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[pschelbert]

I tried DSD to PCM conversion, flac format.
While DSD64 stereo and multichannel works fine, I get failed conversion (status) for DSD128 and DSD256 files.
I tried the 2LNO: http://www.2l.no/hires/index.html
 
038 files
Mozart: Violin concerto in D major - Allegro
Marianne Thorsen / TrondheimSolistene

and the

053 files.
Joseph Haydn: String Quartet In D, Op. 76, No. 5 - Finale - Presto
Engegård Quartet

I get a files with zero bytes: Temporary Convert File - 4368.flac 0kB

Should DSD128 and DSD256 conversion work or not yet implemented?

P.S.: The DSD128 and DSD256 files play correctly, so they are okay.

[dtc]

Both of those files are available as 2x DSD (DSD128). They should both convert fine.

In the Convert dialog, make sure you select DSP Studio and under Output Format put 176K (or whatever you desired sample rate is) in both > 768K and 705K.  The >768K line tells it to convert DSD to PCM, although it does not use the actual number, rather is always converts to 1/8th the original sample rate. That converts 2X DSD to 705K, and the 705K entry then tells it convert that output to 176K.  You should have Use Sox for Resampling turned on in Options - Audio.

[pschelbert]

okay, will try that.
You have to save the settings: "save to"

or its just enough if the tick is on in the DSP settings?

the files are available in DSD128 and DS256 (and DSD64).

DSP studio settings for playback work perfectly, never used it for conversion however.

[dtc]

If you delete the box, it saves the setting.

Note you have to use the DSP Studio settings under Convert. They are different than the ones used for normal playback, although there is no way to know that from looking at the box.

I just tried the Britten 4X DSD and it converted fine.

[dtc]

Oops, they remixed the Mozart piece for MQA and now have it as 4x. I have the original mix as 2X.

The Haydn is only 2X.

[pschelbert]

DSD 4x =?

I am aware only of DSD64, DSD128, DSD256

[dtc]

1x = DSD 64

2x = DSD128

4x = DSD256

1X is the original Sony format. 2X is twice that sampling rate. 4X is 4 times that sampling rate.  I find 1x, 2x and 4x easier to deal with than 64, 128, 256.  64, of course, is the multiple of 44.1K to get to the original 2.8 MHz sample rate.

[kr4]

1x = DSD 64

2x = DSD128

4x = DSD246

256

[dtc]

256

Oops. Typo. Corrected.

[pschelbert]

okay, thanks now it fits together.

[dtc]

Did you get the conversions to work?

[pschelbert]

I can try again today in the evening with the configuration you mentioned.

[pschelbert]

Hi

yes your hints helped. Works up to DSD256 (also DSD128, DSD64). The issue was just options not configured. Doing it as you mentioned solved the problem.
Works great now!

Your hints:
In the Convert dialog, make sure you select DSP Studio and under Output Format put 176K (or whatever you desired sample rate is) in both > 768K and 705K.  The >768K line tells it to convert DSD to PCM, although it does not use the actual number, rather is always converts to 1/8th the original sample rate. That converts 2X DSD to 705K, and the 705K entry then tells it convert that output to 176K.  You should have Use Sox for Resampling turned on in Options - Audio.

[dtc]

Great!

Two more thing to consider.

When converting from DSD to PCM, a lot of high frequency noise is generated at very high frequencies. It needs to be filtered out before it gets to the rest of the electronics.  Sony specified a low pass filter starting at 50 KHz with a slope of 48 dB/octave. You can select the filter to be used in Tools - Options - Audio - Advanced - Configure Input Plug-in - DSD.  MC's default filter is 24 KHz with a slope of 48 KHz. This is a very conservative filter. There are also options for 30 KHz and 50 KHz with a slope of 24 KHz/octave. Unfortunately, the slope on those is not steep enough to fully filter out the noise at higher frequencies. If you want to implement the Sony filter, you can turn off the filtering there and set one in DSP Studio - Parametric Equilizer. If you use DSP Studio, you have to set the filter for conversion and separately for playback if you convert from DSD to PCM on the fly.

In addition, the DSD standard is to record at -6 dB compared to PCM. So, in the Input Plug-in DSD options, there is an check box to boost the level by +6dB. Some people use that, although others think it can tend to clipping in some cases. You need to decide for yourself.

Enjoy.

[pschelbert]

okay , I will try that.

Thanks.

[pschelbert]

some background about DSD

cited from:
https://benchmarkmedia.com/blogs/application_notes/41262017-audio-myth-dsd-provides-a-direct-stream-from-a-d-to-d-a
__________________________
....For this reason, DSD is almost exactly equivalent to a 20-bit 96 kHz PCM system. Notice that I said "20-bit" and not "24-bit".

The Near-Death of the 1-bit System
...In the process of aggressively promoting the failing SACD format, many questionable claims were made. Most claims tried to leverage the appeal of the apparent simplicity of the 1-bit system. Other claims highlighted the bandwidth of the 2.8 MHz sampling system while ignoring the fact that a 50 kHz lowpass filter was required at the output of the D/A converter. These claims produced a loyal DSD following. Many SACD releases were excellent recordings, but their quality has nothing to do with any virtues of the DSD format.

The most common production signal paths for creating a DSD release are as follows:
    24-bit PCM A/D > PCM Workstation > 24-bit PCM > 1-bit Noise Shaper > DSD
    1-bit DSD A/D > 1-bit DSD D/A > Analog Console > 1-bit A/D >DSD
    Analog Tape > Analog Mixing > Analog Master > 1-bit A/D > DSD
    1-bit DSD A/D > DSD to DXD Conversion > DXD PCM Mixing > 1-bit Noise Shaper > DSD

...If the audio exists in a multibit format, the quality will not be improved by noise-shaping down to 1-bit. This multibit to 1-bit conversion should be avoided when possible.


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.
________________________________________
The last sentence is the key why if available go PCM.

[dtc]

This is not the place to have such a discussion. Let me simply say, that many people think DSD sounds better than PCM. 

[michael123]

This is not the place to have such a discussion. Let me simply say, that many people think DSD sounds better than PCM.

Yes, but also because DSD is marketed mostly for the audiophiles, which appreciate and score the albums by both music and sound quality. They invest in sound quality, so finding poorly sounding SACD/DSD is rare, but compressed PCM with dynamic range of 4-7 is a commodity