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: Brian Willoughby <email@hidden>
- Date: Mon, 25 Aug 2008 23:13:46 -0700
On Aug 25, 2008, at 19:15, Mikael Hakman wrote:
Nonsense! Distortion measured under steady-state conditions is -75
dB at 1000 Hz. This is in Apple's spec and I have verified and
reported this in my report. Distortion at 1000 Hz measured right
after the signal starts (reaches the receiver) is not specified by
Apple neither by codec vendor. I have measured it to be around -46
dB or 0.5%. It is quite common for DACs that they exhibit higher
distortion at the very beginning of a signal. Interesting issue is
not whether there is higher distortion but by how much, and how
fast the device stabilizes. Use or not use of windowing function in
analysis has nothing to do with it.
Apple does not publish distortion specs for "right after a signal
starts" because the distortion spec does not change. It is
sufficient for Apple to report the steady-state distortion ratings
and leave it at that.
A DAC cannot have a stabilization time that lasts longer than one
sample period. You're talking about hundreds or thousands of samples
before your measurement code stabilizes, and that has nothing to do
with the D/A or A/D. Note that any filtering (analog or digital) in
the signal path may introduce ringing that lasts longer than one
sample, but this will appear at all times, not just with the first
sample at the start of a signal.
Can you cite an independent source for your claim that it is quite
common for DAC to exhibit higher distortion at the very beginning of
a signal?
Windowing is used because the FT analyses a small part of a signal
as if that part repeated forever. Unless your D/A and A/D line
up perfectly, your windowless analysis is actually looking at a
completely distorted picture of the signal, and thus you'll see
very high distortion as an artifact of your custom tools, not due
to any problem in the Apple audio circuits.
Unless what? A/D doesn't know the signal is coming from another D/
A, it could be a microphone. D/A doesn't know it sends output to an
A/D, it cold be a monitor. So if I do get distorted signal starting
with pure one then they must have distorted it, and this is
precisely what I want to know and measure.
I am saying that your distortion measurements are wrong. You blame
the results on the hardware, I blame it on your custom analysis tools
which go against the entire industry and skip windowing because of
your own personal aesthetic. I will let others comment further on
this, but I think you've really opened a can of worms by publishing a
"Test report" on the MacBook Pro and claiming that it's distortion
levels jump above Apple's advertised specs whenever your pure signals
arrive. You've got a hard sell ahead of you.
In fact, this is worst case when you start the test signal after
silence, because the FT sees a repeating waveform that is very
complex - and this totally explains the high distortion you're
seeing in your custom tools.
Worst case!? Well, about as worst as saxophone starting to play a
note from silence or near silence. About as worst as an artist
playing a slow classic composition on acoustic guitar and letting
every note decay to low levels! What are you talking about?
I am talking about problems with your math and your unique approach.
The Fourier transform has limitations. It's very difficult to get
around them.
Your examples hint at the importance of the attack portion of any
natural sound, and you are correct that the human perception focuses
on these elements. However, it is a fact that linear time-invariant
systems do not treat the beginning of a signal differently than the
middle or the end. It is only Fourier analysis that has trouble with
transitions, and it has the most trouble when you skip windowing.
Because what? I send pure simple sine wave and if it becomes very
complex signal then the device must have made it so. Very complex
signal from simple one = high distortion, which is precisely what I
want to know.
But you are using analysis which does not analyze the exact signal,
it analyzes a different signal which repeats a small subset of the
true signal. The distortion is not created by the device, it is a by-
product of the analysis. You are making assumptions about the true
signal based upon analysis tools which are not precisely dealing with
the true signal.
There isn't any discontinuity; my specially designed signals
guarantee this. I start analysis when and at the very first test
signal sample reaches my receiver. If there were any discontinuity
thereafter, my code would throw an exception. I just designed that
different solution you are asking for and the results are put
forward in my report.
The discontinuity comes before your specially-designed signal, not
after. Even if you try to be clever and throw away all incoming
samples until you reach some special sample that you're looking for,
you still have a discontinuity at the point when you begin your
analysis - there is no way to avoid it. There are always an infinite
number of zero samples before your signal. You can ramp up the
amplitude of the incoming signal with an envelope, in order to
minimize this discontinuity at the beginning, but that's a form of
windowing.
Brian Willoughby
Sound Consulting
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