|
Audio Asylum Thread Printer Get a view of an entire thread on one page |
For Sale Ads |
In Reply to: Re: Do the math... posted by Frank.. on June 14, 2004 at 06:05:30:
.
Follow Ups:
But this paper is a sideline in this discussion.I do recognize that the resulting signal is different from the actual sample value due to the reconstruction filter.
This really means that you simply can't tell much by comparing a graph of a wav that indicates sample values.
That's one off the reasons why I can't dismiss the minimum RMS power and maximum RMS power values. IMO these values ar more telling.
PS. The differences really are minute and could creep in anywhere in the transfer mastering stage and also your soundcard and the used software.
.
Found also another article illustrating the problem.It may be the cause why your samples are clipping.
It can be caused by the dac running out of 'headroom' because the oversampled data from the DF is outside the da converter range or the data is clipped in the DF because the headroom in the DF itself isn't sufficent. (I was referring to the latter)
The analog circuits behind the dac are less likely to cause this trouble. But again this depends on sufficent power rail voltage for the analog circuit. I have seen players with power rails at +/- 8 Volts which leaves not much headroom...
It's possible the clipped waves can be caused in your player.
However your example where the clipped signal level is below a similar signal at a higher level but without clipping contradicts this.What you can do to gain more insight is to record the redbooks spdif out and look if there are signs of clipping similar to the captured wav from the redbook. Look for evidence of clipping below the maximum signal level. If it doesn't clip in the digital output signal then the clipping is caused by your player.
This is a design problem and not a flaw in a format.
(PCM is at an advantage because overflow can be dealt with by having sufficent headroom in processing bits. DSD can't so that's probably why the max out at -6dB)Note that the problem is more likely to occur where high frequencies are at hot levels.
A hot testdisc (cd and dvda) would be a great tool to check a player/dac for this trouble before actual purchase.
If it shows clipping it's just a bad player.It shows once more that any conclusions about dynamic range of this particular dsd and pcm transfer are on thin ice.
It's likely that the difference is caused in your player.Frank
*** However your example where the clipped signal level is below a similar signal at a higher level but without clipping contradicts this. ***Actually, it is also likely that the clipped signal level is at 0dB FS and signals at a higher level are samples of the reconstructed analog waveform at points where it exceeds 0dB FS. remember we are taking a recording of a reconstructed analog waveform, it will be highly unlikely we are sampling at exactly the same points as the original digital recording.
*** PCM is at an advantage because overflow can be dealt with by having sufficent headroom in processing bits. DSD can't so that's probably why the max out at -6dB ***
Your statement is contradicted by those of ted smith and graemme previously on the asylum. given that these people have been involved in designing and building audio equipment, or have direct experience recording and mastering in DSD, i choose to believe them over you.
***Actually, it is also likely that the clipped signal level is at 0dB FS and signals at a higher level are samples of the reconstructed analog waveform at points where it exceeds 0dB FS. remember we are taking a recording of a reconstructed analog waveform, it will be highly unlikely we are sampling at exactly the same points as the original digital recording.***This can be the case. But the wave you showed has **many** samples showing clipping.
The troubles the papers refer to with reconstructed signals between sample points can only occur across a few samples.***Your statement is contradicted by those of ted smith and graemme previously on the asylum. given that these people have been involved in designing and building audio equipment, or have direct experience recording and mastering in DSD, i choose to believe them over you.
***You need to approach this logically. Whatever your guru's have posted about clipping in **recordings** isn't of consequence in this discussion.
With 1 bit at 64*fs you can't turn on (or off) more than that number of bits. So it's ***impossible*** in a pure 1 bit environment to get past 0dB. You simply run out of bits. This is true for recording and processing.
With pcm recording there isn't a need to keep the levels below -6dB.
Maxing out at near 0dB with 24bits isn't creating any problems as long as there are more bits available in the processing environment.
You simply don't need to waist dynamic range during recording.To prevent the problem described in the paper it's only necessairy to keep the maximum output for the production master below -6dB.
In the production stage you can still have the full resolution with pcm.Also this 6dB processing headroom is a big 'waist' of storage space.
With 1 bit is 'costing' you half the number of possible bit patterns.With 24bit pcm 6dB headroom is only costing 1/24th of the available bitspace.
"With 1 bit at 64*fs you can't turn on (or off) more than that number of bits. So it's ***impossible*** in a pure 1 bit environment to get past 0dB. You simply run out of bits. This is true for recording and processing."
I know of no system that can't be overloaded at some point. If you have such a system that can't be overloaded than I suggest you get a patent attorney, and a publicist.
d.b.
This wouldn't get me a fortune.It's very simple once you understand the difference between the recording stage and the processing stage in the production chain.
Create a recording using all the available bits, lets say 24.
Proces the recording in a DAW and apply 12dB gain.
If the processing is done in an 32bit fp environment you are still far from overloading your system.There are already 64bit fp systems becoming available.
This is virtually unlimited headroom in a processing environment.
sorry frank, i don't believe you are right and i continue to choose to believe ted and graemme know what they are talking about and you don't.
I'm right on this one.
yes, of course you are. how wrong of i to have even doubted you. :-)
nt
This post is made possible by the generous support of people like you and our sponsors: