Lasse Laaksonen

Overview of the EVS codec architecture

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.

Scalable superwideband extension for wideband coding

Recent trends in speech and audio codec standardization include scalability and extending the signal bandwidth beyond wideband (WB) to superwideband (SWB). In this paper we introduce a SWB extension for the ITU-T G.718 WB codec. In the SWB extension the high frequency content is generated utilizing the quantized MDCT domain coefficients of the WB core, which enables low additional delay. The proposed implementation is scalable with 4 kbps layers. In the first layer two different coding modes are used depending on the input signal type. The proposed SWB extension is evaluated with listening tests and complexity analysis.

Media coding for the next generation mobile system LTE

Introduction of LTE (Long Term Evolution) brings enhanced quality for 3GPP multimedia services. The high throughput and low latency of LTE enable higher quality media coding than what is possible in UMTS. LTE-specific codecs have not yet been defined but work on them is ongoing in 3GPP. The LTE codecs are expected to improve the basic signal quality, but also to offer new capabilities such as extended audio bandwidth, stereo and multi-channels for voice and higher temporal and spatial resolutions for video. Due to the wide range of functionalities in media coding, LTE gives more flexibility for service provision to cope with heterogeneous terminal capabilities and transmission over heterogeneous network conditions. By adjusting the bit-rate, the computational complexity, and the spatial and temporal resolution of audio and video, transport and rendering can be optimised throughout the media path hence guaranteeing the best possible quality of service.

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.

Quality evaluation of the G.EV-VBR speech codec

ITU-T has selected the candidate submitted by Ericsson, Nokia, Motorola, VoiceAge, and Texas Instruments as the baseline for the G.EV-VBR coding standard. G.EV-VBR is an embedded scalable speech codec that uses state-of-the-art technology to provide the most efficient encoded speech available for various real-time applications. EV-VBR encodes both narrowband (NB) and wideband (WB) speech signals starting at 8 kbps. Near perfect wideband representation is achieved at 32 kbps for all signal types. The bit stream is divided into five robust layers, providing sufficient granularity, in particular for VoIP applications. In addition, an extension to the codec will provide super- wideband and stereo capability by adding layers to the codec. Extensive listening tests were conducted during the ITU-T selection phase to support selection of the best- performing candidate. The selected EV-VBR candidate passed 69 of 70 required and 25 of 28 objective terms of reference.

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.

Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard

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.

Using noise reduction in mode selection and pitch search

We present a novel method of exploiting noise reduction prior to mode selection in classification-based speech coding. Certain parameter estimation and mode selection is performed on a denoised signal to ideally remove the non-speech components before signal classification, although the original input is coded. This method is employed in a recent ITU-T standard, G.718, which provides state-of-the-art performance for narrowband and wideband speech between 8 and 32 kbps. We also present the pitch tracking algorithm of G.718. Open-loop pitch lags are estimated from the denoised signal.

Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS codec

This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.