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DSD Offshoot Discussion From MQA Topic


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For interest's sake I tried putting a WAV track through a PCM -> DSD64 -> PCM conversion path ... hmmm, quite major differences at only 60dB or so down; so, are the 'distortions' intrinsic to that conversion process, or is the software - I used XiSRC - not good enough? Worth trying some variations to that processing chain ...

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The basic process of DSD is trivial - it's the noise shaping where the real meat is, and that's buried in maths normally. Visually, it makes sense why it's needed - with the right "pictures" - something to track down.

 

If one can't get an instinctive understanding of what's happening, then maths does absolutely nothing to clarify the matter.

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I'm a visual sort of the guy, and I found this page to be good as a way of getting a handle on things - http://digitalsoundandmusic.com/5-3-7-the-mathematics-of-dithering-and-noise-shaping/.

 

Yes, it's talking about translating to 4 bits representation, but the principle for 1 bit representation is the same - just think of the digital coding changing state more rapidly with noise shaping, so that the average density can match the actual waveform more precisely at the frequencies that count, that of the audio signal - at the expense of the digital encoding being 'noisier', at frequencies "that don't matter".

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2 hours ago, mansr said:

The glaring error is the claim that DSD is some kind of PDM signal. It isn't.

 

Isn't it about time you corrected the Wikipedia article, https://en.wikipedia.org/wiki/Direct_Stream_Digital,

 

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DSD uses pulse-density modulation encoding - a technology to store audio signals on digital storage media which are used for the SACD. The signal is stored as delta-sigma modulated digital audio, a sequence of single-bit values at a sampling rate of 2.8224 MHz (64 times the CD audio sampling rate of 44.1 kHz, but only at 1⁄32768 of its 16-bit resolution). Noise shaping occurs by use of the 64-times oversampled signal to reduce noise and distortion caused by the inaccuracy of quantization of the audio signal to a single bit. Therefore, it is a topic of discussion whether it is possible to eliminate distortion in one-bit delta-sigma conversion

 

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What's in a name? From https://en.wikipedia.org/wiki/Pulse-density_modulation,

 

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In a PDM signal, specific amplitude values are not encoded into codewords of pulses of different weight as they would be in pulse-code modulation (PCM); rather, the relative density of the pulses corresponds to the analog signal's amplitude. The output of a 1-bit DAC is the same as the PDM encoding of the signal.

 

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However one wants to view the mechanism, and after all the mathematics have been thoroughly digested - what actually counts as regards subjective SQ is the quality of the implementation of the particular method; one tiny poor decision, or a weakness or senstivity to interference effects in some area will be enough to bring down the whole shebang - the circuit never delivers the sound quality that theoretically it's capable of.

 

My attitude is that every technique can deliver "good enough sound" - just get the actual bits on the board to work as they should, assessed by measuring, and then listening; the real arguments should be about how to optimise particular converter approaches to ensure that they produce good sound.

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7 hours ago, Jud said:

 

Yes, that's what I meant. A lot of people are fans of filtering that doesn't cut much, including various NOS DACs, the three-letter word that spawned a thread that spawned this one, and the PONO and Ayre filters.  Does this presumably "warmer," more "organic" sound (at least for some people) provide an appeal that the producer and artist intend as part of the work?

 

"Warmer," more "organic" is actually the true nature of recordings, of all types - one has to deliberately distort at the recording stage to get a cold, "non-organic" feel to a sound.

 

The typical use of heavy digital processing, and inadequate filtering circuitry in the playback area injects artifacts in the sound, which interfere with the 'naturalness' of what one hears. One workaround is the use of NOS DACs, which may enable one to get closer to the true nature of the recording - at the expense of introducing other issues.

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