After reading this, I'm left to ponder what the point of DSD is.
Differentiation in a crowded parity market.
After reading this, I'm left to ponder what the point of DSD is.
Wow. Then DSD has many different ways (bit encoding sequences) to encode the same signal. Because many of the permutations mean the same thing (decode to the same analog wave), it has wasted bits....That's pretty much what the result looks like. ...
What you describe above is what I referred to as "1-bit PCM" in my previous post (#138) in this thread. Quite right about that, it has a large quantization error, and will not distinguish between signals that have the same zero crossings (and largest amplitudes). DSD is a lot better than that because of the noise spread followed by noise shaping.If DSD encodes 0 and 1 whenever the amplitude is below or above zero, how does it encode the amplitude of the signal? For example when we change the volume of a waveform, that doesn't change any of its zero crossing points. More generally, I can devise many different signals that all have the same zero crossing points. And if DSD works like that, it seems it would sample them all the same.
I suspect that DSD must work differently. Instead of simply being 0 or 1 when the signal is below or above zero, somehow, the ratio of 1s to 0s over a particular interval indicates the average amplitude over that interval. But this would mean that sometimes it encodes a 0 when the signal is above zero, and a 1 when it's below zero. Put differently, the probability of encoding a 1 or 0 would be proportional to the amplitude of the signal being encoded at that point. I don't know, just trying to make sense of it.
https://www.audiosciencereview.com/...easuring-distortion.10282/page-12#post-334093What you describe above is what I referred to as "1-bit PCM" in my previous post (#138) in this thread. Quite right about that, it has a large quantization error, and will not distinguish between signals that have the same zero crossings (and largest amplitudes). DSD is a lot better than that because of the noise spread followed by noise shaping.
https://www.audiosciencereview.com/...easuring-distortion.10282/page-12#post-334093
If I understand your description correctly the audio files attached above should be able to illustrate it. There are two files in the attachment and the dithered one uses 0.5 bit TPDF dither with very weak noise shaping to make sure all samples are only encoded with two amplitude values.
It's all 1-bit PCM, just with or without noise shaping.What you describe above is what I referred to as "1-bit PCM" in my previous post (#138) in this thread. Quite right about that, it has a large quantization error, and will not distinguish between signals that have the same zero crossings (and largest amplitudes). DSD is a lot better than that because of the noise spread followed by noise shaping.
Confuse audiophiles and make them pay more, apparently.After reading this, I'm left to ponder what the point of DSD is.
If something you want is only available as DSD, that's of course no reason not to buy it. It's not quite that terrible a format. What you shouldn't do is pay extra for DSD when the same music is available cheaper on another format. The most egregious pricing is probably that of Cookie Marenco who asks $15 for CD quality ($40 for a physical disc) and $50 for DSD256.This discussion prompted me to go into discogs, where I've catalogued my music collection, and see how many SACDs I own.
I found that I own 35, but 29 of them have a unique (and to my ears better) mastering not available on any regular CD pressing. And 4 of the remaining 6 have the same base mastering as a CD version, but for whatever reason the CD version is pushed to clipping while the SACD mastering/layer has the full dynamics.
So without realizing it until now, I've only ever bought two SACDs because of the SACD format - the rest were all for the mastering.
Upon reflection it makes sense, as I believe mastering trumps format every time (at least when it comes to digital formats). But still, I was surprised by this particular fact of my music collecting.
It's all 1-bit PCM, just with or without noise shaping.
Wow. Then DSD has many different ways (bit encoding sequences) to encode the same signal. Because many of the permutations mean the same thing (decode to the same analog wave), it has wasted bits.
That makes DSD an inefficient way to encode the signal. This makes data rate comparisons misleading; DSD having the same data rate as PCM (total # of bits / sec) is lower resolution audio.
PCM only wastes 1 bit per sample, the LSB which is randomized. That's only 1/16 or 1/24 of the bits. And it's not really wasted.
Yes, noise shaping improves the dynamic range in part of the spectrum by making it worse elsewhere. That is not unique to DSD.True enough, they are both binary bitstreams that can both be interpreted as 1-bit PCM. However, the 1-bit PCM without DSM and noise reshaping seemingly only retains one bit depth of amplitude information, while the DSD version apparently retains the equivalent of 18 bits depth of amplitude information (the ENOB) in the audio passband, which should allow it to sound a lot better than the version without the DSM and noise reshaping.
It doesn't, not one bit.This thread made me curious about how DSD differed from PCM in mathematical or algorithmic terms.
It's simple. On a spectrum chart, mark the area deemed important. Typically a rectangle covering frequencies below ~20 kHz and levels above -100 dBFS or so, quantisation noise is not allowed within this area. The minimum sample rate required is twice that of the highest frequency in important area. The lowest level determines the maximum bit depth that might be needed. The trivial solution is to use a sample rate a little higher than the bare minimum (for the filter transition band), whatever bit depth gives the desired dynamic range, and flat TPDF dither. Alternatively, a lower bit depth and higher sample rate can be used along with noise shaping to keep from encroaching on the audio area.I do not know, but I hesitate to pass judgement on the wastefulness of DSD until someone with the math skills analyzes this.
It's simple. On a spectrum chart, mark the area deemed important. Typically be a rectangle covering frequencies below ~20 kHz and levels above -100 dBFS or so, quantisation noise is not allowed within this area. The minimum sample rate required is twice that of the highest frequency in important area. The lowest level determines the maximum bit depth that might be needed. The trivial solution is to use a sample rate a little higher than the bare minimum (for the filter transition band), whatever bit depth gives the desired dynamic range, and flat TPDF dither. Alternatively, a lower bit depth and higher sample rate can be used along with noise shaping to keep from encroaching on the audio area.
As for efficiency, the trivial choice is unbeatable since it encodes exactly the information we want and nothing (well, only a little) more. This can be matched by a high-rate, low-resolution encoding only if the noise shaping 100% effective, i.e. capable of shifting some amount of noise to another part of the spectrum with exactly the same area. No known algorithm can do this. It follows that noise-shaped encodings are always somewhat wasteful.
The precise pattern of bit values _is_ the noise shaping. Since it's not possible to completely fill a section of the spectrum with noise, there is indeed some dynamic range available at higher frequencies. In practice, that's not of much use, though, since the spectral content of music drops off as the frequency rises while the noise level increases. It doesn't take long for the noise to completely swamp whatever signal remains.Agreed. Thanks for the neat logic. I did know that the trivial PCM solution is the most efficient solution. DSD will be somewhat wasteful. I was merely hesitating over whether one could jump to the conclusion that MRC01 seemed to have in mind, that for each OSR number of 1-bit values, the ordering of the values is immaterial, and that it is only the proportion of 1s and 0s within that interval that mattered. As I wondered, would the actual ordering of 1-bit values by the noise shaping algorithm add some resolution at higher frequencies though not to the ENOB that it does at lower frequencies? Such resolution would likely above 20 kHz and thus not be of practical interest, but still of mathematical interest. I guess AnalogSteph probably just answered my question.
The precise pattern of bit values _is_ the noise shaping. Since it's not possible to completely fill a section of the spectrum with noise, there is indeed some dynamic range available at higher frequencies. In practice, that's not of much use, though, since the spectral content of music drops off as the frequency rises while the noise level increases. It doesn't take long for the noise to completely swamp whatever signal remains.
To elaborate on that, if you were to split the sample stream into blocks of 64 (assuming that is the oversampling factor) and collect all the 1 bits at the start of each such block, you'd get basically a PWM signal quantised (in time) with 6-bit precision. DSD can, if one is thus inclined, be seen as oversampled PWM quantised to two widths, zero and 100%, with noise shaping. Not that this interpretation is particularly useful. PCM is much more efficient since it gives each bit position within a block a different meaning, thus allowing the various combinations to express a wider range of values.The precise pattern of bit values _is_ the noise shaping.
I thought it was because, at the time it was introduced, the patents on red book CD were running out, and Sony was hoping to launch a new proprietary system that would continue to generate licensing royalties.Confuse audiophiles and make them pay more, apparently.
None of that necessitates DSD.I thought it was because, at the time it was introduced, the patents on red book CD were running out, and Sony was hoping to launch a new proprietary system that would continue to generate licensing royalties.