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[jazzilike]

I am using MC20 and my library consist of 90% Wma files and 10% DSD (DSF files and sacd iso). I usually play music by random and found the DSD files always playing at -6dB when comparing to other music formats.

To save from changing the volume track by track, is there a way to baseline all music volume output?

[rossop]

As far as I know you cant apply Volume Levelling on DSD files unless you convert them to another file type. Good luck with that. You can do it but, from an audiophile point of view, you may notice a reduction  in sound quality. Its not for me but it maybe for you.

[Hilton]

If your outputting PCM then you can check this box for the DSD input to increase by 6db when converting to PCM.
Options>Audio>Advanced>Configure Input>DSD input plugin

[ken-tajalli]

If your outputting PCM then you can . . . 

. . . do whatever you want.
Apply volume control, levelling, even DSP.
BUT
it won't be DSD anymore!
In reality most DSD's are edited and mixed at recording studios with PCM equipment,
so don't worry, output to PCM at 24bits, at half the sampling rate (176 for 352KHz and 192 for 384KHz)
You will get 99.9% quality.

[6233638]

Analyze your tracks, disable DSD Bitstreaming, and enable Volume Leveling.

If your outputting PCM then you can check this box for the DSD input to increase by 6db when converting to PCM.
Options>Audio>Advanced>Configure Input>DSD input plugin

https://public.dm2301.livefilestore.com/y2pPpZ10N_7Tu4iPlXgKK3dsEYLMOZIcW_6Co0UkKXy2enLksYcVc8JwcixDVVXVk4NzmtGAStIUcQMbINUBOEirfX82JTr7ldkO477tE7OGW8-7bDB1dlSVVpHumHdtMciFB7bdVSYhUp5kz2ZK4pWpA/DSD.JPG

I really don't advise this, it will push many tracks into clipping.

. . . do whatever you want.
Apply volume control, levelling, even DSP.
BUT
it won't be DSD anymore!
In reality most DSD's are edited and mixed at recording studios with PCM equipment,
so don't worry, output to PCM at 24bits, at half the sampling rate (176 for 352KHz and 192 for 384KHz)
You will get 99.9% quality.

There is nothing magical about DSD, it's just another audio format.  24/96 should outperform a 1-bit DSD signal.
You can then encode it back to DSD if your DAC sounds better with a DSD input, but I only recommend it if your DAC actually does sound better.
PCM sounds better than DSD with my Benchmark DAC2.

[kstuart]

I really don't advise this, it will push many tracks into clipping.

It really depends on your SACDs.

I have ticked it, and so far have not heard any clipping.

I think that high quality mastering like Analogue Productions and MFSL and giant labels like CBS/Sony are not going to have excessive levels on SACDs.

When I scanned through the SACD level thread, it wasn't that many that would clip.

[dtc]

. . . do whatever you want.
Apply volume control, levelling, even DSP.
BUT
it won't be DSD anymore!
In reality most DSD's are edited and mixed at recording studios with PCM equipment,
so don't worry, output to PCM at 24bits, at half the sampling rate (176 for 352KHz and 192 for 384KHz)
You will get 99.9% quality.

When upsampling and downsampling PCM to other sample rate rate PCM, the common advice is to do it as an integer multiple of the source, to maintain as many of the original points as possible. However,  from what I have been able to tell, using an integer multiple when going from DSD to  PCM or when going from PCM to DSD provides no advantage. The whole process is very floating point intensive and no integer multiple of the original data survives the process. So, going between 176.4 KHz or 192 KHz and 2.844 MHz (or 5.688 MHz) which PCM sample rate you choice should have not real effect on the quality of the conversion.

There is an issue with the choice of a high frequency filter when going from DSD to PCM. That process produces a lot of high frequency noise above the normal audible range. By default, MC has a low pass filter set to roll off at 24 KHz@48dB/octave. There are also options for 30 KHz or 50 KHz with 24 dB/octave.  People have different opinions on how much of this high frequency to roll off. The conservative MC default was set since some DACs did not handle the higher frequencies well. Others like less aggressive filters. After looking at the noise in my 2xDSD to 192 KHz PCM conversions, I elected to turn the conversion filter off and instead apply a low pass filter at 60 KHz with a 48 dB rolloff. That seems to cut out the obvious noise, while leaving high frequency detail up to 50+ KHz.  Honestly, I cannot hear the difference, but from the frequency spectrums  it seemed to make sense. Whether or not you can hear the difference is debatable and certainly depends on your DAC and the rest of your system.

As others have said, there is no right answer to DSD versus PCM. So much of it depends on the actual implementation in a particular DAC and the handling of the source material that there will never be a universal consensus as to which sounds better.

[6233638]

It really depends on your SACDs.
I have ticked it, and so far have not heard any clipping.
I think that high quality mastering like Analogue Productions and MFSL and giant labels like CBS/Sony are not going to have excessive levels on SACDs.
When I scanned through the SACD level thread, it wasn't that many that would clip.

Well you can check it yourself by analyzing your SACDs and checking to see whether anything has a Peak Level (R128) value greater than -7.0 dB
It's more than a handful in my current library.
 
Of course, if you're converting to PCM anyway, there's really no reason to avoid using Volume Leveling.

There is an issue with the choice of a high frequency filter when going from DSD to PCM. That process produces a lot of high frequency noise above the normal audible range. By default, MC has a low pass filter set to roll off at 24 KHz@48dB/octave. There are also options for 30 KHz or 50 KHz with 24 dB/octave.  People have different opinions on how much of this high frequency to roll off. The conservative MC default was set since some DACs did not handle the higher frequencies well. Others like less aggressive filters.

To be clear, it is not the conversion from DSD to PCM which introduces this high frequency noise.
The high frequency noise is inherent to the DSD format because it's only 1-bit.

Only the 24kHz 48dB/octave filter actually filters it out.
I've requested that the 30kHz filter be modified to also be 48dB/octave rather than 24dB/octave, since that would also be effective while still keeping the HF noise out, but there doesn't seem to be any interest in making changes to the SACD support in MC.

After looking at the noise in my 2xDSD to 192 KHz PCM conversions, I elected to turn the conversion filter off and instead apply a low pass filter at 60 KHz with a 48 dB rolloff. That seems to cut out the obvious noise, while leaving high frequency detail up to 50+ KHz.  Honestly, I cannot hear the difference, but from the frequency spectrums  it seemed to make sense. Whether or not you can hear the difference is debatable and certainly depends on your DAC and the rest of your system.

Yes, with 2xDSD the filters can be extended to double the frequency, so 60kHz 48dB/octave is a good choice.
That's another issue with the SACD support right now DSD at higher rates uses the same filter as 1xDSD, rather than the frequency being adjusted.

[dtc]

To be clear, it is not the conversion from DSD to PCM which introduces this high frequency noise.
The high frequency noise is inherent to the DSD format because it's only 1-bit.

Agreed. The DSD playback system typically filters out the high frequency noise. When converting to PCM, the filtering needs to be done by the conversion software or the PCM playback software.

Only the 24kHz 48dB/octave filter actually filters it out.
I've requested that the 30kHz filter be modified to also be 48dB/octave rather than 24dB/octave, since that would also be effective while still keeping the HF noise out, but there doesn't seem to be any interest in making changes to the SACD support in MC.

You can use the low pass filter option in the parametric equalizer during conversion or during playback to set both the roll off frequency and the slope of the filter. That gives you the control you are looking for. It is actually easier to do it in DSP Studio than in the DSD plugin and you have more options. But you do have to set it appropriately for 1x or 2x DSD conversions. I set my filter both in the conversion option and in the playback option. That does mean that any 192 KHz PCM tracks that did not come from my DSD may have some content filtered out, but I am not too worried about as I have very few 192 KHz files from outside sources. It would be nice to be able to set the filter based on format or by track.

[6233638]

You can use the low pass filter option in the parametric equalizer during conversion or during playback to set both the roll off frequency and the slope of the filter. That gives you the control you are looking for. It is actually easier to do it in DSP Studio than in the DSD plugin and you have more options. But you do have to set it appropriately for 1x or 2x DSD conversions. I set my filter both in the conversion option and in the playback option. That does mean that any 192 KHz PCM tracks that did not come from my DSD may have some content filtered out, but I am not too worried about as I have very few 192 KHz files from outside sources. It would be nice to be able to set the filter based on format or by track.

I don't do offline conversions of say SACD to FLAC, since I play them on different devices and those require different formats.
There's no easy way to apply different filters to different formats in Media Center right now, which is why I stick to the DSD Input Plug-In Filters - but also why I wish that they would be updated so that they adapt to the input format instead of applying one filter across all DSD formats.
 
But I only have two albums in 2xDSD and I don't think either of them actually have any content above 30kHz or so, so it's not that much of a problem for me anyway.
If I applied a filter via DSP Studio, that would get applied to all audio files and not only DSD files which is problematic, as I have a fair amount of content >48kHz PCM.

[dtc]

If I applied a filter via DSP Studio, that would get applied to all audio files and not only DSD files which is problematic, as I have a fair amount of content >48kHz PCM.

Have you looked at how much content there actually is in your PCM files above 60 KHz? I use SPEC to look at the files, and by 60 KHz it is at  -140 dB, although most of my PCM from outside sources is 96 KHz sample rate. My 192 is my vinyl rips.

I agree that something like implementing the low pass filter from DSP Studio in the DSD plug-in would be a great option. And filtering by sample DSD sample rate would be even better. Seems like it would be pretty easy to do. But, as you said, I do not think DSD is a current priority, which is too bad since MC is already one of the leaders is DSD playback and conversion.

[krustycat]

or , you just could somehow raise 6db the volume of your preamplifier and lower -6db the rest of your music that is not DSD

[6233638]

Have you looked at how much content there actually is in your PCM files above 60 KHz? I use SPEC to look at the files, and by 60 KHz it is at  -140 dB, although most of my PCM from outside sources is 96 KHz sample rate. My 192 is my vinyl rips.

I have a lot of 1xDSD (SACD) albums, and two 2xDSD albums. So I would have to use a 30kHz filter, not 60kHz.
 
Whether there is high frequency content there or not (in good recordings there is, in the majority there is not) and whether or not that high frequency content actually matters is beside the point.
If I'm playing high-res files I don't want to blindly filter out everything above 30kHz.
 
That's why the DSD Plug-In has its own filter section. I just wish the 30kHz filter there was updated to be a 48dB/octave filter since 24dB/octave is ineffective, and that the sample rate at which it is applied increased based on the DSD rate. (2xDSD = 60kHz, 4x = 120kHz etc.)
It has been a long time since I looked over things, but I believe the DSD spec calls for a 50kHz 30dB/octave filter rather than the 50kHz 24dB/octave filter currently available.
I don't recommend filtering at 50kHz at all since that does not filter out all the high frequency noise, but if the 50kHz option is going to be there, I'd prefer it matched the spec.

I agree that something like implementing the low pass filter from DSP Studio in the DSD plug-in would be a great option. And filtering by sample DSD sample rate would be even better. Seems like it would be pretty easy to do. But, as you said, I do not think DSD is a current priority, which is too bad since MC is already one of the leaders is DSD playback and conversion.

There really should not be any need for a custom filter. 30kHz at 48dB/octave is the absolute limit for high frequency extension with 1xDSD. Anything higher than that and you are allowing the high frequency noise through.
And the conservative 24kHz filter is a good option as it is.

[ken-tajalli]

All this talk about High frequency content!
I have a Teac UD-501 DSD capable DAC - with variable filtering for different scenarios, also have a Benchmark DAC 2, and a Leema acoustics Elements DAC ( http://www.leema-acoustics.com/product.html?prid=61 ), which is the king of my DAC's ( yes I prefer it to even the benchmark!). The DAC retails for £1200 (same as DAC 2).
Here is a snap-shot of the handbook, that tells it all.

[kstuart]

I hope that was written by a corporate tech writer rather than a designer, because it has very little to do with the actual response.

When designing a filter, the steeper you make it, the more it affects the audio band.

[6233638]

All this talk about High frequency content!
I have a Teac UD-501 DSD capable DAC - with variable filtering for different scenarios, also have a Benchmark DAC 2, and a Leema acoustics Elements DAC ( http://www.leema-acoustics.com/product.html?prid=61 ), which is the king of my DAC's ( yes I prefer it to even the benchmark!). The DAC retails for £1200 (same as DAC 2).
Here is a snap-shot of the handbook, that tells it all.

Yes, but no DAC expects to be sent an unfiltered DSD audio signal.
There is a ridiculous amount of high frequency noise that dwarfs the signal with DSD if left in.
You can't just ignore it, it must be filtered out.
 
If you bitstream a DSD signal to the DAC, the DAC should filter it out itself, but if you were to convert to PCM without filtering it could potentially damage your amplifier/speakers.
So that filter is very important.
 
Currently the only effective filter option in Media Center is the 24kHz 48dB/octave option.

What I have requested is more aggressive filtering, increasing the 30kHz filter from a 24dB/octave filter to a 48dB/octave one.
This would extend the high frequency response while still rejecting all the high frequency noise inherent to 1xDSD.
The current 24dB/octave is too weak and allows too much high frequency noise to pass through the system.

The 50kHz filter - though I don't believe it should be used at all - would be better updated to a 30dB/octave filter to have an option which matches the SACD spec. 24dB/octave is too weak.

However, with these agressive filters, the filter needs to adapt to the DSD rate.
If you are using the 24kHz filter - the only once which is currently effective - there will be zero difference between 1xDSD and 2/4/8x DSD.
The entire point of these higher rates is that they shift the noise into higher and higher frequencies, so the filter should be adjusted to match.
At 2xDSD a 24kHz filter should be moved to 48kHz, the 30kHz filter should be adjusted to 60kHz etc.
 
Now you could debate whether there is any merit to using those signals at all to begin with, but if you're going to support them - and Media Center does - applying the same filter used with 1xDSD to higher DSD rates is not the right thing to do.
 
I actually think that PCM is a better format than DSD to begin with - after all, if you're having to filter at 30kHz, that means 24-bit 88.2kHz PCM would outperform it instead of DSD's 2.8MHz sample rate. (no noise with PCM since it's 24-bit instead of 1-bit)
But SACD is often the only way to get high resolution or multichannel versions of certain albums, and I would like to see it handled as best as possible even if the format is flawed.

[glynor]

I actually think that PCM is a better format than DSD to begin with - after all, if you're having to filter at 30kHz, that means 24-bit 88.2kHz PCM would outperform it instead of DSD's 2.8MHz sample rate. (no noise with PCM since it's 24-bit instead of 1-bit)
But SACD is often the only way to get high resolution or multichannel versions of certain albums, and I would like to see it handled as best as possible even if the format is flawed.

Thanks. That's the most reasonable description of the filter thing I've ever heard.  And I completely agree on this last bit.

I don't have any SACD content (too much work for my blood), but... I get what you're saying here.

[dtc]

Starting with a 2xDSD track that I created by digitizing the vinyl to PCM I created PCM files with MC using the standard 50 KHz/24dB filter and my custom 60 KHz/48dB filter. Hopefully, these will show the filter issue.  I am sure there are better examples, but this is what I came up quickly.

First, the spectrum with no filter applied.  Noise starts to climb at about 60 KHz, as expected.



Next, with the JRiver DSD Plugin 50 Khz/24dB filter. Noise drops above 50 KHz, but flattens out then starts to rise again. The slope of the filter is just not enough to fully get rid of the noise.



Next, with my custom filter 60 KHz/48dB, set in DSP Studio during conversion. Noise drops and stays low.

[kstuart]

Look for the Way Out West that MFSL did as their first CD in the early 1980s.  They put the master tape on a studio two track, connected two wires to the ADC and mastered the result.

All the later ones with better technology don't sound as good because the master tape has lost magnetism since then.

[dtc]

Thanks. I will look for it. My vinyl is a 1992 Doug Sax remaster for Chad Kassem. Most of the rest of my early Sonny Rollins collection is mono releases from the time. Unfortunately, I missed Way Out West when I started collecting him in the late '60s. I do remember his 2 year sabbatical and bought The Bridge as soon as it came out. My CD is a K2 20 bit version from Contemporary done in 2000. The digitized vinyl is much better. I think I also have an OJC version.

Edit: Now I remember. I did look for that CD once, but at $200 or more, it was more than I wanted to pay. But now I'll keep looking for a used version from someone who does not know what he has.