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.

Information spreading processes and advertising strategies are often studied by different epidemic models on a given topology. The goal of this paper is to discover and summarize the effect of underlying network topologies on a general spreading process. A complex set of different networks is studied by computer simulations from regular networks through random networks to different scale-free network topologies. The speed of spreading and the micro-scale features of these systems highlight the differences caused by different network topologies. This may help to plan for example advertising strategies on different social networks.

This paper describes the time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec. The TBE algorithm uses a nonlinear harmonic modeling technique that incorporates principles of time-domain envelope-modulated noise mixing. At 13.2 kbps, the super-wideband coding of speech uses as low as 1.55 kbps for encoding the spectral content from 6.4-14.4 kHz. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing significantly improved quality compared to the other standardized SWB codecs under both clean speech and speech with background noise.

A highly error resilient mode of the newly standardized 3GPP EVS speech codec is described. Compared to the AMR-WB codec and other conversational codecs, the EVS channel aware mode offers significantly improved error resilience in voice communication over packet-switched networks such as Voice-over-IP (VoIP) and Voice-over-LTE (VoLTE). The error resilience is achieved using a form of in-band forward error correction. Source-controlled coding techniques are used to identify candidate speech frames for bitrate reduction, leaving spare bits for transmission of partial copies of prior frames such that a constant bit rate is maintained. The self-contained partial copies are used to improve the error robustness in case the original primary frame is lost or discarded due to late arrival. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing improved quality under channel impairments as well as negligible impact to clean channel performance.

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.

This contribution presents the work in 3GPP on the standardization of a new audio codec for mobile multimedia applications including packet-switched streaming (PSS), multimedia messaging (MMS) and multimedia broadcast/multicast service (MBMS). design constraints, performance requirements, test plans, selection rules were finalized first. Next, extensive subjective listening testing was conducted. The test results showed good performance for the enhanced AAC+ and for the AMR-WB+ candidates. Enhanced AAC+ and AMR-WB+ are recommended for 3GPP Rel6 mobile multimedia services PSS, MMS, and MBMS. Both fixed-point and floating-point specifications are given in 3GPP in form of a C-code for both encoder and decoder. Conformance testing methods are specified as well

This paper examines the application of UDP-lite for unequal error detection in packet-switched speech transmission via Internet protocols (voice-over-IP) over UMTS radio channels. Traditionally, UDP is used as transport layer protocol, which contains a checksum that covers the complete packet. Thus, any packet with residual bit errors is discarded. Speech codecs like AMR, however, can tolerate bit errors in less sensitive parts of the bitstream. A more recent development, UDP-lite, provides unequal error detection with a partial checksum that covers only the sensitive parts of a packet. Thus, only packets with errors in important bits are discarded. We compare the use of UDP-lite for UMTS channels with convolutional and channels with turbo coding. The results show that the achievable quality improvement by applying UDP-lite depends on the residual bit error distribution of the chosen UMTS channel coding method. While we determined a quality improvement for channels with convolutional coding, we did not get an improvement for turbo coding. Furthermore, when combined with header compression, the convolutional coder with use of UDP-lite can reach the performance of the turbo coder with use of UDP

The introduction of packet switched (PS) networks (e.g., IMS) creates a need to evaluate speech transmission quality when using the 3GPP default speech codecs. 3GPP SA4 has created a work item in 3GPP Release 6 on "Performance characterization of default codecs for PS conversational application". France Telecom R&D and Siemens proposed a test framework consisting of a UMTS simulator for the air interface and an IP network simulator. ITU-T recommendation P.800 (1996) is used for the quality estimation of the transmission. The real-time conversational test results show that the AMR-NB (adaptive multirate narrow band) and AMR-WB (AMR wide band) speech codecs are well suited for PS conversational applications. Furthermore, the results clearly show a higher understanding when using AMR-WB rather than AMR-NB.

This contribution reports on the work in 3GPP release 6 on the standardization of a new audio codec for mobile multimedia applications including packet-switched streaming (PSS) and multimedia messaging (MMS). First, the design constraints, performance requirements, test plans, selection rules were finalized for both PSS/MMS audio codecs and for the extended AMR-WB codec (AMR-WB+). The candidate codecs were as follows: MPEG4 HE-AAC codec ("AAC+1') for low and high bit-rate range, coding technologies codec ("Enhanced AAC+") for low and high bit-rate range, and Ericsson, Nokia and VoiceAge AMR-WB+ candidate codec for low bit-rate range. Next, extensive subjective listening testing was conducted. The test results showed good performance for the enhanced AAC+ and for the AMR-WB+ candidates.

In this paper we present a voicing controlled, speech parameter based frame loss concealment for frames that have been encoded with the Adaptive Multi-Rate (AMR) speech codec. The missing parameters are estimated by interpolation and extrapolation techniques that are chosen in dependence of the voicing state of the speech frames preceding and following the lost frames. The voicing controlled concealment outperforms the conventional extrapolation/muting based approach and it shows a consistent improvement over interpolation techniques that do not distinguish between voiced and unvoiced speech. The quality can be further improved if additional information about the predictor states of predictively encoded parameters is available from a redundant transmission in future packets.

In order to make hidden Markov model (HMM) speech recognition suitable for mobile phone applications, Siemens developed a recognizer, Very Smart Recognizer (VSR), for deployment in future mobile phone generations. Typical applications will be name dialling, command and control operations suited for different environments, for example in cars. The paper describes research and development issues of a speech recognizer in mobile devices focusing on noise robustness, memory efficiency and integer implementation. The VSR is shown to reach a word error rate as low as 4.1% on continuous digits recorded in a car environment. Furthermore by means of discriminative training and HMM-parameter coding, the memory requirements of the VSR HMMs are smaller than 64 kBytes.

This paper describes an adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services. The speech codec is based on SB-CELP (splitband-code-excited linear prediction) with five modes operating bit rates from 24 kbit/s down to 9.1 kbit/s. The respective channel coding schemes are based on RSC (recursive systematic code) and UEP (unequal error protection). Both, source and channel codec are designed as homogenous as possible to guarantee robust transmission on current and future mobile radio channels.

We investigate various subband AEC systems in real handsfree situation, including FIR and IIR analysis and synthesis QMF filter banks in a polyphase structure. The adaptation in the subbands is performed by the affine projection algorithm in comparison to the NLMS algorithm. The IIR filters are superior to FIR filters in the sense that they lead to low signal delay and sharp frequency separation. Furthermore the computational complexity is greatly reduced by the use of IIR filters. The results show that splitting the signal into more subbands has advantages. Both wideband and narrowband speech signals have been evaluated.

This paper presents a filter structure which performs implicit decimation of the impulse response. As a result, the number of required operations is reduced or, equivalently, the impulse response length of the filter can be increased. Analysis in the frequency domain shows that this implicit decimation can be applied to systems that exhibit low-pass characteristics or have a smooth transfer function at high frequencies. Such behaviour can be assumed for many technical systems. For the determination of the optimal coefficients many well known algorithms for FIR systems can be used after a slight modification of the signal vector. The performance of implicit decimation is demonstrated for acoustic echo cancellation. Comparison with different algorithms shows that implicit decimation outperforms conventional FIR filtering.