Audio Streaming Services Research Articles

‘It's mostly an accompaniment to something’

‘It's mostly an accompaniment to something’

PEER-TO-PEER AUDIO STREAMING USING THE TOR NETWORK: PROBLEMS, OPPORTUNITIES AND DEVELOPMENT PROSPECTS

This article analyses the potential and challenges associated with the implementation of audio streaming in the anonymous Tor network and the use of peer-to-peer (P2P) technologies. The main focus is on discussing the technical and conceptual issues that arise when integrating audio streaming services with the Tor network, including delays, limited bandwidth and other limitations that affect audio quality. The aim of the study is to find practical solutions to the problem of anonymous audio streaming using Tor technologies or other means to preserve anonymity. By analysing different approaches and tools, it becomes apparent that peer-to-peer broadcasting offers a flexible solution that allows audio to be transmitted over P2P networks, reducing the load on individual nodes and optimising bandwidth usage. This solution offers opportunities for scalability, fault tolerance, bandwidth optimisation and latency reduction, making it an attractive option for audio streaming setups. Meanwhile, the Utopia ecosystem, with its decentralised structure and high level of privacy protection, is another potential solution. Not only does it allow bypassing censorship and firewalls, but it also ensures anonymity in communication and data exchange. With its built-in audio and video communication tools, as well as the ability to create private channels and share files, Utopia can become a reliable platform for streaming audio. Thus, based on the analysis, the solution to the problem of anonymous audio streaming may lie in the integration of technologies such as Peercasting and Utopia, which provide not only the technical capability for streaming, but also a high level of privacy protection for users. It is also important to consider the flexibility of these solutions and the ability to adapt to the specifics of anonymous networks and user requirements. The practical significance of the study of audio streaming over Tor using P2P technologies is to develop methods to increase the anonymity and security of audio communications, which is critical in the face of growing threats to privacy on the Internet. This allows to overcome censorship and Internet restrictions in repressive regimes, providing users with freedom of thought and access to information.

AsQM: Audio Streaming Quality Metric Based on Network Impairments and User Preferences

Audio streaming services have many users because the proliferation of cloud-based audio streaming services for different content. The complex networks that support these services do not always guarantee an acceptable quality on the end-user side. In this paper, the impact of temporal interruptions on the audio streaming reproduction and the user’s preference in relation to audio content are studied. In order to determine the key parameters in the audio streaming service, subjective tests are conducted, and their results show that a user’s quality-of-experience (QoE) is highly correlated with the following application parameters: the number of temporal interruptions or stalls, its frequency and length, and the temporal location where they occur. However, most important, the experimental results demonstrate that users’ preference for audio content plays an important role in users’ QoE. Thus, a preference factor (PF) function is defined and considered in the formulation of the proposed metric named audio streaming quality metric (AsQM). Considering that multimedia service providers are based on web servers, a framework to obtain user information is proposed. Furthermore, the results show that the AsQM implemented in the audio player of an end user’s device has a low impact on energy, processing and memory consumption.

Determinants of Continuance Intention to Use and Electronic Satisfaction with Video and Audio Streaming Services in an emerging country

Determinants of Continuance Intention to Use and Electronic Satisfaction with Video and Audio Streaming Services in an emerging country

Does digitization lead to the homogenization of cultural content?

AbstractUsing weekly music charts data in 10 countries over the period 1990–2015, we analyze whether digitization leads to a trend of homogenization of music content or conversely to a greater acoustic disparity. We split the digitization era in four periods that correspond to four new emblematic distribution models (Napster, iTunes, YouTube, Spotify). Our main result is that while acoustic diversity decreased during the iTunes and the YouTube periods, the period that begins with the introduction of audio streaming services, such as Spotify, represents a turning point and is marked by a significant increase in acoustic diversity.

Local Trends in Global Music Streaming

Audio streaming services have made it easier for countries around the world to listen to each other's music. This expansion in listeners' access to global content, however, has raised questions about streaming's impact on the import and export flows of music between countries and their preferences for local or global content. Here, we analyze five and a half years of all streaming data from Spotify, a global music streaming service, and find that preferences for local content have increased from 2014 through 2019, reversing previously noted trends. Perhaps correspondingly, both common official language and geographic proximity between countries increasingly shape listener consumption during this period, particularly for younger audiences. Further, we show that these trends persist across different genres, listener age groups, and early- and late-adopters of streaming, providing new insights into this newest phase in the continued evolution of music and its impact on listeners around the world.

United Kingdom Copyright Decisions 2019

This report highlights the main UK copyright decisions in 2019, which relate to joint authorship of copyright works; the right to communicate a work to the public in the context of an audio streaming service; blocking injunctions in cases involving a breach of the statutory protections against the circumvention of technological protection measures; the subsistence of copyright in unconventional subject matter; and the availability of Norwich Pharmacal orders against internet service providers.

Quality Measurement of Images on Mobile Streaming Interfaces Deployed at Scale.

With the growing use of smart cellular devices for entertainment purposes, audio and video streaming services now offer an increasingly wide variety of popular mobile applications that offer portable and accessible ways to consume content. The user interfaces of these applications have become increasingly visual in nature, and are commonly loaded with dense multimedia content such as thumbnail images, animated GIFs, and short videos. To efficiently render these and to aid rapid download to the client display, it is necessary to compress, scale and color subsample them. These operations introduce distortions, reducing the appeal of the application. It is desirable to be able to automatically monitor and govern the visual qualities of these small images, which are usually small images. However, while there exists a variety of high-performing image quality assessment (IQA) algorithms, none have been designed for this particular use case. This kind of content often has unique characteristics, such as overlaid graphics, intentional brightness, gradients, text, and warping. We describe a study we conducted on the subjective and objective quality of images embedded in the displayed user interfaces of mobile streaming applications. We created a database of typical "billboard" and "thumbnail" images viewed on such services. Using the collected data, we studied the effects of compression, scaling and chroma-subsampling on perceived quality by conducting a subjective study. We also evaluated the performance of leading picture quality prediction models on the new database. We report some surprising results regarding algorithm performance, and find that there remains ample scope for future model development.

Preserving use case flows in source code: Approach, context, and challenges

Being able to see use case steps directly in code would make the intent expressed by the code more comprehensible and easier to maintain. In this paper, an approach to preserving use case flows (of events) in source code called InFlow is presented within a broader context of its application in the software development process along with the issues that remain as challenges. The approach preserves individual steps of use case flows in the form of their counterpart statements, as well as their ordering. This is achieved by mimicking each use case flow by a sequence of method implementations with each step of the use case flow corresponding to one statement in one of these methods, enabled by the annotation named InFlow. A study of implementing an audio streaming service using InFlow, DCI, and aspect-oriented software development with use cases has been performed and assessed using several metrics in order to determine the level of preserving use case flows in source code. Although InFlow introduces a lot of indirect references to code, the overall results speak in favor of InFlow. DCI adds more complexity to following a use case flow in code than other two approaches. Complexity of making a change to a use case flow ended up with closely matching results. But InFlow guarantees traceability of use case implementation from the domain model implementation.

Relative velocity based vehicle-to-vehicle routing protocol over ad-hoc networks

In this paper, we introduce a new routing protocol supporting high mobility over VANET which is using hop counts and relative velocity between vehicles to find the best routing transmission path. By exchanging vehicles' relative velocities, the more stable and reliable paths are searched as compared to traditional MANET protocols using only hop count information at the time of routing path set-up. To evaluate proposed routing protocol, we compared it with original Ad hoc On-demand Multipath Distance Vector (AOMDV) on NS2. The simulation results showed that proposed routing protocol achieves better performance from about 10% to 50% in terms of normalized routing overhead and packet delivery ratio, when video sequences are transferred on channels. The proposed routing protocol can be useful for the Vehicle-to-Vehicle (V2V) transmission of traffic monitoring data, audio, and video streaming services without the additional global position devices.

Wideband Wireless Access Systems Interference Robustness: Its Effect on Quality of Video Streaming

A necessary requirement incumbent on any information communication system and/or network is the capacity to transmit information with a predefined degree of accuracy in the presence of inevitable interference. The transmission of audio and video streaming services over different conduits (wireless access systems, Internet, etc.) is becoming ever more popular. As should be expected, this widespread increase is accompanied by the attendant new and difficult task of maintaining the quality of service of streaming video. The use of very accurate coding techniques for transmissions over wireless networks alone cannot guarantee a complete eradication of distortions characteristic of the video signal. A software-hardware composite system has been developed for investigating the effect of single bit error and bit packet errors in wideband wireless access systems on the quality of H.264/AVC standard bursty video streams. Numerical results of the modeling and analysis of the effect of interference robustness on quality of video streaming are presented and discussed.

A Context-Aware Network Equipment for Dynamic Adaptation of Multimedia Services in Wireless Networks

B3G (Beyond 3rd Generation) networks is the next step in the wireless world. It will enable the delivery of services to end-users, on different underlying wireless access networks and customised to users' context. Furthermore, seamless mobility between networks and switch of devices will be handled in a transparent way. In this paper, a Context-Aware Network Equipment (CANE) is introduced, that is able to dynamically adapt multimedia services according to the users' context. The term context includes the users' device, the users' preferences, the network conditions as well as the service provider adaptation policies. As a proof of concept, an implementation of a CANE is given for a 802.11 network use-case. An evaluation of the prototype as well as simulations of the proposed solution, both for an audio streaming service and a video streaming service, are presented. The results show that adapting multimedia services in case of network overload can enable to maintain an acceptable quality of service delivered to end-users and even to allow more users to enjoy the services, which will not be possible without the CANE.

Deployment of IP multimedia streaming services in third-generation mobile networks

In this article, an end-to-end quality of service framework for streaming services in 3G mobile networks is considered. Under this scenario, the interaction between UMTS and IETF's protocols and mechanisms for a streaming session is analyzed. By signaling flowcharts, it is shown that both groups of protocols and mechanisms can co-operate to provide seamless end-to-end real-time services. Specifically, the article proposes to make the IP multimedia subsystem aware of the real time streaming protocol, in order to extend its control from SIP to RTSP-based services, such as multimedia streaming services. Supported by this proposed framework, provisioning of audio streaming services over 3G mobile networks is also outlined.

A real-time IMT-2000 audio transmission system

For real-time audio streaming services over next-generation mobile phone systems, such as live concert broadcasting and radio relaying, we have developed a realtime IMT-2000 audio transmission system. This system consists of a real-time encoder and a portable real-time decoder, which are both based on fixed-point DSP chips. TWINVQ technology is used for audio compression, and a scheme to protect audio data against bit-errors was designed based on error sensitivity observations. Self-synchronization and adaptive error concealment are carried out at the decoder. We confirmed that the system can transmit high-quality stereo audio samples at 32 kHz with a total bit rate of 64 kbit/s over a simulated IMT-2000 channel. We also checked that this system can connect to a real IMT-2000 experimental network, and found that it is easily applicable to various audio streaming services.