...60 Hz   Once you find the RMS voltage where the amplifier begins to clip (i.e. the distortion reaches 1%, although manufacturers often use 10% instead), the power output in W can be calculated easily as V * V / R. If there is no excess gain, it might not clip even at the maximum volume...

  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...

  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...

  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 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...

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...

...values), the multiplier is 10, and for dB and voltage/current (i.e. the actual sample levels in digital audio) it is 20. So, for a sine wave it is either 10 * log10(2) (half power relative to a square wave of the same peak level), or 20 * log10(1.4142), and in both cases the result is 3.01 dB.

  That is why the DR rating is based on the loudest (in terms of RMS level) 20% of blocks. If at least 20% of a track is brickwalled, then it will basically ignore the rest, and report only the dynamic range of the compressed parts.

  For matching the loudness of tracks, it is best to use ReplayGain, as it was designed specifically for this purpose. The relative increase in power needed to play the quiet track as loud as the loud one without clipping can then be calculated from the ratio of the peak levels after ReplayGain...

  Actually, you would need to subtract 3.01 from the dB value, rather than divide it by sqrt(2), so it is 6.9 - 3.01 = 3.89 dB. However, the overall RMS level of this track seems to be in fact -6.9 dBFS referenced to a full scale sine wave, and the peak is obviously 0 dBFS as it is clipped, so...

  Your laptop may already have optical S/PDIF output. It is just often shared with one of the 1/8" TRS connectors to save space.

On the other hand, coaxial (rather than optical) S/PDIF is not isolated, and is therefore not protected from ground loops. The expensive USB to coaxial S/PDIF converters claim to reduce jitter, and may or may not implement isolation as well. In other words, they are indeed likely to be snake...

By the way, the signal I used in the test from the second link can be generated with the testgen utility using this input file (for 44100 Hz sample rate):   sine 0 5 11025 45 0.82842 0.82842 sine 0 5 15025 0 0.41421 0.41421   Or use any other software to create it (the parameters after 'sine...

Some DACs (like the CS4398 on this sound card, which clips only at about +2 dBFS) have headroom for them, but not others. Although with music, it is difficult to hear the difference, because for inter-sample peaks above 0 dBFS to occur in significant numbers, it has to be mastered pretty badly...

  Higher (in terms of absolute dB value) is better, and -88 dB crosstalk is plenty good enough. Although it will likely be worse when loaded with low impedance headphones, it still should not be an issue.

  There is a test with simulated crosstalk in this old thread. No ABX results have been posted, though, either because of the lack of interest, or the difference being much more difficult to detect than people expected.   In any case, if you view for example -40 dB crosstalk as 1% narrower...

...    The RMS sum is not based on the dB values, but rather on the power, which is the correct way to combine the RMS level for multiple blocks. That is, the correct overall RMS for the 12 blocks is:   10 * log10((10^(-5 / 10) * 5 + 10^(-5.3 / 10) * 5 + 10^(-5.8 / 10) * 2) / 12) = -5.25 dB

  And for recording, the microphone should ideally have low noise as well, which is not as important for the originally intended application of frequency response measurements. A small capsule generally has a more extended frequency response, but worse SNR.

The foam inserts are also available separately, it is just easier to get the ear pads (as they are sold by many online stores, Amazon, etc.), which you may eventually need anyway.

  They are included with the ear pads (EDT 770 V), if you buy a new pair of those.

  Those who somehow have a DAC that does not could easily find something better at a low cost, such as Realtek HD audio codecs: These graphs of 44.1 kHz digital DAC filters have been generated from the recorded samples of this blind test posted some time ago. So, the audibility of the...