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.
- Full Text:
- Date Issued: 2014
- 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.
- Full Text:
- Date Issued: 2014
A proxy approach to protocol interoperability within digital audio networks
- Authors: Igumbor, Osedum Peter
- Date: 2010
- Subjects: Digital communications , Local area networks (Computer networks) , Computer sound processing , Computer networks , Computer network protocols
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4601 , http://hdl.handle.net/10962/d1004852 , Digital communications , Local area networks (Computer networks) , Computer sound processing , Computer networks , Computer network protocols
- Description: Digital audio networks are becoming the preferred solution for the interconnection of professional audio devices. Prominent amongst their advantages are: reduced noise interference, signal multiplexing, and a reduction in the number of cables connecting networked devices. In the context of professional audio, digital networks have been used to connect devices including: mixers, effects units, preamplifiers, breakout boxes, computers, monitoring controllers, and synthesizers. Such networks are governed by protocols that define the connection management rocedures, and device synchronization processes of devices that conform to the protocols. A wide range of digital audio network control protocols exist, each defining specific hardware requirements of devices that conform to them. Device parameter control is achieved by sending a protocol message that indicates the target parameter, and the action that should be performed on the parameter. Typically, a device will conform to only one protocol. By implication, only devices that conform to a specific protocol can communicate with each other, and only a controller that conforms to the protocol can control such devices. This results in the isolation of devices that conform to disparate protocols, since devices of different protocols cannot communicate with each other. This is currently a challenge in the professional music industry, particularly where digital networks are used for audio device control. This investigation seeks to resolve the issue of interoperability between professional audio devices that conform to different digital audio network protocols. This thesis proposes the use of a proxy that allows for the translation of protocol messages, as a solution to the interoperability problem. The proxy abstracts devices of one protocol in terms of another, hence allowing all the networked devices to appear as conforming to the same protocol. The proxy receives messages on behalf of the abstracted device, and then fulfills them in accordance with the protocol that the abstracted device conforms to. Any number of protocol devices can be abstracted within such a proxy. This has the added advantage of allowing a common controller to control devices that conform to the different protocols.
- Full Text:
- Date Issued: 2010
- Authors: Igumbor, Osedum Peter
- Date: 2010
- Subjects: Digital communications , Local area networks (Computer networks) , Computer sound processing , Computer networks , Computer network protocols
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4601 , http://hdl.handle.net/10962/d1004852 , Digital communications , Local area networks (Computer networks) , Computer sound processing , Computer networks , Computer network protocols
- Description: Digital audio networks are becoming the preferred solution for the interconnection of professional audio devices. Prominent amongst their advantages are: reduced noise interference, signal multiplexing, and a reduction in the number of cables connecting networked devices. In the context of professional audio, digital networks have been used to connect devices including: mixers, effects units, preamplifiers, breakout boxes, computers, monitoring controllers, and synthesizers. Such networks are governed by protocols that define the connection management rocedures, and device synchronization processes of devices that conform to the protocols. A wide range of digital audio network control protocols exist, each defining specific hardware requirements of devices that conform to them. Device parameter control is achieved by sending a protocol message that indicates the target parameter, and the action that should be performed on the parameter. Typically, a device will conform to only one protocol. By implication, only devices that conform to a specific protocol can communicate with each other, and only a controller that conforms to the protocol can control such devices. This results in the isolation of devices that conform to disparate protocols, since devices of different protocols cannot communicate with each other. This is currently a challenge in the professional music industry, particularly where digital networks are used for audio device control. This investigation seeks to resolve the issue of interoperability between professional audio devices that conform to different digital audio network protocols. This thesis proposes the use of a proxy that allows for the translation of protocol messages, as a solution to the interoperability problem. The proxy abstracts devices of one protocol in terms of another, hence allowing all the networked devices to appear as conforming to the same protocol. The proxy receives messages on behalf of the abstracted device, and then fulfills them in accordance with the protocol that the abstracted device conforms to. Any number of protocol devices can be abstracted within such a proxy. This has the added advantage of allowing a common controller to control devices that conform to the different protocols.
- Full Text:
- Date Issued: 2010
A grid based approach for the control and recall of the properties of IEEE 1394 audio devices
- Authors: Foulkes, Philip James
- Date: 2009
- Subjects: IEEE 1394 (Standard) , Computer sound processing , Digital communications , Local area networks (Computer networks) , Sound -- Recording and reproducing -- Digital techniques , Computational grids (Computer systems)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4594 , http://hdl.handle.net/10962/d1004836 , IEEE 1394 (Standard) , Computer sound processing , Digital communications , Local area networks (Computer networks) , Sound -- Recording and reproducing -- Digital techniques , Computational grids (Computer systems)
- Description: The control of modern audio studios is complex. Audio mixing desks have grown to the point where they contain thousands of parameters. The control surfaces of these devices do not reflect the routing and signal processing capabilities that the devices are capable of. Software audio mixing desk editors have been developed that allow for the remote control of these devices, but their graphical user interfaces retain the complexities of the audio mixing desk that they represent. In this thesis, we propose a grid approach to audio mixing. The developed grid audio mixing desk editor represents an audio mixing desk as a series of graphical routing matrices. These routing matrices expose the various signal processing points and signal flows that exist within an audio mixing desk. The routing matrices allow for audio signals to be routed within the device, and allow for the device’s parameters to be adjusted by selecting the appropriate signal processing points. With the use of the programming interfaces that are defined as part of the Studio Connections – Total Recall SDK, the audio mixing desk editor was integrated with compatible DAW applications to provide persistence of audio mixing desk parameter states. Many audio studios currently use digital networks to connect audio devices together. Audio and control signals are patched between devices through the use of software patchbays that run on computers. We propose a double grid-based FireWire patchbay aimed to simplify the patching of signals between audio devices on a FireWire network. The FireWire patchbay was implemented in such a way such that it can host software device editors that are Studio Connections compatible. This has allowed software device editors to be associated with the devices that are represented on the FireWire patchbay, thus allowing for studio wide control from a single application. The double grid-based patchbay was implemented such that it can be hosted by compatible DAW applications. Through this, the double grid-based patchbay application is able to provide the DAW application with the state of the parameters of the devices in a studio, as well as the connections between them. The DAW application may save this state data to its native song files. This state data may be passed back to the double grid-based patchbay when the song file is reloaded at a later stage. This state data may then be used by the patchbay to restore the parameters of the patchbay and its device editors to a previous state. This restored state may then be transferred to the hardware devices being represented by the patchbay.
- Full Text:
- Date Issued: 2009
- Authors: Foulkes, Philip James
- Date: 2009
- Subjects: IEEE 1394 (Standard) , Computer sound processing , Digital communications , Local area networks (Computer networks) , Sound -- Recording and reproducing -- Digital techniques , Computational grids (Computer systems)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4594 , http://hdl.handle.net/10962/d1004836 , IEEE 1394 (Standard) , Computer sound processing , Digital communications , Local area networks (Computer networks) , Sound -- Recording and reproducing -- Digital techniques , Computational grids (Computer systems)
- Description: The control of modern audio studios is complex. Audio mixing desks have grown to the point where they contain thousands of parameters. The control surfaces of these devices do not reflect the routing and signal processing capabilities that the devices are capable of. Software audio mixing desk editors have been developed that allow for the remote control of these devices, but their graphical user interfaces retain the complexities of the audio mixing desk that they represent. In this thesis, we propose a grid approach to audio mixing. The developed grid audio mixing desk editor represents an audio mixing desk as a series of graphical routing matrices. These routing matrices expose the various signal processing points and signal flows that exist within an audio mixing desk. The routing matrices allow for audio signals to be routed within the device, and allow for the device’s parameters to be adjusted by selecting the appropriate signal processing points. With the use of the programming interfaces that are defined as part of the Studio Connections – Total Recall SDK, the audio mixing desk editor was integrated with compatible DAW applications to provide persistence of audio mixing desk parameter states. Many audio studios currently use digital networks to connect audio devices together. Audio and control signals are patched between devices through the use of software patchbays that run on computers. We propose a double grid-based FireWire patchbay aimed to simplify the patching of signals between audio devices on a FireWire network. The FireWire patchbay was implemented in such a way such that it can host software device editors that are Studio Connections compatible. This has allowed software device editors to be associated with the devices that are represented on the FireWire patchbay, thus allowing for studio wide control from a single application. The double grid-based patchbay was implemented such that it can be hosted by compatible DAW applications. Through this, the double grid-based patchbay application is able to provide the DAW application with the state of the parameters of the devices in a studio, as well as the connections between them. The DAW application may save this state data to its native song files. This state data may be passed back to the double grid-based patchbay when the song file is reloaded at a later stage. This state data may then be used by the patchbay to restore the parameters of the patchbay and its device editors to a previous state. This restored state may then be transferred to the hardware devices being represented by the patchbay.
- Full Text:
- Date Issued: 2009
An investigation into the hardware abstraction layer of the plural node architecture for IEEE 1394 audio devices
- Authors: Chigwamba, Nyasha
- Date: 2009
- Subjects: IEEE 1394 (Standard) , Digital communications , Computer sound processing , Local area networks (Computer networks) , Computer network architectures , Sound -- Recording and reproducing -- Digital techniques
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4598 , http://hdl.handle.net/10962/d1004841 , IEEE 1394 (Standard) , Digital communications , Computer sound processing , Local area networks (Computer networks) , Computer network architectures , Sound -- Recording and reproducing -- Digital techniques
- Description: Digital audio network technologies are becoming more prevalent in audio related environments. Yamaha Corporation has created a digital audio network solution, named mLAN (music Local Area Network), that uses IEEE 1394 as its underlying network technology. IEEE 1394 is a digital network technology that is specifically designed for real-time multimedia data transmission. The second generation of mLAN is based on the Plural Node Architecture, where the control of audio and MIDI routings between IEEE 1394 devices is split between two node types, namely an Enabler and a Transporter. The Transporter typically resides in an IEEE 1394 device and is solely responsible for transmission and reception of audio or MIDI data. The Enabler typically resides in a workstation and exposes an abstract representation of audio or MIDI plugs on each Transporter to routing control applications. The Enabler is responsible for configuring audio and MIDI routings between plugs on different Transporters. A Hardware Abstraction Layer (HAL) within the Enabler allows it to uniformly communicate with Transporters that are created by various vendors. A plug-in mechanism is used to provide this capability. When vendors create Transporters, they also create device-specific plug-ins for the Enabler. These plug-ins are created against a Transporter HAL Application Programming Interface (API) that defines methods to access the capabilities of Transporters. An Open Generic Transporter (OGT) guideline document which models all the capabilities of Transporters has been produced. These guidelines make it possible for manufacturers to create Transporters that make use of a common plug-in, although based on different hardware architectures. The introduction of the OGT concept has revealed additional Transporter capabilities that are not incorporated in the existing Transporter HAL API. This has led to the underutilisation of OGT capabilities. The main goals of this investigation have been to improve the Enabler’s plug-in mechanism, and to incorporate the additional capabilities that have been revealed by the OGT into the Transporter HAL API. We propose a new plug-in mechanism, and a new Transporter HAL API that fully utilises both the additional capabilities revealed by the OGT and the capabilities of existing Transporters.
- Full Text:
- Date Issued: 2009
- Authors: Chigwamba, Nyasha
- Date: 2009
- Subjects: IEEE 1394 (Standard) , Digital communications , Computer sound processing , Local area networks (Computer networks) , Computer network architectures , Sound -- Recording and reproducing -- Digital techniques
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4598 , http://hdl.handle.net/10962/d1004841 , IEEE 1394 (Standard) , Digital communications , Computer sound processing , Local area networks (Computer networks) , Computer network architectures , Sound -- Recording and reproducing -- Digital techniques
- Description: Digital audio network technologies are becoming more prevalent in audio related environments. Yamaha Corporation has created a digital audio network solution, named mLAN (music Local Area Network), that uses IEEE 1394 as its underlying network technology. IEEE 1394 is a digital network technology that is specifically designed for real-time multimedia data transmission. The second generation of mLAN is based on the Plural Node Architecture, where the control of audio and MIDI routings between IEEE 1394 devices is split between two node types, namely an Enabler and a Transporter. The Transporter typically resides in an IEEE 1394 device and is solely responsible for transmission and reception of audio or MIDI data. The Enabler typically resides in a workstation and exposes an abstract representation of audio or MIDI plugs on each Transporter to routing control applications. The Enabler is responsible for configuring audio and MIDI routings between plugs on different Transporters. A Hardware Abstraction Layer (HAL) within the Enabler allows it to uniformly communicate with Transporters that are created by various vendors. A plug-in mechanism is used to provide this capability. When vendors create Transporters, they also create device-specific plug-ins for the Enabler. These plug-ins are created against a Transporter HAL Application Programming Interface (API) that defines methods to access the capabilities of Transporters. An Open Generic Transporter (OGT) guideline document which models all the capabilities of Transporters has been produced. These guidelines make it possible for manufacturers to create Transporters that make use of a common plug-in, although based on different hardware architectures. The introduction of the OGT concept has revealed additional Transporter capabilities that are not incorporated in the existing Transporter HAL API. This has led to the underutilisation of OGT capabilities. The main goals of this investigation have been to improve the Enabler’s plug-in mechanism, and to incorporate the additional capabilities that have been revealed by the OGT into the Transporter HAL API. We propose a new plug-in mechanism, and a new Transporter HAL API that fully utilises both the additional capabilities revealed by the OGT and the capabilities of existing Transporters.
- Full Text:
- Date Issued: 2009
Connection management applications for high-speed audio networking
- Authors: Sibanda, Phathisile
- Date: 2008 , 2008-03-12
- Subjects: Flash (Computer file) , Computer networks , Computer networks -- Management , Digital communications , Computer sound processing , Sound -- Recording and reproducing -- Digital techniques , Broadcast data systems , C# (Computer program language) , C++ (Computer program language) , ActionScript (Computer program language)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4634 , http://hdl.handle.net/10962/d1006532 , Flash (Computer file) , Computer networks , Computer networks -- Management , Digital communications , Computer sound processing , Sound -- Recording and reproducing -- Digital techniques , Broadcast data systems , C# (Computer program language) , C++ (Computer program language) , ActionScript (Computer program language)
- Description: Traditionally, connection management applications (referred to as patchbays) for high-speed audio networking, are predominantly developed using third-generation languages such as C, C# and C++. Due to the rapid increase in distributed audio/video network usage in the world today, connection management applications that control signal routing over these networks have also evolved in complexity to accommodate more functionality. As the result, high-speed audio networking application developers require a tool that will enable them to develop complex connection management applications easily and within the shortest possible time. In addition, this tool should provide them with the reliability and flexibility required to develop applications controlling signal routing in networks carrying real-time data. High-speed audio networks are used for various purposes that include audio/video production and broadcasting. This investigation evaluates the possibility of using Adobe Flash Professional 8, using ActionScript 2.0, for developing connection management applications. Three patchbays, namely the Broadcast patchbay, the Project studio patchbay, and the Hospitality/Convention Centre patchbay were developed and tested for connection management in three sound installation networks, namely the Broadcast network, the Project studio network, and the Hospitality/Convention Centre network. Findings indicate that complex connection management applications can effectively be implemented using the Adobe Flash IDE and ActionScript 2.0.
- Full Text:
- Date Issued: 2008
- Authors: Sibanda, Phathisile
- Date: 2008 , 2008-03-12
- Subjects: Flash (Computer file) , Computer networks , Computer networks -- Management , Digital communications , Computer sound processing , Sound -- Recording and reproducing -- Digital techniques , Broadcast data systems , C# (Computer program language) , C++ (Computer program language) , ActionScript (Computer program language)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4634 , http://hdl.handle.net/10962/d1006532 , Flash (Computer file) , Computer networks , Computer networks -- Management , Digital communications , Computer sound processing , Sound -- Recording and reproducing -- Digital techniques , Broadcast data systems , C# (Computer program language) , C++ (Computer program language) , ActionScript (Computer program language)
- Description: Traditionally, connection management applications (referred to as patchbays) for high-speed audio networking, are predominantly developed using third-generation languages such as C, C# and C++. Due to the rapid increase in distributed audio/video network usage in the world today, connection management applications that control signal routing over these networks have also evolved in complexity to accommodate more functionality. As the result, high-speed audio networking application developers require a tool that will enable them to develop complex connection management applications easily and within the shortest possible time. In addition, this tool should provide them with the reliability and flexibility required to develop applications controlling signal routing in networks carrying real-time data. High-speed audio networks are used for various purposes that include audio/video production and broadcasting. This investigation evaluates the possibility of using Adobe Flash Professional 8, using ActionScript 2.0, for developing connection management applications. Three patchbays, namely the Broadcast patchbay, the Project studio patchbay, and the Hospitality/Convention Centre patchbay were developed and tested for connection management in three sound installation networks, namely the Broadcast network, the Project studio network, and the Hospitality/Convention Centre network. Findings indicate that complex connection management applications can effectively be implemented using the Adobe Flash IDE and ActionScript 2.0.
- Full Text:
- Date Issued: 2008
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
- Full Text:
- Date Issued: 2003
- 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
- Full Text:
- Date Issued: 2003
A distributed approach to surround sound production
- Authors: Smith, Adrian Wilfrid
- Date: 1999
- Subjects: Surround-sound systems , Computer sound processing , Music -- Data processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4602 , http://hdl.handle.net/10962/d1004855 , Surround-sound systems , Computer sound processing , Music -- Data processing
- Description: The requirement for multi-channel surround sound in audio production applications is growing rapidly. Audio processing in these applications can be costly, particularly in multi-channel systems. A distributed approach is proposed for the development of a realtime spatialization system for surround sound music production, using Ambisonic surround sound methods. The latency in the system is analyzed, with a focus on the audio processing and network delays, in order to ascertain the feasibility of an enhanced, distributed real-time spatialization system.
- Full Text:
- Date Issued: 1999
- Authors: Smith, Adrian Wilfrid
- Date: 1999
- Subjects: Surround-sound systems , Computer sound processing , Music -- Data processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4602 , http://hdl.handle.net/10962/d1004855 , Surround-sound systems , Computer sound processing , Music -- Data processing
- Description: The requirement for multi-channel surround sound in audio production applications is growing rapidly. Audio processing in these applications can be costly, particularly in multi-channel systems. A distributed approach is proposed for the development of a realtime spatialization system for surround sound music production, using Ambisonic surround sound methods. The latency in the system is analyzed, with a focus on the audio processing and network delays, in order to ascertain the feasibility of an enhanced, distributed real-time spatialization system.
- Full Text:
- Date Issued: 1999
An investigation into the use of IEEE 1394 for audio and control data distribution in music studio environments
- Authors: Laubscher, Robert Alan
- Date: 1999 , 2011-11-10
- Subjects: Digital electronics , Sound -- Recording and reproducing -- Digital techniques , MIDI (Standard) , Music -- Data processing , Computer sound processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4619 , http://hdl.handle.net/10962/d1006483 , Digital electronics , Sound -- Recording and reproducing -- Digital techniques , MIDI (Standard) , Music -- Data processing , Computer sound processing
- Description: This thesis investigates the feasibility of using a new digital interconnection technology, the IEEE-1394 High Performance Serial Bus, for audio and control data distribution in local and remote music recording studio environments. Current methods for connecting studio devices are described, and the need for a new digital interconnection technology explained. It is shown how this new interconnection technology and developing protocol standards make provision for multi-channel audio and control data distribution, routing, copyright protection, and device synchronisation. Feasibility is demonstrated by the implementation of a custom hardware and software solution. Remote music studio connectivity is considered, and the emerging standards and technologies for connecting future music studio utilising this new technology are discussed. , Microsoft Word , Adobe Acrobat 9.46 Paper Capture Plug-in
- Full Text:
- Date Issued: 1999
- Authors: Laubscher, Robert Alan
- Date: 1999 , 2011-11-10
- Subjects: Digital electronics , Sound -- Recording and reproducing -- Digital techniques , MIDI (Standard) , Music -- Data processing , Computer sound processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4619 , http://hdl.handle.net/10962/d1006483 , Digital electronics , Sound -- Recording and reproducing -- Digital techniques , MIDI (Standard) , Music -- Data processing , Computer sound processing
- Description: This thesis investigates the feasibility of using a new digital interconnection technology, the IEEE-1394 High Performance Serial Bus, for audio and control data distribution in local and remote music recording studio environments. Current methods for connecting studio devices are described, and the need for a new digital interconnection technology explained. It is shown how this new interconnection technology and developing protocol standards make provision for multi-channel audio and control data distribution, routing, copyright protection, and device synchronisation. Feasibility is demonstrated by the implementation of a custom hardware and software solution. Remote music studio connectivity is considered, and the emerging standards and technologies for connecting future music studio utilising this new technology are discussed. , Microsoft Word , Adobe Acrobat 9.46 Paper Capture Plug-in
- Full Text:
- Date Issued: 1999
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