Originally Posted by CoiL
Actually it is not hard at all to setup. And the sound quality might improve noticeably depending on which DAC/amp being used. Directsound under windows and pulseaudio under linux for example are poor quality imho if You are looking for quality sound and are using dedicated dac/amp since that data/audio goes through OS internal kernel mixer and it affects sound in bad way. For example jumping from directsound to ASIO I had noticeably quality improvement with Aune T1, more detailed, better separation and soundstage etc. But this is only my experience with my rig. Many say they hear no difference. Media player classic should be also quite easy to setup but I don`t get the point if You are already using foobar. What concerns Youtube - I don`t really care about its sound since You can`t get high sample/bps audio from there anyway. For me personally, easiest is using ASIO4ALL. Here is a pic sample how it`s set up: http://cdn.head-fi.org/3/35/35c8322c_AuneT1_ASIOWAOPsettings.jpeg
Will test different plugin options tonight. Was too tired yesterday.
One more thing - my recommendation with Sound Unlocked 1.2.2 is to turn off irr bandwidth (for me seems to make whole spectrum too harsh sounding),
turn on lossless only and use slow roll-off DF.
So far it has been best FW regarding to SQ. Tested with Fidelio X1, Piston v2.1 and modified HD-681 and was comparing with Aune T1 (detailed setup on profile).
MSESE didn`t impress me at all, even stock 1.5.0 sounded better to me and with my setup. All I would like from SU1.2.2 is that gapless mode would work with lossless only but it`s minor thing that doesn`t bother me. When I get bored will try more FW`s but atm seems I`m going to stay on SU1.2.2 - highly recommend it to others! Thanks for that awsome FW Dmitry ;)
I don't care about media player classic or youtube audio quality, I DO care about video and audio sync which apparently needs additional syncing... can I just "tell" wasapi (or whatever is better to use) to just increase/decrease buffer when DX50 is being used for everything not just foobar? it would be much easier to just get that audio half a second earlier so it isn't late any more...
Originally Posted by Yanec
If you use "normal" - you bypass the WASAPI, so you should choose "event"(e.g. pull) which is the best option, or if it is not working correctly - use "push".
I personally prefer ASIO4ALL with "high process priority", "64 bit driver" set in foobar preferences ; and in ASIO4ALL advanced settings window only "allow pull mode (wave rt)" checked with buffer offset = 4ms and ASIO Buffer Size = 72 samples. These buffer settings should work equally good for anything from mp3 to 24/192 flac.
the problem I encountered is that when I use WASAPI (be it event or push) sometimes audio just stops working for youtube and also media player classic (O know right, messing with utput settings in foobar shouldn't be affecting other programs, but appearently, it is...)
also, there is this problem - when using wasapi, I can't set bitrate to 24, as it won't play anything in foobar and if I set it at 16 - it doesn't play anything over 16/44.1 I hgave lot's of stuff that's high bitrate and everything works when I choose just "normal" ibasso mango player" as output in foobar...
Originally Posted by CoiL
Yanec, actually for PLAYBACK only it is recommended to set ASIO buffer size 512 and above. Low buffer and low latency is needed only for recording/mixing purposes. Setting buffer size higher doesn`t affect SQ, only avoids possible jitter/distortion/clicks/pauses because larger payload to/from RAM (like data on queque waiting to be played instead taking it straight and getting possible lag in data stream). I`ve had some few cases where I had to set larger buffer size to play hi-res lossless without problems.
More into topic again. I`ve been doing intensive comparing between irr bandwidth off vs. on and what I`ve found is - with OFF setting soundstage moves further away and has more depth towards front and it`s noticeably larger with more air/decay/reverb. Separation is still good. With irr ON soundstage is much closer and despite everything being clearer and with littlebit better instrument separation I personally find it degrading soundstage size and spaceful imaging. Maybe it is just my taste and fit for Fidelio X1 but I prefer SU1.2.3 with irr bandwith OFF + lossless only + DF filter slow roll-off + DX50 gain set to mid.
If someone bothers to compare, then additional opinion would be appreciated.
Btw, someone should add Dmitry`s Sound Unlocked 1.2.3 and link to first post: https://github.com/dm1try/dx50_sound_unlocked/blob/master/README.md
that's good to know. if I ever manage to make wasapi work with 24 bit stuff for DX50 :D
anyways, I just want to use DX50 as my DAC nonstop, without having to turn to inbuilt PC audio card for videos, because DX50 lags the sound about half a second and watching videos in this manner is quite frustrating...