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To be fair...

Posted by Charles Hansen (M) on June 10, 2017 at 18:52:19

In Reply to: Only in your head... posted by J. Phelan on June 10, 2017 at 16:01:02:

>> If 'temporal blurring' exists, do we have any studies, in the past 35 years, that shows this is a problem ? If so -who, before Meridian, tried to address it ? <<

To the best of my knowledge, the chronology of "temporal blurring" (MQA's marketing term for a phenomenon referred to by various names over the years) is as follows:

1) It was first identified in 1984 by engineers from Studer and Soundstream (world's first commercially available digital audio recorder with a 50kHz sampling rate - later morphed into a car audio company) in an AES paper, "Dispersive Models for A-to-D and D-to-A Conversion Systems".

2) Wadia was the first (late 1980s) commercial example of a product that traded a small rolloff in the top octave (ie, frequency domain) for dramatically reduced "ringing" (ie, time domain).

3) Pioneer popularized Wadia's approach in the early 1990s and called it Legato Link. Their production volume was so large that soon all Burr-Brown DAC chips (Pioneer's supplier) included a "slow rolloff" digital filter option.

4) Sakura System's "47 Lab" was the first digital product to eliminate the reconstruction filter in the D/A converter altogether. This was first announced in the Japanese DIY magazine "MJ" (roughly equivalent to "The Audio Amateur") in a series of articles in 1996 and 1997. This represented an extreme exploration in the audible effects of digital filtering, at least on the playback side.

5) Ayre was the first audio manufacturer (in 1999) to allow the user to select between the "standard" approach and Wadia's approach to digital filters. There was a rear-panel switch labeled "Listen/Measure", the owner's manual recommended the time-optimized "Listen" position and all known reviews agreed.

6) In the late 1990s Sony and Philips developed the SACD format, specifically designed to replace the expiring patents of the CD format. Perhaps the greatest difference between the DSD format and convention PCM is the elimination of anti-aliasing filters on the A/D side, due to the extremely high sample rate. The D/A side used a 3rd-order (18dB/octave) analog filter at 50kHz to reduce the levels of out-of-band noise that could disrupt or damage downstream equipment including both electronics and transducers. (For comparison purposes the 1st-generation Sony CD player used a 9th-order [54dB/octave] analog reconstruction filter.)

This was the basis for one of the prominent claims for SACD - its "superior" pulse response. In my opinion these claims were misleading as while they accurately showed the superior (narrower) pulse response of the A/D converter, they apparently did not show the broadening of that pulse after it passed through the low-pass filter in the D/A converter. Nonetheless, my personal opinion is that the major sonic differences between DSD and *conventionally-implemented* PCM is due to the filtering used (brickwall anti-aliasing on the A/D side and brickwall reconstruction on the D/A side).

This was another signpost pointing out the important sonic impacts that digital filtering imparts. (As a side note, SACD forbade access to the unencrypted digital data stream - just as with MQA.)

7) In 2004 Peter Craven published an AES paper ("Antialias Filters and System Transient Response at High Sample Rates") on so-called "apodizing" filters, which was commissioned by Meridian. Craven noted that any "pre-ringing" created by a steep linear-phase filter in the A/D converter could be filtered out later in the chain. However at the popular 44.1kHz sampling rate the apodizing filter exhibited about the total energy of the original anti-aliasing filter. The claimed advantage was that by using a minimum-phase filter that the energy in the "pre-ringing" would be shifted to the "post-ringing".

This would seem to be a much less objectionable effect, as all sounds in nature create "post echoes" due to the presence of any reflective surfaces near the sound source. Psychoacoustic testing has confirmed that the ear/brain is distinctly more sensitive to "pre-ringing" than "post-ringing"

8) Meridian introduced products that included apodizing filters in 2008/9, I believe. Around the same time, having read Craven's 2004 paper and conducting further independent research, Ayre released products allowing the user to select between either an apodizing (sharp minimum-phase) filter or a slow rolloff minimum-phase filter.

9) In 2012 Ayre developed the worlds first PCM A/D converter that used a digital filter with no time-domain artifacts whatsoever for dual- and quad-sample rates. Single-sample rates had the minimal amount of time smear, achieved at the cost of a slight rolloff in the top octave. All recordings made with this converter have no "errors" for which MQA could possibly correct.

10) Meridian began working on MQA at least in 2013 and possibly earlier. The original focus appeared to be to reduce the file size of high-res files, largely for storage with portable players. An AES paper on the subject was published in 2014 ("A Hierarchical Approach to Archiving and Distribution").

11) Since the original announcement MQA's goals seemed to have expanded, first to "correct" for errors made by the A/D converter used to create the original files, and later to "compensate" for the signature of the particular digital filter/converter chip (apparently one needs to sign a non-disclosure agreement to know precisely what is involved) in a particular D/A converter. MQA's method of "de-blurring" A/D errors appears to closely resemble Ayre's slow rolloff minimum-phase digital filter from 2009.

The bottom line is that there has been a fairly broad awareness of time-domain issues involved with digital audio for over 30 years. It has been examined in some detail, both in academic journals and by equipment manufacturers. I hope this overview is helpful.

As always, strictly my personal opinions and not necessarily those of my employer or lap-dog.

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