Ingemar Johansson

Bandwidth efficient AMR operation for VoIP

An example of a bandwidth efficient adaptive multi rate (AMR) system for Voice over IP (VoIP) is presented. In VoIP, packet losses cause degradation of the synthesized speech. The distortions may propagate over several consecutive frames, since predictors in the codec exploit inter-frame correlations to gain coding efficiency. To reduce the effects of packet loss, forward error correction (FEC) that adds redundant information to voice packets can be used. However, while FEC can reduce the effects of packet loss, it will increase the amount of bandwidth used by the voice stream, which is not desirable. In this paper we propose FEC methods like partial redundancy, selective redundancy for the most sensitive frames and parameter interpolation in conjunction with AMR codec mode adaptation, which secure the speech quality when using AMR for VoIP without increasing the bandwidth substantially.

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.

Method and arrangement for producing comfort noise in a linear predictive speech decoder

Comfort noise is produced in a linear predictive speech decoder which operates discontinuously, i.e., treats data frames which alternately represent speech information and background noise. During decoding of received data frames which contain background noise-describing parameters, a first number of these data frames which have been received directly before a speech frame are excluded and replaced with one or more background noise describing frames which have been received earlier. Another number of the background noise-describing frames which have been received immediately after a sequence of speech frames are also left out during the decoding and replaced by one or more background noise-describing frames which have been received before the sequence of speech frames. This results in a minimized degradation of the background noise information and gives an optimal comfort noise on the receiver side.

Self-clocked rate adaptation for conversational video in LTE

This paper describes a rate adaptation framework for conversational video services. The solution conforms to the packet conservation principle and uses a hybrid loss and delay based congestion control algorithm. The framework is evaluated over both simulated bottleneck scenarios as well as in a LTE system simulator and is shown to achieve both low latency and high video throughput in these scenarios, something that improves the end user experience.

Conex lite for mobile networks

The paper proposes ConEx Lite to ease the deployment of congestion bitrate policing in existing 4G and WiFi mobile networks with initial objectives to improve end user experience and backhaul network performance and dimensioning. ConEx Lite is a congestion management solution working at the bearer or tunnel layer independent from UE terminals and Internet endpoints or other transport protocol (e.g. TCP) implementations. It consists of simple functions that can be implemented on existing radio access and core nodes without negatively impacting the performance of the mobile network. ConEx Lite provides faster response to congestion and allows mobile operators to control the congestion volume policies according to their radio access technology and/or service mix.

Network assisted rate adaptation for conversational video over LTE, concept and performance evaluation

This work investigates rate adaptation of conversational video in a mobile system using Long Term Evolution, LTE, and where the adaptation is assisted by the radio network. The conventional way to do rate adaptation is through adaptation in the end-points where the transmission rate is selected based on measurements of received packets. This study investigates two network-based algorithms for rate adaptation, a rate fair algorithm that assigns the same rate to all conversational video users in a cell, and a resource fair algorithm that aims to give all users in the cell a fair amount of resources. Both algorithms are combined with delay based scheduling. Both network-based algorithms perform excellently. The delay stays low even when the resource utilization is close to 100%, and the video rate is adapted to the system load. As could be expected, the average user rates are higher with the resource fair algorithm. An end-point based adaptation algorithm is investigated as well, but it cannot keep a low delay at high load.

Adaptive Video with SCReAM over LTE for Remote-Operated Working Machines

Remote operation is a step toward the automation of mobile working machines. Safe and efficient teleremote operation requires good-quality video feedback. Varying radio conditions make it desirable ...