Daniel J. Sinder

Improved error resilience for volte and VoIP with 3GPP EVS channel aware coding

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.

Super-wideband bandwidth extension for speech in the 3GPP EVS codec

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.

Recent Speech Coding Technologies and Standards

This chapter presents an overview of recent developments in conversational speech coding technologies, important new algorithmic advances, and recent standardization activities in ITU-T, 3GPP, 3GPP2, MPEG and IETF that offer a significantly improved user experience during voice calls on existing and future communication systems. User experience is determined by speech quality, hence network operators are very concerned about quality of speech coders. Operators are also concerned about capacity, hence coding efficiency is another important measure. Advanced speech coding technologies provide the capability to both improve coding efficiency and user experience. One option to improve quality is to extend the audio bandwidth from traditional narrowband to wideband (16 kHz sampling) and super-wideband (32 kHz sampling). Another method is in increasing the robustness of the coder against transmission errors. Error concealment algorithms are used which substitute the missing parts of the audio signal as far as possible. In packet-switched applications (VoIP systems), special mechanisms are included in jitter buffer management (JBM) algorithms to maximize sound quality. It is of high importance to ensure standardization and deployment of speech coders that meet quality expectations. As an example of this, we refer to the Enhanced Voice Services (EVS) project in 3GPP that is developing the next generation speech coder in 3GPP. The basic motivation for 3GPP to start the EVS project was to extend the path of codec evolution by providing super-wideband experience at around 13 kb/s and better quality for music and mixed content in conversational applications. Optimized behavior in VoIP applications is achieved through the introduction of high error robustness, jitter buffer management, inclusion of source-controlled variable bit rate operation, support of various audio bandwidths, and stereo.

eAMR: Wideband speech over legacy narrowband networks

This paper introduces eAMR (enhanced-AMR), a novel technique for delivering wideband speech over existing narrowband networks. Instead of using a completely new wideband speech coder which would require new infrastructure, as is the case e.g. for AMR-WB or EVS, eAMR is based on the existing AMR (narrowband) codec, which is already widely deployed. eAMR uses an efficient coding model to represent the high frequencies of the speech signal, and combines it with watermarking technology to hide this data within a normal narrowband AMR bitstream. As a result, eAMR is a wideband codec which is fully compatible with the existing AMR network infrastructure, and therefore can be deployed as a handset-only feature.

An Objective Evaluation Methodology for Blind Bandwidth Extension.

In this paper we introduce an objective evaluation methodology for Blind Bandwidth Extension (BBE) algorithms. The methodology combines an objective method, POLQA, with a bandwidth requirement, based on a frequency mask. We compare its results to subjective test data, and show that it gives consistent results across several bandwidth extension algorithms. Additionally, we show that our latest BBE algorithm achieves quality similar to AMR-WB at 8.85 kbps, using both subjective and objective evaluation methods.