Perbandingan Kualitas Audio Aplikasi Telekonferensi Menggunakan Metode Subjective Test ITU-R BS.1116-3

Kebutuhan manusia untuk saling berkomunikasi dan saling terhubung disaat pandemi COVID-19 menyebabkan penggunaan aplikasi telekonferensi meningkat. Ada banyak aplikasi telekonferensi yang tersedia dan pengguna perlu memperhatikan kualitas suara saat memilih aplikasi telekonferensi. Kualitas suara dipengaruhi oleh perbedaan codec yang digunakan dan dikembangkan oleh aplikasi. Tugas akhir ini membahas tentang perbandingan kualitas audio antar aplikasi telekonfensi dengan mengevaluasi codec yang digunakan secara Subjective Test ITU-R BS.1116-3. Penilaian secara subjective dipilih karena merupakan penilaian kualitas audio secara perceptual dimana estimasi output dari komponen sinyal yang dipersepsikan memperhitungkan kompleksitas proses pendengaran dan hasil akhir pengukuran dapat dikatakan sebagai representasi dari pendengaran manusia. Standar subjective test ITU-R BS.1116-3: Methods for The Subjective Assessment of Small Impairments In Audio Systems digunakan untuk mengetahui perbedaan kecil antar kualitas audio. Penilaian dilakukan dengan membandingkan audio asli dan audio terkompresi dari 5 jenis aplikasi telekonferensi (Zoom ,Skype, Microsoft Teams, Cisco Webex dan RingCentral). Sampel audio asli direkam dari sisi pembicara dan sampel audio terkompresi direkam dari sisi penerima. Kedua sampel dinilai dengan metode subjective test oleh 20 orang subjek secara offline. Hasil penilaian dianalisa untuk mendapatkan nilai SDG setiap aplikasi. Dari penilaian Subjective test, didapati bahwa audio RingCentral memiliki nilai rata-rata yang lebih tinggi dibandingkan audio aplikasi lain dengan rentang nilai dari -0,97 sampai -1,67 dan berada di kategori perceptible but not annoying dan slightly annoying. Sedangkan Audio Aplikasi Zoom memperoleh nilai rata-rata yang paling rendah dengan rentang nilai -1,71 sampai -2,32 dan berada di kategori slightly annoying

The contrast effect: QoE of mixed video-qualities at the same time

In desktop multi-party video-conferencing videostreams of participants are delivered in different qualities, but we know little about how such composition of the screen affects the quality of experience. Do the different videostreams serve as indirect quality references and the perceived video quality is thus dependent on other streams in the same session? How is the relation between the perceived qualities of each stream and the perceived quality of the overall session? To answer these questions we conducted a crowdsourcing study, in which we gathered over 5000 perceived quality ratings of overall sessions and individual streams. Our results show a contrast effect: high quality streams are rated better when more low quality streams are co-present, and vice versa. In turn, the quality p

Multimedia

The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications

A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths