Real-time audio spectrum analyser research, design, development and implementation using the 32 bit ARMR Cortex-M4 microcontroller
- Authors: Just, Stefan Antonio
- Date: 2017
- Subjects: Spectrum analyzers , Sound -- Recording and reproducing -- Digital techniques , Real-time data processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: http://hdl.handle.net/10962/50536 , vital:25997
- Description: This thesis describes the design and testing of a low-cost hand-held real-time audio analyser (RTAA). This includes the design of an embedded system, the development of the firmware executed by the embedded system, and the implementation of a real-time signal processing algorithms. One of the objectives of this project was to design an alternative low-cost audio analyser to the current commercially available solutions. The device was tested with the audio standard test signal (pink noise) and was compared to the expected at-spectrum response corresponding to a balanced audio system. The design makes use of an 32-bit Reduced Instruction Set Computer (RISC) processor core (ARM Cortex-M4), namely the STM32F4 family of microcontrollers. Due to the pin compatibility of the microcontroller (designed and manufactured by STMicroelectronics), the new development board can also be upgraded with the newly released Cortex-M7 microcontroller, namely the STM32F7 family of microcontrollers. Moreover, the low-cost hardware design features 256kB Random Access Memory (RAM); on-board Micro-Electro-Mechanical System (MEMS) microphone; on-chip 12-bit Analogue-to-Digital (A/D) and Digital-to-Analogue (D/A) Converters; 3.2" Thin-Film-Transistor Liquid-Crystal Display (TFT-LCD) with a resistive touch screen sensor and SD-Card Socket. Furthermore, two additional expansion modules were designed and can extend the functionality of the designed real-time audio analyser. Firstly, an audio/video module featuring a professional 24-bit 192kHz sampling rate audio CODEC; balanced audio microphone input; unbalanced line output; three MEMS microphone inputs; headphone output; and a Video Graphics Array (VGA) controller allowing the display of the analysed audio spectrum on either a projector or monitor. The second expansion module features two external memories: 1MB Static Random Access Memory (SRAM) and 16MB Synchronous Dynamic Random Access Memory (SDRAM). While the two additional expansion modules were not completely utilised by the firmware presented in this thesis, upgrades of the real-time audio analyser firmware in future revisions will provide a higher performing and more accurate analysis of the audio spectrum. The full research and design process for the real-time audio analyser is discussed and both Problems and pitfalls with the final implemented design are highlighted and possible resolutions were investigated. The development costs (excluding labour) are given in the form of a bill of materials (BOM) with the total costs averaging around R1000. Moreover, the additional VGA controller could further decrease the overall costs with the removal of the TFT-LCD screen from the audio analyser and provided the external display was not included in the BOM.
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- Authors: Just, Stefan Antonio
- Date: 2017
- Subjects: Spectrum analyzers , Sound -- Recording and reproducing -- Digital techniques , Real-time data processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: http://hdl.handle.net/10962/50536 , vital:25997
- Description: This thesis describes the design and testing of a low-cost hand-held real-time audio analyser (RTAA). This includes the design of an embedded system, the development of the firmware executed by the embedded system, and the implementation of a real-time signal processing algorithms. One of the objectives of this project was to design an alternative low-cost audio analyser to the current commercially available solutions. The device was tested with the audio standard test signal (pink noise) and was compared to the expected at-spectrum response corresponding to a balanced audio system. The design makes use of an 32-bit Reduced Instruction Set Computer (RISC) processor core (ARM Cortex-M4), namely the STM32F4 family of microcontrollers. Due to the pin compatibility of the microcontroller (designed and manufactured by STMicroelectronics), the new development board can also be upgraded with the newly released Cortex-M7 microcontroller, namely the STM32F7 family of microcontrollers. Moreover, the low-cost hardware design features 256kB Random Access Memory (RAM); on-board Micro-Electro-Mechanical System (MEMS) microphone; on-chip 12-bit Analogue-to-Digital (A/D) and Digital-to-Analogue (D/A) Converters; 3.2" Thin-Film-Transistor Liquid-Crystal Display (TFT-LCD) with a resistive touch screen sensor and SD-Card Socket. Furthermore, two additional expansion modules were designed and can extend the functionality of the designed real-time audio analyser. Firstly, an audio/video module featuring a professional 24-bit 192kHz sampling rate audio CODEC; balanced audio microphone input; unbalanced line output; three MEMS microphone inputs; headphone output; and a Video Graphics Array (VGA) controller allowing the display of the analysed audio spectrum on either a projector or monitor. The second expansion module features two external memories: 1MB Static Random Access Memory (SRAM) and 16MB Synchronous Dynamic Random Access Memory (SDRAM). While the two additional expansion modules were not completely utilised by the firmware presented in this thesis, upgrades of the real-time audio analyser firmware in future revisions will provide a higher performing and more accurate analysis of the audio spectrum. The full research and design process for the real-time audio analyser is discussed and both Problems and pitfalls with the final implemented design are highlighted and possible resolutions were investigated. The development costs (excluding labour) are given in the form of a bill of materials (BOM) with the total costs averaging around R1000. Moreover, the additional VGA controller could further decrease the overall costs with the removal of the TFT-LCD screen from the audio analyser and provided the external display was not included in the BOM.
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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.
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- 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.
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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:
- 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.
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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.
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- 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.
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An investigation into the application of the IEEE 1394 high performance serial bus to sound installation contro
- Authors: Klinkradt, Bradley Hugh
- Date: 2003 , 2013-05-24
- Subjects: Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4612 , http://hdl.handle.net/10962/d1004899 , Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Description: This thesis investigates the feasibility of using existing IP-based control and monitoring protocols within professional audio installations utilising IEEE 1394 technology. Current control and monitoring technologies are examined, and the characteristics common to all are extracted and compiled into an object model. This model forms the foundation for a set of evaluation criteria against which current and future control and monitoring protocols may be measured. Protocols considered include AV/C, MIDI, QSC-24, and those utilised within the UPnP architecture. As QSC-24 and the UPnP architecture are IP-based, the facilities required to transport IP datagrams over the IEEE 1394 bus are investigated and implemented. Example QSC-24 and UPnP architecture implementations are described, which permit the control and monitoring of audio devices over the IEEE 1394 network using these IP-based technologies. The way forward for the control and monitoring of professional audio devices within installations is considered, and recommendations are provided. , KMBT_363 , Adobe Acrobat 9.54 Paper Capture Plug-in
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- Authors: Klinkradt, Bradley Hugh
- Date: 2003 , 2013-05-24
- Subjects: Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4612 , http://hdl.handle.net/10962/d1004899 , Digital electronics , Sound -- Recording and reproducing -- Digital techniques , Music -- Data processing , Computer sound processing
- Description: This thesis investigates the feasibility of using existing IP-based control and monitoring protocols within professional audio installations utilising IEEE 1394 technology. Current control and monitoring technologies are examined, and the characteristics common to all are extracted and compiled into an object model. This model forms the foundation for a set of evaluation criteria against which current and future control and monitoring protocols may be measured. Protocols considered include AV/C, MIDI, QSC-24, and those utilised within the UPnP architecture. As QSC-24 and the UPnP architecture are IP-based, the facilities required to transport IP datagrams over the IEEE 1394 bus are investigated and implemented. Example QSC-24 and UPnP architecture implementations are described, which permit the control and monitoring of audio devices over the IEEE 1394 network using these IP-based technologies. The way forward for the control and monitoring of professional audio devices within installations is considered, and recommendations are provided. , KMBT_363 , Adobe Acrobat 9.54 Paper Capture Plug-in
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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
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- 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
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