Zareli

I just have an opinion about listening tests in the matter and I didn't expect someone else to persuade me regarding that aspect. I just asked for recommendations about articles on theoretic and/or instrumental measurements regarding higher DSD rates improvement of DSD64 limitations. I'm sorry if I expressed myself in a misleading way.

Although listening tests are helpful sometimes, I believe anything above 48khz/24bit pcm is beyond what humans can detect or differentiate in AB listening tests. About DSD, I belive DSD64 still can get more dynamic resolution but I don't know if DSD128 can provide that extra resolution from a theoric point of view and anyway it's difficult to provide the conditions for a listening test with EFFECTIVE 120db of dynamic range and above (peaks above 120db that can be harmful), so ultimately one can assume that humans can probably detect the benefits of having more than 120 db of dynamic range, but the conditions are practically impossible (a room with <30 db floor noise and very specific and expensive lab components) and it would hurt the listeners anyway. After that considerations it is safe to assume that DSD128 really doesn't offer practical benefits over DSD64 just like 96/24 pcm doesn't offer practical benefits over 48/20 pcm for the end listener and listening tests would be pretty much "useless" to answer the original questions. And if any difference can be heard at all it is probably an artifact rather than a real difference between the formats (you're hearing the downsample processing artifacts instead of the true difference between DSD256 and DSD64, similar to what was hypotethised in the Horus experiment.

Hi community: I've been intrigued by Hi-Res DSD formats lately. I already know DSD's main claim is the extended dynamic range, being around 120db (although somewhere I read on non-music sound tests it could reach 180db potentially). Also everyone seems to agree that PCM 24/48 is technically superior than DSD64 and more efficient (raw uses less than DSD, losslessly comrpessed uses less than DST). I want to know how do higher rate DSD compares to PCM but no one seems to be making that comparison. I read the whole experiment of the Mendelssohn Session and heard the tracks. I heard little if any difference (I can't play native DSD256 anyway) but I was more interested in the discussion than the raw result. So conceptually this is what I "learned": Higher rate DSD allows noise shaping into a higher frequency realm, which allows a higher cutoff, which means extended frequency response. But what about dynamic range ? From my basic knowledge about Sigma-Delta converters, the oversampling (over nyquist frecuency) is what ultimately allows to reduce noise floors and increase dynamic range. I'm inclined to say that DSD128 then would have a higher dynamic range as well as a higher frequency range. If that is so, Then what would be the closest "PCM mathematical equivalent" to DSD128 and DSD256 ? (I know there's no such thing but let's say 20bit/44.1khz or 20bit/88.2khz can more or less transparently represent DSD64 and vice versa). How much dynamic range increase you get by doubling the DSD sampling rate ?

Yeah thanks, I've been trying that although I don't understand it very well. I don't know how you can convert files, so far only been able to play the files. I've only been able to convert through audiogate and teac.

Thanks for the suggestion. I haven't noticed any distortion on the MK2 version when endpoint volume is locked at 0db. Maybe it's because I flashed the latest firmware from the forums and I'm using the latest drivers. Anyway both DACs look similar but they're quite different internally. I usually try to use both endpoint and knob volumes at 0db when on speakers. On headphones I go around -15db knob volume because I don't own any heavy inefficient headphones.

I ran the tool: So this confirms my DAC has a second *hidden* hardware volume control alongside the main one I guess. Thanks for your answer.

I was reading about the Horus/Grimm experiment and how everything ended up being processed by the Signalyst's Sigma-Delta Modulator (SDM) because it was better (more transparent) than the conversion made through merging technologies SDM. I started to wonder if the SDM I've been using was not absolutely obsolete. I currently have set up the SDM type D from the SACD foobar plugin. As it is documented, is an old SDM defined by Phillips. The other one I have available is on the TEAC converter, although I have no idea of what kind of SDM algorithm is it running internally (maybe it's an old one). In this regard I trust newer implementations because there's little reason to even create and implement a new SDM if it's not going to be worth it in terms of transparency and/or computational efficiency. I'm ignorant in this area so I anything you can teach me is highly appreciated, I haven't been able to find trustworthy information in this technical aspects. So what is the currently most transparent SDM algorithm for DSD manipulation in windows ?

I ran this utility to map audio device windows drivers suggested somewhere else: https://blogs.msdn.microsoft.com/matthew_van_eerde/2014/11/20/walking-the-idevicetopology-tree-to-see-audio-driver-settings/ I got this output, not completely sure about this but should it mean the Driver/DAC has another internal digital volume control entirely for windows (or other OS) to control ? I'd assume this processing is happening inside the DAC, alongside the main volume control. The knob Gain control in the dac goes from -49db to 0db in 1db steps (this one is also monitored and controlled through the drivers). The windows Gain control output says -63 to 0 in steps of 0.5db (this one is only controlled and reported by windows). eRender endpoint Name: Speakers (MUSILAND Monitor 02 US mark2 Audio Device) Endpoint ID: {0.0.0.00000000}.{55afa541-560c-4e47-8e2a-2b102fd723b1} 0x10000: {2}.\\?\muaudio#vid_1fc9&pid_6235#8&8e7630e&0#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\topology 0x10001: Speakers 0x20001: Master Mute Mute node: NOT MUTED 0x20000: Speakers Channel 0 volume, -63 dB to 0 dB in steps of 0.5 dB: 0 dB Channel 1 volume, -63 dB to 0 dB in steps of 0.5 dB: 0 dB 0x10000: {2}.\\?\muaudio#vid_1fc9&pid_6235#8&8e7630e&0#{6994ad04-93ef-11d0-a3cc-00a0c9223196}\wave 0x10001: 0x20000: DAC 0x10000:

Yeah, I know you can't control the volume of DoP or Native 1bit DSD streams with the built in PCM algorithms. My DAC is as Musiland Monitor US 02 MK. As I mentioned the DAC's internal volume is controlled by a knob or the drivers and remains completely unaffected by the windows volume setting.

Hello Like many folks I like to bypass DirectSound on W10 for music playback. I preferred to stick to ASIO but it's buggy on every DAC/Soundcard I've ever tried, producing noise when putting load on the CPU and having problems at certain sampling rates so later I decided to use WASAPI whenever it was possible because it's less glitchy in my experience. On Windows 10 (W10) the global volume control is not bypassed by WASAPI, KS or ASIO. Currently Foobar also allows volume control on PCM on ASIO, WASAPI or KS. Only the independent w10 app-mixer is disabled and muted for the rest of the system. It's no big deal since both are supposed to have fp-higher-than-bit-depth gain algorithms so they *shouldn't* compromise quality, still out of OCD I prefer to send the exact source bit by bit stuff to my USB DAC so I was just assuming bitperfect stream was achieved with foobar and windows at full gain (0db) through WASAPI or ASIO. Now this is the real puzzler for me: Windows 10 can still control volume (reduce gain) when either Jriver or Foobar are streaming DoP and it works properly. More bizarre is the fact that Windows 10 can control the global volume of the Native DSD 1bit stream sent through ASIO. Not even foobar2000 or Jriver can control the volume of DSD streams (Native DSD or DoP). Why ? Because PCM algorithms (gain in this case) are inherently incompatible with DSD and dedicated direct DSD algorithms have not been implemented in this players. I don't know any software that has implemented this on-the-fly volume control DSP for DSD streams. I completely doubt that microsoft decided to implement this proprietary algorithms that probably would need to buy licenses for and/or develop themselves for a functionality that ultimately 99.9% of the users don't care about and the other 0.1% that actually know about DSD don't actually need or want. I already checked, W10's global volume is totally independent from the DAC's gain settings. Any ideas on what's going on ? So far this are my guesses: - W10 can handle native DSD streams at least for On-The-Fly Gain control, even though it doesn't support SACD or DSD formats in the built in player (this is not a VAIO). - The DAC's drivers have an algorithm that allows the windows audio mixer to perform DSD processing that's applied before the stream is sent over USB. This is unlikely since the drivers are very barebones, only allow basic monitoring of the DAC sample rate and switching between the outputs. Any ideas on what is going on here ? TL DR Windows 10 can properly control the global volume of a system streaming bitperfect DSD through either Native DSD on ASIO or DoP on WASAPI. I explained why it doesn't make any sense to me.