Sorry, I misunderstood your original comment. When you said: > Up, flat, down, flat. Guess what: you've just introduced enormous amounts of aliasing. I though

Sorry, I misunderstood your original comment. When you said: > Up, flat, down, flat. Guess what: you've just introduced enormous amounts of aliasing. I thought you were suggesting that the jagged, stair-step output of a DAC is what causes aliasing, which is a common misconception. I was trying to point out that stair-step output is just the reconstruction filter of the DAC operating as intended, and that a reconstruction filter is an anti-aliasing filter. In the case of a DAC's sample-and-hold, it's an easy way to select the baseband alias. [1] I think one reason this thread went off the rails is that there are two anti-aliasing filters in a digital synthesizer. I was initially referring to the hardware anti-aliasing filter at the output of the DAC, which is also called a reconstruction filter or an anti-imaging filter. Everybody else was referring to the digital anti-aliasing filter that you need in order to create a band-limited approximation of an infinite bandwidth ideal signal so that you can actually sample it, because attempting to sample the ideal signal directly is always incorrect. Given that the article is about the precursor to digital synthesis, I suppose I should have realized that people were going to be more interested in discussing the latter. However, now we're getting to the second reason this thread went off the rails (and the reason I usually regret participating in audiophile threads): Differences in terminology, points of focus, and "Well, Ackchyually" audiophile-grade condescension cause people to read past each other and continue to argue despite largely being in agreement. If you read my downstream responses (https://news.ycombinator.com/item?id=25601970 https://news.ycombinator.com/item?id=25601970 and https://news.ycombinator.com/item?id=25602267 https://news.ycombinator.com/item?id=25602267), I hope it's clear that I both understand and agree with the actual argument you wanted to make. [1] Aside: If you were to insert sinusoids at twice the sampling frequency instead of flats in between each sample, you'd be selecting the alias centered around twice the sampling frequency instead of the one at baseband. This is (roughly) how a mixer works.