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Posts by stv014

You need these for the measurement: 1. a simulated load (or even real load if you do not mind your headphones playing extremely loud sine waves) 2. a signal generator 3. an analyzer with an input that is suitable for testing the output of the amplifier at its maximum level 4. a digital multimeter that can ideally be used for measuring the output level of the amplifier, or at least to calibrate the analyzer (if it is just a PC sound card and software) This older thread...
 Another (in my opinion not unlikely) possibility is that the "clear" differences you heard are mainly the result of expectation bias and simple factors by level differences.  That is why the test should ideally be performed by people who do expect to hear differences. You could try some ABX tests yourself, although I guess you most probably will not.  Actually, there is a new revision of the ODAC that no longer uses an ESS DAC.
 Another solution is to delay the input by half sample with sinc interpolation. This is basically the same as upsampling to 88.2 kHz, except the output does not include samples that would be the same (with linear phase filtering) as the input signal. Inter-sample peaks are not necessarily exactly halfway between the samples, as it can be seen above, but the result is reasonably accurate.
 With synthetic test signals, the inter-sample peaks can be made very high, but such peaks are unlikely to occur in real music that is not already heavily compressed and clipped (and in that case, it is not even obvious whether having the inter-sample peaks above 0 dBFS is correct or not, as the peak limiting/clipping algorithm running in continuous time would not have produced peaks that exceed 0 dBFS). It can be a real problem with some special signals used in...
 The upsampling is only used to find out what the inter-sample peaks would be (approximately) at the original sample rate. The exact level of the peaks depends on the reconstruction filter used, so it is actually not the same for all DACs.
 It is likely to be clipping if runs of multiple samples are at 0 dBFS, and I think Audacity has an option for detecting this. Also, if a histogram of the sample values peaks at 0 dBFS, then it suggests clipping as well (as it can be seen on the 2008 version of this track). Without any dynamic compression or clipping the histogram would look approximately like a bell curve. By the way, even when it is technically not "clipping", extreme brickwall compression can still...
I was originally referring to the clipping that possibly occurs when decoding lossy compressed audio.   Nevertheless, clipping is not uncommon in commercial music either, because of the loudness war.
 In other words, they can. Of course, converting the output to an integer format limits it to 0 dBFS, but that is already clipping of the actual sample values.
With the lossy codecs, it is not just inter-sample clipping, but clipping can occur already on the PCM stream as some samples could exceed 0 dBFS.
 Sample and hold is not the same as linear interpolation. If we consider all samples being represented as "infinitely" short impulses (so that in the frequency domain the passband is mirrored between the Nyquist frequency and the sample rate, and then this repeats infinitely at Fs intervals with no roll-off) as having no reconstruction filter, then sample and hold is a filter with a square impulse response, and linear interpolation has triangle impulse response (length = 2...
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