I had been hoping to get the wedding blog post series done by now, but as usual life/work gets in the way... time for an audio intermission... !
Some of the viewers of this blog may be aware that when not at my full-time job, I spend rather a lot of time working on audio... way back in, crikey... must be 2001, I was commissioned to develop the software for the Zero One Ti48. This was one heck of a way to do my first commercial product, and is where I really cut my teeth on doing practical, high quality audio DSP, after my dalliance with digital crossovers back at university.
In some ways the Ti48 was way ahead of its time. It allowed you to rip CDs to an internal hard drive back before the concept of a music server had materialised in general use. While it may have been based on a PC architecture, and it was criticised in some quarters because of that, it was to misunderstand the work that had gone into the concept and how the quality was far beyond what a typical "PC player" could achieve and could offer truly "high-end" sound quality, through a combination of the right hardware and software.
One thing that was particularly unusual about the Ti48 as an audio transport was its ability to play up to 192kHz material. The only problem was that back in 2002, there wasn't any 192kHz material to play! Audio A/D converters capable of doing 192K back then were rare, and probably custom designed, or re-purposed from another intended use.
96kHz capability had been around for much longer, probably hitting the mainstream back in 1993 with the Pioneer D-05... I do remember when this came out, and it seemed very exciting to be able to cover well beyond the hearing range to allow for improved digital processing and avoid the hairiness in the top octave - the reviewer marvelled at how much more natural the tape hiss sounded... It took a lot longer to make it to other recording equipment, though...
While it was very cheap to make an existing 48kHz delta sigma A/D do 96kHz - you just do less decimation at the end... this wasn't really optimal for performance as you ended up with a lot of shaping noise where your new octave was meant to be. It really required a redesign of the modulators and in some cases faster bit clocks to achieve a "true" 96kHz performance, but the potential was there.
This came in very useful for the advent of DVD... you may ask why? Because DVD was the first "HiRes" digital format in wide public consumption... the story goes that the chaps at Pioneer managed to sneak in 24-bit 96kHz support to the official DVD specification... given their previous form with early 96kHz products, this makes a lot of sense - they felt it was beneficial, and having the main delivery format for films supporting it would put a huge number of players out there. Very wise.
Players were not forced to play 96kHz directly as I recall (they were allowed to downsample to 48K) but all must be able to play a 24/96 disc.
I got my first DVD drive in perhaps 1998 or so from Creative... it was bundled with a big Dxr2 MPEG2 decoder card, as most PCs of the time were too weak to be able to decode smoothly by their own. Standalone DVD players were still fairly expensive at this time, so adding one to a PC was a reasonable solution for DVD watching.
At the time, I wasn't aware of the 24/96 capability - I was mostly buying it to watch films in a quality never encountered before at home... but some people were looking into what was possible...
One in particular was David Chesky... Chesky Records (along with Classic Records too) put out some of the earliest 24/96 DVD-Vs... these basically consisted of a static video frame which was then followed with pure 24/96 audio... DVD couldn't guarantee the bandwidth to offer more than 24/96 stereo in PCM, but this was a massive step up technically from what was available in the past.
Playing one of these discs on a computer used to be a proper pain in the backside. What I ended up doing was extracting the raw data from the VOB files and then running a bit rearranger as the samples were packed into a strange order - this was worked out through trial and error on my part! Then I had a normal 24/96 WAV file... while I had been able to record 24/96 since 1999, this was my first opportunity to see what a professionally recorded hi res recording looked like... indeed, on the FFT there was life above 22K after all!
While most large diaphragm microphones struggle to remain flat, there is still plenty of energy going up there particularly for impulsive/percussive sounds, and while we may not be able hear these through our ears very well (bone conduction is another matter), humans are remarkable at hearing inter-channel differences... so it seems worthy to try moving up to a higher sampling rate from a delivery point of view.
What's the catch? Well, you need a lot more storage, and you make the jitter problem worse. Combined with the requirement to optimise a converters' characteristics for the higher rate, this means a converter may well sound better at a lower sampling rate. An interesting test of this is to downsample high resolution audio... I developed a program called FinalCD to do just that. It is a clunky, old-school command line program but is fairly well regarded in terms of its sound quality.
Certainly, I designed the sharp filter to capture as much as possible of the original 96kHz signal into the 44.1kHz sampling rate limitation of Compact Disc. While it would be possible to go more precise still, it is really pushing close to the limit of what can be crammed on there and is technically close to perfect. Many years ago, perhaps around 2004, I used FinalCD to compare a 24/96 recording to a 44.1 downconversion of the same material. In the same player, the 44.1 sounded better... in a different player with completely different transport/DAC architecture? Same result. The 44.1 just sounded more musical.
This didn't make any sense at the time, but as mentioned above, this is not hugely surprising when everything is taken into account... running a D/A at a lower sampling rate increases the tolerance to jitter for reproducing the waveform correctly. You are trading the ability to time the signal transitions correctly effectively against the settling time or amplitude precision of the D/A... this is precisely why delta-sigma converters suffer so much from jitter, as they need to run much faster to make up for their lack of raw resolution, often only composed of 31 or so elements... less than 5 bits.
In any case, time moves on. Since developing the Discrete DAC many years ago and combining it with custom digital filters and dither running on my Ti48 equivalent, I've been fairly content with the quality of my CD playback, with no big steps for improvement... the limitation seemed to mainly fall on the source. Now I am an advocate of the potential of 16/44.1 and feel that it has been hard done by for many years with some truly terrible recordings and masterings (perhaps done under duress in the latter case), but there was always the nagging feeling that a bit more bandwidth could help if done right...
Aside from the work done on Sunrise, improving my analogue replay massively over the past year has perhaps shown better where CD would ideally be than any high-res recording had done so to date... so it was time to do some more investigation into the possible reasons. To do this, I'd need some more "Hi-Res" material... ideally material I was familiar with and already had on multiple formats - it might help to shed some light...
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