![]() #Potplayer audio driversIf the drivers aren't working correctly, or you're using a cable that's too long, it's quite possible that your computer "thinks" your DAC doesn't support the higher sample rates because the USB connection negotiations didn't finish properly. #Potplayer audio driverIt's also possible that you have a driver issue going on. (It could be that the player you're using is set to resample - many offer the option. If this has happened, then your player program may be automatically resampling what it plays to accommodate what it believes is a limitation of your DAC. If this information exchange didn't complete properly, which can happen if you have driver or cable issues, then the computer may "think" that your DAC only supports 44k. You should double check the output settings on your player.Ģ) When you connect a USB device to your computer, the two devices exchange information about what sample rates they both support. ![]() Many players can be manually configured to resample certain sample rates (this is intended to allow them to play high sample rate files through DACs that don't support them). This can happen for several reasons:ġ) Your player could be set to do so. The most likely choice is that your player program is resampling the files. I would also assume that the display on the DAC itself is probably correct. #Potplayer audio windowsIf you're using WASAPI, then WINDOWS should not be resampling the files you're playing. (A specific ASIO driver may also include the option to resample "on the way through".) In general, ASIO should play files at their native sample rate, and Kernel Streaming will NOT do so. The fact that something hasn't been proven to be true in no way proves that it is UNTRUE - that's just flawed logic. (I've seen plenty of tests that showed that, under certain circumstances, the majority of a certain group of subjects didn't notice a specific difference, and I've seen tests that proved that specific people who believed they were able to hear specific differences under specific conditions were in fact unable to do so, but I've never seen a comprehensive test anywhere near compelling enough to be worthy of being claimed to prove the general case.) To be totally truthful, I've never seen ANY of what I would call "well conducted rigorous testing" on that subject at all. (This is in direct contrast to converting a FLAC file to a WAV, then back to a FLAC, as many times as you like - and ending up with a file that IS identical to the original.) What you end up with won't be bit-for-bit the same as the original then either. Likewise, start with any 16/44 file you like, convert it to 24/96 using your favorite program, then convert it back to 16/44 using the same program. ![]() Even beyond that, most of the higher end programs offer several different choices of which filter slope and type to use, so, even with the same program, you can produce an infinite number of close-but-not-identical results. Start with a 24/96 file (you pick it) re-sample it to 16/44 using any two audio programs (you pick them) and do a bit comparison between the results. and there do seem to be quite a few people, including myself, who do claim to hear differences - at least in some specific situations.) (Well, let's say that the fact that differences exist can be proven easily enough. No - it is, in fact, technically correct. Basically, this will play high sample rate files at their proper sample rate, and up-sample those recorded at a lower sample rate (which, while it can't add information, and so can't actually improve them, generally won't do much damage). ![]() If you prefer not to do that, the next easiest thing is to set your default output sample rate to either 24/96 or 24/192 and just leave it there. (Incidentally, WASAPI mode will take care of matching the sample rate the bit depth generally takes care of itself in most cases.) If you really want the best sound quality when playing high-res music, the easiest thing is to switch to a program that does support WASAPI mode. The catch is that not all programs support WASAPI mode, including most of the ones that come with Windows, (although all "real audiophile music players" do). ![]() Windows has a special audio mode called WASAPI and, when you use WASAPI mode, each file will automatically be played back at its native sample rate. By default, Windows expects you to set a particular sample rate, and plays everything at that sample rate - and setting that sample rate to match the files you're playing would allow you to play them back without conversion. ![]()
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