Ari Lakaniemi

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

VoIP in 3G networks: an end-to-end quality of service analysis

This paper presents the results of a quality of service (QoS) study for the VoIP service over 3G WCDMA networks. An end-to-end simulation platform has been used for this purpose. The simulations have been using an adaptive multi-rate (AMR) speech codec at 12.2 kbit/s with combination of RTP, UDP and IPv6 protocols. The simulated transmission path includes two radio links (uplink and downlink), connected with a packet switched core network and UTRAN radio access networks with several different radio transmission conditions. Furthermore, robust header compression is applied in both radio links. The results include buffering statistics, end-to-end delay estimates, and packet loss statistics.

Subjective VoIP speech quality evaluation based on network measurements

A system for measuring application-level voice over IP (VoIP) quality of service (QoS) is introduced. We also present a simple method for combining measurement data with speech coding simulation software, thus enabling reliable evaluation of subjective speech quality of the measured connection. An example measurement with mean opinion score (MOS) test results is presented and results are summarized.

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications

Variable model order LPC quantization

This paper presents a new method to apply variable bit-rate predictive quantization of the variable model order LPC parameters. In addition, the method is employed to interpolate the parameters within the analysis frame. The LPC model order selection algorithm is based on the characteristics of the input signal and on the performance of the LPC model. Hence, the variable bit-rate LPC quantization is source controlled. The number of quantized parameters needs to be identical in successive frames to be able to apply the predictive quantization and to interpolate parameters inside the frame. Therefore, the order of the LPC model of the previous frame needs to be expanded or reduced to be the same as the current frame LPC model. The advantage of variable model order LPC quantization is the lowered average bit-rate compared to a fixed rate while the speech quality remains the same.

Low Complex Audio Encoding for Mobile, Multimedia

Computational resources in a mobile terminal and in wireless transmission channels set strict requirements for audio coding in mobile multimedia applications. This paper introduces, a low complex audio encoding method enabling high-quality audio for mobile multimedia. The presented method is the low complexity version of the extended AMR-WB audio encoder in the 3GPP standards. The encoder was designed based on requirements of the 3GPP multimedia messaging service (MMS)

A general method for analyzing and synthesizing loss patterns

A method for producing synthetic packet loss patterns for voice over IP (VoIP) speech quality testing is described that is based on actual measurements of packet losses. The method has been designed to make no assumptions of network loss behaviour and can produce synthetic test data for different packet durations.