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Dolby Laboratories

Projects

Rapid acoustic and DSP prototyping of a new conference phone

Being a member of Dolby's audio DSP feature team for our interactive voice products, I was well placed to help objectively and perceptually evaluate acoustic prototypes for new designs.

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I collaborated with Dolby's industrial design, mechanical and electrical engineering teams to concept product architectures. From Sydney, I 3D printed representative acoustic prototypes,  placed transducers and designed the spatial capture beamforming for each device.

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Once the early design phase was complete, I debugged earliest product builds by identifying and solving key issues related to electronic amplifiers, transducer coupling, embedded drivers and the beamforming design.

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Surround sound from arbitrary microphone arrays

Dolby identified a market trend whereby most modern audio endpoints adopting MEMs microphone technology for audio capture. This is because they had a much smaller form factor, are cheap, and often still boasted sufficient signal-to-noise ratios for communications applications.

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Since we were also building our own MEMs endpoints, it made sense for the our team to commit focused effort building a method to bring Dolby's spatial capture technology to this enormous range of devices. This served our immediate product needs but also unlocked new business partnerships with other consumer electronics organisations.

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I led the design of:

- A frequency-domain based optimization routine for producing ambisonic soundfields, with the ability to tune and trade-off white noise gain with spatial fit as a function of frequency

- A python-based acoustic characterisation package that captures the polar response of the transducers in a form that is useful for the optimization.

- Additional optimization methods to improve the ability of the devices to reduce the impact of non-linear coupling and other sources of noise.

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Audio DSP features for embedded, PC and mobile targets

At Dolby, my team was responsible primarily for developing the core audio DSP algorithms behind Dolby Voice - a real-time, low latency voice communication stack for capture and render on mobile, PC, the browser and embedded devices.

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I made significant contribution to the design, implementation and test of a wide range of DSP features. To build Dolby Voice, there was constant attention paid to the core audio capture technology. I impacted the product portfolio by:​

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- Improving the duplex performance of the protocol by making our echo prediction more robust for all frequencies, especially those at low SNR.

- Improving noise suppression and voice gating features for new conferencing devices and new communication modes.

- Improving and testing various machine learning algorithms, including our voice activity detection and typing noise suppression algorithms.

- Improving network resilience through forward error correction (FEC) and jitter buffering.

- Implementing various ITU standard codecs, such as AMR-WB, G729, and Dolby Voice's own standards.

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As a part of DSP feature engineering, I was also involved in the functional testing of all these algorithms, and worked with our consumer electronics groups as they brought the same technology to a wider range of capture applications.

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Factory acceptance test design, yield management and operational handover

 Because I had plenty of early exposure to electroacoustic design, electrical debugging and key performance metrics of the conference endpoints, I was given the task of ensuring our devices were able to be manufactured reliably at scale.

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This involved:

- Being on-site in Guangzhou, China as the earliest devices were being built to inform operational process and devise early factory tests.

- Consistently monitoring the health of devices as they were produced.

- Collaborating with our contract manufacturers to pragmatically design device pass-fail conditions for trading off quality with yield.

- Communicate the implications of manufacturing process changes to management and other business stakeholders.

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