Don’t get me wrong. The Bridge II is not a bad solution, but according to some posts in this forum, it seems that users are getting better results using USB, I2S and even Toslink.
I agree, it is absolutely necessary to place an optical break in the Ethernet feed. It’s all about cutting out your noisy neighbors that sit in your home network (Server, NAS, IP-TV Box as well as the Ethernet Switch itself).
Ethernet seems to be such a big problem, that @tedsmith does not want to deal with it on his future reference system, the TSS DAC.
This is my own speculation, but I believe that there are the following reasons for leaving Bridge functions out of the future TSS DAC:
SQ. Get rid of every component, that could introduce noise. Ethernet is one of the worst. Keep it as far away from the DAC as possible or when used as service interface, allow it to be disabled.
Complexity. Implementing Bridge functions means dealing with additional software complexity (server/player software, Roon endpoint etc.). Having separate devices per function will also increase flexibility. Simply build the best sounding DAC. No cumbersome additional software.
Control. By focussing on the ONE core competency, you have full control of the product.
Looking at the statements from Ted, it seems that having a rock solid Ethernet solution, that fits audiophile requirements, is absolute key. So you should hand this problem over to people, who do have deep knowledge of the problem and deal with Ethernet based audio streaming for years (experience). The opticalRendu is Sonore’s third generation of audio optimized streaming platforms (not counting the SonicorbiterSE). I really believe that they know exactly what they are doing (because this is ALL they do)… it’s their ONE core competency.
Coming back to placing an optical break in the Ethernet feed: By using Matrix and I2S, this is not necessary the case (and who takes care of the Ethernet implementation?) Also, going optical on the Matrix (Toslink) will limit my possibilities streaming DSD files. I have some nice DSD128 that I really love…
USB on the other hand, gives me the most flexibility here. I did not have the chance to play with the opticalRendu yet, but I do have experience with the microRendu and the ultraRendu. The concept of the opticalRendu makes sense to me. Yes, I agree: You should really use a high quality USB cable to feed your DAC.
Aloha Ted - could you please clarify this for me. If with HQPlayer I upsample both PCM and DSD to DSD 128, so that what my DSJ receives in DSD, does doing this in anyway bypass or negate the DSD upsampling done by the DSJ? Does the DSJ say “oh, this is already DSD, there’s nothing for me to do?” Or does everything that enters the DSJ or DS get treated by your (Ted’s) DSD upsampling in inherent to you DAC creations?
I ask this because I frequently read about HQPlayer that it relieves the DAC from having to do the upsampling, relying instead on a more powerful computer (Mac of PC). Thus wondering if that’s the case?
I wouldn’t want to bypass the upsampling benefits you’ve built into your DACs.
There are two types of upsampling in the DS: PCM is upsampled to 352.8k or 384k and 352.8k PCM, 384k PCM and DSD are upsampled to 56.448MHz. No work is saved by pre-upsampling, the upsamplers are always running. FPGA’s just don’t work like the processors people are used to. Everything is running all of the time and you just choose the answers you want… And a general purpose processor isn’t more powerful (at upsampling) than the FPGA.
But ignoring the theoretical, upsampling to 352.8k externally and then using the DS for further upsampling is a fine thing to do if you like the sound you get or upsampling to DSD externally is a fine thing to do if you like the sound. All that really matters is if you like what you are hearing.
Upsampling PCM to any rate lower than 352.8k externally probably isn’t optimal for sound, but there’s nothing wrong with trying it and seeing what you think.
I am not a Ted, obviously, but I have been playing around with HQPlayer quite a bit. I have found that, in my setup, upsampling all PCM to 352.8 sounds better using the xtr filter and NS5 noise shaper versus any DSD filter.
Cool, thanks for the tip. Since the Matrix arrived, I’ve been preferring to take everything to DSD in HQP before sending it over to Ted and the DSJ. Closed Form 16M has been my latest landing point…
Yes, I’ve been listening to the same filter this evening, poly-sinc-long-LP (2s) and it has a nice diffused, full sound. I really like it. Large soundstage as well. I like it better than Closed-Form-Fast which was my previous big soundstage champ…
I tried the poly-sinc-ext2 and poly-sinc-long-lp-2s filters back to back and prefer the former. The music has a bit more fullness and depth with the poly-sinc-ext2 filter. The soundstage might be a tad more natural in presentation too. You should give it a try.
But the big difference here for me is the ASDM7EC modulator. This modulator got rid of a little bit more of that “digital edge”.
Thanks for the tips. Honestly, both filters sound quite good to me. But the ASDM7EC modulator truly does elevate things to levels I’ve yet to experience. Garcia/Grisman DSD is my go to test album, sounds incredible with the ASM7EC. Surprised my old MacBook could pull it off, but it’s chugging through it just fine. Are you still upsampling all PCM to DSD?
Sinc-L is my new (current) mistress. It’s non-apodizing though so not the best choice for poorly recorded/mastered content. Ext2 is better for helping that 80s glare.
I only upsample to 192kHz and hear the value vs not. (192 is I2S chip limitation)