Re: Test report MBP built-in audio device
Re: Test report MBP built-in audio device
- Subject: Re: Test report MBP built-in audio device
- From: "Mikael Hakman" <email@hidden>
- Date: Thu, 28 Aug 2008 02:52:57 +0200
- Organization: Datakonsulten AB
On Wednesday, August 27, 2008 3:54 PM, James Chandler Jr wrote:
From: "Mikael Hakman" <email@hidden>
I'm not physically, electronically or even logically switching the device
on. The device is on for a long period of time and it is feed by a
continuous signal. The signal consists of silence (zeros) up to a certain
time. At that time the signal becomes pure sine wave with phase 0 (i.e.
first sine sample is sin(0), next is sin(dt) etc). I continuously record
output from the device and I know the exact delay between my output into
device and device's output into my program. Therefore I know which of
recorded samples that corresponds to the first sine wave sample. I start
my analysis from that sample. When I say that I know the exact delay I
mean the exact number of samples, which of course vary a little between
the runs but is measured at the very beginning of each run, earlier
(before silence and sine wave) in the test signal so to speak.
The reason I'm using such a test signal is that it is a crude simulation
of what happens when a musical instrument is played. Because proper
reproduction of signal during instrument's attack time (first few
milliseconds after you hit a string, start blowing etc) has been shown to
be the second most important factor after harmonic content to our
perception of timbre, I decided to measure distortion in experiments that
are close to this musical reality. Perhaps such experiments could explain
why 2 audio devices having the same or very close specs, may sound so
differently, one sounds right, the other doesn't.
I know very little about it, but if you for instance duplicate this
experiment, feeding your signal (silent head + sudden onset sine wave)--
If you process your test signal thru a simple IIR digital HiPass DC
Blocker filter--
Visually examine the output, and the signal onset will not look like a
clean sine wave for quite awhile after the signal onset, until the DC
Blocker settles.
You can see the same thing with about any kind of filtering, including the
low-pass anti-alias filtering on ADC inputs and DAC outputs (though those
filters settle quicker than a DC Blocker). A sudden onset interacts
strongly with the impulse response of the filter.
Dunno what an FFT of that first sudden-onset 'odd-looking' DC Blocker sine
cycle would show. A short un-windowed FFT on that first wave cycle might
make you think it indicates distortion. Dunno. Never tried it.
But such linear filters usually don't make harmonic distortion, only 'time
distortion' contributed by the impulse response.
Maybe you are looking at some other artifact, and have a way of ignoring
linear filter settling artifacts?
Audio DAC outputs and ADC inputs, will almost invariably have high-pass,
DC Blocking characteristic. A short-term harmonic distortion test should
have some way of ignoring the time-distortion of such filters' impulse
responses.
That is one advantage of a windowed steady-state test-- In that case any
linear impulse responses in the system have presumably settled, and so
they will not confuse the distortion measurement?
Hello James,
My aim is to measure how far from or close to an ideal system, the actual
system is. Therefore I cannot ignore all those distortions you propose to
ignore because it is the magnitude of these very distortions that make one
system better (closer to ideal) than another.
Without here going into the math of convolutions and transfer functions, a
system under investigation produces an output signal (response) given an
input signal. Any change to the original input, that isn't constant time
shift, isn't constant attenuation, and isn't adding constant offset, is by
the very definition a distortion. This includes distortion produced by your
IIR digital HiPass DC Blocker filter, and LowPass, BandPass, and StopBand
filters and of course FIR dunno.
There are many components in a system that contribute to this distortion.
Imperfections in DAC/ADC algorithms, various filter effects (whether filters
are inherent to DAC/ADC construction or not), output/input circuitry,
variation of power or reference voltages, temperature changes, other effects
in semiconductors etc.
While knowledge about reasons why such distortion appear and what components
that contribute and in what proportions to the total distortion may be
interesting to system's vendor, it is not required when you want to compare
systems and assess relative quality of audio that is (re)produced.
Both research and experience shows that steady-state distortion measurements
cannot explain some by humans perceived differences in audio quality. To put
it simply, given 2 devices with the same or very similar specifications, one
may sound right while the other not. This ambiguity has been observed both
at consumer, prosumer, audiophile, and studio levels.
It has also been shown that proper reproduction of signal during attack
phase of an instrument's voice is second most important factor in human
perception of timbre (first being harmonic content). Duration of this attack
phase vary with instrument. For most instruments it is in a range of few
milliseconds. Therefore I'm studying distortion produced by a system during
this period of time. This cannot be done using steady-state measurements
because during attack time there is no steady-state yet (otherwise it wouldn't
be attack anymore).
I hope this answers your questions.
Regards/Mikael
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