Stefan Bruhn

AMR-WB+: a new audio coding standard for 3rd generation mobile audio services

Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase, 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides a unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

Extended AMR-WB for high-quality audio on mobile devices

This article presents the architecture, performance, and application scenarios of the AMR-WB+ (extended AMR-WB) audio codec, which provides high quality at exceptionally low rates, and consistent quality over all audio types. This codec was recently selected by 3GPP and DVB to support low-bit-rate audio and audiovisual applications on mobile networks

Partial spectral loss concealment in transform coders

A novel error concealment technique for partial spectral loss in transform coders is presented. Based on amplitude and phase inter- and intra-frame correlations, an algorithm for missing spectral coefficient restoration is derived. The algorithm restores the missing spectral coefficients by predicting the amplitude using energy matching and the phase using group delay conservation principles. Results from listening tests illustrate the performance of the proposed algorithm.

ITU-T G.EV-VBR baseline codec

We present the Q.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.

Adaptive thresholds for AMR codec mode selection

The speech codecs from the adaptive multi-rate (AMR) codec family enable provisioning of excellent speech quality, at the same time providing a way forward towards state-of-the-art, spectrally efficient, high capacity cellular networks. One straightforward way to characterize the benefit of AMR speech codecs is that the robustness to interference and noise in radio networks is increased and that this advantage over other, nonadaptive, speech codecs can be capitalized on in several different ways, e.g., by enhancing speech quality or improving system capacity. In this paper, improved mode adaptation, where codec mode switching thresholds are adaptive to radio conditions, is discussed. Example simulations show that an adaptive thresholds algorithm applied to GSM can significantly improve objective speech quality. Corresponding improvements were also found in informal listening tests.

Standardization of the new 3GPP EVS codec

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 3GPPs 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 3GPPs 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.

Packet-loss concealment technology advances in EVS

EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results.

Joint capacity and quality evaluation for AMR telephony speech in WCDMA systems

The adaptive multi-rate (AMR) speech codec is the mandatory speech codec for WCDMA systems. The codec supports eight different source rates ranging from 12.2 kbit/s down to 4.75 kbit/s. This paper evaluates different ways of allocating AMR rates to users on the downlink in a WCDMA system. A novel system performance concept is introduced based on a user satisfaction metric that tries to combine the user experience from different speech quality levels as well as events like blocking and dropping. System simulations show that the AMR codec introduces a significant trade-off between capacity and quality for the speech service. By allocating AMR modes based on the system load, the quality and capacity trade-off can be efficiently balanced and high system performance can be achieved for a wide range of offered loads.

Matrix product quantization for very-low-rate speech coding

Efficient block coding methods for LPC information play an essential role in very-low-rate speech coding systems. The subject of this contribution is a new suboptimal matrix quantization scheme for LPC parameters, called matrix product quantization (MPQ), which operates at bit rates between 300 and 700 b/s. MPQ encodes sequences of LPC parameter vectors using a product formulation of two matrices which describe the average parameter vector and the temporal contour. In fixed-rate coding systems for mobile communication, MPQ achieves a very high coding efficiency at a low coding delay. Compared to the multi-frame coding method (MFC) of Kemp et al. (1991), which causes a delay of 8 frames, the MPQ scheme operates more efficiently even at a coding delay of only 3 frames. Applying MPQ to a variable-rate segment vocoder, a bit rate reduction of 50% compared to memoryless VQ is obtained at a frame period of 20 ms.

Continuous and discontinuous power reduced transmission of speech inactivity for the GSM system

Discontinuous transmission (DTX) or variable bit rate operation during speech inactivity is applied in various cellular speech communication systems. This is a technique in essence to turn off the transmission during periods of speech silence. The purpose is to reduce interference caused to other users concurrently transmitting on the air interface and to save battery power in the mobile stations. A major problem with this technique arises for emerging communication systems with link adaptation as both continuous channel tracking and continuous inband back channels are disabled when transmission is turned off. This paper presents a novel alternative method to conventional DTX which allows to keep the transmission continuous or almost continuous but which maintains or even exceeds the gain of the conventional method.