Originally from St. John’s, Newfoundland, Geoff Martin completed his Bachelor of Music in pipe organ at Memorial University of Newfoundland in 1990. He is a graduate of McGill University’s Masters of Music in Sound Recording and, in 2001, he completed his doctoral research in which he developed a method of using digital signal processing to simulate reflections from quadratic residue diffusers for multichannel acoustic environments.
Following completion of his doctorate, Geoff was a Faculty Lecturer for McGill’s Music Technology area, where he taught courses in new media, electronics, and electroacoustics. In addition, he was a member of the development team for McGill’s new Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT). He taught electroacoustic music composition and conducted the contemporary music ensemble at the University of Ottawa. He has also been a regular member of the visiting faculty in the Music and Sound Department at the Banff Centre for the Arts in Alberta, Canada.
Geoff was a member of the Audio Engineering Society (AES) from 1990 to 2013 and served on the executive for the Montreal Student Chapter for two years. He has published and presented numerous papers on microphone techniques, multichannel audio and virtual acoustics. In addition, he has been an invited presenter at many AES conferences, speaking on subjects including recording studio practices, signal processing and automotive audio. Geoff served as the Papers Chair for the 24th International Conference of the AES titled “Multichannel Audio: The New Reality” held at The Banff Centre. He was the chair of the Technical Committee on Microphones and Applications and the Chair of the Conference Policy Committee. He was also a member of the review board for the Journal of the AES.
He moved to Denmark in the Fall of 2002 to begin work in the research and development area of the Automotive Audio division of Bang & Olufsen a/s. He is currently the Tonmeister and Senior Technology Specialist in Sound Design in B&O’s acoustics department, and has sat on the company’s Board of Directors as an Employee-Elected Representative.
Since starting at B&O he has been the sound designer or assistant sound designer for the following products:
- BeoLab 90, 50, 28, 20, 19, 18, 17, 15/16, 14, 12-x, 11, 8
- BeoVision 11-x, 12, 14, Avant & Avant NG series, Horizon
- BeoPlay V1, A3, A9
- Advanced Sound System for the Audi A8 (Gen. 1)
- Sound System for the Audi R8 (Gen. 1)
- BeoSound 1, 2, 8, 35, Theatre
- BeoLit 12
- BeoPlay earphones and headphones: H3, H4, H5, H6, H7, H8, H9, E8, H95
He is the inventor of the Adaptive Sound Technology (AST) system in the BeoSystem 3 and the True Image upmixing algorithm in Bang & Olufsen televisions, as well as the signal flow designer for the audio processing in the BeoVision 11, 14, & Avant series. He also wrote the Technical Sound Guide advanced user manuals for the BeoVision televisions and the BeoLab 90 & 50, available on the B&O website.
In addition, Geoff is also continually writing a textbook on sound recording which is available (by making a donation to a charity of your choice) at his website at www.tonmeister.ca.
In his spare time, he makes bespoke fountain pens, restores vintage pens, and repairs antique pocket watches.
Some additional and more detailed information can be found here:
External Articles
Bang & Olufsen’s Geoff Martin Pursues the Perfect Sound
Brian says:
Hey Geoff, though I’ve departed professional audio (in leaps and bounds – McGill is a distant memory) I still enjoy visiting your pages. Good work! Keep it up.
Mauro says:
hi geoff, i want subscribe to newsletter of your blog, so i will have real time notification about new articles on it.
is it possible ?
thank you
geoff says:
Hi Mauro,
I’ve just added the option to do this. On the bottom of the right-hand side of the page, you should now see a section titled “Subscribe to Blog via Email”.
Cheers
-geoff
Mauro says:
thak you, but it regards the entire blog ?
geoff says:
I think so.
Mauro says:
ok, so we will see :)
Mauro says:
last question : to see all your post from the start to the and what can i do ?
Mauro says:
ok, all rigth, thanks for all
geoff says:
hi Mauro,
Sorry, I don’t understand this question. The easiest way to see all the postings is to go to http://www.tonmeister.ca/wordpress/ and start going backwards in time using the numbered links at the bottom of the page. However, this means that you’ll have to go through EVERYTHING – a lot of which is nonsense… :-)
Cheers
-geoff
Mauro says:
yes geoff, all right, like you write.
bye
Mauro says:
hi geoff, i confirm that subscription work well.
bye
Gary Eickmeier says:
Are you the same Geoff who posts in the rec.audio.pro newsgroup? Also, what is this list of test tracks that is posted here – a B&O disc that is available?
geoff says:
Hi Gary,
I used to post in rec.audio.pro – but I don’t think that I’ve sent anything to them for at least 8 years or so.
The list is just a list – there is no disc. I update it occasionally, but only on a Tidal playlist….
Cheers
-geoff
geoff says:
Hi again Gary,
I just did a Google search for “rec.audio.pro geoff” to see if I recognised anything.
I found lots of relatively recent posts from someone named geoff – but that person is definitely not me.
Cheers
-geoff
Mark Edmiston says:
Hi Geoff,
I just stumbled on your blog – American here into high end for ages, but conversely hate piles of equipment. I currently have Sound Lab electrostats, but am intrigued by powered/dsp driven speakers. Have had Meridian in the past – loved the aesthetics but underwhelmed by the sound. I am a multichannel music guy, and happy to see you address this. I have a number of questions about processing, DSD, etc. Would you be willing to explore this with me? I have an integrated music/video system.
geoff says:
Hi Mark,
I’d be happy to try to answer any questions you have – as long as the answers don’t violate any contractual/secrecy issues I have with my employer. :-)
Cheers
-geoff
John Gouk says:
Hi Geoff, totally loving the Digital Audio distortion articles. I’ve constructed some active speakers using a commercial DSP unit, which I’m feeding with analogue at the moment, but I could modify it to use a digital input, which will at some point require sample rate conversion given the range of source material. The device has a rate converter chip fitted, so I’m just wondering if you have any experience with the approach and thus quality of those generally or specifically e.g. given a specific chip, are there different implementations that would make a difference? It would be easy to save effort and justify leaving things as D->A->D->A rather than D->D->A if it is going to be “better” i.e. glitch-free. I’m not sure I could hear the difference of removing the step anyway… although I could do it and fix up a switch for an A-B comparison… Cheers, thanks
geoff says:
Hi John,
Thanks! The best answer to your question is one-half “I don’t know” and one-half “it depends”. I’ll take the second of these without encroaching on the first…
At the simplest level, this boils down to the performance of the ADC vs the Sampling Rate Converter (SRC) in your loudspeaker. I would at least start by looking at the frequency response performance and the signal-to-noise specifications in the data sheets for the two chips.
Generally speaking, the “weakest links” in such a system are the ADC and a poorly-made SRC. However, the only way to find out the answer to your question is to look at the data sheets, and to do some measurements.
Of course, the SNR of the ADC is heavily dependent on the level of the signal entering it. For example, if you build your system to be capable of accepting a 2.0 V RMS input, and your source can only deliver 1.0 V RMS maximum, then you’ve lost 6 dB on your signal-to-noise.
There are also potential issues with the implementation of the analogue signal ahead of the ADC. For example, I have, in the past, purchased one DSP unit that had XLR outputs. If you connect these to a balanced input, then the performance of the system will be acceptable. However., if you just connect pins 1 and 2 of the XLR output to an unbalanced device, there’s a significant amount of hum/noise in the signal. This is the result of noise on the ground plane of the device, and unfortunately, there’s no way around this without some re-engineering of the box…
I hope that this helps a little…
Cheers
-geoff