Tuesday, November 26, 2013

Where was the "NSA" when Lanza was plotting the Sandy Hook massacre?


While the vast majority of persons interviewed had no explanation for the shooter’s actions, a review of electronic evidence or digital media that appeared to belong to the shooter, revealed that the shooter had a preoccupation with mass shootings, in particular the Columbine shootings and a strong interest in firearms. For example, there was a spreadsheet with mass murders over the years listing information about each shooting.

from Adam Lanza's "Obsession" With Columbine....

So, if the NSA is constantly monitoring everything for signs of terrorist intent, I think they have some 'splaining to do. But considering the reality of "the NSA" which really does have access to nearly everything on our online computers, it's not surprising that it would "overlook" signs of a mass shooting in the works, because the reality of "the NSA" which does this sort of thing (as opposed to the official NSA) is that it's the signals-intelligence division of organized Satanism, given special access to the internet and our operating systems, including the ability to hamper or deny us access to the internet at the ISP level. (The image above is a calling card they left on one of my Google blog "control panels.") Woodrow Wilson, in his famous 1913 quote about some mysterious, watchful force, was referring to organized Satanism. All that's changed in this regard are the tools at its disposal.

I suspect, although I cannot prove, that Sandy Hook shooter Lanza was driven by the Spirits of Darkness, which seem like the sort of being which would run organized Satanism. These beings can have very subtle effects, such as by simply planting a thought, a feeling, or urge in us. At worst, they can turn us into puppets, use us to commit some monstrous deed, and then have us commit suicide to keep us from talking about what was going through out minds. Those with weak will-power such as Lanza are the easiest prey.  So, why would they use their "NSA" to report on themselves?

Monday, November 11, 2013

CD-player interpolation/oversampling in a nutshell

Rev A  (see Notes)

I haven't seen a good explanation of interpolation/oversampling for the uninitiated, so I scribbled the following one:

"Oversampling" in the D-to-A process of a CD player is essentially a matter of calculating extra samples to be inserted between samples from CDs to make it easier to filter out the sampling-related stuff at the output of the DAC, leaving just the audio signal. I don't understand how these calculations are performed - one look at a DSP text and I think you'll agree that this is best left to large groups of mathematical geniuses. Suffice it to say that these calculations can be performed more accurately with DSP chips than in the digital filter section of DAC chips, and that the greater accuracy sounds much better.

The resulting samples don't represent a waveform with higher frequencies (which is theoretically impossible anyways - additional frequencies would imply the creation of additional information out of nothing), but the same waveform with a higher sampling rate than is required for its bandwidth - i.e. "oversampled." Again, the purpose of oversampling is to create greater frequency-separation at the DAC output between the stuff to be kept and the stuff to be filtered out, so that analog filters with acceptable phase characteristics can be used for this purpose.

As good as CDs can sound, they're still not as good as high-resolution PCM, or DSD, which is the gold standard of digital audio. Part of this reason is that the data from a CD has 16-bit resolution, and the process of upsampling it to 24 bits assumes that the  data from the CD has a resolution of 24 bits and just uses it as is. (What other choice is there?) So, this in itself introduces distortion. But, because so much of our favorite music is available on CD at best, we have to make the best of this flawed format, and this includes DSP-chip-based interpolation which uses minimum-phase filter-algorithms.

While performing research into whether a certain A/V receiver also makes use of DSP chips for interpolation, I realized that using external digital filters in conjunction with DAC chips is nothing new. However, the use of digital filters which have minimum-phase characteristics, which Robert Harley of the Absolute Sound called "nothing short of revolutionary," is new.  It's not external filtering or minimum phase alone, but the combination, that has produced the dramatic improvements in CD playback over the last few years.

Notes
Rev A: Added last paragraph

Sunday, November 10, 2013

Error in deleted post requires separate correction

Although the deleted post on DSP-based oversampling contained some glaring errors (most of which were nullified by removing it, and explaining the source of the misconception), it contained one error that might take on a life of its own unless I address it specifically.  That error is the assumption that a digital filter affects only frequencies above the cutoff frequency, as if a formula and its implementation in digital hardware could create a physical low-pass analog filter. The fact is that a digital filter rejects the frequencies above the cutoff frequency by calculating the samples representing the DESIRED waveform. The additional precision of a DSP-chip-based filter creates more precise samples, and hence superior sound.

The deleted post was otherwise one of the better layman-oriented explanations of oversampling I've seen. So, I'll revise it and eventually repost it somewhere.

Alternate explanation of Denon al32 interpolation provides useful insights

Denon's Advanced AL32 Processor expands audio data to 32 bits and uses a proprietary algorithm to interpolate the data and perform up-conversion and sampling, achieving a playback sound that is close to the original source.

Since high-performance devices capable of large-capacity processing read data samples across a wide spectrum and process them in a single stage, they interpolate signals with greater precision compared with multi-stage digital filters and other such devices. In addition, the use of algorithms ideal for frequency characteristics outside the audible range to filter sudden bursts of musical data or continuous sound at high frequencies protects sound quality from the adverse effects of aliasing noise or drops in high-range response.

The Advanced AL32 Processor reproduces the delicate nuances of music, as well as spatial information such as the position of the artist and the breadth, height, and depth of the stage, in a more natural manner. High-precision 32-bit, 192-kHz D/A converters have been used to bring out the maximum performance of the Advanced AL32 Processor.
from Denon Asia DCD-2020AE page

I've taken a circuitous route in an attempt to understand the quality of sound I heard from the CD layer of the BIS Records' hybrid disc of Beethoven's 9th symphony. I assume that the CD player was a Denon, since Denon is a favorite of broadcasters. Since the Denon explanation didn't satisfy me, I researched advances in CD playback in general, and hit a wall when trying to perform research into Cambridge's ATF2 technology, which seems to imply that it corrects for errors in the INPUT data by performing a curve-fitting operation, which has a tinge of hype to it.

So, I went back to researching the Denon system, and found a better explanation on the Denon Asia website, which I excerpted above. It seems to indicate that the audible superiority of al32 is due to the use of a DSP to implement a one-pass digital filter with high precision and with a characteristic optimized for its transient response, which is very similar to, if not exactly the same as, the approach used by Ayer Acoustics (whose products are highly regarded by the likes of Stereophile) as explained here. The benefit provided by using a DSP chip therefore boils down to greater precision, and perhaps the ability to implement a more complex and accurate algorithm, and not, as I was assuming, the difference between linear (or no) interpolation, and interpolation based on digital filtering.

DAC chips contain digital filters, after all

I've removed my post entitled "DSP interpolation for the masses" because it was partly based on the misconception that DACs do not actually attempt to implement digital filtering internally. That misconception was based on misreading an interview with Ayre Acoustics' chief designer Charles Hansen, who was commenting on an "interpolation" process which takes place AFTER the interpolation performed by the digital filter.

However, it turns out that, according to Hansen, the digital filtering performed in DAC chips is performed in stages, and that round-off errors accumulate.  External filters use a "one pass" approach so that round-off errors occur only once.  Perhaps it is this reduction in round-off errors which provides the dramatic improvement I've noticed with DSP-based filtering. (Even Wolfson's high-end DAC chips have inputs for external filters.) I'll continue to study the issue.