder standard at around 16 kbit/sec. This standard was jointly developed by VoiceAge and Nokia.

The Third Generation Partnership Project (3GPP) and European Telecommunications Standards Institute (ETSI) have carried out development and standardisation of a wideband speech codec for GSM and the third generation mobile communication WCDMA system since 1999. The Adaptive Multi-Rate Wideband (AMR-WB) codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. The AMR-WB codec was jointly developed by Nokia and VoiceAge. AMRWB extends the audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to existing 2nd and 3rd generation mobile communication systems. The wideband speech service provided by the AMRWB codec will give mobile communication speech quality that even exceeds (narrowband) wireline quality.

Estimation of the spectral envelope in the frequency domain allows to avoid some problems of linear prediction (LP) algorithms for voiced speech. We present a low complexity method of spectral envelope estimation from harmonics for low rate coding. The method consists in computing the harmonic amplitude spectrum using pitch-synchronous DFT with length depending on voicing, modifying this spectrum outside the telephone bandwidth to simplify modeling of the useful bandwidth and interpolating it by a frequency-domain low-pass filter. An all-pole model is then fitted to this modified smoothed version of the harmonic spectrum. The method was implemented on the harmonic-stochastic excitation (HSX) vocoder and the performance was compared with the LP algorithm similar to that used in the G.729 speech coding standard. A-B comparative tests show an important increase in perceptual quality.

CELP speech coders have proved to be efficient for the coding of speech at medium and low bit rates. The NSA has recently introduced the Federal Standard 1016, CELP speech coder at 4800 bps. We propose two algorithms that reduce the complexity of this coder by a factor 2. The first algorithm uses the particular structure of the standardised code book to simplify the calculation of the cross-correlation term. The second algorithm is a multistage method by subsampling that eliminates nonrelevant code book sequences for the calculation of the energy term.>