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 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 a low bit-rate MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the performance of the coder is illustrated with some listening test results.

Currently, various approaches to trying the digitization of education is going on in the world. OECD has issued a report on an international comparison about the digitization progress. Accordingly in Japan, the Ministry of Education, Culture, Sports, Science and Technology has promoted discussions on the digitization while many boards of education at each local government have been discussing it. Each of them has different circumstances and a great difference in realities in its educational field. In this paper, we describe practical examples for the digitization of education tried in Oita prefecture using a cloud service and SNS, considering information technology in the field of education.

We worked toward development of a low bit-rate encoding algorithm and a low-complexity encoding algorithm for G.718/G.718-SWB, a new standard of International Telecommunication Union - Telecommunication Standardization Sector (ITU-T). Many key technologies including Band-Selective Shape-Gain Coding (BS-SGC) were developed for the standard. BS-SGC can encode the audio signal efficiently at a low bit-rate. Great improvement was confirmed by official subjective testing. Furthermore, our low-complexity technologies can reduce the computational complexity substantially with no quality degradation of the output signal. They were adopted as featured components of G.718/G.718-SWB, which were approved, respectively, in June 2008 (G.718) and in March 2010 (G.718-SWB).

The authors present and evaluate a technique for synchronizing the internal states of a code-excited-linear-prediction (CELP) encoder and decoder after the occurrence of frame erasure. The designed technique, called ldquoduplicated transmission (DT),rdquo uses some redundant information for realizing synchronization. The encoder performs encoding processes twice and sends two codes for each frame. One code is encoded by an encoder that is initialized. The code is used in cases where the previous frame is erased. An onset detector is combined with the DT technique to select the frames to which the DT should be applied. Subjective test results suggest that, by introducing DT selectively, the number of DT frames is reducible by about 80% without degrading the subjective quality. Results demonstrate that synchronization of the internal states is effective in cases of erasure of onset. The DT technique requires no additional algorithmic delay. For that reason, it would a better choice for particular applications for which the delay has a significant impact.

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.

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.

Predictive vector quantization (PVQ) for a bandwidth scalable LSP (line spectrum pairs) quantizer was studied. In the quantizer, LSP parameters obtained from a wide-band (16-kHz sampling) speech signal were quantized using LSP parameters encoded in a narrow-band (8-kHz sampling) layer. We designed several configurations of 16-bit quantizer utilizing PVQ and compared their performance. Simulation results showed that effective exploitation of interframe and intra-frame prediction improved objective performance. Compared with a baseline algorithm, PVQ using normal inter-frame prediction achieved better SD (spectral distortion) performance by 0.15 dB, and further improvement of 0.05 dB was gained by introducing novel interframe and intra-frame prediction. Robustness against frame erasure was considered by adopting switched prediction and "forgetting" capability. The designed memory-based PVQ algorithms were found to be robust up to 20% frame erasure rate.

We propose an energy extrapolation-based concealment algorithm for an excitation signal of a CELP decoder. In the concealment algorithm, adaptive codebook excitation is generated to maintain continuity of excitation energy on consecutive frames, and high-frequency regions are replaced with random excitation processed through a high-pass filter. The cut-off frequency of the filter is switched according to mode information. The algorithm was implemented on the 3GPP AMR standard decoder, and its performance was compared with that of the standard algorithm. Listening test results suggested that improved subjective quality was produced by the new algorithm.

The paper presents an efficient spectrum coding method for super-wideband (beyond 7 kHz, e.g. 10 kHz or 15 kHz bandwidth) speech signals based on a bandwidth expansion technique. By using a 7 kHz bandwidth speech signal, its frequency band over 7 kHz is generated by the expansion technique without violating the harmonics structure of the speech signal. The bandwidth expansion is performed by pitch filtering in a frequency domain. A 7 kHz bandwidth spectrum is used as a pitch filter state, and pitch filtering is performed toward a frequency band over 7 kHz. We adopted this pitch filtering based spectrum coding (PFSC) to our proposing 7/10/15 kHz bandwidth scalable coder. The scalable coder consists of an existing standard wideband coder as a base-layer and two PFSC coders as an enhancement-layer. One PFSC coder encodes a 7-10 kHz band spectrum at 4.4 kbit/s and the other a 10-15 kHz band spectrum at 2.2 kbit/s. When the AMR-WB coder at 15.85 kbit/s is used as the base-layer, the total bitrate of the scalable coder is 22.45 kbit/s and the total algorithmic delay is 30 ms. We conducted degradation category rating (DCR) tests for both 10 kHz and 15 kHz bandwidth signals. The results show that the DMOS score of the proposed coder is better than that of the 7 kHz bandwidth original signals in both bandwidth clean speech conditions. In addition, when G.722 at 56 kbit/s is used as the base-layer instead of the AMR-WB coder, the DMOS score of this scalable coder is close to that of the 7 kHz bandwidth original signals in both bandwidth audio conditions.

This paper presents the 4 kbit/s speech coding candidate submitted by AT&T, Conexant, Deutsche Telekom, France Telecom, Matsushita, and NTT for the ITU-T 4 kbit/s selection phase. The algorithm was developed jointly based on the qualification version of Conexant. This paper focuses on the development carried out during the collaboration in order to narrow the gap to the requirements in an attempt to provide toll quality at 4 kbit/s. This objective is currently being verified in independent subjective tests coordinated by ITU-T and carried out in multiple languages. Subjective tests carried out during the development indicate that the collaboration work has been successful in improving the quality, and that meeting a majority of the requirements in the extensive selection phase test is a realistic goal.

This paper presents a dispersed-pulse codebook for CELP coder. This codebook generates an excitation vector by convoluting dispersion vectors with signed pulses in an algebraic codevector. The dispersion vectors are obtained through training so the coding distortion to be reduced. An objective evaluation result shows that the coding distortion with this codebook is smaller than that with an algebraic codebook. The dispersed-pulse codebook is applied to a 4 kb/s CELP coder. Subjective evaluation results show that: (1) the fundamental performance of the 4 kb/s coder is equivalent to that of G.726 32 kb/s coder, and (2) the performances of the 4k b/s coder under some error and background noise conditions are equivalent to those of G.729 8 kb/s coder.