16 bit files almost unlistenable now...

[esldude]

4 hours ago, AudioDoctor said:

Well there you have it folks, GUTB has spoken...

 

GUTB, I give your pronouncements the exact same amount of merit I give those of Donald Trump...

I think you are giving GUTB too much credit in that case. 

[esldude]

20 hours ago, GUTB said:

 

MP3 = OUT OF THIS BODY SATAN

 

Well, to be honest -- which I always am -- modern psycho-acoustic processing on these lossy compression schemes work wonders. About 2 years ago I tested VBR MP3 with the latest encoder vs FLAC and I seriously could tell NO difference. If there was a difference it was so small it couldn't break past memory effect / test stress.

 

STILL, it's better to store music in a lossless format simply due to the cheapness of storage, future flexibility and easier CPU load. 

 

Your speakers and amp are pretty entry level, so I wouldn't go crazy cleaning up your source, but some clean-up is recommended. At the very least a Jitterbug on your USB connection to the Teac, and I strongly recommend power conditioning, even inexpensive ones (check china-hifi and AliExpress, Panamax on eBay) will help the Teac, and POSSIBLY the Marantz. If you can't afford even this, at LEAST run an extension from another room's power circuit to power the Teac and Marantz -- in fact, you should do that anyway even with a power conditioner.

 

You should do the bolded part above if you wish to create the chance for ground loops which may cause hum or noise that otherwise would not be there.  

 

Better advice is don't take advice from GUTB.  He usually is wrong.  

[esldude]

18 hours ago, Sonicularity said:

 

Thank you for the advice.

See my post just above. 

[esldude]

39 minutes ago, Sonicularity said:

 

I am not weak in my mathematical knowledge and deductive reasoning.  I don't recall establishing my financial means; though, I suppose it was simply assumed from my comments.   Neither of us will probably ever change GUTB's perspective on audio gear and its relationship with cost.  I recognized the futility of continuing on in that direction.  

Giving advice based upon price is one thing, and has pitfalls of course.  Giving advice that is obviously wrong for technical reasons and likely to induce problems from following that advice is another.  Using an extension cord probably makes sense to GUTB because of his idea about how digital things are all noisy.  He's fairly wrong about that. Using an extension cord for another circuit for that wrong-headed reason has the additional problem of likely increasing system noise and hum over ground connections.  So definitely bad advice.  He does this often enough I don't care about changing his mind.  I hate to see his advice be taken when it is so often wrong. 

[esldude]

52 minutes ago, Cary said:

 

Thanks for ruining my evening.  Now I have to go read about mp3 v aac.  

Actually more like AAC is better at lower bitrates like 128k or less.  Maybe a tiny bit better at middling bit rates.  Little if any categorical benefit between them at 220k or more.  If you really want to be bothered throw in Ogg Vorbis to compare. 

 

If possible, yes lossless is what you want. FLAC or ALAC or WAV or AIFF.

 

Here is a chart showing quality in one simple graphic.  No one is claiming 128 kbps is transparent vs lossless.  But they get near that once you starting going above 128 kbps.  Therefore none of the formats have a decisive advantage at the upper rates.   Opus btw is an open source codec from xiph.org.  Soundcloud can/does use Opus. 

 

http://opus-codec.org/comparison/

 

[esldude]

12 minutes ago, sandyk said:

 Even at 187Kilobits, .aac and other lossy formats sound way below what is obtainable from16/44.1 .wav or ,flac unless your system isn't very resolving .

Well these results are from carefully done evaluations of codecs.  

 

Can I hear a quality deficit at 160 kbps?  Yes I can.  At 220 kbps it gets harder, but is still there.  At higher rates I have trouble hearing the difference, but can.  Obviously that is what one would expect the only issue being how low is acceptable.  So your post doesn't really disagree with anything I've posted.  

 

By the time you get above 128 kbps the difference window between that and lossless is getting smaller and necessarily the difference in various lossy codecs is less.  At 128 kbps and less AAC is better than MP3, and maybe Opus is better than both.  Much above that point, it is less important which codec you use for lossy.  

[esldude]

16 minutes ago, sandyk said:

Dennis

 I can't stand 128Kilobits .aac etc. With a previous Video Downloader if I wanted Audio to better match the quality of the Video I would DL both the 1080 and 1920  versions, and replace the audio of the lower bitrate 1920 x 1080  version with the 187Kilobits .aac of the 1080 x 720 version.

 

Alex

 

 

 P.S.

 Further improved differences in that other area now.

I don't like 128 either for any length of time.  To me, and this likely is variable in people, 160 kbps is a large subjective improvement.  It still is clearly not lossless, but the annoyance is noticeably reduced.  The difference between hearing a lack of resolution and hearing an annoying additive is brought down. When I listen to internet radio, it has to be something of special interest for me tolerate 128 kbps.  I like to stick to 192k and above. 

[esldude]

7 minutes ago, GUTB said:

 

I'm neurotypical. My advice is based on personal experience; sometimes I have the theoretical knowledge to explain my experiences and sometimes I don't. I'm not a prophet, just an audiophile.

 

Running an extension from another room with an unused circuit is because you want your audio gear off the circuit with all the trash that would be on a general household circuit such as switching PSUs, AC motors (fridges, fans, etc), lights, dimmers, etc and so on.  If you have a religious belief that power delivery doesn't matter than it might seem like segregating the power circuits this way is bad advice. 

Yes, quite often, despite the imagined hash on the circuity you measure little or no difference in signal quality coming out of gear. 

 

However, connecting grounds between separate circuits of the wiring system in your house is nearly guaranteed to increase ground current between devices and cause an increase in hum and noise.  It might be hardly noticeable (though quite easily measured) or might be an obvious hum or buzz heard at the speakers.  

 

You give technically, demonstrably, wrong advice at a 10:1 ratio to the bad if not worse.  This is just one more example.  You are the one with the religious power belief.  Unfortunately it is a wrong belief..........again.  You are unwilling to entertain the idea you are incorrect.  So the best thing is to warn people against your wrong-headedness.  

[esldude]

4 hours ago, gmgraves said:

Pedanticism, thy name is Firedog! The point was that DSD cannot be edited as DSD, It has to be edited in another format. There is nothing that says that the DSD file couldn't be converted to 33.3 RPM stereo lacquer transcriptions to be A/B roll edited (like video tape) to a third 33.3 RPM stereo lacquer transcription, before being converted back to DSD. But I'm reasonably sure that a similar number of record companies do that as transfer their DSD recordings to analog tape to edit them. 

Actually there are at least a few who do have a mix and tape step in the middle with DSD on each side. 

[esldude]

28 minutes ago, mansr said:

Better than what?

Better than actual high fidelity.

[esldude]

2 hours ago, GUTB said:

 

Pick up a Mapleshade CD -- every singe one is mastered from a tape --  and compare it to ANY of your digital-sourced music you'd like.

 

The only exception to this is native high-rate DSD recordings played back on a native DSD DAC. Native high-rate DSD recordings present a realness and inner-detail, clarity, etc, that is unmatched in the realm of digital media.

You are talking at cross purposes.  Highest fidelity is not always unerringly preferred by audiophiles.  Tape is a lower fidelity medium.  Plain and simple.  That is that. 

 

Tape has a sound of its own.  You like it, many people like it, doesn't make it of higher fidelity.  Were CD not of good fidelity it wouldn't transmit the preferred qualities of a tape source.  

[esldude]

8 minutes ago, numlog said:

So far you either have people saying  there is no audible difference at all or it's 'nearly identical'.

 

If its not identical then your post has little meaning on an audiophile forum. the difference is small but extremely important, it is the absolute beginning of the chain after all.  Isnt the whole point of being an audiophile to strive for the ultimate sound? why discount a possible improvement just because it's small?

 

You may appreciate it's importance more with time or as your system improves but at least you and a few a others can admit there is something there.

 

I think you are over-simplifying. 

 

"..........or it's 'nearly identical'."  Hardly fits with 16 bit files almost unlistenable.

 

So you say its almost unlistenable and then say don't discount a small difference.  Which is it, something you just almost can't bear to listen to, or a bit of a difference you finally can detect?

 

If its the latter, probably not too many people are going to disagree.  16 bit done well is probably indistinguishable for playback.  But it has less room for error and often possibly even usually isn't done well.  So hearing a difference isn't far out there. 

 

On the other hand, the idea 16 bit, even at its finest sounds obviously inferior to the point you hardly can stand listening to it.....not very believable.  Or you have a personal problem where the tiniest little barely perceptible difference from perfection drives you crazy.  And if the latter, good luck, as music in any format will leave not even 1 in 1000 available recordings acceptable to you. 

[esldude]

38 minutes ago, numlog said:

Question: How many people use planar magnetic headphones here?

I don't listen too much to headphones.  I've owned some Stax Lamdas, Koss ESP9's and some more normal dynamic headphones like DT880s.  

 

I listen to Soundlab ESL speakers.  Don't know how that compares to planar magnetic headphones not having listened to any.  I would think the electrostat phones could work as well as planar magnetics.  They have some advantages in speakers over planar magnetic speakers like Magnepans.  

[esldude]

45 minutes ago, numlog said:

You are right the title is misleading, its mostly the obsession with perfection that makes it 'almost unlistenable'. It's sickening to have spent so much time improving something then to have no control over the core part.

snippage........

 

Don't ever start recording then.  I've done a little recording.  Making high fidelity recordings isn't really much of a puzzle.  However, there are a myriad of reasons such things are a commercial non-starter.  And almost as many as why even non-commercially you can't get there from here.  Most available recordings are a shadow of the full fidelity they could be.  But you can't listen in the car, you can't listen over your phone when out and about, and you aren't constantly bombarded by notable sounds nobody wants them.  Frustrating to know what is possible and what you can get people to accept.  Simple recordings that aren't processed can sound so nice, calm, and offer good verisimilitude of reality.

 

If you want to try some nice ungimmicked recordings try those Mario Martinez has offered for download to members here on CA.  Play Classics.  Here is a recent offering. 

 

 

[esldude]

4 minutes ago, Summit said:

 

“Modern delta-sigma converters rely on a combination of internal high-speed upsampling and noise-shaping algorithms (digital feedback wrapped around the switch array) to attain their performance. Without dither + noise-shaping algorithms, you only get 12-bits to 14-bits of resolution, and we'd all still be using ladder converters.”

 

"Single-bit converters (and the SACD/DSD system) rely on dither and noise-shaping even more. Without noise shaping, SACD/DSD at 64fs (2.8224MHz) would only have 6-bits, or 36dB, of dynamic range, about the same as a not-very-good telephone connection."

 

[R2R DAC]…: “The ideal noise spectra is a "triangular" spectra centered on the Nyquist frequency of the physical converter—in other words, if the converter is running at 352.8MHz, the center of the dither-noise spectra should be 176.4kHz.”

 

https://positive-feedback.com/Issue66/dsd.htm

 

“A Sigma-Delta based DA converter realizes a high SNR with the use of a DAC with few quantization levels and noise-shaping techniques. In the digital domain the input signal to the DAC is shaped, such that the quantization noise of the DAC is moved to high frequencies. In the analog domain a passive low-pass filter removes the quantization noise, resulting in a clean baseband signal. The structure of a Sigma-Delta DAC is, except for some special PWM systems, a feed-forward solution, i.e. there is no feed-back from the analog output into the noise-shaping filter. Because the noise-shaping feed-back signal is not crossing the analog-digital boundary, the name Sigma-Delta DAC is confusing and misleading. A Sigma-Delta DAC is the combination of a DD converter and a high-speed few-bit DAC. In Fig. 2.6 the complete Sigma-Delta DAC structure is shown. The digital n-bit input signal is passed to a DD converter which upsamples the input to N × Fs before an all digital SDM reduces the word-length. The noise-shaped m-bit signal is passed to the mbit DAC which converts the digital signal to the analog domain. Finally the analog signal is filtered to remove the out-of-band quantization noise.”

 

file:///C:/Users/AS/Downloads/9789400713864-c2.pdf

 

Is there some point to your post related to the thread topic?

[esldude]

29 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. If the OP thinks my posts are OT I will ask Chris to erase them. Ok?

I am still searching for the point.  You referring to noise shaping in sigma-delta DACs, or noise-shaping during mastering before being sent to the redbook format?  Does this have something to do with noise shaping making redbook unlistenable or sound worse?

 

 

[esldude]

19 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.

Well we hear some 10 or 15 or maybe even 20 db into noise, but we don't hear noise 20 db into music?  Why?  Look at how masking works. 

 

One of things our ear does is break up things into maybe 30 or 32 bands of frequencies.  So some music can be at a level so that SPL wise it is 20 db below the noise level over the whole band we can hear, but the music content is still above the part of the noise in that band our ears are filtering with.  Something like the way a tone can be unmasked in an FFT even though the signal level of the tone is below the noise level for the entire band.   The ear is more advanced than a static FFT in that it moves those filters around.  Still it moves those around looking for tones which within the ear's filters is above noise in that filter band.  

 

The reverse with our ear hearing noise above music doesn't work.  No doubt that is why our ears work like they do.  Eventually as the signal level of music descends far enough into noise yes the noise interferes with us hearing it at all much less details of it. 

 

The noise levels in 16 bit even with dither and noise shaping are low enough it is difficult to come up with a situation where it can interfere with low level detail.  Not impossible, but very difficult.  Very, very few recordings have a noise floor low enough for 16 bit to become an issue in terms of noise. And by very few I mean like probably not even .1% maybe not even .01% so basic noise issues with 16 vs 24 bit isn't why someone is dissatisfied with 16 bit. 

[esldude]

1 minute ago, adamdea said:

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?

Instantaneous dynamic range of the ear is around 60 db.  The rest is what you are talking about.  How quickly other effects expand that and how audible things are gets complicated.  But yes, even if you had playback peaking pretty often at 120 db and noise 90 db down, which by itself would be audible, you'll never hear it with that signal as it masks that level even for a period of time after it drops away.  

[esldude]

14 minutes ago, mansr said:

That's not a meaningful question. Noise, in the sense we're discussing here, is a wide-band signal with a smooth spectrum. When noise level is given as a single figure, it is the integrated power, possibly with weighting, over the entire frequency range of interest. A tone or music, which is a collection of tones, with the same SPL has the power concentrated in a few narrow regions. The spectral intensity, measured in dB/Hz, in these regions is much higher than that of the noise. That is why we can hear the signal even in the presence of noise with higher total power.

 

I think you were saying something similar, if in a roundabout way.

Yes that is exactly what I was saying.  

[esldude]

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

[esldude]

3 hours ago, danadam said:

I have no words to describe the first 20 s but after that it's almost unlistenable

 

Would by nice to have, let's say, -60 dBFS version to compare.

I could add that later today.

[esldude]

Okay these are FLAC files.  - 40 dbFS pink noise in one and -60 dbFS pink noise in the other. 

 

 

06 Heat Wave 1.flac

 

06 Heat Wave 2.flac

[esldude]

2 hours ago, adamdea said:

snip...

 

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. 

Depends upon the microphone, recording, mixing etc. 

 

Plenty of condenser microphones have a self noise equivalent to 10 dbSPL, and can stay at less than 1% THD to 140 db.  Of course any preamp gain will amplify that noise.  The more limiting factor is where are you going to record that has a noise level down at the same low leveL?  Also the distribution of noise can be such that in the midrange 3-5 khz it might be that low. 

 

Let's assume a good condition.  I think you'll need 30 db gain to record say a good orchestra which raise that 10 dbSPL equivalent to 40 dbSPL.  You'll probably have that much noise in the room anyway.  So you're lucky to get 70-80 db dynamic range above the noise floor.  Again at midrange frequency that region can be fairly low in level.  

[esldude]

Yes.  Not sure what I did, but going by peak levels I had noise at -33 dbFS instead of -40 dbFS.  Sorry, a mistake was made somewhere.  It looks like I did a mix and render on the -40 db noise twice.  Which works out to 33 or 34 db on the end result.  Oops!

 

So the FLAC files look as I intended.  So you can say where the noise ends via listening. 

 

HINT: it isn't the same place the noise ended in the mp3.  

[esldude]

Funny how this disclaimer on nearly all early CD's is forgotten.  Nearly all early CD were AAD.  This is a simple statement of fact about the two mediums. 

 

"The Music on this Compact Digital Disc was originally recorded on analog equipment. We have attempted to preserve, as closely as possible, the sound of the original recording. Because of its high resolution, however, the Compact Disc can reveal limitations of the source tape."