[dankoss]

I tried to find the context for that image but there wasn't anything on the site. The image is most certainly incorrect -- or more likely, looking at the output of the analog electronics after the DAC.

Sampling theory says that a perfect square wave can be represented at any frequency below Nyquist. That doesn't mean that the codec or the analog electronics are capable of responding instantly at those frequencies, but that has nothing to do with the fact that a 1kHz square wave can be perfectly sampled with a 44.1kHz sample rate. The image is simply incorrect.

Transient response in the real world is generally limited only by the acoustic transducer response of the system, because everything else has the ability to respond much faster than audio rates. With Pono, this means that the earbuds or headphones you use with the player will have a greater affect on the transient response than the electronics inside.

[rrrooss]

This image is from RME a sound card maufaturer. And the image is of the analog output after dac. So still in the electrical domain. Short comings of a playback system has what to do with the accuracy of a recoding medium? There are SO many factors that stop and audio signal reproduction from being perfect. But the signal domain is the easiest to make better. Transducers that have to fight the law of physics for accuracy are the obvious weak point. But as with anything rubish in = rubbish out. Before you even get to the weakest point of an audio system the transducer (the speaker) transistors and amplification have their own short comings the accuracy of an amplifiers transient response is measured by its slew rate. A square wave is impossible for a speaker cone to reproduce in theory the rising and falling edge is describing an instantaneous movement. A speaker cone can't do this, inertia says no. And this is called distortion. As every part of the system introduces its own distortion the accuracy gets less and less. To accept distortion at the signal level in the persuit of accuracy is counterintuive. But if your system can't take advantage of this extra accuracy, go ahead and use a lesser format, no sense in the extra file size. If you want a reference quality signal, use the high res formats. To be clear here I'm not saying I want to listen to digitally encoded square waves, a square wave is a tool for showing the accuracy of input=output. A square wave being the hardest analog waveform to capture and reproduce. I am objecting to the idea that hi res formats have no advantage over what we are used to as consumers. It is the content distributors that are taking advantage saying that 24bit/96k is more expensive. The bandwidth and storage does not make this a more premium product you should be paying for the album not the format. But why limit choice because the masses can't see the benefits. If you buy the album you as the consumer can choose the format. I'm not sure if all this arguing stands up to pick whatever format suits your needs. This argument should be if I bought the album why can I choose the format most applicable to my playback needs/desires. And if your concern is that company's charge more for the high res, that refeclts poorly of the distribution company. If you think it's a waste of space just download the lesser format. If I had a choice I'd download it as it left the studio/mastering house.

[dankoss]

Not sure what point you're trying to make here, but I was responding to your post implying that 192kHz sampling rate improves transient response. It doesn't improve it at all. While there is sampling "distortion" by way of quantization error, this error is inaudible because we're sampling at 24 bits, beyond the dynamic range of human hearing. High sample rates actually REDUCE audio quality because they sample ultrasonics that will interact within the audible range and potentially damage speakers or pick up EMI interference.

It has been proven over and over via ABX testing that high res formats are completely indistinguishable from a 24b/44.1kHz master. And further studies have proven most audio engineers can't distinguish between lossless audio and 320kbps MP3. That's what the Xiph link elsewhere in the thread shows.