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Timmy Fox

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About Timmy Fox

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  • Birthday October 5

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    Tech, music, photography, furry fandom

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  1. Uh, just a curiosity here since I don't feel it's been made clear enough; What's the difference between this (706/707) and the "older" beta (790)? Haven't tried running either yet but judging by the version numbering I'd assume 706/707 are relatively minor updates from the, at this point, rather old 705 build (mainly for compatibility purposes and some minor bugfixes) as opposed to being further updates to the rather more experimental build 790 which dropped rather recently as well. Is this correct?
  2. This is incorrect. Based on your descriptions it sounds like you've simply downloaded different versions of the same song from different sources and equated that because there is a difference it must be because of the format. This is untrue. One common misconception is that louder is better; This is false as well. If you take two identical audio tracks and make one louder than the other, it will psychologically be perceived as better due to being louder despite being otherwise identical. Now, remember there will be differences between different tracks; The same song will not necessarily always sound the same depending on where it comes from. This is due to differing masters; The same song on 2 different albums or even the same song on the same album printed at different times can actually have differences. This is because different so-called "masters" are often used when producing an album and they aren't always exactly identical. There's also the marketing aspect of it where places that sell so-called "high-res" audio will actually try and find some of the better masters and even intentionally modify the song slightly so that it will sound different. Thus you can't just download a song in 24 bit and another in 16 and compare them directly because they may already be intentionally made to be different. So, how to give this a fair comparison? Well, take that Gorillaz song you mentioned and make sure it's 24 bit (because that's supposed to be better you say, yes?) and then convert it yourself down to 16 bit. Foobar is quite good at doing this. Then compare your original file to the resulting converted file and I'm sure you'd be hard pressed to tell the difference (spoiler alert, you won't be able to). Also, if using foobar, one good tool to compare them is Foo ABX which lets you directly compare the files and eliminates bias and placebo because you won't actually know which one is which; You'll just be asked to compare them and guess which one you think is which and then it will tell you if you're right or not. If you still don't agree with me, feel free to send me the tracks in PM and I'll compare them to see if there really is as big a difference you claim there to be. As for MP3, in case you weren't aware, this an open-source format with a large variety of different codecs and implementations available, some with different focus than others. It is true that mp3 originally was intended to degrade the audio to save space but several codecs, most notably the LAME one, rather aims to preserve as much quality as possible whilst still maintaining a reasonably small file size. LAME-encoded mp3 files at around 256kbit and higher are shown to be practically transparent; Meaning that the only loss in quality compared to the source PCM data is so small that it's next to impossible to hear it with your own ears. This has been put to the test numerous times through blind tests and there's hardly anyone who, through proper scientific method, been able to actually tell the difference.
  3. @ACE7F22 @ToaneeM Then I invite you two to do a full double blind test and tell me if you can actually tell the difference because I sure can't with notably higher performing equipment than what a mere phone is capable of. Even IF there is a difference you speak of, it's so small that if I and a bunch of other people can't make it out on any of the tests I've conducted on a variety of dedicated DAC/AMP combinations with numerous different high-end headphones including some HD800, Hifiman, STAX etc. you'd very likely impossibly be able to tell the difference using a mobile integrated dac/amp and essentially any portable headphones. Either way, don't take my word from it, here's one of several popular papers that go more into detail about it written by someone who's actually done this type of work professionally for many years; https://people.xiph.org/~xiphmont/demo/neil-young.html
  4. I'm sorry but there is several things that are factually incorrect about this. Firstly, the "Only sampling 2 times" has been proven through the Nyqvist theorem to be enough to reproduce the full soundwave indetically to its source. Surely we can capture more points and such but that's completely redundant because only 2 samples is actually needed to perfectly reproduce an identical analog signal. Sure the real world isn't flawless but the little bit of extra leeway added with 44.1 and 48kHz (over the theoretical requirement of 40kHz) is enough to eliminate these. Thus, there is no other reason than the "bigger numbers must be better"-mentality to have higher sample rates for mere audio reproduction. Secondly, I think you misunderstand bit depth. It has little to do with sample size and is all about amplitude and dynamic range. With 16 bits we can seamlessly reproduce 120dB of dynamic range which is plentiful unless you want to go loud enough to cause yourself permanent hearing damage in a matter of minutes, if not even seconds, whilst simultaneously trying to hear a mosquito or needle falling in the background. You can't compare it to video that way because your eyes work quite differently and there are very different sensitivities to be applied; Most shortcomings in video is limited by costs and technology whereas audio is limited only by your ears. But sure, if you absolutely wish to compare your eyes and ears in this manner then the more fair comparison would be these: Take an IR-emitter (like a remote controller) or a true blacklight. Most likely you won't actually be able to see any real light yet you know it's there. If you photograph it with a camera you'll see it's probably really bright even if your eyes can't see anything. That is a limitation of your eyes. You physically cannot see it. This is a much better analogy to why higher sample rates won't actually make any difference. You know how, if you've looked at a bright light and gotten dazzled it can get to see anything, especially things that are dark? This is the limitation of your eyes' dynamic range; Your eyes are physically limited in how much difference they can make out between bright and dark. You usually don't notice it very much however because your eyes constantly adjust themselves much in the same was as a camera adjusts itself depending on whether your trying to photograph something bright or something dark. In other words, there is a limit to how big the difference between bright and dark can be for your eyes to be able to clearly make out any details (also why it's so hard to see anything in a dark room after you have just been in a bright room). It's the exact same for your ears, there's a limit to the loudest and quietest sound you are able to make out at the same time, though for your ears they don't have to adjust nearly as much as your eyes will and the limit is closer to where it's so loud it starts hurting and you'll eventually go deaf due to sheer loudness. 16 bits is enough to cover this range practically seamlessly.
  5. More than likely that difference is because it's two different versions of the same track and not just a plain 16bit vs 24bit. If you instead take the 24 bit file and downconvert it to 16bit and do a blind test to compare, I doubt you'd be able to hear the difference. All the bit depth does is affect the amplitude range. 16 bit is plenty for playing at a loud rock concert at full volume. By the time you would need more than 16 bits you would already be at the point where you'd have turned the volume up so loud either the speakers would have already broken or your permanently damaged your ears (if you're not already completely deaf by that point), whichever would come first.
  6. Oh pardon my misspelling. Harry Nyquist was born in Sweden as Harry Theodor Nyqvist. I'm not entirely sure why his last name is more commonly spelled as Nyquist and not Nyqvist but he may have changed it as he emigrated to the US. Either way here in Sweden the spelling with a v is the most common one and is an old/alternative spelling for the Swedish word "Kvist", meaning a piece of a wooden branch or a twig. Despite being spelled with a u, it is still pronounced with a hard v-sound due to said origin. A lot of Scandinavian last names have their origin in nature, actually. Either way, that is essentially a simplified version of how I was thought back in Electronics and Computer Sciences class. His proof shows that the lowest sampling rate required to accurately reproduce any given frequency is one that's twice of said frequency. Going higher (so called oversampling) does have some use in certain situations due to certain factors but as far as digital audio reproduction goes, 44.1 or 48 kHz has been proven time after time to be plenty.
  7. This is incorrect. More samples does mean more data but it does not equate to more detail. This does not work the same as, say, frames per second in a movie or something of the like. Having more samples per second is effectively just more of the same. As per the Nyqvist theorem, 2 samples per reproduced frequency has been mathematically proven to be enough to reproduce the recorded sound in its entirety. Adding more samples is practically just adding more dots to fill in an already perfect sine wave. This is the basically adding extra redundant points to a mathematical graph when you already have the proper points required to make a perfect sketch of said graph. Or with an oversimplification; Adding more samples (say 4X the samples instead of 2X) is practically the same as adding 5+5+3+3 = 16 instead of 10 + 6 = 16. It's adding redundant samples to fill in an already perfect sine wave.
  8. No. Flac is like a zip file, you can compress a wav into flac and then extract the identical wav file back from the flac file again just fine. The audio data that comes out of your headphones or speakers will be identical regardless if you use wav or flac. You're telling me my Sennheiser HD800 are not good enough headphones to hear the difference? Ok, let me ask you if you actually know what you are talking about. First off, let's address 192kHz vs 48Khz. If you're familiar with the Nyqvist theorem it is a scientific theorem that proves you need a sample frequency that is double that of which you wish to reproduce. This means that with 192kHz audio you can accurately reproduce sound with frequencies up to 96kHz. With 48kHz you can reproduce up to 24kHz frequency sounds. For a reference, the limit of human hearing is reached at around 19-20kHz though it gets lesser with age. If you've ever heard a mosquito, they emit sounds at around 17kHz (and is thus close to the limit of the very brightest sound that your ears are physically able to perceive). A dog whistle, depending on type, emits sounds anywhere between 23 and 54kHz and are known to be impossible to hear by just about any human (the same way you can not see infrared not UV light). Thus proves that both 44.1 and 48kHz is, per the Nyqvist theorem, more than enough to accurately reproduce any sounds that fall within the audible range of human hearing. Any more is only necessary if you wish to reproduce sounds of dog whistles. As for bit depth (16, 24, 32 etc.) 16 bit audio is enough to natively reproduce up to 96dB of dynamic range. That is the difference between the loudest and quietest sound. 96dB is about the difference between a concert and complete silence. And I don't mean as in a quiet room in the middle of the night, that's still around 10 dB or so (a whisper is ~20dB), I mean complete silence which is only really achievable in special extremely insulated anechoic chambers. This 96dB figure however is not a true limit; There is also a technique called dithering which is applied to (without any loss of quality) increase this to an effective 120dB of dynamic range. This is the difference the complete silence I already explained and a sticking your ear right next to a jet engine. 120dB is loud enough to give you permanent hearing damage in seconds. Many consumer headphones and speakers will likely break before you exceed these levels of loudness anyway. Thus, 16 bit audio is enough to playback music loud enough to either break your headphones/speakers or permanently damage your hearing in seconds (whichever comes first).
  9. To be fair, high-res audio is more of a marketing ploy and a myth. I mean, unless you can actually hear dog whistles there is very little reason to go above the standard 44.1kHz ...
  10. Gingerbread? Damn, I hate to say it but you're probably due for a new phone.. That OS is over 5 years old and I don't expect any real support coming for it - rather more and more apps will stop being compatible with it. Supporting older Android versions, for developers, is just a major hassle that causes a bunch of unnecessary issues and time wasting with the only end result that people running a version that's like half a decade old can use the app.
  11. Huh, cause I have a faint memory of accidentally setting a rating on my PC on a song that showed up in Poweramp after I copied the file to my device.. I might be mistaken though, or perhaps it just reads rating tags but doesn't write to them?
  12. I thought this already was implemented? Perhaps it depends on what file type you're using?
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