Talk:Direct Stream Digital

message: Has Direct Stream Digital Video ever been proposed? Photographic images of any kind, including those captured electronicly by TV/Video camaras, etc, are an analog phenomenen, similar to sound.
:Yes. DSD is pulse-density modulation, a generalization of the pulse-width modulation used in Laserdisc. --Damian Yerrick (☎) 01:53, 1 February 2006 (UTC)
thread_title: Untitled

message: This sentence is ambiguous: "The process of creating a DSD signal is conceptually similar to taking a 1-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator, which converts the 1-bit bitstream into multibit PCM." It is also a terrible way to describe the process in an introductory article. It should be described in a simpler fashion, from basic principles, instead of relying on the reader first to understand a more complicated system.  — Preceding unsigned comment added by Oscarruitt (talk • contribs) 06:46, 23 June 2011 (UTC)
replies: text: A better description would be this: "The 1-bit sigma-delta converter keeps a running total of the proceeding single bit DSD stream (the sigma-part), where a zero is -1 and a one is +1, and compares this is to a sample of the input signal (at, say, 24-bit resolution using a analogue-to-digital converter or ADC) (the delta-part) to calculate whether the next bit should be a +1 (if the input is greater than sum) or a -1 (if the input signal is less than the sum). If the sum is the same as the input signal the converter still has to choose +1 or -1, creating noise while "hunting" for the precise value. If the input signal is sampled at 96 kHz, for example, there are 29.4 DSD samples (at 2.8224 MHz for SACD) for every PCM sample." Derekjc (talk) 12:01, 1 June 2013 (UTC)
thread_title: Sentence is confusing; also, a terrible way to describe DSD

message: The first figure, the one showing waveforms for PCM and DSD, is just wrong and horrible and should be removed immediately. It looks like it was drawn by a child, a child who doesn't understand sampling. Someone remove it now.  — Preceding unsigned comment added by Oscarruitt (talk • contribs) 05:46, 23 June 2011 (UTC)
thread_title: Figure is wrong

message: when i run spectral analysis on pcm wav ripped from CDs, many times the frequency rate peaks below 44.1, also consider the generations of recording on dats. have any recordings actually utilized the extremely high frequency possible with this format? i'd like to have moar information on this subject plz. --AlexOvShaolin 01:08, 4 August 2007 (UTC)
replies: text: This is a fairly standard thing - studios often run 20khz low pass filters on their final mixes to avoid any ringing artefacts on CD from frequencies near/over the 22.05khz Nyquist limit of the 44.1khz media, and probably to reduce treble noise on analogue media (don't think I've come across a tape out of any I've analysed that mustered 20khz, regardless of what was prerecorded on it, or what the sales pitch on the shrinkwrap of blank cassette may have said). A number of them do go pretty much all the way up to 22khz, the only suggestion I can make for these differences is the age of the recording - newer equipment will have better quality and largely digital filters that can make a very effective, narrow 'cut', say with the filter output dropping sharply from 0db at 21.5khz to -96dB at 22.0khz, but older mixers will have much looser, cheaper, analogue filters that might need a good couple khz of bandwidth to muster a -60dB drop, and so will start rolling off from around 19khz in order to knock out enough of the high frequencies. Also any discs made from older material will suffer similar problems simply from lower fidelity master material, when it didn't matter much to preserve the very highest frequencies as vinyl couldn't easily represent them even on the first play, and rapidly lost treble fidelity on repeat plays. (I have no citations, but a lot of this knowledge came from wikipedia and pages linked from it - do a search :) 82.46.180.56 (talk) 02:35, 27 December 2007 (UTC)
thread_title: quality

message: :Image:DSDlogo.gif is being used on this article. I notice the image page specifies that the image is being used under fair use but there is no explanation or rationale as to why its use in this Wikipedia article constitutes fair use. In addition to the boilerplate fair use template, you must also write out on the image description page a specific explanation or rationale for why using this image in each article is consistent with fair use.
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BetacommandBot (talk) 06:36, 2 January 2008 (UTC)
thread_title: Fair use rationale for Image:DSDlogo.gif

message: These variants of DSD are of interest and should be added by someone who knows about them much better than me.  —Preceding unsigned comment added by 78.105.11.183 (talk) 00:28, 11 April 2008 (UTC)
replies: text: DSD128 is now added (DSD64 is regular DSD) while DXD has its own article (to which a reference is now added). The Seventh Taylor (talk) 23:10, 9 March 2009 (UTC)
thread_title: DSD64, DSD128, and DXD

message: As original contributor of the DSD-CD article I do not object to merging it with this article as suggested. I even volunteer doing it myself. The main reason for making it  a separate article was to be able to include a link to it in Template:Audio formats but I suppose that can be achieved otherwise too. The Seventh Taylor (talk) 22:27, 30 May 2009 (UTC)
thread_title:  Proposal to merge DSD-CD article with DSD article 

message: I'm fairly sure that the picture explaining the difference between PCM and PDM is incorrect, as it shows the (admittedly similar) PWM signal, not PDM. In PWM the length of the pulse denotes amplitude, as in the image. However in PDM, as the pulses are not contiguous, so it would appear to the casual observer more like higher frequency. Example: In PWM (NOT DSD) we can denote the lowest amplitude as 0000, slightly higher signal amplitude as 0001, then 0011, then 0111 and the highest amplitude as 1111 (not to be confused with binary of course), so a continuous low amplitude would be 0000/0000/0000, and a rising signal would be represented as 0000/0001/0011/0111/1111. Each slash-parenthesis represents one sample. In contrast, PDM, as used in DSD, should be (in ascending order) 0000/0001/0101/1111. Note additionally that there is less information transmitted with a given master clock rate. Can somebody either change or replace the photo? It's great having images, people like me can't figure stuff out without 'em :) Arctic hobo (talk) 21:41, 25 September 2009 (UTC)
thread_title:  DSD/PCM explanation image 

message: About the last link in the article: This time there seems to be a connection with DSD but I still think it's link spam. Does anybody (other than the poster) disagree? The Seventh Taylor (talk) 12:38, 3 March 2010 (UTC)
thread_title:  External links 

message: The first paragraph of the DSD vs. PCM had a long explanation with no citations and at least a few fundamental inaccuracies.(eg: it states that sound is a transverse wave and light is longitudinal, in reality they are reversed)
I'm removing this until it is reviewed, preferably by the original author.  — Preceding unsigned comment added by 2601:A:1700:6700:0:0:0:E96 (talk) 13:59, 16 May 2015 (UTC)
thread_title:  DSD vs. PCM 

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thread_title:  External links modified 

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thread_title:  External links modified 

message: I flagged that section because after several attempts I was simply not able to sift through the audio equipment jargon and abbreviations to understand how DSD works. And I generally understand how audio sampling works. Someone who knows more about DSD would do well to clarify what is being said with less jargon and clearer explanations of the mechanism of DSD. There are links to related concepts but even after looking at those I still don't really know what's being said. For example this bit: 
"The process of creating a DSD signal is conceptually similar to taking a one-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator, which converts the 1-bit bitstream into multi-bit PCM. Instead, the 1-bit signal is recorded directly and, in theory, only requires a lowpass filter to reconstruct the original analog waveform. In reality it is a little more complex, and the analogy is incomplete in that 1-bit sigma-delta converters are these days rather unusual, one reason being that a one-bit signal cannot be dithered properly: most modern sigma-delta converters are multi-bit."
Creating a DSD signal is "conceptually similar" to [jargon jargon jargon], which [jargon] but [qualification with jargon] and also the concept isn't really simmilar because [jargon]. 
Dcb2124 (talk) 16:09, 23 March 2016 (UTC)
thread_title:  Flagged DSD technique section for incomprehensible, abbreviations, and technicality 

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thread_title:  External links modified 

message: The second paragraph states that the "sequence of single-bit values ... [is stored] ... only at  of its 16-bit resolution)."  
This seems incorrrect.  A resolution reduction from 16 bits to 1 bit is 2-16, so the correct reduction should be . If the reduction is from a stereo pair to the single-bit stream (???), this number would be . 
algocu (talk) 14:32, 18 November 2017 (UTC)
replies: text: Thought so as well. But the sentence talks about a "reduction": From 16bits to 1bit is a reduction of 15 bits, which is equivalent to 2^-15. 212.35.11.31 (talk) 12:20, 11 September 2022 (UTC)
thread_title:  Resolution Reduction Calculation is Wrong 

message: I have reverted  lead changes. These are not clear improvements and the edit summary does not help me understand what improvements are intended. ~Kvng (talk) 22:02, 13 January 2023 (UTC)
thread_title:  Lead improvements 

message: DSD64 sampling is NOT 64x that of a CD. It’s DOUBLE. CD is 16-bit/44.1kHz. Multiplied together to get the total number of bps, it’s 705.6kbps PER CHANNEL. Comes out to 1,411.2kbps total. That’s 1.4112MHz. DSD64 is 2.8224MHz which is EXACTLY DOUBLE because it comes to the same 16-bit depth and twice the sampling frequency of 88.2kHz. DSD64 advertising that it samples twice as much as CD is FALSE. This information needs to be updated! 2600:1700:B291:F30:E5EA:53F7:F0A1:FEE8 (talk) 05:51, 26 August 2023 (UTC)
replies: text: The sample rate (the rate at which the audio signal is examined) is 64x. The information rate is 2x. ~Kvng (talk) 14:19, 28 August 2023 (UTC)
thread_title:  DSD64 is NOT 64x a CD 

message: If SACD is DSD64, it cannot have a frequency response of 80-90kHz! It’s the equivalent of 16-bit/88.2kHz. It has to sample a wave TWICE to be able to see a wave. That gives it a maximum frequency response of 44.1kHz. There are MANY errors in this article and it needs to be rewritten. 2600:1700:B291:F30:E5EA:53F7:F0A1:FEE8 (talk) 05:59, 26 August 2023 (UTC)
replies: text: Nyquist–Shannon sampling theorem says it has a maximum frequency response of 1.4112 MHz. Of course noise shaping buries most of that so determining frequency response is somewhat subjective. ~Kvng (talk) 14:25, 28 August 2023 (UTC)
thread_title:  SACD cannot have 80-90kHz response