https://hdl.handle.net/2123/14848 2026-06-07T01:50:26Z 2026-06-07T01:50:26Z Wang, Andrew Anxu https://hdl.handle.net/2123/15689 2025-10-21T01:01:25Z 2016-06-07T00:00:00Z

modQ: a modular EQ engine with asymmetric filter profiles Wang, Andrew Anxu The modQ modular EQ engine pairs a novel filter topology with a highly-scalable engine framework to deliver a platform that is equally suited to precision tone-shaping as it is to processing dozens of simultaneous audio streams. The ability to create asymmetrical filter profiles eliminates the complexity associated with overlaying traditional filters while the processing framework delivers resiliency and scalability features unlike any other platform currently available, allowing customers and end users to focus on content and product delivery.

2016-06-07T00:00:00Z Jaworski, Joshua https://hdl.handle.net/2123/15033 2025-10-21T01:01:21Z 2016-06-06T00:00:00Z

Final written review - Polygrain Jaworski, Joshua Polygrain is a powerful compositional tool developed to allow time-based manipulation of an input signal. Based the principles of granular synthesis it breaks an audio signal down into micro-segments or ‘grains’ using them as building blocks in the creation of an output signal. Through an explanation of the Digital Signal Processing (DSP) involved in its implementation and the subjective evaluation of output sounds produced by Polygrain, further development of the plug-in is justified due to its compositional validity.

2016-06-06T00:00:00Z Stephen, Joshua Mark https://hdl.handle.net/2123/15032 2025-10-21T01:01:16Z 2016-06-06T00:00:00Z

The FocusAmp - Multiband Control Over Distortion Based Effects Stephen, Joshua Mark The following final review details the basic components, implementation and evaluation of a multi-band distortion unit called the FocusAmp. Designed to give an end user a large amount of flexibility, the FocusAmp allows users to choose where crossover points between frequency bands are, what type of distortion is used and how much in each band is applied. The unit is controlled via a text based graphical interface and has been implemented within the Matlab architecture. This package contains the final review, in addition to a possible future GUI and audio examples pre and post processing.

2016-06-06T00:00:00Z Park, Yeong Min https://hdl.handle.net/2123/15045 2025-10-21T01:01:18Z 2016-06-06T00:00:00Z

FUZZ FACE OVERDRIVEN BY TUBE-SCREAMER Park, Yeong Min Distortion effects implemented using digital signal processing software MATLAB, problem description about past and current issues we deliver final product specification, implementation, evaluation of final product development.

2016-06-06T00:00:00Z Ashpole, James https://hdl.handle.net/2123/15043 2025-10-21T01:01:21Z 2016-06-06T00:00:00Z

Multi-Band Compressor Ashpole, James Multi-band effects give engineers much more control over how they affect their signal. In conjunction with compressors they allow precise sculpting of a sounds dynamics to ensure optimal quality in a recording. A Multi-band compressor was designed through implementation through MATLAB. Through examination of this code, we can see the effects that this can have on our signal. With adjustments made to the code the user can easily alter parameters to effect their signal how they want and suit any application. A Written Review has been supplied along with all the MATLAB files required for running the multi-band compression system. Examples and test audio files have been supplied also.

2016-06-06T00:00:00Z Fong, Osborn https://hdl.handle.net/2123/15044 2025-10-21T01:01:13Z 2016-06-06T00:00:00Z

Adaptive Pitch Detection employing the use of Fast Fourier Transform and Autocorrelation Function Fong, Osborn

2016-06-06T00:00:00Z Bechara, Matthew https://hdl.handle.net/2123/15046 2025-10-21T01:01:28Z 2016-06-06T00:00:00Z

Ear Exciting Exciter Final Review Bechara, Matthew

2016-06-06T00:00:00Z Jancovich, Benjamin https://hdl.handle.net/2123/15040 2025-10-21T01:01:26Z 2016-06-06T00:00:00Z

SAMPLE LIBRARY PREPARATION TOOL Jancovich, Benjamin The modern audio production process is highly dependent on efficient workflows and pre-preparation of content. Fast turnaround is crucial. For this reason, it is standard practice for producers and sound designers to maintain libraries of pre recorded sounds. Whether those sounds have been purchased as sample libraries or recorded by the end user, it may be necessary that they undergo various types of signal processing to minimize any additional mixing required once they are loaded into to a production. A common example is that of a commercially available drum sample library. Often the hi-hat samples are monaural, or feature a very narrow stereo image, and can also contain unnecessary low frequency information. These issues can be remedied in the production session once the samples have been added, but this interruption to workflow can limit creativity and reduce operational efficiency. This product has been developed as a solution to this problem. It is intended to batch process large numbers of audio files to prepare them for later use. The signal processing modules are implemented in what is known as a multiband processing network, which consists of several sets of parallel signal processing chains that each work on a different part of the frequency spectrum. This allows for a higher degree of control than processing the entire broadband signal. The central concept is that what is considered a desirable sonic characteristic for one part of a signal’s spectrum, may not be desirable for another. An exaggerated stereo width effect for example, may be desirable in the high-mid frequency range of a bass synthesizer sound, and this can be created using an inter- aural time difference. In the low range however, the phasing introduced by this delay between channels will result in an unintended loss of low frequency information. Multiband processing eliminates this problem.

2016-06-06T00:00:00Z Xi, Peng https://hdl.handle.net/2123/15029 2025-10-21T01:04:17Z 2016-06-06T00:00:00Z

Zen-Flanger Xi, Peng

2016-06-06T00:00:00Z Li, Junting Jr https://hdl.handle.net/2123/15038 2025-10-21T01:01:16Z 2016-06-06T00:00:00Z

Simple HRIR Filter Achieved with the Data of KEMAR Li, Junting Jr The HRIR filters are widely used in many fields today, for example, some ASMR recordings and stereo songs mixing. My function provides a simple HRIR filter that could make a mono recording mixed into a stereo, binaural recording with relatively accurate positioning using the data of Knowles Electionics Mannequin for Acousitc Research (KEMAR).

2016-06-06T00:00:00Z Adlington, Isabella https://hdl.handle.net/2123/15036 2025-10-21T01:04:22Z 2016-06-05T00:00:00Z

Auto-wah envelope filter software Adlington, Isabella

2016-06-05T00:00:00Z Pereira, Frederico https://hdl.handle.net/2123/15035 2026-03-11T02:08:57Z 2016-06-06T00:00:00Z

Final Written Review, Digital Audio Systems DESC9115, 2016 Pereira, Frederico A granular synthesis prototype is described The prototype applies the granulation algorithm to a given input audio signal, and outputs up to four channels of audio. The signals in the different output channels can be decorrelated, in order to achieve a superior subjective spatial impression.

2016-06-06T00:00:00Z Vyas, Neil https://hdl.handle.net/2123/15027 2026-03-11T02:08:58Z 2016-06-06T00:00:00Z

Bandpass Limiter Vyas, Neil Bandpass Limiter is a limiter that has a bandpass filter included in its design. When an audio signal is input into the limiter, it band-passes the audio signal before it gets to the limiter stage. The Bandpass Limiter can process mono as well as stereo files and it gives the user an option to use it as a mastering limiter as well as a bandpass limiter, where the low cut and high cut frequencies are defined by the user. All the parameters of the limiter like limiter threshold, attack time and release time can be defined by the user. Main function, supporting function, working example script and example audio files can be found attached.

2016-06-06T00:00:00Z Holmes, Jonothan https://hdl.handle.net/2123/15034 2026-03-11T02:08:54Z 2016-06-06T00:00:00Z

Four Input Amplitude Modulator Holmes, Jonothan This report presents the requirements and assessment of the various processing necessary for an audio effect described here as the four input amplitude modulator. An overview, accompanied by description of the implementation and the audible effects of amplitude and frequency modulation is given. The reader is provided with matlab and audio files that demonstrate the concepts presented.

2016-06-06T00:00:00Z Arcila, Javier https://hdl.handle.net/2123/15041 2026-03-11T02:08:54Z 2016-06-06T00:00:00Z

Basso-Forte DSP Product Proposal Arcila, Javier Much of the music produced nowadays is heard through headphones or small loudspeakers. This implies a considerable lost in sound quality when using this type of devices for music reproduction, especially for the low frequencies. Coincidentally, a good bass performance has become one of the leading features for customers to choose a device over another one. This is why the BASSO-FORTE is introduced as a Digital Audio App designed to enhance the perceived sound level in low frequencies through the loudspeakers of a phone.

2016-06-06T00:00:00Z Brooker, Lucas https://hdl.handle.net/2123/15026 2026-03-11T02:08:57Z 2016-06-06T00:00:00Z

Final Written Review - High and Low Shelf Tone Shaper Brooker, Lucas The characteristics of analogue audio processors are useful and valued for musical applications. A function has been implemented that allows the user to easily and intuitively introduce and control non linear behaviours and produce an output signal that has some of the characteristics that are valued in the analogue processing domain.

2016-06-06T00:00:00Z Love, James https://hdl.handle.net/2123/15039 2026-03-19T06:49:57Z 2016-06-06T00:00:00Z

A Pitch-Controlled Tremolo Stomp Box Love, James

2016-06-06T00:00:00Z Smith, Trenton https://hdl.handle.net/2123/15028 2026-03-10T00:18:33Z 2016-06-06T00:00:00Z

Live Multiband Speaker Delay Compensation. Smith, Trenton live multiband speaker delay compensation using Generalised Cross-Correlation and Phase Transform. Please Note: Uploaded is the .M MATLAB files to run the multi-band delay element of the project along with audio examples of such.

2016-06-06T00:00:00Z Back, Michael https://hdl.handle.net/2123/15024 2026-05-07T00:53:15Z 2016-06-06T00:00:00Z

Multidist - A Distortion Unit with Optional Multiband Functionality Back, Michael Multidist is a new, exciting distortion effects processer developed by Trentone. The Multidist software offers users the freedom and variety that other processors do not. It has optional multiband capabilities, six separate distortion options, graphical user interfaces and it all runs in under 10 seconds! The software is evolving and it is shaping up to become very, very powerful All of the required files have been supplied. A zip with all the files is supplied, as well as individual files

2016-06-06T00:00:00Z Shi, Zhe https://hdl.handle.net/2123/15037 2026-05-07T00:53:11Z 2016-06-06T00:00:00Z

Implement Artificial Reverberation by using filterbank methods Shi, Zhe

2016-06-06T00:00:00Z Cameron Jeffs, Nash https://hdl.handle.net/2123/15025 2026-05-07T00:53:12Z 2016-06-06T00:00:00Z

Stretch Effect: A Sound Design Tool Cameron Jeffs, Nash The "Stretch Effect" is a digital audio signal processing tool used to create interesting and bizarre sound design effects. A complete written review of the system specification, implementation and evaluation has been included as has the MATLAB operation code and system's user interface. Audio examples of the system presets have also been included.

2016-06-06T00:00:00Z Nugroho, Alexander Christian https://hdl.handle.net/2123/15042 2026-05-07T00:53:16Z 2016-06-06T00:00:00Z

Multi-Band Noise Gate Product Proposal Nugroho, Alexander Christian This final written review will focus on the Multi-Band Noise Gate that is developed as a plug-in for a Digital Audio Workstation (DAW). This report will describe the noise problems that many sound engineers face, the technical specification of this product, the implementation of the developed algorithm and the performance evaluation of the product.

2016-06-06T00:00:00Z Li, Changlong https://hdl.handle.net/2123/15031 2026-05-07T00:53:05Z 2016-06-05T00:00:00Z

All-in-one Ring Modulation Application Li, Changlong This final written review introduces the brief history of ring modulation application and the basic modulation principle with a mathematical formula in introduction part. Then two potential problems are proposed, including the problem of the integration of different types of ring modulators and the limitation of selectivity of inputs based on different durations of signals, which is solved by DSP and specified in specification part. By using the demonstrated parameters and audio signals (saxophone and harp1), the whole script is implemented and the relevant exported files have uploaded. Finally, two evaluation methods (physical and performance evaluations) are proposed with diagrams.

2016-06-05T00:00:00Z Perry, David https://hdl.handle.net/2123/15030 2026-05-07T00:53:06Z 2016-06-05T00:00:00Z

Final Review Perry, David

2016-06-05T00:00:00Z Natoli, Daniel John https://hdl.handle.net/2123/15008 2026-02-26T13:27:26Z 2016-06-05T00:00:00Z

distCONV: A Revolutionary Time-Variant, Dynamic Convolution/Distortion Processor! Natoli, Daniel John distCONV is a revolutionary dynamic convolution/distortion processor that allows a user to convolve two audio files (.wav) whilst applying time-augmentation and distortion processing to the convolved signal over time. The distCONV function offers multiple, interchangeable distortion algorithms (from subtle saturation to buzz- saw distortion!), time stretching, impulse reversal and FFT-based convolution in two variable systems. The entire process has been streamlined for computational efficiency, and is controlled through a graphical user interface (GUI), offering complete control over the system’s parameters. All of the appropriate MATLAB files (including test .wav files and convolved examples) have been supplied. The "distCONV - FULL SYSTEM.zip" file contains all of the uploaded files, organised into folders for a single convenient download.

2016-06-05T00:00:00Z Leer, Rachael https://hdl.handle.net/2123/15007 2026-05-07T00:53:03Z 2016-06-04T00:00:00Z

Doppler_Toolbox Leer, Rachael The Doppler Shift is defined as a change in pitch as a result of relative motion between a sound source and a receiver. As the source and receiver become closer together the sound waves arrive at closer intervals, and thus the receiver perceives a higher pitch than the original emitted frequency. As the source moves further away the receiver hears a lower frequency than what was emitted. The relative velocity also produces a change in magnitude (Nave 2012). The Doppler Shift occurs between ANY sound source and receiver when there is relative motion. It is most noticeable when a plane flies overhead or an ambulance drives passed, however the Doppler Shift is always occurring, giving important auditory cues that allow us to so accurately and usually subconsciously, localize sound. This principle forms the basis of many spatial related technologies, such as sonar, radar and acoustic mapping. Thus, it is of the utmost importance for anyone studying engineering, science, audio or acoustics to be aware of, and understand the Doppler Shift. The computer program Matlab is used extensively by university students and professionals in these fields for coding purposes. The software includes many in-built functions for performing calculations, but none of the in-built functions address the doppler shift. I believe the addition of a “Doppler Toolbox” would be a very effective and usable addition to the software for future releases. This submission includes four functions - POLICE_RADAR, DOPPLER_APPROACHING, DOPPLER_RECEDING & DOPPLER_PASSING. They can all be run using the script Doppler_Toolbox_script.

2016-06-04T00:00:00Z