Thursday, 23 July 2015
Well, I guess I am flattered that the results of my recent Digital Filters Test got attention on AudioStream (Stereophile affiliate). The other day, Michael Lavorgna posted an entry entitled "The Trouble With Audio Tests" including a few quotes originating from my INVITATION to the test as well as my ANALYSIS posts.
I'm going to start today's entry addressing some of his thoughts on the matter and how I view tests like this.
First, let's just start with the quote from Ernest Rutherford (1871-1937) brought up by Mr. Lavorgna: "If your experiment needs statistics, you ought to have done a better experiment."
Thursday, 16 July 2015
This is part 3 of the results from the test of digital filters which began with Part I (RESULTS) and follows Part II (ANALYSIS).
As usual with these tests, I provided the respondents with an opportunity to describe what they heard when comparing the two samples (linear phase vs. minimum phase upsampling). In this portion I will collate the responses into what people described as the sonic difference and try to classify the responses to the type of upsampling setting for each of the samples... By doing this, I think one can get a sense of whether there is any subjective consistency and also an idea of the variability of subjective opinion. I have put my editorial comments in brackets and italicized.
Friday, 10 July 2015
This post is a continuation of RESULTS: The Linear vs. Minimum Phase Upsampling Filters Test (Part I) where I had already summarized the rationale, procedure, and description of the 45 test respondents including basic demographics, equipment, and raw results.
IV. AnalysisIn this segment, let's try to ask some questions to see if we can come up with answers on the significance of the findings themselves. I think the best way to interrogate the data might be to ask a few questions and see if an answer can be teased out...
I. BackgroundWell, the time has arrived to open up the covers and see what the data reveals!
As a recap, I direct you to the post "INTERNET BLIND TEST: Linear vs. Minimum Phase Upsampling Filters" where the test was introduced and invitations sent out for participants to be involved. In preparation for some of the discussions here, I invite you to read up on an excellent "primer" on digital signal processing done by Kieran Coghlan ("Up-sampling, Aliasing, Filtering, and Ringing: A Clarification of Terminology") published on Secrets of Home Theater and High Fidelity in May. Note that digital processing affects both audio and visual technology, hence the discussion applies to 4K video as much as it does to hi-fi sound. In it, he talks about the "Fourier pairs"; functions have both time and frequency domain effects. Simply put for us in audio as it relates to this test, the steeper the DSP function in the frequency domain (eg. a steep "brick wall" filter), the more the effect in the time domain (ie. ringing). Here's a chapter in the book The Scientist and Engineer's Guide to Digital Signal Processing for those who want to go into even more of the mathematics.
The reason I want to explore this in a blind test is simply because time domain plots of discontinuous signals as produced by DAC upsampling antialiasing filters (generally presented as an "impulse" / Dirac delta function plots) are often used to portray the reputed benefits of various digital filters in the audio world. Furthermore, there are those who write about and suggest that differences in upsampling digital filter parameters affect the sound in very substantial ways. The idea that if we decrease ringing, especially the pre-ringing prior to the main impulse signal, could lead to significant improvements in sonic quality and that it is desirable to aim for the use of minimum-phase filters (and by extension, perhaps it would be good for the audiophile to purchase a DAC that has this feature). Post-ringing is said to be less problematic as auditory masking reduces audibility.
Tuesday, 7 July 2015
Let's think for a moment what "audiophilia" is about...
'We' love music so the likelihood is that 'we' have lots of albums to listen to. That's one of the great things about computer audio - a unified, easily accessible library. (This is of course not necessarily the case for everyone.)
'We' enjoy talking about hardware that can make the sound better. 'We' in fact often spend a lot of time considering how to match the pieces of an audio system to extract the most out of the collection above.
'We' argue about what's "best" in terms of the hardware because 'we' are passionate about the pastime and it is fun to find ways of making things sound better. When passionate about something, natural human biases (especially among men!) will result in arguments, disagreements, and debates, does it not? Just ask the guys rooting for their favourite team or the "friendly" competition between nations in international sports... The way I see it, there's nothing wrong with this. Debate sparks thought, hopefully ideas which come to fruiting with advancement. Of course, sometimes things do turn ugly and we see unfortunate brawls or riots with alleged fouls or outright unsportsmen-like anger or hatred. Last I checked, nobody got murdered or trampled to death in audio-related debates thankfully.
Wednesday, 1 July 2015
|A cool example of the "Spectral Frequency" display - from Adobe.|
Okay guys, I actually wrote the text that follows 2 months ago before the Digital Filters Test. As happens sometimes, a post can get lost in the "draft" bin and later found. It refers back to the DSD-to-PCM analysis series from April:
ANALYSIS: DSD-to-PCM Conversion 2015 - Windows & Mac OS X
ANALYSIS: DSD-to-PCM 2015 - foobar SACD Plug-In, AuI ConverteR, noise & impulse response...
Consider this as part 3 of the 'trilogy' for this week as I continue to work on the results of the recently-closed Digital Filters Test.
I was reminded recently by Wombat in his post here that we can have a look at the spectral frequency display as well when assessing sonic data in the audio editor. Also, Mnyb talked about the low-pass filtering in SACD players and what "standards" were used. Well, I don't know about the formal standards, but the first Sony SCD-1 SACD player back in 1999 had a defeatable analogue filter placed at 50kHz and according to the DSD Wiki, all SACD players were supposed to include this "optional" filter. I know a few people on audiophile forums suggest they subjectively preferred the filter be turned off.