[maty]

Because the soft player prevents    There are a limit with the consumer O.S. to play audio without problems, cuts... the latency.

Some years ago the best sound I got with foobar2000 was 60 - 90 ms. As I was optimizing my Win 10 Pro I got down to 50 ms without being shrill. At 50 ms it sounds perfect in my system from few years ago, a sign that I have done things well.

If your O.S. is not very well optimized you will not like the sound at 50 ms I suspect.

In Linux you have Daphile and others compilations that can work in Real Time. When I fixed that the soft players work in RT in my windows the sound is a little shrill, more detailed by fatigued.

By the way, a very useful tool is Process Lasso. I can not live without it.

https://bitsum.com/

Now that nobody reads us: I have never gotten the superb sound I have with W10 Pro (very optimized to play multimedia) in Linux, it is the/my sad reality.

I do not know all the answers!

https://www.google.com/search?q=Process+Lasso+audio

[maty]

New screen capture to you:

[IMG] http://maty.galeon.com/WP-imagenes/soft/Process-lasso-rules-multimedia.png

Yes, I can not live without Total Commander in all my windows from... Win 3.0 !  The Windows Explorer is a *&%$#

[charmerci]

JRiver defaults to 100ms. I'll try the lower settings too.


I'll try bitsum.


Thanks.

[charmerci]

I've tried JRiver at 5/10/25 ms. - forget it! Don't know what's happening but a lot of digital "screeching"! 


50ms - hard to distinguish the difference between that and 100ms (default). 250ms sounds much more "spacious" - less restricted soundstage than 50ms.


Anyway, just decided to leave it at the 100ms.


500ms keeps the program buffering the music so that nothing plays.

[maty]

50ms - hard to distinguish the difference between that and 100ms (default). 250ms sounds much more "spacious" - less restricted soundstage than 50ms.

=> you have a bottleneck in your O.S. wich I suppose is not modified by you to play multimedia like 99.99% of users -> many audiophiles say that their CD/SACD players sound a lot better than computer files.

It is the usual thing, most just spend and spend to try to improve the sound instead of locating the system's bottlenecks and eliminating them before spending a lot of money.

[charmerci]

=> you have a bottleneck in your O.S. wich I suppose is not modified by you to play multimedia like 99.99% of users

This is correct. But I'm not going to spend hours figuring it out. 

[maty]

If you use ASIO or ASIO4ALL (both are the best choice in front wasapi or... ), it is very good idea try to minimize the buffer size. In my system, the best sound and without cuts is 64 samples.

[IMG] http://maty.galeon.com/WP-imagenes/soft/ASIO4ALL-v213-buffer-size-64-samples.png

Best = more analogue, without "digitis". With 96, 128, 256... the sound is more "digital", more strident, fatigue...

Note: ASIO4ALL v2.14 sounds worse than v2.13 in my system. Always it is the same: trial and error.

[Yun66]

I use both JRiver and HQPlayer. JR is mainly used for managing files and HQP for playing. Sound wise, HQP sounds obviously better than JR in my system. But HQP needs a lot of computing power if up sampling to DSD512.

[maty]

JRMC 23 64 bits to play PCM files from the computer has very good. I listen very good recordings with high DR from analogue masters, why to convert to DSD?

[maty]

http://www.thewelltemperedcomputer.com/KB/DSD.htm

-> http://www.audiodesignguide.com/PureDSD/dsd2pcm.html


PCM test signal 1 kHz

PCM to DSD:

I can't help but wonder if what's happening here is like tube amps and analogue playback (eg. vinyl). Objectively the DSD conversion adds distortion but the anomalies are not perceived as objectionable and in some material, the added noise and imprecision actually makes it sound less "sterile", "clinical", more "real" (conversely being in an anechoic chamber is disturbingly unreal due to the profound silence). It would make sense to me that some people could prefer DSD64 over DSD128 upconversion since DSD64 will give you more of that distortion. Even though the noise is ultrasonic in nature as measured off the DAC, nonlinearities in the playback system like your headphones and speakers (perhaps certain amps as well) could create audible intermodulation. Maybe for certain music, this could be especially beneficial.

If you listen bad recordings like 99% of people maybe the noise of converting PCM -> DSD disguise the poor quality of the recording and be more enjoyable.

But if your PCM files are from very good recordings you do not want to add noise.

[maty]

Audio Myth -"DSD Provides a Direct Stream from A/D to D/A" by John Siau - August 27, 2015

https://benchmarkmedia.com/blogs/application_notes/41262017-audio-myth-dsd-provides-a-direct-stream-from-a-d-to-d-a

Conclusions

In virtually all cases, DSD adds significant processing to the music production chain. DSD absolutely fails to deliver a "direct" path between the A/D and the D/A. Any such claims are marketing spin. DSD offers no advantages over a conventional 24-bit 96 kHz system, and it fails to leverage the massively parallel architecture of modern converters. DSD cannot match the measured performance of a conventional 24-bit 96 kHz system. Conventional PCM systems provide the most direct and transparent signal path between the A/D in the studio and the D/A in the consumer's home. For these reasons, there is no compelling reason to pay extra for a DSD recording if a 96 kHz version is available. If the choice is between a CD and a DSD version, the DSD version may offer some improvement.

Benchmark recognizes that there are many fine high-resolution recordings that are only available in DSD format. For this reason, Benchmark DAC2 converters are designed to directly accept 24-bit PCM or 1-bit DSD without adding any internal format conversions. This versatility makes it easy to play both high-resolution formats to their fullest potential.

Usually I listen 24/96 FLAC from very good recordings with high/very high DR and from analogue masters. I do not need PCM -> DSD.

[maty]

In other forums, hidding to searchers, I have an old thread about "the advantages" of DSD.

If the digital master was created directly in DSD without further processing in the studio then the sound is better.

Question/problem: how many recordings are available with that premise? Very very very few

[Yun66]

YMMV should have been included in my previous post.  The main reason I convert some of my files to DSD is that I like to try different ways to play my favorite music and attempt to get the best sound out of them to my ears.  Some files sound smoother after conversion. Some sound a lttle dull.  I also have a alot of DSD64 files. Some of them sound better when played in 512. I've to say that a lot of times I can't hear differences. Regarding JR vs HQP, I played a few HiRes PCM files with both on the same system. I felt that JR sound a little compressed compared with HQP. Of course, it depends on settings, there are alot of them.  Again YMMV 

[maty]

To play DSD or DSD -> PCM, foobar2000 (with plugins) has better sound than JRMC.

Very few SACD sound better than other formats. It is not a problem of technology but that they do it badly.

Old albums from Rolling Stones, Credence Clearwater Revival and few more in Japanese SACD sound better than CD or vinyl.

Usually, the commercial SACD and DSD files have been reprocessed, they have lowered the DR and have worse sound than the originals.

You spend a lot of money on a good SACD player and it turns out that the records sound worse than their vinyl counterparts!!!!!!!!!!


Loudness war.

https://en.wikipedia.org/wiki/Loudness_war

http://dr.loudness-war.info/


And many HR digital files are from the CD 16/44 and not from the analogue/digital master. A fraud.

Now the same thing happens with the MQA, as fashionable as before the DSD, but it is not the thread. If you listen bad recordings 16/44, maybe after the MQA sound better. And the source, again, usually is not the master!


I finish, if you/we almost always listen to badly recorded music, why spend a lot of money on the music equipment? It is a disaster for the high end industry, now and in the future.

[maty]

Coming back on the added noise, years ago, when I had not alleviated my enormous problems with my dirty mains, when I made a resampler with SoX I liked more the sound with

• Allow aliasing:

As soon as I improved my problems a bit, the sound was better without aliasing, which is what is relevant.

[Pete Schumacher]

How do you recommend the DSD processor be set up?

[maty]

44100, 88200, 176400... only 44100 multiples with SACD, DSD...

A problem: more load the CPU has, greater the noise/jitter generated!

To play a SACD ISO like PCM, I would choose 44100 (bad recording) or 88200 (good recording).

If you have a bad recording with low DR and/or from digital master between 1980-199*, you can try with more big numbers. Adding noise to something already noisy will not be so problematic. Well, if the OS and the soft players are very optimized to play multimedia.

Trial and error. And only 44100 multiples with DSD and SACD.