1,766 research outputs found
Software Defined Media: Virtualization of Audio-Visual Services
Internet-native audio-visual services are witnessing rapid development. Among
these services, object-based audio-visual services are gaining importance. In
2014, we established the Software Defined Media (SDM) consortium to target new
research areas and markets involving object-based digital media and
Internet-by-design audio-visual environments. In this paper, we introduce the
SDM architecture that virtualizes networked audio-visual services along with
the development of smart buildings and smart cities using Internet of Things
(IoT) devices and smart building facilities. Moreover, we design the SDM
architecture as a layered architecture to promote the development of innovative
applications on the basis of rapid advancements in software-defined networking
(SDN). Then, we implement a prototype system based on the architecture, present
the system at an exhibition, and provide it as an SDM API to application
developers at hackathons. Various types of applications are developed using the
API at these events. An evaluation of SDM API access shows that the prototype
SDM platform effectively provides 3D audio reproducibility and interactiveness
for SDM applications.Comment: IEEE International Conference on Communications (ICC2017), Paris,
France, 21-25 May 201
Early adductive reasoning for blind signal separation
We demonstrate that explicit and systematic incorporation of abductive reasoning capabilities into algorithms for blind signal separation can yield significant performance improvements. Our formulated mechanisms apply to the output data of signal processing modules in order to conjecture the structure of time-frequency interactions between the signal components that are to be separated. The conjectured interactions are used to drive subsequent signal separation processes that are as a result less blind to the interacting signal components and, therefore, more effective. We refer to this type of process as early abductive reasoning (EAR); the “early” refers to the fact that in contrast to classical Artificial Intelligence paradigms, the reasoning process here is utilized before the signal processing transformations are completed.
We have used our EAR approach to formulate a practical algorithm that is more effective in realistically noisy conditions than reference algorithms that are representative of the current state of the art in two-speaker pitch tracking. Our algorithm uses the Blackboard architecture from Artificial Intelligence to control EAR and advanced signal processing modules. The algorithm has been implemented in MATLAB and successfully tested on a database of 570 mixture signals representing simultaneous speakers in a variety of real-world, noisy environments. With 0 dB Target-to-Masking Ratio (TMR) and no noise, the Gross Error Rate (GER) for our algorithm is 5% in comparison to the best GER performance of 11% among the reference algorithms. In diffuse noisy environments (such as street or restaurant environments), we find that our algorithm on the average outperforms the best reference algorithm by 9.4%. With directional noise, our algorithm also outperforms the best reference algorithm by 29%. The extracted pitch tracks from our algorithm were also used to carry out comb filtering for separating the harmonics of the two speakers from each other and from the other sound sources in the environment. The separated signals were evaluated subjectively by a set of 20 listeners to be of reasonable quality
Studies on noise robust automatic speech recognition
Noise in everyday acoustic environments such as cars, traffic environments, and cafeterias remains one of the main challenges in automatic speech recognition (ASR). As a research theme, it has received wide attention in conferences and scientific journals focused on speech technology. This article collection reviews both the classic and novel approaches suggested for noise robust ASR. The articles are literature reviews written for the spring 2009 seminar course on noise robust automatic speech recognition (course code T-61.6060) held at TKK
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