Determination of speaker configuration for an immersive audio content creation system
- Authors: Lebusa, Motebang
- Date: 2020
- Subjects: Loudspeakers , Surround-sound systems , Algorithms , Coordinates
- Language: English
- Type: Academic theses , Master's theses , text
- Identifier: http://hdl.handle.net/10962/163375 , vital:41034
- Description: Various spatialisation algorithms require the knowledge of speaker locations to accurately localise sound in 3D environments. The rendering process uses speaker coordinates to feed into their algorithms so that they can render the immersive audio content as intended by an artist. The need to measure the loudspeaker coordinates becomes necessary, especially in environments where the speaker layouts change frequently. Manually measuring the coordinates, however, tends to be a laborious task that is prone to errors. This research provides an automated solution to the problem of speaker coordinates measurement. The solution system, SDIAS, is a client-server system that uses the capabilities provided by the Ethernet Audio Video Bridging standard to measure the 3D loudspeaker coordinates for immersive sound systems. SDIAS deploys commodity hardware and readily available software to implement the solution. A server sends a short tone to each speaker in the speaker configuration, at equal intervals. A microphone attached to a mobile device picks up these transmitted tones on the client side, from different locations. The transmission and reception times from both components of the system are used to measure the time of flight for each tone sent to a loudspeaker. These are then used to determine the 3D coordinates of each loudspeaker in the available layout. Tests were performed to determine the accuracy of the determination algorithm for SDIAS, and were compared to the manually measured coordinates. , Thesis (MSc) -- Faculty of Science, Computer Science, 2020
- Full Text:
- Date Issued: 2020
- Authors: Lebusa, Motebang
- Date: 2020
- Subjects: Loudspeakers , Surround-sound systems , Algorithms , Coordinates
- Language: English
- Type: Academic theses , Master's theses , text
- Identifier: http://hdl.handle.net/10962/163375 , vital:41034
- Description: Various spatialisation algorithms require the knowledge of speaker locations to accurately localise sound in 3D environments. The rendering process uses speaker coordinates to feed into their algorithms so that they can render the immersive audio content as intended by an artist. The need to measure the loudspeaker coordinates becomes necessary, especially in environments where the speaker layouts change frequently. Manually measuring the coordinates, however, tends to be a laborious task that is prone to errors. This research provides an automated solution to the problem of speaker coordinates measurement. The solution system, SDIAS, is a client-server system that uses the capabilities provided by the Ethernet Audio Video Bridging standard to measure the 3D loudspeaker coordinates for immersive sound systems. SDIAS deploys commodity hardware and readily available software to implement the solution. A server sends a short tone to each speaker in the speaker configuration, at equal intervals. A microphone attached to a mobile device picks up these transmitted tones on the client side, from different locations. The transmission and reception times from both components of the system are used to measure the time of flight for each tone sent to a loudspeaker. These are then used to determine the 3D coordinates of each loudspeaker in the available layout. Tests were performed to determine the accuracy of the determination algorithm for SDIAS, and were compared to the manually measured coordinates. , Thesis (MSc) -- Faculty of Science, Computer Science, 2020
- Full Text:
- Date Issued: 2020
A convenient approach to the deterministic routing of MIDI messages
- Authors: Shaw, Brent Roy
- Date: 2018
- Subjects: MIDI (Standard) , Microcontrollers , XMOS Limited , Computer architecture , Embedded computer systems
- Language: English
- Type: text , Thesis , Masters , MSc
- Identifier: http://hdl.handle.net/10962/63256 , vital:28387
- Description: This research investigates the design and development of a Wireless MIDI Connection Management solution in order to create a deterministic MIDI transmission system. A investigation of the MIDI protocol show it to have certain limitation that can be overcome through the use of transmission solutions. These solutions can be used to improve on the versatility of MIDI while overcoming the MIDI's notorious cable length limitation. XMOS's deterministic XS1 microcontrollers are used to enable the design of a real-time system. The MIDINet system is investigated to identify both the strengths and weaknesses of such a connection management system, while other systems for network transmission of MIDI messages are reviewed. These investigations lead to a design concept for a new network MIDI transmission system that allows for the remote management of connections. The design and subsequent implementation of both the transmission system and the connection management system are then detailed. A testing methodology is then devised to allow for the newly created connection management system to be compared to the MIDINet system. The findings show the deterministic system to have lower latency than that of the MIDINet system, while utilising more compact and power efficient hardware.
- Full Text:
- Date Issued: 2018
- Authors: Shaw, Brent Roy
- Date: 2018
- Subjects: MIDI (Standard) , Microcontrollers , XMOS Limited , Computer architecture , Embedded computer systems
- Language: English
- Type: text , Thesis , Masters , MSc
- Identifier: http://hdl.handle.net/10962/63256 , vital:28387
- Description: This research investigates the design and development of a Wireless MIDI Connection Management solution in order to create a deterministic MIDI transmission system. A investigation of the MIDI protocol show it to have certain limitation that can be overcome through the use of transmission solutions. These solutions can be used to improve on the versatility of MIDI while overcoming the MIDI's notorious cable length limitation. XMOS's deterministic XS1 microcontrollers are used to enable the design of a real-time system. The MIDINet system is investigated to identify both the strengths and weaknesses of such a connection management system, while other systems for network transmission of MIDI messages are reviewed. These investigations lead to a design concept for a new network MIDI transmission system that allows for the remote management of connections. The design and subsequent implementation of both the transmission system and the connection management system are then detailed. A testing methodology is then devised to allow for the newly created connection management system to be compared to the MIDINet system. The findings show the deterministic system to have lower latency than that of the MIDINet system, while utilising more compact and power efficient hardware.
- Full Text:
- Date Issued: 2018
An investigation of protocol command translation as a means to enable interoperability between networked audio devices
- Authors: Igumbor, Osedum Peter
- Date: 2014
- Subjects: Streaming audio Data transmission systems Computer network protocols Computer networks -- Management Command languages (Computer science)
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4689 , http://hdl.handle.net/10962/d1011128
- Description: Digital audio networks allow multiple channels of audio to be streamed between devices. This eliminates the need for many different cables to route audio between devices. An added advantage of digital audio networks is the ability to configure and control the networked devices from a common control point. Common control of networked devices enables a sound engineer to establish and destroy audio stream connections between networked devices that are distances apart. On a digital audio network, an audio transport technology enables the exchange of data streams. Typically, an audio transport technology is capable of transporting both control messages and audio data streams. There exist a number of audio transport technologies. Some of these technologies implement data transport by exchanging OSI/ISO layer 2 data frames, while others transport data within OSI/ISO layer 3 packets. There are some approaches to achieving interoperability between devices that utilize different audio transport technologies. A digital audio device typically implements an audio control protocol, which enables it process configuration and control messages from a remote controller. An audio control protocol also defines the structure of the messages that are exchanged between compliant devices. There are currently a wide range of audio control protocols. Some audio control protocols utilize layer 3 audio transport technology, while others utilize layer 2 audio transport technology. An audio device can only communicate with other devices that implement the same control protocol, irrespective of a common transport technology that connects the devices. The existence of different audio control protocols among devices on a network results in a situation where the devices are unable to communicate with each other. Furthermore, a single control application is unable to establish or destroy audio stream connections between the networked devices, since they implement different control protocols. When an audio engineer is designing an audio network installation, this interoperability challenge restricts the choice of devices that can be included. Even when audio transport interoperability has been achieved, common control of the devices remains a challenge. This research investigates protocol command translation as a means to enable interoperability between networked audio devices that implement different audio control protocols. It proposes the use of a command translator that is capable of receiving messages conforming to one protocol from any of the networked devices, translating the received message to conform to a different control protocol, then transmitting the translated message to the intended target which understands the translated protocol message. In so doing, the command translator enables common control of the networked devices, since a control application is able to configure and control devices that conform to different protocols by utilizing the command translator to perform appropriate protocol translation.
- Full Text:
- Date Issued: 2014
- Authors: Igumbor, Osedum Peter
- Date: 2014
- Subjects: Streaming audio Data transmission systems Computer network protocols Computer networks -- Management Command languages (Computer science)
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4689 , http://hdl.handle.net/10962/d1011128
- Description: Digital audio networks allow multiple channels of audio to be streamed between devices. This eliminates the need for many different cables to route audio between devices. An added advantage of digital audio networks is the ability to configure and control the networked devices from a common control point. Common control of networked devices enables a sound engineer to establish and destroy audio stream connections between networked devices that are distances apart. On a digital audio network, an audio transport technology enables the exchange of data streams. Typically, an audio transport technology is capable of transporting both control messages and audio data streams. There exist a number of audio transport technologies. Some of these technologies implement data transport by exchanging OSI/ISO layer 2 data frames, while others transport data within OSI/ISO layer 3 packets. There are some approaches to achieving interoperability between devices that utilize different audio transport technologies. A digital audio device typically implements an audio control protocol, which enables it process configuration and control messages from a remote controller. An audio control protocol also defines the structure of the messages that are exchanged between compliant devices. There are currently a wide range of audio control protocols. Some audio control protocols utilize layer 3 audio transport technology, while others utilize layer 2 audio transport technology. An audio device can only communicate with other devices that implement the same control protocol, irrespective of a common transport technology that connects the devices. The existence of different audio control protocols among devices on a network results in a situation where the devices are unable to communicate with each other. Furthermore, a single control application is unable to establish or destroy audio stream connections between the networked devices, since they implement different control protocols. When an audio engineer is designing an audio network installation, this interoperability challenge restricts the choice of devices that can be included. Even when audio transport interoperability has been achieved, common control of the devices remains a challenge. This research investigates protocol command translation as a means to enable interoperability between networked audio devices that implement different audio control protocols. It proposes the use of a command translator that is capable of receiving messages conforming to one protocol from any of the networked devices, translating the received message to conform to a different control protocol, then transmitting the translated message to the intended target which understands the translated protocol message. In so doing, the command translator enables common control of the networked devices, since a control application is able to configure and control devices that conform to different protocols by utilizing the command translator to perform appropriate protocol translation.
- Full Text:
- Date Issued: 2014
An investigation into the control of audio streaming across networks having diverse quality of service mechanisms
- Authors: Foulkes, Philip James
- Date: 2012
- Subjects: Streaming audio -- Testing Data transmission systems -- Testing Computer networks -- Management Computer networks -- Evaluation Computer network protocols -- Standards
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4607 , http://hdl.handle.net/10962/d1004865
- Description: The transmission of realtime audio data across digital networks is subject to strict quality of service requirements. These networks need to be able to guarantee network resources (e.g., bandwidth), ensure timely and deterministic data delivery, and provide time synchronisation mechanisms to ensure successful transmission of this data. Two open standards-based networking technologies, namely IEEE 1394 and the recently standardised Ethernet AVB, provide distinct methods for achieving these goals. Audio devices that are compatible with IEEE 1394 networks exist, and audio devices that are compatible with Ethernet AVB networks are starting to come onto the market. There is a need for mechanisms to provide compatibility between the audio devices that reside on these disparate networks such that existing IEEE 1394 audio devices are able to communicate with Ethernet AVB audio devices, and vice versa. The audio devices that reside on these networks may be remotely controlled by a diverse set of incompatible command and control protocols. It is desirable to have a common network-neutral method of control over the various parameters of the devices that reside on these networks. As part of this study, two Ethernet AVB systems were developed. One system acts as an Ethernet AVB audio endpoint device and another system acts as an audio gateway between IEEE 1394 and Ethernet AVB networks. These systems, along with existing IEEE 1394 audio devices, were used to demonstrate the ability to transfer audio data between the networking technologies. Each of the devices is remotely controllable via a network neutral command and control protocol, XFN. The IEEE 1394 and Ethernet AVB devices are used to demonstrate the use of the XFN protocol to allow for network neutral connection management to take place between IEEE 1394 and Ethernet AVB networks. User control over these diverse devices is achieved via the use of a graphical patchbay application, which aims to provide a consistent user interface to a diverse range of devices.
- Full Text:
- Date Issued: 2012
- Authors: Foulkes, Philip James
- Date: 2012
- Subjects: Streaming audio -- Testing Data transmission systems -- Testing Computer networks -- Management Computer networks -- Evaluation Computer network protocols -- Standards
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4607 , http://hdl.handle.net/10962/d1004865
- Description: The transmission of realtime audio data across digital networks is subject to strict quality of service requirements. These networks need to be able to guarantee network resources (e.g., bandwidth), ensure timely and deterministic data delivery, and provide time synchronisation mechanisms to ensure successful transmission of this data. Two open standards-based networking technologies, namely IEEE 1394 and the recently standardised Ethernet AVB, provide distinct methods for achieving these goals. Audio devices that are compatible with IEEE 1394 networks exist, and audio devices that are compatible with Ethernet AVB networks are starting to come onto the market. There is a need for mechanisms to provide compatibility between the audio devices that reside on these disparate networks such that existing IEEE 1394 audio devices are able to communicate with Ethernet AVB audio devices, and vice versa. The audio devices that reside on these networks may be remotely controlled by a diverse set of incompatible command and control protocols. It is desirable to have a common network-neutral method of control over the various parameters of the devices that reside on these networks. As part of this study, two Ethernet AVB systems were developed. One system acts as an Ethernet AVB audio endpoint device and another system acts as an audio gateway between IEEE 1394 and Ethernet AVB networks. These systems, along with existing IEEE 1394 audio devices, were used to demonstrate the ability to transfer audio data between the networking technologies. Each of the devices is remotely controllable via a network neutral command and control protocol, XFN. The IEEE 1394 and Ethernet AVB devices are used to demonstrate the use of the XFN protocol to allow for network neutral connection management to take place between IEEE 1394 and Ethernet AVB networks. User control over these diverse devices is achieved via the use of a graphical patchbay application, which aims to provide a consistent user interface to a diverse range of devices.
- Full Text:
- Date Issued: 2012
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
High speed end-to-end connection management in a bridged IEEE 1394 network of professional audio devices
- Authors: Okai-Tettey, Harold A
- Date: 2006
- Subjects: IEEE 1394 (Standard) Digital communications Computer networks Sound -- Recording and reproducing -- Digital techniques Computer sound processing
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4653 , http://hdl.handle.net/10962/d1006638
- Description: A number of companies have developed a variety of network approaches to the transfer of audio and MIDI data. By doing this, they have addressed the configuration complications that were present when using direct patching for analogue audio, digital audio, word clock, and control connections. Along with their approaches, controlling software, usually running on a PC, is used to set up and manage audio routings from the outputs to the inputs of devices. However one of the advantages of direct patching is the conceptual simplicity it provides for a user in connecting plugs of devices, the ability to connect from the host plug of one device to the host plug of another. The connection management or routing applications of the current audio networks do not allow for such a capability, and instead employ what is referred to as a two-step approach to connection management. This two-step approach requires that devices be first configured at the transport layer of the network for input and output routings, after which the transmit and receive plugs of devices are manually configured to transmit or receive data. From a user’s point of view, it is desirable for the connection management or audio routing applications of the current audio networks to be able to establish routings directly between the host plugs of devices, and not the audio channels exposed by a network’s transport, as is currently the case. The main goal of this work has been to retain the conceptual simplicity of point-to-point connection management within digital audio networks, while gaining all the benefits that digital audio networking can offer.
- Full Text:
- Date Issued: 2006
- Authors: Okai-Tettey, Harold A
- Date: 2006
- Subjects: IEEE 1394 (Standard) Digital communications Computer networks Sound -- Recording and reproducing -- Digital techniques Computer sound processing
- Language: English
- Type: Thesis , Doctoral , PhD
- Identifier: vital:4653 , http://hdl.handle.net/10962/d1006638
- Description: A number of companies have developed a variety of network approaches to the transfer of audio and MIDI data. By doing this, they have addressed the configuration complications that were present when using direct patching for analogue audio, digital audio, word clock, and control connections. Along with their approaches, controlling software, usually running on a PC, is used to set up and manage audio routings from the outputs to the inputs of devices. However one of the advantages of direct patching is the conceptual simplicity it provides for a user in connecting plugs of devices, the ability to connect from the host plug of one device to the host plug of another. The connection management or routing applications of the current audio networks do not allow for such a capability, and instead employ what is referred to as a two-step approach to connection management. This two-step approach requires that devices be first configured at the transport layer of the network for input and output routings, after which the transmit and receive plugs of devices are manually configured to transmit or receive data. From a user’s point of view, it is desirable for the connection management or audio routing applications of the current audio networks to be able to establish routings directly between the host plugs of devices, and not the audio channels exposed by a network’s transport, as is currently the case. The main goal of this work has been to retain the conceptual simplicity of point-to-point connection management within digital audio networks, while gaining all the benefits that digital audio networking can offer.
- Full Text:
- Date Issued: 2006
Software quality assurance in a remote client/contractor context
- Authors: Black, Angus Hugh
- Date: 2006
- Subjects: Computer software -- Quality control , Software engineering , Information technology
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4648 , http://hdl.handle.net/10962/d1006615 , Computer software -- Quality control , Software engineering , Information technology
- Description: With the reliance on information technology and the software that this technology utilizes increasing every day, it is of paramount importance that software developed be of an acceptable quality. This quality can be achieved through the utilization of various software engineering standards and guidelines. The question is, to what extent do these standards and guidelines need to be utilized and how are these standards and guidelines implemented? This research focuses on how guidelines developed by standardization bodies and the unified process developed by Rational can be integrated to achieve a suitable process and version control system within the context of a remote client/contractor small team environment.
- Full Text:
- Date Issued: 2006
- Authors: Black, Angus Hugh
- Date: 2006
- Subjects: Computer software -- Quality control , Software engineering , Information technology
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4648 , http://hdl.handle.net/10962/d1006615 , Computer software -- Quality control , Software engineering , Information technology
- Description: With the reliance on information technology and the software that this technology utilizes increasing every day, it is of paramount importance that software developed be of an acceptable quality. This quality can be achieved through the utilization of various software engineering standards and guidelines. The question is, to what extent do these standards and guidelines need to be utilized and how are these standards and guidelines implemented? This research focuses on how guidelines developed by standardization bodies and the unified process developed by Rational can be integrated to achieve a suitable process and version control system within the context of a remote client/contractor small team environment.
- Full Text:
- Date Issued: 2006
A remote interactive music keyboard tuition system
- Authors: Newton, Mark Brian
- Date: 2005
- Subjects: Computer-assisted instruction , Keyboard instrument music -- Instruction and study , Music -- Computer assisted instruction , Music in education
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4603 , http://hdl.handle.net/10962/d1004860 , Computer-assisted instruction , Keyboard instrument music -- Instruction and study , Music -- Computer assisted instruction , Music in education
- Description: A networked multimedia system to assist teaching music keyboard skills to a class is described. Teaching practical music lessons requires a large amount of interaction between the teacher and student and is thus teacher intensive. Although there is a range of computer software available for learning how to play the keyboard, these programs cannot replace the guidance of a music teacher. The possibility of combining the music applications with video conferencing technology for use in a keyboard class is discussed. An ideal system is described that incorporates the benefits of video conferencing and music applications for use in a classroom. A design of the ideal system is described and implemented. Certain design and implementation decisions are explained and the performance of the implementation examined. The system would enable a music teacher to effectively teach a music class keyboard skills.
- Full Text:
- Date Issued: 2005
- Authors: Newton, Mark Brian
- Date: 2005
- Subjects: Computer-assisted instruction , Keyboard instrument music -- Instruction and study , Music -- Computer assisted instruction , Music in education
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4603 , http://hdl.handle.net/10962/d1004860 , Computer-assisted instruction , Keyboard instrument music -- Instruction and study , Music -- Computer assisted instruction , Music in education
- Description: A networked multimedia system to assist teaching music keyboard skills to a class is described. Teaching practical music lessons requires a large amount of interaction between the teacher and student and is thus teacher intensive. Although there is a range of computer software available for learning how to play the keyboard, these programs cannot replace the guidance of a music teacher. The possibility of combining the music applications with video conferencing technology for use in a keyboard class is discussed. An ideal system is described that incorporates the benefits of video conferencing and music applications for use in a classroom. A design of the ideal system is described and implemented. Certain design and implementation decisions are explained and the performance of the implementation examined. The system would enable a music teacher to effectively teach a music class keyboard skills.
- Full Text:
- Date Issued: 2005
A comparative study of the Linux and windows device driver architecture with a focus on IEEE1394 (high speed serial bus) drivers
- Authors: Tsegaye, Melekam Asrat
- Date: 2004
- Subjects: Microsoft Windows (Computer file) , Linux , Operating systems (Computers) , DOS device drivers (Computer programs) , Linux device drivers (Computer programs)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4591 , http://hdl.handle.net/10962/d1004829 , Microsoft Windows (Computer file) , Linux , Operating systems (Computers) , DOS device drivers (Computer programs) , Linux device drivers (Computer programs)
- Description: New hardware devices are continually being released to the public by hardware manufactures around the world. For these new devices to be usable under a PC operating system, device drivers that extend the functionality of the target operating system have to be constructed. This work examines and compares the device driver architectures currently in use by two of the most widely used operating systems, Microsoft’s Windows and Linux. The IEEE1394 (high speed serial bus) device driver stacks on each operating system are examined and compared as an example of a major device driver stack implementation, including driver requirements for the upcoming IEEE1394.1 bridging standard.
- Full Text:
- Date Issued: 2004
- Authors: Tsegaye, Melekam Asrat
- Date: 2004
- Subjects: Microsoft Windows (Computer file) , Linux , Operating systems (Computers) , DOS device drivers (Computer programs) , Linux device drivers (Computer programs)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4591 , http://hdl.handle.net/10962/d1004829 , Microsoft Windows (Computer file) , Linux , Operating systems (Computers) , DOS device drivers (Computer programs) , Linux device drivers (Computer programs)
- Description: New hardware devices are continually being released to the public by hardware manufactures around the world. For these new devices to be usable under a PC operating system, device drivers that extend the functionality of the target operating system have to be constructed. This work examines and compares the device driver architectures currently in use by two of the most widely used operating systems, Microsoft’s Windows and Linux. The IEEE1394 (high speed serial bus) device driver stacks on each operating system are examined and compared as an example of a major device driver stack implementation, including driver requirements for the upcoming IEEE1394.1 bridging standard.
- Full Text:
- Date Issued: 2004
Multiprotocol control of networked home entertainment devices
- Authors: Siebörger, David Robert
- Date: 2004
- Subjects: Home entertainment systems , Home video systems
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4642 , http://hdl.handle.net/10962/d1006585 , Home entertainment systems , Home video systems
- Description: Networks will soon connect a wide range of computing devices within the home. Amongst those devices will be home entertainment devices. Remote control over the network will be a key application for networked entertainment devices, and requires a protocol for communication understood by both controller and controlled device. Devices capable of communication using multiple control protocols will be compatible with a wider range of controllers than those which implement only one control protocol. This work examines home networks and a number of control protocols. The implementations of the UPnP and AV/C protocols for an AV receiver are described. The issues involved in the concurrent use of multiple control protocols to control a device are considered, possible methods of concurrent control discussed, and a solution which simulates virtual copies of the device is implemented and tested.
- Full Text:
- Date Issued: 2004
- Authors: Siebörger, David Robert
- Date: 2004
- Subjects: Home entertainment systems , Home video systems
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4642 , http://hdl.handle.net/10962/d1006585 , Home entertainment systems , Home video systems
- Description: Networks will soon connect a wide range of computing devices within the home. Amongst those devices will be home entertainment devices. Remote control over the network will be a key application for networked entertainment devices, and requires a protocol for communication understood by both controller and controlled device. Devices capable of communication using multiple control protocols will be compatible with a wider range of controllers than those which implement only one control protocol. This work examines home networks and a number of control protocols. The implementations of the UPnP and AV/C protocols for an AV receiver are described. The issues involved in the concurrent use of multiple control protocols to control a device are considered, possible methods of concurrent control discussed, and a solution which simulates virtual copies of the device is implemented and tested.
- Full Text:
- Date Issued: 2004
An investigation into tools and protocols for commercial audio web-site creation
- Authors: Ndinga, S'busiso Simon
- Date: 2000
- Subjects: Web sites -- Design , Digital libraries , Internet -- Security measures
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4621 , http://hdl.handle.net/10962/d1006488 , Web sites -- Design , Digital libraries , Internet -- Security measures
- Description: This thesis presents a feasibility study of a Web-based digital music library and purchasing system. It investigates the current status of the enabling technologies for developing such a system. An analysis of various Internet audio codecs, streaming audio protocols, Internet credit card payment security methods, and ways for accessing remote Web databases is presented. The objective of the analysis is to determine the viability and the economic benefits of using these technologies when developing systems that facilitate music distribution over the Internet. A prototype of a distributed digital music library and purchasing system named WAPS (for Web-based Audio Purchasing System) was developed and implemented in the Java programming language. In this thesis both the physical and the logical component elements of WAPS are explored in depth so as to provide an insight into the inherent problems of creating such a system, as well as the overriding benefits derived from the creation of such a system.
- Full Text:
- Date Issued: 2000
- Authors: Ndinga, S'busiso Simon
- Date: 2000
- Subjects: Web sites -- Design , Digital libraries , Internet -- Security measures
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4621 , http://hdl.handle.net/10962/d1006488 , Web sites -- Design , Digital libraries , Internet -- Security measures
- Description: This thesis presents a feasibility study of a Web-based digital music library and purchasing system. It investigates the current status of the enabling technologies for developing such a system. An analysis of various Internet audio codecs, streaming audio protocols, Internet credit card payment security methods, and ways for accessing remote Web databases is presented. The objective of the analysis is to determine the viability and the economic benefits of using these technologies when developing systems that facilitate music distribution over the Internet. A prototype of a distributed digital music library and purchasing system named WAPS (for Web-based Audio Purchasing System) was developed and implemented in the Java programming language. In this thesis both the physical and the logical component elements of WAPS are explored in depth so as to provide an insight into the inherent problems of creating such a system, as well as the overriding benefits derived from the creation of such a system.
- Full Text:
- Date Issued: 2000
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
An object-oriented toolkit for music notation
- Authors: Eales, Andrew Arnold
- Date: 1999 , 2000-04-26
- Subjects: Musical notation , Object-oriented programming (Computer science) , Computer music -- History and criticism , Musical notation -- Software
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4618 , http://hdl.handle.net/10962/d1006473 , Musical notation , Object-oriented programming (Computer science) , Computer music -- History and criticism , Musical notation -- Software
- Description: This thesis investigates the design and implementation of an object-oriented toolkit for music notation. It considers whether object-oriented technology provides features that are desirable for representing music notation. The ability to sympathetically represent the conventions of music notation provides software tools that are flexible to use, and easily extended to represent less common features of music notation. The design and implementation of an object-oriented class hierarchy that captures the structural and semantic relationships of music notation symbols is described. Functions that search for symbols, and update symbol positions are also implemented. Traditional context-sensitive and spatial relationships between music symbols may be maintained, or extended to provide notational features found in modern music. MIDI functionality includes the ability to play music notation and to allow step-recording of MIDI events. The toolkit has been designed to simplify the creation of applications that make use of music notation; example applications are created to demonstrate its capabilities. , Microsoft Word , Adobe Acrobat 9.46 Paper Capture Plug-in
- Full Text:
- Date Issued: 1999
- Authors: Eales, Andrew Arnold
- Date: 1999 , 2000-04-26
- Subjects: Musical notation , Object-oriented programming (Computer science) , Computer music -- History and criticism , Musical notation -- Software
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4618 , http://hdl.handle.net/10962/d1006473 , Musical notation , Object-oriented programming (Computer science) , Computer music -- History and criticism , Musical notation -- Software
- Description: This thesis investigates the design and implementation of an object-oriented toolkit for music notation. It considers whether object-oriented technology provides features that are desirable for representing music notation. The ability to sympathetically represent the conventions of music notation provides software tools that are flexible to use, and easily extended to represent less common features of music notation. The design and implementation of an object-oriented class hierarchy that captures the structural and semantic relationships of music notation symbols is described. Functions that search for symbols, and update symbol positions are also implemented. Traditional context-sensitive and spatial relationships between music symbols may be maintained, or extended to provide notational features found in modern music. MIDI functionality includes the ability to play music notation and to allow step-recording of MIDI events. The toolkit has been designed to simplify the creation of applications that make use of music notation; example applications are created to demonstrate its capabilities. , Microsoft Word , Adobe Acrobat 9.46 Paper Capture Plug-in
- Full Text:
- Date Issued: 1999
Routing MIDI messages in a shared music studio environment
- Authors: Mosala, Thabo Jerry
- Date: 1995
- Subjects: MIDI (Standard) , Computer networks
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4672 , http://hdl.handle.net/10962/d1006695 , MIDI (Standard) , Computer networks
- Description: The Rhodes Computer Music Network (RHOCMN) is a network which allows main studio resources to be shared. RHOCMN is growing into a multi-workstation environment and additional devices are being incorporated into the system. A star configuration is used for transmitting MIDI from a MIDI patch bay to the workstations and MIDI devices. This imposes several disadvantages on the use of the studio, such as wiring problems. In a quest to avoid problems related to MIDI in RHOCMN, the MIDINet system was developed. The idea was to acquire a viable solution to MIDI's main problems which does not involve a redefinition of the MIDI specification.
- Full Text:
- Date Issued: 1995
- Authors: Mosala, Thabo Jerry
- Date: 1995
- Subjects: MIDI (Standard) , Computer networks
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4672 , http://hdl.handle.net/10962/d1006695 , MIDI (Standard) , Computer networks
- Description: The Rhodes Computer Music Network (RHOCMN) is a network which allows main studio resources to be shared. RHOCMN is growing into a multi-workstation environment and additional devices are being incorporated into the system. A star configuration is used for transmitting MIDI from a MIDI patch bay to the workstations and MIDI devices. This imposes several disadvantages on the use of the studio, such as wiring problems. In a quest to avoid problems related to MIDI in RHOCMN, the MIDINet system was developed. The idea was to acquire a viable solution to MIDI's main problems which does not involve a redefinition of the MIDI specification.
- Full Text:
- Date Issued: 1995
The analysis of a computer music network and the implementation of essential subsystems
- Authors: Wilks, Antony John
- Date: 1995
- Subjects: Computer networks , Computer music , MIDI (Standard)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4666 , http://hdl.handle.net/10962/d1006682 , Computer networks , Computer music , MIDI (Standard)
- Description: The inability to share resources in commercial and institutional computer music studios results in non-optimal resource utilisation. The use of computers to process, store and communicate data can be extended within these studios, to provide the capability of sharing resources amongst their users. This thesis describes a computer music network which was designed for this purpose. Certain devices had to be custom built for the implementation of the network. The thesis discusses the design and construction of these devices.
- Full Text:
- Date Issued: 1995
- Authors: Wilks, Antony John
- Date: 1995
- Subjects: Computer networks , Computer music , MIDI (Standard)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4666 , http://hdl.handle.net/10962/d1006682 , Computer networks , Computer music , MIDI (Standard)
- Description: The inability to share resources in commercial and institutional computer music studios results in non-optimal resource utilisation. The use of computers to process, store and communicate data can be extended within these studios, to provide the capability of sharing resources amongst their users. This thesis describes a computer music network which was designed for this purpose. Certain devices had to be custom built for the implementation of the network. The thesis discusses the design and construction of these devices.
- Full Text:
- Date Issued: 1995
The synthesis of sound with application in a MIDI environment
- Authors: Kesterton, Anthony James
- Date: 1991
- Subjects: Computer sound processing -- Research , Music -- Data processing -- Research , MIDI (Standard)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4678 , http://hdl.handle.net/10962/d1006701 , Computer sound processing -- Research , Music -- Data processing -- Research , MIDI (Standard)
- Description: The wide range of options for experimentation with the synthesis of sound are usually expensive, difficult to obtain, or limit the experimenter. The work described in this thesis shows how the IBM PC and software can be combined to provide a suitable platform for experimentation with different synthesis techniques. This platform is based on the PC, the Musical Instrument Digital Interface (MIDI) and a musical instrument called a digital sampler. The fundamental concepts of sound are described, with reference to digital sound reproduction. A number of synthesis techniques are described. These are evaluated according to the criteria of generality, efficiency and control. The techniques discussed are additive synthesis, frequency modulation synthesis, subtractive synthesis, granular synthesis, resynthesis, wavetable synthesis, and sampling. Spiral synthesis, physical modelling, waveshaping and spectral interpolation are discussed briefly. The Musical Instrument Digital Interface is a standard method of connecting digital musical instruments together. It is the MIDI standard and equipment conforming to that standard that makes this implementation of synthesis techniques possible. As a demonstration of the PC platform, additive synthesis, frequency modulation synthesis, granular synthesis and spiral synthesis have been implemented in software. A PC equipped with a MIDI interface card is used to perform the synthesis. The MIDI protocol is used to transmit the resultant sound to a digital sampler. The INMOS transputer is used as an accelerator, as the calculation of a waveform using software is a computational intensive process. It is concluded that sound synthesis can be performed successfully using a PC and the appropriate software, and utilizing the facilities provided by a MIDI environment including a digital sampler.
- Full Text:
- Date Issued: 1991
- Authors: Kesterton, Anthony James
- Date: 1991
- Subjects: Computer sound processing -- Research , Music -- Data processing -- Research , MIDI (Standard)
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4678 , http://hdl.handle.net/10962/d1006701 , Computer sound processing -- Research , Music -- Data processing -- Research , MIDI (Standard)
- Description: The wide range of options for experimentation with the synthesis of sound are usually expensive, difficult to obtain, or limit the experimenter. The work described in this thesis shows how the IBM PC and software can be combined to provide a suitable platform for experimentation with different synthesis techniques. This platform is based on the PC, the Musical Instrument Digital Interface (MIDI) and a musical instrument called a digital sampler. The fundamental concepts of sound are described, with reference to digital sound reproduction. A number of synthesis techniques are described. These are evaluated according to the criteria of generality, efficiency and control. The techniques discussed are additive synthesis, frequency modulation synthesis, subtractive synthesis, granular synthesis, resynthesis, wavetable synthesis, and sampling. Spiral synthesis, physical modelling, waveshaping and spectral interpolation are discussed briefly. The Musical Instrument Digital Interface is a standard method of connecting digital musical instruments together. It is the MIDI standard and equipment conforming to that standard that makes this implementation of synthesis techniques possible. As a demonstration of the PC platform, additive synthesis, frequency modulation synthesis, granular synthesis and spiral synthesis have been implemented in software. A PC equipped with a MIDI interface card is used to perform the synthesis. The MIDI protocol is used to transmit the resultant sound to a digital sampler. The INMOS transputer is used as an accelerator, as the calculation of a waveform using software is a computational intensive process. It is concluded that sound synthesis can be performed successfully using a PC and the appropriate software, and utilizing the facilities provided by a MIDI environment including a digital sampler.
- Full Text:
- Date Issued: 1991
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