Abstract
This paper addresses the problem of automatic identification of vowels uttered in isolation by female and child speakers. In this case, the magnitude spectrum of voiced vowels is sparsely sampled since only frequencies at integer multiples of F0 are significant. This impacts negatively on the performance of vowel identification techniques that either ignore pitch or rely on global shape models. A new pitch-dependent approach to vowel identification is proposed that emerges from the concept of timbre and that defines perceptual spectral clusters (PSC) of harmonic partials. A representative set of static PSC-related features are estimated and their performance is evaluated in automatic classification tests using the Mahalanobis distance. Linear prediction features and Mel-frequency cepstral coefficients (MFCC) coefficients are used as a reference and a database of five (Portuguese) natural vowel sounds uttered by 44 speakers (including 27 child speakers) is used for training and testing the Gaussian models. Results indicate that perceptual spectral cluster (PSC) features perform better than plain linear prediction features, but perform slightly worse than MFCC features. However, PSC features have the potential to take full advantage of the pitch structure of voiced vowels, namely in the analysis of concurrent voices, or by using pitch as a normalization parameter. (C) 2007 Acoustical Society of America.

Abstract
This paper introduces new ideas on wideband stationary/non-stationary noise removal for audio signals. Current noise reduction techniques have generally proven to be effective, yet these typically exhibit certain undesirable characteristics. Distortion and/or alteration of the audio characteristics of primary audio sound is a common problem. Also user intervention in identifying the noise profile is sometimes necessary. The proposed technique is centered on the classical Kalman filtering technique for noise removal but uses a novel architecture whereby advanced signal processing techniques are used to identify and preserve the richness of the audio spectrum. The paper also includes conceptual and derivative results on parameter estimation, a description of multi parameter Signal Activity Detector (SAD) and our new found improved results.

Abstract
Consumer audio applications such as satellite radio broadcasts, multi-channel audio streaming and playback systems coupled with the need to meet stringent bandwidth requirements are eliciting newer challenges in parametric multichannel audio coding schemes. This paper describes the continuation of our research concerning the Immersive Soundfield Rendition (ISR) system. In particular we present detailed subjective result data benchmarking the ISR system in comparison to MPEG Surround and also characterizing the audio quality level at different sub-modes of the system. We also describe enhancements to various algorithmic components in particular the blind 2-to-5 channel upmixing algorithm and describe a novel scheme for providing enhanced stereo downmix at the receiver for improved decoding by conventional matrix decoding systems.

Abstract
Low bit rate audio coding often results in the loss of a number of key audio attributes such as audio bandwidth and stereo separation. Additionally, there is also typically a loss in the level of details and intelligibility and/or warmth in the signal. Due to the proliferation, e.g. on Internet, of low bit rate audio coded using a variety of coding schemes and bit rates over which the listener has no control, it is becoming increasingly attractive to incorporate processing tools in the player which can ensure a consistent listener experience. We describe a novel post-processing toolkit which incorporates tools for (i) Stereo Enhancement, (ii) Blind Bandwidth Extension, (iii) Automatic Noise Removal and Audio Enhancement, and, (iv) Blind 2-to-5 channel upmixing. Algorithmic details, listening results, and audio demonstrations are presented.

Abstract
XCP-b proposes a modification to the XCP router algorithm that computes the spare bandwidth. The modification removes the need for an XCP router to know the exact capacity of the channel, making it possible to use the XCP-b variant in transmission media where the capacity is hard to measure. An example of this kind of medium is the IEEE 802.11. Previous work shows that XCP-b behaves well in single-hop wireless networks and that it Outperforms TCP in terms of fairness, queuing delay, stability and efficiency when the bandwidth delay product of the network grows. In this paper we extend the validation and evaluation of XCP-b to the case of multi-hop wireless networks, both stand-alone and as access networks to other wired networks. The results show that XCP-b maintains its fundamental characteristics in wireless multi-hop scenarios, such as stable throughput and low standing queues, while distributing the bandwidth fairly and using the available capacity efficiently. The simulations also show that XCP-b produces congestion window values that are closer than TCP to the theoretical upper-bound which maximizes spatial reuse.

Abstract
Emerging access networks will use heterogeneous wireless technologies such as 802.11, 802.16 or UMTS, to offer users the best access to the Internet. Layer 2 access networks will consist of wireless bridges (access points) that isolate, concatenated, or in mesh provide access to mobile nodes. The transport of real time traffic over these networks may demand new QoS signalling, used to reserve resources. Besides the reservation, the new signalling needs to address the dynamics of the wireless links, the mobility of the terminals, and the multicast traffic. In this paper a new protocol is proposed aimed at solving this problem-the QoS Abstraction Layer (QoSAL). Existing only at the control plane, the QoSAL is located above the layer 2 and hides from layer 3 the details of each technology with respect to the QoS and to the network topology. The QoSAL has been designed, simulated, and tested. The results obtained demonstrate its usefulness in 4G networks.