oldkidLG

oldkidLG t1_j6j6zp5 wrote

Reply to comment by Solypsist_27 in Loss-less by TooSmalley

Again, you completely disregard the impact of digital filters on the sound. To mitigate this negative impact, the higher the sampling rate, the better

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oldkidLG t1_j6j2i7o wrote

Reply to comment by No-Bother6856 in Loss-less by TooSmalley

Both DSD and PCM capture the same live stream from the microphone, but DSD playback is more straightforward with far less digital filtered steps involved. I have just mentioned Sony's S-Master technology in another comment, which sends DSD directly to the amplification stage without digital to analog conversion. And the output is music

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oldkidLG t1_j6j0hxl wrote

Reply to comment by No-Bother6856 in Loss-less by TooSmalley

This would only be that simple if capture and reproduction of sound were perfect. In reality, digital filters alter the signal. DSD avoid steep filters and retains the harmonics, whether you think they are audible or not

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oldkidLG t1_j6izcu5 wrote

Reply to comment by Solypsist_27 in Loss-less by TooSmalley

Go check the frequencies produced by real musical instruments. You will see that they by far exceed 20khz. Of course, we cannot hear these, but as they are harmonics, they interact with the audible range of sound and we are perfectly able to notice when they're missing

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oldkidLG t1_j6iyvys wrote

Reply to comment by klogg4 in Loss-less by TooSmalley

>1 bit samples, might you. Which do not replicate sound wave in any way, unlike PCM.

That's wrong. To replicate dynamic range of the analog signal, each sample is encoded to be played back at higher or lower frequency than the one before it. With at least 2.8 million samples per second, this creates a much better capture of the sound than anything PCM

That's funny that you chose ESS as an example, because recent AKM chips, (pre and post factory fire) all include a direct DSD path with a simple low pass filter.

There are also Sony's S-Master class D amp technology that send DSD directly to the amplification stage. That wouldn't be possible if DSD wasn't a faithful representation of the analog signal.

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oldkidLG t1_j6ircu6 wrote

Reply to comment by klogg4 in Loss-less by TooSmalley

>DSD is an archiving format for analog sources, it does not have any other use cases. It does not do anything better than PCM in terms of sound. "Better approximation", "more information" and "less digital processing and filtering" - all of this is complete nonsense.

No, it's not. Take any recent DAC chip schematics, and you will see that the DSD circuit is shorter than the PCM one.

DSD take far more samples per second during recording. Of course, it's going to retain more information. You can't argue that

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oldkidLG t1_j6illgj wrote

Reply to comment by klogg4 in Loss-less by TooSmalley

There is a misconception about the term "lossless" in digital audio. People think that the capture of the real performance is lossless. It isn't, because it is technically not yet possible. Even binaural recordings are a mere approximation of what it is to be there.

However, recording to DSD instead of PCM is a better approximation than PCM, because DSD retains more information and requires less digital processing and filtering.

As a matter of fact, DSD is almost analog. It's a digital continuous stream. Most delta sigma DACs use a multibit bitstream internally that is very comparable to DSD, but something is lost during the unnecessary PCM encoding and decoding steps.

Of course, if the music is digitally produced to begin with, there is no benefit in using DSD. But anything that involves live instruments and/or vocals will sound better if recorded and played back in DSD

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oldkidLG t1_j6ih466 wrote

Reply to comment by klogg4 in Loss-less by TooSmalley

Literally every PCM DAC is compatible with DSD via DoP. Direct recording to DSD or analog to DSD conversion are both vastly superior sonically to PCM. There is literally no point to own an audiophile grade DAC if you never use it to listen to DSD.

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oldkidLG t1_itrnt5q wrote

I have a Focal Elex with fenestrated sheepskins pads, so I think I should reply even if I haven't had the pleasure to listen to Utopia yet. With these pads and if I use something like MorphIt to EQ them with Utopia as a target, I would get very close to the real deal, but they will never match the speed of the beryllium drivers. So, yes technicalities are definitely a think when frequency response is determined by the physical properties of an unusual driver material.

I don't have the time or knowledge to answer for all types of headphones technicalities, but I can speak of slam or punch.

I believe slam is unrelated to the frequency response. It has to do with the volume of air the driver can move at once. If the whole audio chain from file format to DAC, amp then headphones is optimized to maximize dynamics, slam is definitely perceptible.

See this thread

or this post for more details.

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