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DSD: Explain it to me because I'm not getting it


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But haven't your PCM files already been filtered?

I can see upsampling first if you want to do additional processing, so that has less effect on the original audio, but I'm not really sure that there are any filtering benefits for files which are already in a PCM format.

 

Yes, it has been decimation filtered already. But for upsampling/oversampling it is in most cases anyway going to happen. It is more about the choice where and how you want that to happen.

 

I can see someone perhaps wanting to upsample if their DAC uses linear-phase upsampling and you want to use minimum-phase upsampling.

 

Or use different kind of linear-phase upsampling, or use different kind of delta-sigma modulator instead of the one inside the DAC...

 

But I feel like most people that prefer DSD like it because it colors the sound.

 

Why do you think similar process inside a $10 DAC chip wouldn't have such effect? Most I've measured are horribly compromised designs because there's just not enough DSP processing power to do things properly.

 

Because DSD is a 1-bit format, there are a lot of compromises to be made so each modulator may have its own type of "sound".

 

So do the modulators inside ADC and DAC chips... For ADC side you have no choice, but at DAC side you may be able to bypass the built-in modulator by using DSD.

 

I hear this argument a lot, and obviously you are very knowledgeable on the subject, but I'd much rather send it a multi-bit input than convert everything to 1-bit first.

 

When you send PCM to the DAC chip, in most cases it will use digital filters to first go to 352.8/384 kHz sampling rate. And then use sample-and-hold (zero-order-hold) oversampling and delta-sigma modualtor to reach 5.6/6.1 MHz D/A conversion rate.

 

When you send DSD to the DAC chip, in best cases no digital processing is applied at all and it goes straight to the D/A conversion stage. Or some minimal digital processing is used (ESS Sabre) before going to the D/A conversion stage.

 

While some people may not consider them to be "multi-bit" DACs, they are not 1-bit DACs either, and do all of their internal processing in a multi-bit format.

 

So does the software that converts PCM to DSD... So yes and no, you can use same "multi-bit" DAC to natively convert DSD to analog. I have demonstrated this with my open DSC1 design.

 

So while you may prevent the DAC from doing any internal upsampling, it's going to go through remodulation which is more likely to have a detrimental effect on the sound, in my opinion.

 

The most detrimental effect on the sound is any SAH/ZOH or linear interpolation oversampling used in a DAC (due to resource shortage), which is most of the DACs.

 

So depends on which DAC you use. If you use for example DAC based TI/BB chips, it is not going through remodulation. Or if you use a DAC based on Cirrus Logic / Wolfson which enables the Direct DSD mode.

 

I've measured pretty nice linearity improvements on ESS Sabre with DSD inputs compared to PCM inputs.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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I've measured pretty nice linearity improvements on ESS Sabre with DSD inputs compared to PCM inputs.

 

My ears agree. :) Miska, would it be possible to publish that measurements?

 

Just now I have also iFi iDSD Nano (TI DSD1793) at home for test, the effect is the same.

 

HQPlayer does digital processing much better than our DAC chips.

i7 11850H + RTX A2000 Win11 HQPlayer ► Topping HS02 ► 2x iFi iSilencer ► SMSL D300 ► DIY headamp DHA1 ► HiFiMan HE-500
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L

Does this effect on the sound stage show it self more with the headphones than with the speakers, or it's even worse with the speakers ?

 

What sort of room treatment do you have ?

 

I'll have to test that out. I've done pretty much all my critical listening on speakers. Room treatments are filled bookshelves, a rug and some heavy furniture. It's a small room (13x13)and it would be intolerably "live" without these items. Definitely not an ideal space, but it's what I've got.

 

Peter's technical stuff went right over my head, but his description of the stretched and smeared sound is exactly what I'm hearing.

 

Gerard

Mac Mini (+Tidal +Roon) -> WiFi -> Lyngdorf TDAI1120 ->JM Reynaud Lucia (Tellurium Q Black v2)

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For proper calculations better use floating point format. Better 64-bit.

 

It allow:

 

1. Decrease quantization noise

 

2. Avoid overloading. We can normalize signal only of last stage.

 

Releasing 32-bit floating point math at budget FPGA or microcontrollers is non-trivial task.

 

Need bigger time of development. Possible using C++ (single chip computer) or VHDL (for FPGA) languages. But it is not make this task more easy.

 

Many troubles with debugging. Here possibly using "hardware debugger" - oscillograph :)

 

When you use PC - no problem. Except CPU's resources consuming of course.

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But I feel like most people that prefer DSD like it because it colors the sound.

 

No. My experience is that DSD leads to more focus to instruments, less smudged details - less sibilance, less sharpness/aggresivity in trebles.

 

More "air" and generally better imaging and more resolving timbre of instruments is coming because of higher detail level. That can be only result of less distortion. Higher level of detail is prerequisite to hear more reverb and thus to get better spatial information about depth and the recording "room" in general. With PCM, especially 44.1k, recordings sound me more flat, it's much harder for me to perceive depth.

 

With PCM you can do a lot of processing without affecting how the file sounds, because it's a multi-bit format.

 

I agree ... I would like to use a crossfeed or binaural processing with HQPlayer. With Foobar it is possible to apply DSP plugins before on the fly PCM to DSD conversion is applied. HQPlayer contains convolution engine also for DSD, so it could be possible, but I don't know how to do it.

i7 11850H + RTX A2000 Win11 HQPlayer ► Topping HS02 ► 2x iFi iSilencer ► SMSL D300 ► DIY headamp DHA1 ► HiFiMan HE-500
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But I feel like most people that prefer DSD like it because it colors the sound.

 

It depend on quality of filtration and sigma delta modulation.

 

Theoretically in audible range no difference between PCM and DSD (as audio stream in file). Both restored via analog filter to uninterrupt analog signal. Both has absolutelly identical spectrum in the range.

 

But in system "audio file" - "player" - "hardware" many variants with different quality.

 

 

With PCM you can do a lot of processing without affecting how the file sounds, because it's a multi-bit format.

 

For right processing also no difference. Bit depth here compensated via sample rate and noise management. A lot processing distortion level absolutelly same to PCM.

AuI ConverteR 48x44 - HD audio converter/optimizer for DAC of high resolution files

ISO, DSF, DFF (1-bit/D64/128/256/512/1024), wav, flac, aiff, alac,  safe CD ripper to PCM/DSF,

Seamless Album Conversion, AIFF, WAV, FLAC, DSF metadata editor, Mac & Windows
Offline conversion save energy and nature

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When you send PCM to the DAC chip, in most cases it will use digital filters to first go to 352.8/384 kHz sampling rate. And then use sample-and-hold (zero-order-hold) oversampling and delta-sigma modualtor to reach 5.6/6.1 MHz D/A conversion rate.
This is news to me.

 

OK, we need to replace USB2 with HDMI, Thunderbolt, or some other high-bandwidth connection and send a multi-bit 5.6MHz signal straight from the PC.

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I mean better solution than any galvanic connection would be to have a new standard based on serious LAN protocol with error detection and correction. It would solve high speeds required for multichannel hires as well as galvanic isolation. I mean to substitute UAC and SPDIF with usual computer communication over LAN. DACs would do digital audio data buffering and then would provide data directly to DAC on a short galvanic connection (via I2S) without any need for USB or S/PDIF transport.

i7 11850H + RTX A2000 Win11 HQPlayer ► Topping HS02 ► 2x iFi iSilencer ► SMSL D300 ► DIY headamp DHA1 ► HiFiMan HE-500
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As I see it, the current trend is to buy a dac that doesn't play your audio files. Instead you must buy a hot rod of a computer to convert your files into a format that this new dac can play. This is supposed to yield better results? How? At the end of the day aren't you just playing back what you started with?

 

This is basically a bunch of guys who couldn't get a job in mainline DSP trying to outguess and diminish the work of those who do have jobs with DAC companies. They sell into the section of the market who believe that engineers are constantly trying to put one across the unsuspecting consumer, and present themselves as 'the people's engineer' by pandering to every fantasy, no matter how extreme.

Mike zerO Romeo Oscar November

http://wakibaki.com

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L

 

I'll have to test that out. I've done pretty much all my critical listening on speakers. Room treatments are filled bookshelves, a rug and some heavy furniture. It's a small room (13x13)and it would be intolerably "live" without these items. Definitely not an ideal space, but it's what I've got.

 

Peter's technical stuff went right over my head, but his description of the stretched and smeared sound is exactly what I'm hearing.

 

Gerard

 

Hi Gerald,

 

I would suggest trying out some absorbers on the walls at the four first reflection points of speakers, it may well help your soundstage problem significantly. I had the experience when I changed to more detailed transparent amps, I had serious sound stage problems which was then cured by additional room treatments. Your Job 225 is one of the most transparent amps out there ;-)

 

Do you have any hard surfaces between your speakers ? ( equipment rack, flat screen TV)

Sound Test, Monaco

Consultant to Sound Galleries Monaco, and Taiko Audio Holland

e-mail [email protected]

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This is basically a bunch of guys who couldn't get a job in mainline DSP trying to outguess and diminish the work of those who do have jobs with DAC companies. They sell into the section of the market who believe that engineers are constantly trying to put one across the unsuspecting consumer, and present themselves as 'the people's engineer' by pandering to every fantasy, no matter how extreme.

 

Wakibaki, looking at your posts in this forum, it would seem that you have not personally heard the effects of "dirty" power on sound quality of audio systems. I therefore suspect that you have not been able to discern the sound quality differences that different modulators and filters can make to sound quality. You are indeed fortunate in not needing all these enhancements, but please do not denigrate the efforts of digital practitioners who have developed algorithms that many audio enthusiasts find enhancing to the perceived sound quality.

 

Until 10 months ago, I was very reluctant to get into computer audio, but then I heard JRMC19 do PCM to DSD128, and I was floored by the improvement. Not every audiophile likes the change, but it would seem a majority who have tried it do.

 

There is a broad spectrum of listening perception, it's uniquely personal, and each persons taste is just as valid and valuable as anyone else's

 

My fantasy is to get my system to sound like a live performance, and I am extremely grateful to those digital practioners who have created hardware and software that allows to to perceive a sound that is getting closer and closer to a live performance.

Sound Test, Monaco

Consultant to Sound Galleries Monaco, and Taiko Audio Holland

e-mail [email protected]

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This is basically a bunch of guys who couldn't get a job in mainline DSP trying to outguess and diminish the work of those who do have jobs with DAC companies. They sell into the section of the market who believe that engineers are constantly trying to put one across the unsuspecting consumer, and present themselves as 'the people's engineer' by pandering to every fantasy, no matter how extreme.

 

I guess you don't know much about me or others, at least you are demonstrating lack of knowledge. I would advise not to make statements about other people whom you don't know.

 

You don't need to go further than DAC datasheets or use some measurement gear to see where those fall short. I also publish measurement data and you are free to make your own measurements too. Plus I published my own discrete delta-sigma DAC design. I don't need to use DAC chips from anybody at all. I know how the stuff works inside out.

 

So how about publishing some objective data of your own to back your argument instead of trying to insult other people, thank you.

 

 

P.S. There are number of restrictions with DAC chips in terms of DSP. For example TDP of the chip, low clock speed, synchronous MCLK plus chip area and manufacturing process. And it is just plain stupid to put large DSP less than a millimeter away from a very noise sensitive analog part.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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It shows. Wakibaki's signature spells MORON. That shows as well.

 

I know how the stuff works inside out.

Forrest:

Win10 i9 9900KS/GTX1060 HQPlayer4>Win10 NAA

DSD>Pavel's DSC2.6>Bent Audio TAP>

Parasound JC1>"Naked" Quad ESL63/Tannoy PS350B subs<100Hz

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There is a broad spectrum of listening perception, it's uniquely personal, and each persons taste is just as valid and valuable as anyone else's

 

Over the years I have never understood statements like this. It is as if mankind development of ears has resulted in different hearing capacities (not withstanding adverse medical conditions). The sound coming from an audio system is the same regardless of who is listening to it. It is either there are it isn't there.

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That is because you have a medical adversity, just like I said in my post. But just because you can't see without your glasses doesn't mean an object isn't there. By contrast, just because someone doesn't need glasses to see doesn't enable them to see something that isn't there. In audio, 'talking heads' generally claim to hear something that isn't there.

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Over the years I have never understood statements like this. It is as if mankind development of ears has resulted in different hearing capacities (not withstanding adverse medical conditions). The sound coming from an audio system is the same regardless of who is listening to it. It is either there are it isn't there.

 

Yep, but how that sound is perceived is very individual. Do you like the same music as your neighbor? Why not? It is exactly the same sound, no?

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Agree, it is the same sound whether I like it or not. Perception doesn't change the sound, however. The musical content is either there or it isn't. If someone's hearing is normal he is going to hear the same sound as anyone else, although he may not prefer it.

 

I disagree, in part. The same sound waves may present themselves at two people's ears, but what they "hear" may be entirely different. I think we disagree in syntax, not in concept.

 

Example - I love Baritone Horns, French Horns, and Trombones. I "hear" these when other people only "hear" strings or strings and a subset of the brass. It isn't that they don't get the same physical stimuli, it is what they are listening for. If that makes sense to you. :)

 

-Paul

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Agree, it is the same sound whether I like it or not. Perception doesn't change the sound, however. The musical content is either there or it isn't. If someone's hearing is normal he is going to hear the same sound as anyone else, although he may not prefer it.

 

From my point of view, 'perception' is the "Philosopher's Stone" and very different from people to people. Some other senses are involved, not hearing only, because it's a complicated brain system, that we could call it our 'hearing system'.

 

Of course could be the 'same sound' if ALL the conditions are the same, but some of this conditions could vary from day to day, like mood. Paul mentioned 'taste' too and I agree.

 

Roch

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Paul, I love to look at beautiful women but don't ask me to describe one. However, they are going to look the same whether or not I can describe them or not. PCM or DSD? If they are engineered correctly and the same binary content is there they are going to same the same.

 

Two different oversampling filters won't produce the same output data and two different delta-sigma modulators won't produce the same output data.

 

Two different D/A conversion stage architectures won't produce same analog output,. and two different analog reconstruction filters won't produce same analog signal.

 

If you measure carefully enough, even two DAC chips of identical make and model won't produce exactly same output signal in same circuit because there are manufacturing tolerances. Two seemingly identical things existing in physical world won't ever be exactly the same. Monte Carlo method is a good way to evaluate worst case boundaries of component tolerances in complex circuits.

 

PCM and DSD won't have same binary content because they are fundamentally different formats.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Paul, I love to look at beautiful women but don't ask me to describe one. However, they are going to look the same whether or not I can describe them or not. PCM or DSD? If they are engineered correctly and the same binary content is there they are going to same the same.

 

Again, yes. In a perfect world. But, I occasionally butt heads with another CA'er over this topic. It is difficult to describe.

 

Consider please, presenting a 16/44.1 signal to a DAC. Most DACs, in the reconstruction phase, are going to upsample/oversample that, hopefully using some kind of interpolating filter as well as sample and hold, to wind up with a very different and much much higher sample rate to convert from digital to analog. Depends upon the specific DAC and DAC chip of course, but some are already converting up to the 5mhz range, obviously not a PCM format.

 

Now, if you take all that processing out of the DACs purview, and present the DAC with a signal at a sample rate and format it is going to convert to anyway, the chances are that it will sound different. Different algorithms, different choices, different output created, different analog sound output. Although that difference can be minor, it can also be fairly large. Add to that the vagaries of each system, where some aspects of the audio are enhanced, some suppressed, and it is even more likely to sound different.

 

In short, yep, you are correct, but only under special conditions. It happens to be the conditions we all wish we had in our systems, but very very rarely accomplish. ;)

 

By the way, describing the beautiful women I know is one of my favorite pastimes. And proves often that two sets of eyes with photons impinging on them, often cause people to see very different things. :)

 

-Paul

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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From my point of view, 'perception' is the "Philosopher's Stone" and very different from people to people. Some other senses are involved, not hearing only, because it's a complicated brain system, that we could call it our 'hearing system'.

 

Of course could be the 'same sound' if ALL the conditions are the same, but some of this conditions could vary from day to day, like mood. Paul mentioned 'taste' too and I agree.

 

Roch

I agree, but there is not only psycoacustic at work. There are two components at work.

We can have exactly the same audio signal and the same equipment to play it through, but it may be perceived differently because the listener is different (and all that is related to him/her both physically and mentally) and because the roomand its associated conditions - I would say - in 90% of the cases (10% is for those who use headphones) is different. The room acts as a filter that masks or enhance certain frequencies in our listening experience.

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