16 bit files almost unlistenable now...

[adamdea]

 

3 hours ago, numlog said:

So over the past few months I've improved my system in all areas, its not the holy grail of sound but any further improvements will be very subtle refinements.

DXD and DSD128 is pretty spectacular but redbook makes up 99% of the music. As the system improved the limitations of redbook really start to show which is a disappointing surprise.

 

Sample rate is important but I did not realise just how critical higher bit depths are, 16 bits is just way to low.

Even with the best 16bit recordings and I can now notice this distinct hazy and compressed sound , it's really surprising how clearly audible it is once you get to moderately high volumes.

 

Luckily 24 bit 44.1kHz are pretty common now but there is so much music, particularly electronic and hip hop, where its true essence will be lost inside this inferior format.

 

Have a go at downloading the test files here and see whether you have a preference for each pair. If so note them down.

https://archimago.blogspot.com/2014/04/internet-test-24-bit-vs-16-bit-audio.html

There is no haze or compression with 16 bits relative to 24 bits, just a higher quantisation/dither noise level. [this was covered in a thread recently] However for most recordings the background noise level is much higher than the 16 bit quantisation noise. If you want to (actually) hear the difference its best to use stupidly amplified digital silence. 

 

There was an article by Bob Stuart years ago, supposedly arguing for hi rez where he showed that if you ignored masking [which is absurd] you could just hear triangular pdf dither noise in a 16 bit recording if you played it at 120dB peak level [which you can't or you would be deaf]. Even then you wouldn't be able to hear the quantisation noise if you used noise shaped dither.

[adamdea]

15 hours ago, mansr said:

You must use dither. Otherwise there may well be audible degradation.

and triangular dither not rectangular. Noise shaped dither might be better but on the other hand it might not, so perhaps best avoided for a 24/44  comparison with 16/44.

[adamdea]

1 minute ago, mansr said:

Shaped dither at 44.1 kHz is tricky as it easily becomes audible. At 88.2 kHz and up, there's a wider range of completely inaudible frequencies where noise can be dumped without issue.

It's a risky business trying to dump quantisation noise in such a small, potentially audible band. Especially when the only point in it is to ensure that you couldn't hear the quantisation noise  in a silent passage with the volume on max.

[adamdea]

1 hour ago, Summit said:

One of the pros of oversampling is that the noise-shaping, filtering etc can be done at much higher frequency than what the recording has natively.  

Do you mean oversampling in a dac? If so I'm not sure I follow about the noise shaping. Or do you mean oversampling in an adc (ie using a higher sampling rate than the final distribution format). If so I follow.

[adamdea]

14 hours ago, Summit said:

 

Yes I meant oversampling in a DAC, although filtering and noise-shaping can be made in both an ADC, DAC or in computer software like HQPlayer.

 

Any signal, can be frequency shaped using filters. This is what filters do – they change amplitudes of frequency components of a signal that passes through them. They can make high frequency components weaker (low-pass filter), or can make them stronger (high-pass filter) or can frequency-shape a signal any way you want.

 

We already used some frequency shaping in our oversampling chapter – to cut off high frequencies from a quantized signal (we weakened high-frequency components down to zero).

 

The quantization noise can be shaped to improve signal-to-noise ratio of a quantized signal. But how do we do it?

 

We cannot simply place a filter at the output of our quantizer because the quantizer must be the last element in our digitalization chain – if we place a filter after it, the output signal from filter will not be quantized any more and we will not be able to use it for a digital computer.

 

Imagine that you oversampled your signal and now you have a lot of frequency space at higher frequencies that are not used by your actual signal. Only the quantization noise occupies this region. Now we could simply cancel that noise portion by filtering out these frequencies as explained in the oversampling chapter. However we can do much better – let’s re-shape the quantization noise so it will mostly be pushed away from lower-frequency regions and forced into higher-frequency regions and just then we cancel it out. That would be really charming.

 

http://charming.awardspace.com/delta_sigma/delta_sigma.html

This is about A/D and what they are talking about it shaping the quantisation noise from the delta sigma conversion from analogue. 

I don’t think it applies to a dac except in relation to requantisation- eg the additional noise in the delta sigma conversion when turning the (usually) 16 bit pcm data into low bitrate data. 

But this does not afaik affect the shape of the quantisation noise implicit in the original data. 

Happy to be corrected if I have misunderstood. 

[adamdea]

 

2 minutes ago, Summit said:

 

Yes it’s an explanation that noise-shaping, also can, and is often used for Red book because of oversampling to higher frequency.

 

Well sort of... It's just a pair of quotes explaining how a ds dac works. Of course the oversampling  and noise shaping is "used for red book" in the sense that it used to convert redbook to analogue but it has nothing to do with noise shaping the 16 bit quantisation noise.  TBH the distinction between quantisation noise in the recording and requantisation noise caused by the conversion process is a bit tricky and is not clearly explained in most generalist texts.

It's easy to hop from a/d conversion noise to 16 quantisation noise to d/a conversion noise in an explanation because of  course the chain is only as strong as the weakest link. Without da conversion noise shaping,  (most?) ds dacs would not achieve 16 bit performance in the audible band. Equally if you really want to see the a/d converter noise floor then you need to buy 24/96 or ideally 24/192 (in fact that's pretty much all you get going from 24/96 to 24/192)

[adamdea]

25 minutes ago, Jud said:

 

I of course agree with most of this (can't do otherwise, since it's factual).  The single piece I'd like to have a little conversation over is the thing about ignoring masking being absurd.  Certainly, ignoring masking from ambient sound is absurd, and that is what we mostly are talking about regarding noise floor.

 

But I also have read people saying that music masks noise/distortion from the equipment or recording.  While also obviously true, since we are trying to hear the music rather than the noise, I would think we'd be concerned about noise/distortion masking (interfering with clearly discerning) low level details in the music, rather than vice versa.

All I was getting at was that if you have music playing so that the peaks are at 120dB then your chances of hearing any noise (or anything else) at 90dB below peak is pretty slight. Your ear has a sort of variable dynamic range so when it hears loud things it reduces its sensitivity  to avoid damage. So the quietest sound can hear i the presence of a loud signal is very different from the quietest sound you can hear period. 

 

Putting it another way- masking is the main reason why perceptual codecs work.

 

Noise could mask signal in principle but

1) you can hear a tone 20 db below the overall noise level because it will still be higher than the noise level in the relevant bin. so 16 bit allows your to hear tones at -110 dB below (theoretical) peak

2) with a changing signal (aka music) and a constant noise won't your brain focus on the signal?

[adamdea]

1 hour ago, esldude said:

Here is a quick example.  

Pink noise at about - 40 dbFS and less than that from peak levels in the music.  Noise alone for the first few seconds and then with the music. Bonus points if you can tell me when the noise goes away.  It isn't impossible, you can hear when it is removed partway thru the snippet, but I think it is harder than some might expect.  This level of noise would be like bad cassette tape with no Dolby noise reduction.

 

 06 Heat Wave.mp3

11-12s?

[adamdea]

6 minutes ago, Jud said:

So from the examples @semente has come up with, maybe 40dB to just slightly more for really dynamic material (short of some symphonic music with modern instruments - the performances with period instruments that I've heard tend to be a little quieter).

I’m not sure where you are going with this.

 

I do however feel confident you will struggle to hear 16 bit quantisation noise except by turning up the volume in a silent passage (of a test tone “recording”. For most real recordings the noise in the recording is 10 dB higher or more.  If you are interested playing at 100 dB peak level at home then the room noise level is around 70 dB down in a quietish room. 

 

What is the level below peak of self noise in most mics -80dB? Genuine question- I’m not sure but I have that figure in the back of my mind. 

[adamdea]

In relation to background noise and intelligibility-

Is it significantly more difficult to hold a conversation in a quiet room in the day than it is to hold one in a quiet room at night? 

[adamdea]

On 7/20/2018 at 8:20 PM, Jud said:

 

Interesting thought. You mean not being able to see the speaker's face in the dark - and when replaying over an audio system?  Or were you thinking of something else?

I meant that background noise level would probably be much lower at night. But I don’t reckon it would make that much difference. Come to think of it, it was probably a crap point. 

[adamdea]

11 hours ago, numlog said:

I discovered something interesting today regarding 24+ bit files.

 

The fre:ac converter alpha allows you to package 16 bit files in 24 or 32 bit container, I was using it to convert FLAC to WAV initially but ended up exploring this feature... Im a big fan of insignificant improvements

 

Comparing a few 16 bit WAV conversions of 16 bit FLAC to  32 bit WAV conversions of 16 bit FLAC I feel that the differences are close if not the same as a 24 bit file and 16 bit down conversion of 24 bit file - the 32 bit version is superior

 

Anyone who notices an audible difference between 24 and 16 bit files should give this a try and see if they can hear anything. This could mean that the  benefit of higher bit depths has nothing to do with extra recorded data, instead it's additional unused headroom of a 24 bit package that makes it better (depending on your DAC/PC/USB Cable etc.).

So potentially any file could be 'converted' to 24 bit.

if your system can already process those original 16 bits natively without an issue then they're may not be much point.

 

please dont kill me

 

 

 

 

 

 

 

 

There is a school of thought that when comparing eg 24/96 with 16/44 you have to upsample/convert the 16/44 to 24/96 so thatt you are then comparing the content of the files rather than (possibly) the different ways in which the dac treats 24 bit files vs 16 bit files and/or different ways the dac treats 44.1lkhz files from 96khz files.

[adamdea]

23 minutes ago, PeterSt said:

 

So my analysis software would show differences if they were there at all, but I don't even wanted to know about it because I wouldn't be able to explain differences if they were there to begin with. I know enough of computers to know that no storage mechanisms exist to store the data which would be able to incur for different sound, unless what I just described for physical representation but which I eliminated as explained (OK, told about).

 

Kind regards,

Peter

 

I have no idea what your are saying but i would love someone to set this to music, possibly with dancing, or at least mime.

[adamdea]

13 minutes ago, Richard Dale said:

I am a 'subjectivist' and I believe in all sorts of daft things such as preferring tube amplifiers to transistor ones, and thinking RCA interconnects sound different and so on. But I am also a computer programmer and I know what a Unix/Linux 'file' is. I am amazed that we have not been able to explain to sandyk why his ideas about files can't possibly make any sense. So perhaps I'll attempt to explain to non-computer science experts what a 'file' is.

 

A file is a sequence of bytes, where each byte is a pattern of eight '1's and '0's. That is all the 'file' is, and it is the definition of a 'file', no more and no less. A simpler sort of 'file' might be called a 'byte file', which is just a single byte consisting of eight bits where each bit is either '1' or '0'. You can write down the contents of a 'byte file' on a piece of paper, for instance as '01110011' would be a value, and then you could  pass that piece of paper to someone else.

 

You could also represent this 'byte file' using a line of eight people holding flags where a person representing a '0' would hold the flag downwards, and a person representing a '1' would hold the flag in the air. It would be then very easy to transpose the 'byte file' represented by the line of people to a sequence of '1's and '0's written on a piece of paper and pass that piece of paper to someone else. 

 

That other person could decide to represent the 'byte file' as a sequence of eight glass bottles on a wall, which were either green or white glass, where the green bottles would stand for a '1' and the white bottles would stand for a '0'. You could look at the eight bottles and call over to a line of eight people holding flags and tell them how to hold their flags, either 'up' or 'down' to represent the same 'byte file' as the bottles version.

 

Because the meaning of a 'byte file' is independent of how it is represented it doesn't make any sense to say that it is possible that there could be extra values associated with each 'byte file' beyond the sequence of eight values, or that a perfect copying mechanism for transposing one representation to another must always fail in the same way. That is what would need to happen if copying computer files, was in some way like making imperfect photo copies. It isn't a physical thing like that.

 

Maybe I've still failed, but flying pigs are physical things and computer files are not - they are an abstraction entirely independent of any particular physical representation.

 

A very famous computer scientist and mathematician called Alan Turing described a very simple computer in terms of a paper tape reader and some operations that it could make when reading the current symbol on the paper tape. In his theory he was able to say that this very simple computer was exactly equivalent to any type of more complicated computer. That is on the same lines of my descriptions of representing what I have called a 'byte file' with humans with flags, lines of bottles or with digits on paper. They are exactly equivalent, there is never anything more and never anything less when you transpose one into another.

Good heavens, do you mean to say that the information content of a digitally encoded signal may be largely independent of the accidents of the medium in which it is stored or transmitted? Perhaps this may have implications for digital audio.