The ability to provide seamless and adaptive quality of service (QoS) in heterogeneous environment is the key to the success of the next generation wireless communication systems. There has been a considerable amount of QoS research going on recently but the main portion still occurred in the context of individual architecture components and much less progress has been made in addressing the issue of an overall QoS architecture for the mobile Internet. In this article, the state-of-the-art QoS techniques and standardization activities are summarized. Important challenges in building ubiquitous QoS framework over the heterogeneous environment are examined in detail. Then a three-plane QoS framework based on QoS signaling architecture and policy framework is proposed to provide seamless support for future applications.

A gradient-descent based optimization procedure is applied to the window sequence used for linear prediction (LP) analysis of the ITU-T G.729 CS-ACELP coder. By replacing the original window of the standard by the optimized versions, similar subjective quality is obtainable at reduced computational cost and / or lowered coding delay. In addition, an optimization strategy is described to find the line spectral frequency (LSF) interpolation factor. The t gradie reflec outlin w[k],

This paper presents a new Analysis-by-Synthesis (AbS) technique for joint optimization of the excitation and model parameters based on minimizing the closed loop synthesis error instead of the linear prediction error. By minimizing the synthesis error, the analysis and synthesis stages become more compatible. Using a gradient descent algorithm, LSPs for a given excitation are optimized to minimize the error between the original and the synthesized speech. Since the optimization starts from the LPC solution, the synthesis error is guaranteed to be lower than that obtained using the LPC coefficients. For the ITU G.729 codec, there is about 1dB of improvement in the segmental SNR for male and female speakers over 4 to 6 second long sentences. By adding an extra optimization step, the technique can be incorporated into the LPC, multi-pulse LPC and CELP-type speech coders.

The ability to provide seamless and adaptive quality of service (QoS) in a heterogeneous environment is the key to the success of the next generation wireless communication systems. There has been a considerable amount of QoS research going on recently but the main portion still occurred in the context of individual architecture components and much less progress has been made in addressing the issue of an overall QoS architecture for the mobile Internet. In this article the state-of-the-art QoS techniques and standardization activities are summarized. Important challenges in building a ubiquitous QoS framework over the heterogeneous environment are examined in detail. Then a three-plane QoS framework based on QoS signaling architecture and policy framework is proposed to provide seamless support for future applications.

This paper presents a new Analysis-by-Synthesis (AbS) technique for joint optimization of the excitation and model parameters based on minimizing the closed loop synthesis error instead of the linear prediction error. By minimizing the synthesis error, the analysis and synthesis stages become more compatible. Using a gradient search in the root domain, model parameters for a given excitation are optimized to minimize the error between the original and the synthesized speech. Since the optimization starts from the LPC solution, the synthesis error is guaranteed to be lower than that obtained using the LPC coefficients. For multipulse LPC, there is a 0.5–1 dB improvement in the segmental SNR for male and female speakers over 4 to 6 second long sentences. Listening tests and objective MOS scores confirm the improved speech quality. By adding an extra optimization step, the technique can be incorporated into the LPC, multi-pulse LPC and CELP-type speech coders.

A pitch synchronous innovation-CELP (PSI-CELP), proposed by NTT DoCoMo in 1993, is adopted as the Japanese half-rate PDC (personal digital cellular) speech codec standard. This algorithm is based on CELP (code excited linear prediction) with a pitch synchronized excitation source. It uses 3.45 kbit/s out of 5.6 kbit/s for speech coding and the remaining 2.15 kbit/s for error protection. The paper summarizes the standardized PSI-CELP algorithm. The techniques used to improve speech quality, to reduce computational complexity, and to reduce memory requirements are mentioned. A real time operating prototype based on this algorithm is also described.>

Robust speech coding and an efficient error correction coding is indispensable for toll-quality voice transmission in mobile radio because of degradation due to multipath fading. A decision procedure for the bit-selective forward-error-correction (BS-FEC) scheme is proposed as a promising solution to this problem, and backward-type prediction speech coding is shown to be more robust against transmission errors than forward-type prediction. Combinations of BS-FEC (channel coding) and backward-type prediction speech coding are designed, and improvements in the SNRs of transmitted voice signals are examined. Simulation results show that BS-FEC can provide good speech quality even at a bit error rate (BER) of 10/sup -2/ in Rayleigh fading environments, at the cost of a slight degradation in the SNR (signal-to-noise ratio) at low BERs.>

A new decision method for band-limited digital FM that employs discriminator detection is presented. This new method adopts a multilevel decision scheme to soften the intersymbol interference effects caused by severe band limitation. It is shown that applying this method significantly improves bit error rate (BER) performance of digital FM using discriminator detection. Theoretical BER evaluation proves the superiority of this method, and establishes the optimum values for fundamental transmitter parameters. Experimental results are provided to support the superiority of this method.

Through laboratory simulation tests and field experiments in the Tokyo metropolitan area, 16 kbit/s Gaussian filtered minimum shift keying (GMSK) transmission performance has been experimentally clarified in the 920 MHz land mobile radio environment. The experimental results agree closely with theory, and they show that fast multipath fading severely degrades average bit error rate (BER) performance in GMSK transmission. However, a space diversity reception technique using a postdetection selection combining scheme is able to efficiently mitigate the fast multipath fading.