An investigation into the application of Distributed Endpoint Processing to 3D Immersive Audio Rendering
- Authors: Devonport, Robin Sean
- Date: 2020
- Subjects: Uncatalogued
- Language: English
- Type: thesis , text , Masters , MSc
- Identifier: http://hdl.handle.net/10962/163258 , vital:41022
- Description: Thesis (MSc)--Rhodes University, Faculty of Science, Computer Science, 2020.
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An investigation into the use of intuitive control interfaces and distributed processing for enhanced three dimensional sound localization
- Authors: Hedges, Mitchell Lawrence
- Date: 2016
- Subjects: Human-computer interaction , Acoustic localization , Sound -- Equipment and supplies , Acoustical engineering , Surround-sound systems , Wireless sensor nodes
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4724 , http://hdl.handle.net/10962/d1020615
- Description: This thesis investigates the feasibility of using gestures as a means of control for localizing three dimensional (3D) sound sources in a distributed immersive audio system. A prototype system was implemented and tested which uses state of the art technology to achieve the stated goals. A Windows Kinect is used for gesture recognition which translates human gestures into control messages by the prototype system, which in turn performs actions based on the recognized gestures. The term distributed in the context of this system refers to the audio processing capacity. The prototype system partitions and allocates the processing load between a number of endpoints. The reallocated processing load consists of the mixing of audio samples according to a specification. The endpoints used in this research are XMOS AVB endpoints. The firmware on these endpoints were modified to include the audio mixing capability which was controlled by a state of the art audio distribution networking standard, Ethernet AVB. The hardware used for the implementation of the prototype system is relatively cost efficient in comparison to professional audio hardware, and is also commercially available for end users. the successful implementation and results from user testing of the prototype system demonstrates how it is a feasible option for recording the localization of a sound source. The ability to partition the processing provides a modular approach to building immersive sound systems. This removes the constraint of a centralized mixing console with a predetermined speaker configuration.
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The development of a discovery and control environment for networked audio devices based on a study of current audio control protocols
- Authors: Eales, Andrew Arnold
- Date: 2016
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: http://hdl.handle.net/10962/539 , vital:19968
- Description: This dissertation develops a standard device model for networked audio devices and introduces a novel discovery and control environment that uses the developed device model. The proposed standard device model is derived from a study of current audio control protocols. Both the functional capabilities and design principles of audio control protocols are investigated with an emphasis on Open Sound Control, SNMP and IEC-62379, AES64, CopperLan and UPnP. An abstract model of networked audio devices is developed, and the model is implemented in each of the previously mentioned control protocols. This model is also used within a novel discovery and control environment designed around a distributed associative memory termed an object space. This environment challenges the accepted notions of the functionality provided by a control protocol. The study concludes by comparing the salient features of the different control protocols encountered in this study. Different approaches to control protocol design are considered, and several design heuristics for control protocols are proposed.
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Network simulation for professional audio networks
- Authors: Otten, Fred
- Date: 2015
- Subjects: Sound engineers , Ethernet (Local area network system) , Computer networks , Computer simulation
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4713 , http://hdl.handle.net/10962/d1017935
- Description: Audio Engineers are required to design and deploy large multi-channel sound systems which meet a set of requirements and use networking technologies such as Firewire and Ethernet AVB. Bandwidth utilisation and parameter groupings are among the factors which need to be considered in these designs. An implementation of an extensible, generic simulation framework would allow audio engineers to easily compare protocols and networking technologies and get near real time responses with regards to bandwidth utilisation. Our hypothesis is that an application-level capability can be developed which uses a network simulation framework to enable this process and enhances the audio engineer’s experience of designing and configuring a network. This thesis presents a new, extensible simulation framework which can be utilised to simulate professional audio networks. This framework is utilised to develop an application - AudioNetSim - based on the requirements of an audio engineer. The thesis describes the AudioNetSim models and implementations for Ethernet AVB, Firewire and the AES- 64 control protocol. AudioNetSim enables bandwidth usage determination for any network configuration and connection scenario and is used to compare Firewire and Ethernet AVB bandwidth utilisation. It also applies graph theory to the circular join problem and provides a solution to detect circular joins.
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An investigation of parameter relationships in a high-speed digital multimedia environment
- Authors: Chigwamba, Nyasha
- Date: 2014
- Subjects: Multimedia communications , Digital communications , Local area networks (Computer networks) , Computer network architectures , Computer network protocols , Computer sound processing , Sound -- Recording and reproducing -- Digital techniques
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4725 , http://hdl.handle.net/10962/d1021153
- Description: With the rapid adoption of multimedia network technologies, a number of companies and standards bodies are introducing technologies that enhance user experience in networked multimedia environments. These technologies focus on device discovery, connection management, control, and monitoring. This study focused on control and monitoring. Multimedia networks make it possible for devices that are part of the same network to reside in different physical locations. These devices contain parameters that are used to control particular features, such as speaker volume, bass, amplifier gain, and video resolution. It is often necessary for changes in one parameter to affect other parameters, such as a synchronised change between volume and bass parameters, or collective control of multiple parameters. Thus, relationships are required between the parameters. In addition, some devices contain parameters, such as voltage, temperature, and audio level, that require constant monitoring to enable corrective action when thresholds are exceeded. Therefore, a mechanism for monitoring networked devices is required. This thesis proposes relationships that are essential for the proper functioning of a multimedia network and that should, therefore, be incorporated in standard form into a protocol, such that all devices can depend on them. Implementation mechanisms for these relationships were created. Parameter grouping and monitoring capabilities within mixing console implementations and existing control protocols were reviewed. A number of requirements for parameter grouping and monitoring were derived from this review. These requirements include a formal classification of relationship types, the ability to create relationships between parameters with different underlying value units, the ability to create relationships between parameters residing on different devices on a network, and the use of an event-driven mechanism for parameter monitoring. These requirements were the criteria used to govern the implementation mechanisms that were created as part of this study. Parameter grouping and monitoring mechanisms were implemented for the XFN protocol. The mechanisms implemented fulfil the requirements derived from the review of capabilities of mixing consoles and existing control protocols. The formal classification of relationship types was implemented within XFN parameters using lists that keep track of the relationships between each XFN parameter and other XFN parameters that reside on the same device or on other devices on the network. A common value unit, known as the global unit, was defined for use as the value format within value update messages between XFN parameters that have relationships. Mapping tables were used to translate the global unit values to application-specific (universal) units, such as decibels (dB). A mechanism for bulk parameter retrieval within the XFN protocol was augmented to produce an event-driven mechanism for parameter monitoring. These implementation mechanisms were applied to an XFN-protocol-compliant graphical control application to demonstrate their usage within an end user context. At the time of this study, the XFN protocol was undergoing standardisation within the Audio Engineering Society. The AES-64 standard has now been approved. Most of the implementation mechanisms resulting from this study have been incorporated into this standard.
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An investigation of the XMOS XSl architecture as a platform for development of audio control standards
- Authors: Dibley, James
- Date: 2014
- Subjects: Microcontrollers -- Research , Streaming audio -- Standards -- Research , Computer sound processing -- Research , Computer network protocols -- Standards -- Research , Communication -- Technological innovations -- Research
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4694 , http://hdl.handle.net/10962/d1011789 , Microcontrollers -- Research , Streaming audio -- Standards -- Research , Computer sound processing -- Research , Computer network protocols -- Standards -- Research , Communication -- Technological innovations -- Research
- Description: This thesis investigates the feasiblity of using a new microcontroller architecture, the XMOS XS1, in the research and development of control standards for audio distribution networks. This investigation is conducted in the context of an emerging audio distribution network standard, Ethernet Audio/Video Bridging (`Ethernet AVB'), and an emerging audio control standard, AES-64. The thesis describes these emerging standards, the XMOS XS1 architecture (including its associated programming language, XC), and the open-source implementation of an Ethernet AVB streaming audio device based on the XMOS XS1 architecture. It is shown how the XMOS XS1 architecture and its associated features, focusing on the XC language's mechanisms for concurrency, event-driven programming, and integration of C software modules, enable a powerful implementation of the AES-64 control standard. Feasibility is demonstrated by the implementation of an AES-64 protocol stack and its integration into an XMOS XS1-based Ethernet AVB streaming audio device, providing control of Ethernet AVB features and audio hardware, as well as implementations of advanced AES-64 control mechanisms. It is demonstrated that the XMOS XS1 architecture is a compelling platform for the development of audio control standards, and has enabled the implementation of AES-64 connection management and control over standards-compliant Ethernet AVB streaming audio devices where no such implementation previously existed. The research additionally describes a linear design method for applications based on the XMOS XS1 architecture, and provides a baseline implementation reference for the AES-64 control standard where none previously existed.
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An investigation into the application of the IEEE 1394 high performance serial bus to sound installation contro
- Authors: Klinkradt, Bradley Hugh
- Date: 2003 , 2013-05-24
- Subjects: Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4612 , http://hdl.handle.net/10962/d1004899 , Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Description: This thesis investigates the feasibility of using existing IP-based control and monitoring protocols within professional audio installations utilising IEEE 1394 technology. Current control and monitoring technologies are examined, and the characteristics common to all are extracted and compiled into an object model. This model forms the foundation for a set of evaluation criteria against which current and future control and monitoring protocols may be measured. Protocols considered include AV/C, MIDI, QSC-24, and those utilised within the UPnP architecture. As QSC-24 and the UPnP architecture are IP-based, the facilities required to transport IP datagrams over the IEEE 1394 bus are investigated and implemented. Example QSC-24 and UPnP architecture implementations are described, which permit the control and monitoring of audio devices over the IEEE 1394 network using these IP-based technologies. The way forward for the control and monitoring of professional audio devices within installations is considered, and recommendations are provided. , KMBT_363 , Adobe Acrobat 9.54 Paper Capture Plug-in
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