SACD Player/Transport Choices

I can confirm mahler’s assertion. My OPPO UDP-205 sends DSD via HDMI to my ARCAM AVR30 receiver. ARCAM display shows “DSD” at 2.8 MHz, both for stereo and multichannel. Sounds great!😊

@f1a thanks for the confirm. Appreciated.

@bergenma 

but given the guts of the X110 are inferior to the model you have

To be fair to Raevon, most of the internals of the X110 and X200 are identical.  The X200 has the addition of analogue audio outputs and it looks to me that the power supply has been upgraded to suit analogue, with a well shielded toroidal transformer.

Both boxes have a pretty low component count which should enhance reliability of the electronics.  The MediaTek chips are in the highest classification of integrated circuit densities.

The Magnetar UDP800 is essentially a Raevon X200 without the second Burr Brown DAC, which delivers 7.1 and 5.1 channel analogue over RCA interconnects.  Raevon and Magnetar are a bit like Marantz and Denon - sister companies.

According to Reavon’s website

"DSD is natively transported via HDMI as a fully digital signal up to 5.1 multichanel"

But the website also claims

"The best possible picture and sound technologies for an incredible home cinema experience"

Down-converting SACD to CD quality on analogue playback is not the best possible sound technology but full SACD quality is delivered over HDMI to a suitable DAC or pre-amplifier. No disc player should output full volume noise though

@faustuss

Now that you have educated yourself, you may be able to appreciate that when we say that a transport delivers Direct Stream Digital (DSD) natively over High-Definition Multimedia Interface (HDMI), we really mean it!

DSD

To be absolutely clear, DSD is a brilliantly simple way of encoding sound waves.  Each successive bit indicates whether to notch the digital sound pressure level up or down by one quantum, when compared to the real analogue sound pressure level, Silence is represented by an endless series of (up, down) at Megahertz rates.  Rising sound pressure will have more ups than downs, while falling has more downs than ups.  The analogue to digital conversion should not be more than half a quantum out. 

Moreover the quantisation noise can be removed by passing the bit stream through a gentle low-pass filter rolling off in the MHz region.  In principle the output can be played as an analogue signal without further processing except for a volume control!

Conventionally geeks refer to ups as 1 and downs as 0 in the digital domain.

PCM

By contrast, Pulse Code Modulation (PCM) samples the sound pressure level about 44,000 times per second (using CDs as an example).  At each sample, it measures the sound pressure level on a linear scale from about -32,000 to +32,000.  This range can be encoded into 16 binary bits (0 or 1) where each bit represents twice the level of the previous bit.

PCM cannot encode frequencies higher than half the sampling rate, and the low pass filters needed to remove digital noise must operate close to audible high frequencies.

Binary

You can see how this works by counting in binary using your fingers.  Using only the four fingers on your left hand, you can count from decimal 0 to 15.  The first finger represents 0 or 1, the next 0 or 2, then 0 or 4 etc.  You add up all the fingers to get the decimal equivalent. Keep going with the four fingers on your right hand and you get anywhere from decimal 0 to 255.  (This is the range of one computer byte, and looks like part of a four byte Internet Protocol version 4 address, eg 192.255.1.201)

Add in four toes from each foot, giving you 16 bits, and you can count from 0 to 65535.  Or -32767 to +32768 if one finger represents the - sign, which is handy because sound waves go both up and down from silence (0).  Why didn’t our Neanderthal relatives invent counting this way?

PCM linearity

So what is the issue with PCM?  Well, the most significant bit represents a value 16384 times bigger than the least significant bit.  At some point, when the sound pressure rises by 1, all fifteen lower bits switch off and the biggest one switches on.  It is almost impossible to manufacture any analogue device, like a resistor, to this degree of accuracy.

Imagine feeding every number between -32767 and +32768 in sequence to a Digital to Analogue Converter (DAC). Ideally the output would increase with every sample – known as a monotonic increase.  But resistors cannot be made accurate enough.

Philips knew this from the get-go, and just used the top 14 bits in their early CD players.  It gets even more ridiculous with high resolution at 24 bits, which needs 256 times more accuracy still, or 32 bits which needs 65,536 times the accuracy in order to resolve every bit.

Sigma-Delta

Enter the sigma-delta technique.  If you take the difference between consecutive PCM  levels, you can in theory add or subtract that many quanta to get from one sample to the next. So in effect, you convert the differences between consecutive sample to a local DSD stream.  It is easier to add a number of near-identical charges than to accurately trim a resistor.

It is similarly easy to create a PCM sequence from a DSD stream – just count the net number of ups and downs over 64 bits (for DSD64) and add or subtract from the previous sample.  But you cannot go the other way without guessing where each up or down should fit in time.  You lose information.  This is why a DAC that handles a DSD stream natively is better than a DAC chip that needs DSD to be externally converted to PCM.

Transporting digital formats

Any digital format can be chunked up into packets.  The packets can be any size.

If you want to send say a DSD stream over I2S, just chunk it up into 16-bit chunks and push each chunk into the pipeline.  The receiving end has to know what format is being delivered so it can unpack the chunks into a new stream identical to the original DSD stream.  It is not really over but inside and nothing is lost. 

The network technology can be anything that supports chunks (known as packets).  Internet Protocol, Ethernet, Universal Serial Bus (USB); anything will do if it is error free and fast enough.

On the other hand, if the DSD stream has to be converted to the two-channel 16-bit PCM samples I2S expects, then the timing detail gets lost and most likely the conversion will not be monotonic.

 

No, it does not! Please note the following from my previous post, do with it what you will -

"DSD over USB

In 2012, representatives from many companies and others developed a standard to represent and detect DSD audio within PCM frames; the standard, commonly known as DSD over PCM (DoP), is suitable for other digital links that use PCM. The 1.1 revision added protocol support for higher DSD sample rates without requiring an increase the underlying PCM sample rate."

Native DSD
"The definition of native DSD playback is somewhat a matter of philosophy. Generally speaking, it avoids the conversion of DSD data into multibit PCM anywhere along the reproduction chain. Many commercially available DACs now support native DSD."

"Now that you have educated yourself, you may be able to appreciate that when we say that a transport delivers Direct Stream Digital (DSD) natively over High-Definition Multimedia Interface (HDMI), we really mean it!"