I wrote a little while ago about the relationship between DSD and PCM – how DSD is a specific implementation of SDM (Sigma-Delta Modulation), and how both ADCs and DACs for PCM are built around SDM engines. I also wrote about the algorithms that convert data between the two formats, and how the conversions are not entirely lossless.
The picture I left you with was that – for all practical
purposes – there is no such thing as “Pure” PCM. Any PCM music data is derived from SDM at
some point in its creation, and has therefore undergone at least one conversion
I want to ramble further on DSD, and whether music stored
and replayed in the DSD format can be any more “Pure”. The problem is the inescapable fact that –
until somebody comes up with a truly significant breakthrough – you cannot
“edit” music in the SDM domain. This has
huge ramifications for the recording industry, where recording, mixing, and
mastering can involve very profound manipulation of the music. Indeed something as simple as volume control
– a fade-out, for example – cannot be done in the SDM format. And the recording industry routinely employs way
more elaborate effects that would make your hair curl (have you noticed how many recording artists have unnaturally curly
To my knowledge, there are only two studio-grade recording
desks out there capable of producing commercial DSD recordings – Sonoma and
Pyramix. Of the two, Sonoma is the
oldest, and least functional. Sonoma
drops the signal out to PCM for fade-in and fade-out, but apart from that
offers no sound manipulation capability.
Pyramix is modern and quite progressive, but it does all its mixing in
“DXD” which is 24-bit 352.8kHz PCM, so all Pyramix DSD is derived from what are
essentially DXD masters.
Is there any such thing as a pure DSD recording? Well, yes, there is, but you have to restrict
yourself to transcriptions of analog tape, where no further audio processing is
required. To be fair, there is a fair
amount of archival material out there which could benefit greatly from
transcription to DSD for re-release.
But, for new recordings, you would need to record to analog tape and mix
on an analog deck if you wanted to create true DSD recordings. Some boutique studios do follow this
And as far as it goes, that sounds all well and good. But then I came across an interesting
paragraph in a 10-year old technical paper from Philips in Holland (who,
together with Sony, were the driving force behind SACD). Here they talk about the typical “DAC”
configuration used in a SACD player, and I was rather surprised to read
it. According to this paper, the
low-pass analog filters that are required to convert pure DSD to analog do not
possess the impulse response characteristics they consider to be necessary for
high-end audio performance. However,
digital low-pass filters are more than up to the task. Therefore, the first thing the DAC does is to
use a digital low-pass filter to convert the DSD to a 2.822MHz multi-bit PCM
signal. This PCM signal is then fed into
a SDM to generate a multi-bit (typically between 3 and 5 bits) SDM signal at
5.6MHz or even 11.3MHz. This multi-bit
SDM can finally be passed through a low-pass analog filter without having to
sacrifice the impulse response characteristic.
So, who’d-a thunk it? DSD gets converted to PCM and back again in
the DAC of a SACD player! It would be
interesting to find out whether modern DSD DACs utilize a similar
approach. If so, then arguably, as well
as there being no such thing as “Pure” PCM, there could be no such thing as
“Pure” DSD either!