Android resampling

Oct 24, 2019 at 4:48 AM Post #31 of 124
[1] So no one able to answer my question with reliable sources......
[2] at the point that many songs get mastered at 24bit going down to 16bit is just making that file lossy a little part of songs will get chopped off. It also effects volume of sound but that's not it.
[3] If higher than 16bit audio file is and snake oil why don't you guys just sell all your dap, dac and etc apple lighting gender and Samsung c type dongle has arround 100db snr dnr that's already higher than 16bit dnr
[4] The main point of hifi system is to reappearence original sound that has recorded and having higher audio file makes the sound be near to original sound since digital signal isn't curve.
[5] If you look at the files that has higher than 20khz bitrate you'll be able to see spectrum that is higher than 20k and sometimes goes up to 50khz that shows there is also some sound in that range that can effect other sounds.
[6] Measurement and spectrum shows is that there is difference on resampled audio and higher bitrate audio and most simple thing to find out is simply blind testing to your self
[7] and especially for resampled audio because of resampling filter it's very easy to be audible ...

Oh dear, and again!

1. Firstly, if you're going to contradict the accepted facts/science then YOU have to provide "reliable sources", please do so. As to our reliable sources, any text book on digital audio, for example 'The Principles of Digital Audio" - Pohlmann, "Digital Audio Explained" - Nika Aldrich and many others. In the case of the double blind testing of higher than 16/44.1 (hi-rez) played back at 16/44.1, there are numerous sources but the most commonly quoted is the Mayer & Moran study published in 2007. Now it's your turn!

2. Firstly, songs are NEVER mastered "at 24bit going down to 16bit"! Songs today are mastered at 64bit (float) and from that "go down to" a 24bit version and a 16bit version. And prior to "today" (many years ago), songs were mastered either in the analogue domain or at either 32bit (float) or 48bit. Secondly, little parts of the song were NOT "chopped off", they are "dithered off" (source: "Mastering Audio ..." - Bob Katz and numerous others)! Secondly, it does NOT affect the "volume of sound" (sources: All the text books already quoted and numerous others). Lastly, how is any of this relevant? Your android device is not mastering, it's simply resampling!

3. 100dB DR is NOT higher than 16bit. If we're talking about a 16bit master version derived from a digital mastering process, then it theoretically has a dynamic range of 120dB but again this is irrelevant as virtually all digital masters at any bit depth have a dynamic range of 60dB and typically much less! And secondly, if we sell our daps, dacs, etc., what are we going to replace them with? No one makes 16bit high quality DAC chips any more, they're all at least 24bit!

4. Firstly, Hi-Fi systems cannot/do not reproduce a digital signal, they ONLY reproduce an acoustic signal (otherwise we obviously couldn't hear it) and this is derived from an analogue signal reconstructed from digital data, which IS A CURVE!!! (same sources as previously).

5. No it does NOT! It shows that there is some sound in that range but NOT that it can effect other sounds and obviously whatever is between 20kHz and 50kHz is outside of human hearing (sources: Every medical text book that deals with hearing written in about the last 100 years!).

6. The most simple and accurate thing is just to have a look at the scientific evidence but double blind testing yourself is good, if for some strange reason you doubt the scientific evidence and that's something that I and countless others have done over the course of decades. Why don't you have a look on Hydrogen Audio, which has numerous examples?

7. Again, unless you provide some reliable evidence of your own, then as your claim contradicts a wealth of reliable evidence, it cannot be viewed as anything other than a false statement by a deluded audiophile!!

G
 
Oct 24, 2019 at 4:58 AM Post #32 of 124
So if my room snr is 30db isn't it 96-30 to make the 1db of noise there need to me 66db of signal? Why is it + not -

It's + not - because you cannot hear the noise floor of a recording 30db below the noise floor of your listening environment! If a 16/44.1 file had a dynamic range of 96dB then the quietest part of that dynamic range would have to be roughly equal to or louder than the noise floor of your listening environment. So, if the noise floor of your listening environment is say 30dB, the quietest part of the recorded dynamic range would also need to be roughly 30dB and therefore the loudest part of the recording's dynamic range would be 126dB (96dB higher than the noise floor). It's for this reason that extremely few commercial music recordings have a dynamic range of more than 60dB and the vast majority have less than 50dB!

G
 
Oct 24, 2019 at 11:56 AM Post #33 of 124
That's not the point...... yes resampled 16bit and 24bit 192khz can be easily heard

How are you doing the resampling? A bump down to 16/44.1 should sound exactly the same as the original. If it doesn't, something is wrong. No details should be lost in downsampling because 16/44.1 is overkill to fully contain commercially recorded music anyway.

I think the problem here is that you are starting from assumptions that just aren't true. If you really are hearing a difference in a level matched, direct A/B switched blind comparison, either the software has become corrupted or the equipment is faulty. It also could be that you aren't applying to controls to your ABX properly and your bias is skewing the results.

In any case, if I was you, I would focus on verifying this difference you hear in carefully controlled conditions, not trying to explain why a difference exists. Once you're positive there is a problem, then the next step would be to get another android device and see if it degrades the signal in the same way. You need to be more systematic in the way you approach this.
 
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Oct 24, 2019 at 5:51 PM Post #34 of 124
Oh dear. It should be obvious that if you are going to post statements of fact in a Sound Science forum, then you really need to have some understanding of sound science. If you are not sure or don't understand the science then by all means ask, but then don't post incorrect/false statements of fact. So to address your false statements:

1. The amount of "details lost" obviously varies here. As the name indicates, a lossy codec (say aac or mp3) looses a lot more detail than a lossless format, say 16bit. The question is whether these lost details are audible, a question that's been answered countless times over the course of many years. However in your given scenario, an android device resampling to say 16/48, we're not dealing with "lossy" encoding/decoding! And no, you could not only call these lost details "dnr".

2. This statement is completely false and demonstrates a fundamental lack of understanding of both digital audio and sound recording! When we record an instrument playing we use microphones and mic pre-amps (typically several). These are analogue devices (that are far from perfect) and are not positioned where your ears would be when listening to a "live instrument playing". Therefore, you SHOULD hear a difference between a recorded instrument and a live instrument but it has nothing to do 16/44.1 digital audio!

3. This too is unfortunately completely false! 16bit or 24bit doesn't have any "steps", the output is a continuous analogue waveform (without any steps) that is NOT closer (or further) to pure analogue sounds, it IS pure analogue sounds!!

4. There's a couple of serious errors in your statement. Firstly, there is no such thing as "pure" 16bit or 24bit, ALL 16bit and 24bit recordings are resampled through a digital resampling filter, MORE than once, BEFORE you even load it into your android device!

5. Whether it's different or not is irrelevant, whether that difference is audible is what's relevant. The reason that whether it's different or not is irrelevant is because it's ALWAYS different. Even the exact same song file played twice in succession without changing any setting or playback parameters will be different because all the analogue components in your playback chain (amp and headphones for example) produce thermal noise and as thermal noise is random noise, it will be different every time!

6. What "part of it" is it loosing? In this example, no part of it will be lost! Some part of it may have been lost using 25+ year old resampler but I'm assuming your android device is newer than 25 years?!

7. You seem to have linked to the wrong test. The test you've posted demonstrates that one or two test subjects were able to successfully ABX the analogue output of two devices, a 1996 soundblaster computer card and a 1991 Yamaha CD player. I'm assuming your android device does not contain either a 23 year old soundblaster card or a 28 year old CD player?!

G
Ok you took long time to write that but only thing that i want to see from you is sources that resampling 44.1to 44 doesn't matter now a days even it mattered 20years also pls stop giving be useless information and pls give me some useful information that i asked
 
Oct 24, 2019 at 5:56 PM Post #35 of 124
Bitrate involves the noise floor. 24 bit has a noise floor of -144 dB. 16 bit has a noise floor of -96 dB. Your living room has a noise floor of about -30dB at best. Simple math. 30 dB plus 96 dB is 126 dB. 120 dB is the threshold of pain. The only way to hear a noise floor below 16 bit is to incur hearing damage. The noise floor of a good recording studio is significantly higher than 16 bit.

Sampling rate involves frequency response. According to the Nyquist Theory, in order to perfectly recreate a frequency, you need a sampling rate of double the frequency. So 44.1 can reproduce up over 20 kHz. 20 kHz is the limit of human hearing. Anything above that is inaudible. If you can't hear it, it doesn't matter. So 48 is overkill. 48 is mainly used for video because it divides neatly with video frame rates.. It doesn't improve quality.

There you go, my man. Look it up for yourself. Google Nyquist Theory and bitrate noise floor. Have fun with it. I'm happy to provide you with breadcrumbs if you are interested in following them. If not, don't let the door hit you in the butt!
If my headphone or iem can isolate noise around 30db like atimotic ER series it wouldn't be 120 it will be 120 because it can isolate most of the noises
 
Oct 24, 2019 at 6:00 PM Post #36 of 124
What about the noise floor of your player? What about the noise floor of the studio the music was recorded in? What about the fact that the music probably doesn't have more than 50dB of dynamic range anyway? You probably never raise the volume much over 70dB, and that is quite loud. 96 dB would be VERY hard to listen to for more than a couple of seconds.

In the real world, the difference between 24 bit and 16 bit can't be heard. The AES has published studies that prove that. If you think they're wrong and you can easily discern that, set up a controlled test and prove it. If you do that and succeed, submit your findings for verification. I'm sure there are plenty of engineers who would happy to be involved in a game changer of a finding like that.
 
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Oct 24, 2019 at 6:03 PM Post #37 of 124
What about the noise floor of your player? What about the noise floor of the studio the music was recorded in? What about the fact that the music probably doesn't have more than 50dB of dynamic range anyway?

In the real world, the difference between 24 bit and 16 bit can't be heard. The AES has published studies that prove that.
Can you link does studies?
 
Oct 24, 2019 at 6:05 PM Post #38 of 124
Sure... this is the source of most scientific research on sound reproduction and engineering...

http://www.aes.org/e-lib/
 
Oct 24, 2019 at 7:26 PM Post #40 of 124
The Audio Engineering Society is a peer reviewed scientific journal that publishes studies. You can search their database for one that looks interesting by putting in the pertinent keywords, and purchase them one at a time. Or you can join the Society and get access to their whole library.

You can also use google to search for "16 bit vs 24 bit audibility test" or something like that and you'll find several casual online reports of tests.

But to understand it, you might want to start by reading this article on what bit depth actually is. Gregorio is here to answer your questions if you have any.
https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/

It's a really good idea to set time aside every week to learn about things that interest you. Basic digital audio information, like bitrate, sampling rate, Nyquist Theory, decibel scale, frequency response, distortion, dynamic range, etc. would be a good place to start. Having a basic understanding will make it easier for you to understand what people are talking about. You can probably find resources in your native language that would be easier for you to read than English.
 
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Oct 24, 2019 at 7:28 PM Post #41 of 124
What about the noise floor of your player? What about the noise floor of the studio the music was recorded in? What about the fact that the music probably doesn't have more than 50dB of dynamic range anyway? You probably never raise the volume much over 70dB, and that is quite loud. 96 dB would be VERY hard to listen to for more than a couple of seconds.

In the real world, the difference between 24 bit and 16 bit can't be heard. The AES has published studies that prove that. If you think they're wrong and you can easily discern that, set up a controlled test and prove it. If you do that and succeed, submit your findings for verification. I'm sure there are plenty of engineers who would happy to be involved in a game changer of a finding like that.
I'm looking for the study that aes published
 
Oct 24, 2019 at 7:31 PM Post #42 of 124
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Oct 25, 2019 at 2:03 AM Post #43 of 124
Ok you took long time to write that but only thing that i want to see from you is sources that resampling 44.1to 44 doesn't matter now a days even it mattered 20years also pls stop giving be useless information and pls give me some useful information that i asked

Even more “oh dear”, because not only don’t you have any understanding of the basic relevant science but you can’t read, even what you yourself wrote!

You asked for the science and sources, I provided that. I also gave you the source of a published study that resampling from “high resolution” to 44/16 was inaudible! You on the other hand quoted a Hydrogen Audio thread comparing the analogue stages of 25+ year old devices, which was nothing to do with digital resampling. So YOU are being a HYPOCRITE asking me to stop posting useless information, when that’s exactly what YOU did and I provided exactly what you asked for, including sources! You though did not, you have STILL NOT provided any sources or reliable evidence to support your FALSE claims of the technical differences between bit depths/sample rates OR that resampling is audible!

Please stop being a hypocrite, it indicates that you are either a troll, a deluded, ignorant audiophile or both, and you presumably don’t want to give that impression of yourself, do you?

G
 
Oct 25, 2019 at 2:50 AM Post #44 of 124
He’s got a direct link to the study he wants, and all it costs home to learn is $33. If it isn’t worth that to him to know what he’s talking about, then I guess he’s gotten all of his other ideas from spurious sources for free too. Even the Wikipedia article on bit depth would be enough to point him to the truth and he hasn’t bothered to look at that. Whatever....
 
Oct 25, 2019 at 5:06 AM Post #45 of 124
Even more “oh dear”, because not only don’t you have any understanding of the basic relevant science but you can’t read, even what you yourself wrote!

You asked for the science and sources, I provided that. I also gave you the source of a published study that resampling from “high resolution” to 44/16 was inaudible! You on the other hand quoted a Hydrogen Audio thread comparing the analogue stages of 25+ year old devices, which was nothing to do with digital resampling. So YOU are being a HYPOCRITE asking me to stop posting useless information, when that’s exactly what YOU did and I provided exactly what you asked for, including sources! You though did not, you have STILL NOT provided any sources or reliable evidence to support your FALSE claims of the technical differences between bit depths/sample rates OR that resampling is audible!

Please stop being a hypocrite, it indicates that you are either a troll, a deluded, ignorant audiophile or both, and you presumably don’t want to give that impression of yourself, do you?

G
You, and others, took the time to share your knowledge and help the OP to understand his resampling query. Amazingly, rather than understanding what you wrote, he replied rudely and yet still expects you to make further efforts to educate him. What a dude.
 

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