I ordered the DirectStream about a week ago and had a few questions about the volume control. I plan on using the DS as a DAC only.
Thanks
The DS has more headroom than typical DACs (i.e., more bits that can be thrown away without audible effect) and you should not lose resolution if the volume control is not set to 100. As for the gain, that depends on your system. Set it so that the gain best suits your amp (if connected directly) or preamp (e.g., so the usable preamp volume range is somewhere in the middle). The volume control is digital but there is no PCM conversion.
asiano said I ordered the DirectStream about a week ago and had a few questions about the volume control. I plan on using the DS as a DAC only.
- Should I set the Volume to 100 to avoid any loss of resolution?
- Do I set the Gain to High or Low?
- Is the volume control analogue or digital? I've read that if the volume control is digital the signal must be converted to PCM.
Thanks
asiano saidOne cannot change volume on a 1-bit data stream, be it PCM or DSD/PDM. It is the 1-bit format which is the issue, not PDM or PCM.
- Is the volume control analogue or digital? I've read that if the volume control is digital the signal must be converted to PCM.
The DirectStream converts incoming data to 28.22MHz at 30 bits (calling this 10x DSD is a bit confusing; the sampling rate is 10 times that of standard DSD, but referring to DSD implies a 1-bit architecture.)
The 30 bit data stream allows for volume changes. After the volume change, the data is down converted and noise shaped to 2x DSD 1-bit and the FPGA output goes to the low pass filter for conversion to analog. The output is transformer coupled (like most tube amps).
As a side note, the DirectStream oversamples 10x DSD rates as this allows integer conversions (no numbers to the right of the decimal point) for both sample rate families of 44.1 base rate and 48 base rate. Integer conversion provides for simpler math and no rounding errors. This is why there is no loss of resolution. It is a very cool design.
Slightly expanding on Elk’s post:
I’ve put a FAQ on the site about worrying about PCM vs DSD: http://www.psaudio.com/ps_faq/converting-dsd-to-pcm-bad/
It’s a matter of definition as to whether the volume is implemented with PCM or DSD.
If the volume control wasn’t in the DS the DS would still have the same architecture and do the same math.
When we say PCM, noised shaped samples at a 2.8224MHz or a 28.224 MHz sampling rate isn’t what we usually think of.
When we say DSD, a 55 bit noise shaped encoding isn’t what we usually think of.
The volume control is done with 55 bits at a 2.8224MHz rate but in the multibit part of the sigma delta remodulator. As I alluded to, that remodulator is in integral part of going from single rate DSD to double rate DSD - a process that some might argue is not an intrinsically PCM process.
Thanks, Ted, for clarifying and correcting.
I almost got it right. 55 bits at a 2.8224MHz is very fun as a concept.
My recollection is the FPGA is 30 bits. Am I misremembering? If not, where is the additional 25 bits introduced? I am probably a bit confused as to the data path.
Thank you once again for putting up with us and so actively participating in the forum.
Elk said I almost got it right. 55 bits at a 2.8224MHz is very fun as a concept.You aren't misremembering - I use a 24 bit volume (with 2^20 being unity) so I need at least 54 bits there (the extra bit is a technical detail.)My recollection is the FPGA is 30 bits. Am I misremembering? If not, where is the additional 25 bits introduced? I am probably a bit confused as to the data path.
One of the advantages of an FPGA is that you can use whatever resources you want when you need them. The width of the samples changes all over the place as needed. For example in a FIR filter the most obvious implementation needs a precision of (width-of-input * width-of-coefficients + log2(number-of-coefficients)) but the result is no more accurate than the input.
Similarly the upsampler uses more the 30 bits internally (at 28.224MHz), and the sigma delta modulator then needs more bits than just the 55 bits to avoid saturation…
Got it.
Thanks!
Am I allowed to resurrect a 6-year old thread? 
The title and the info is completely on point to my question.
OK, here goes. I wired the DirectStream (balanced) direct into an AR Ref 75 (not SE, not that that matters). I was really paranoid of putting full volume into this very beefy amp and destroying my ears and speakers. So I cranked the volume way down to single digits and hit mute. I started up a track (Bridge II), unmuted, and then slooooooowly increased the volume. About nothing happened for the first 30 or 40 increments. I didn’t start hearing something until somewhere in the 20’s or 30’s and the volume hardly went up at all (if it went up at all) until the 50’s somewhere. “Normal” useful volume range is in the 60-85 range for tracks of average loudness.
Just curious, why is the DS volume control so non-linear? It doesn’t have a useful range in a large part of the lower half. It isn’t a usability issue, it has a useful range in the upper end but it seemed weird that the lower end had so little effect on the volume coming out of the speakers.
By the way, this is an amazing combination, it sounds simply wonderful. (Speakers are Thiel CS .5’s, Jim Thiel’s baby speakers). Those things image like the dickens.
It’s actually dB based (1/2 dB per click) which means that the relative change is the same for all steps. In most systems the relative percieved change is quite even from 1 to 100.
Make sure you don’t have the 20dB attenuator engaged on the DS DAC (40 clicks on the volume button, be careful!). If you aren’t familiar with this feature, on the silver remote it is the bottom button of the three between the DAC volume and the Pre Volume buttons. It’s marked “DAC Level”.
I have the black remote, where is it on that one? (I upgraded a PerfectWave to a DirectSttream, that was a fun project). What does “40 clicks on the volume button” mean? Seems like I must have this engaged or otherwise the volume control wouldn’t be so non-linear.
P.S. Hi Ted, thanks for the post.
The volume control on the directstream dac is lossless by design. The only thing that can be affected is signal to loss ratio.
I just looked and I don’t see that button on the old PWT/PWD remote but should should be able to get to it through the touchscreen just turn the volume all the way down before you go looking for it. I/2 db per click so 40 clicks is 20 db for the attenuator.
Hopefully you’ve figured out if the attenuator was engaged, but just in case… Dirk correctly identified the button, I thought about including that information but thought, nah, not necessary. Wrong! With respect to the 40 clicks, as Ted stated each press/click of the volume Button changes the volume 0.5 dB. The attenuator is 20 dB, so engaging it drops the volume The equivalent of 40 clicks, disengaging it raises the volume the equivalent of 40 clicks. The warning is if you have the attenuator engaged (volume dropped 20 dB) and playing music at a reasonable level and then disengage the attenuator the volume will most definitively be unreasonably loud!
With my system, no pre-amp, the DS sr. sounds better without the attenuator and volume level 30-35 compared to vl 70-75 with the attenuator.
