16 bit files almost unlistenable now...

[mansr]

I have both CD quality and hi-res music in my library. Now and again I'll be listening to something with unusual clarity and think to myself, maybe there's something to hi-res after all. Then I look at the info display and notice it's plain old CD quality, just very well recorded.

[mansr]

4 minutes ago, kumakuma said:

My own experience of comparing a number of recordings with the same mastering in different resolutions has convinced me that the differences between Red Book and higher resolution formats are quite subtle WHEN everything else is kept constant.

It's actually very easy to test this. Take your hi-res recording of choice and reduce it to CD quality using any of the numerous programs available, then compare the result to the original. This way you know for sure that the same master is used for both formats.

[mansr]

1 minute ago, kumakuma said:

Hello Kitty is Japanese. Japanese audiophiles are the best. Therefore, these headphones must be awesome. Or awful. Engrish is very tricky language.

[mansr]

2 minutes ago, fas42 said:

It's highly instructive to take some hi res tracks from a recognised source - I use those from 2L - and subtract out the extra stuff after creating a CD quality version; and then look at, and listen to the hi res elements after amplifying by enormous amounts - ummm, absolutely nothing of special value. Sometimes, just transients from accidental, non-musical bumps and scrapes in the recording space.

 

If it sounds obviously different, then it's different masterings, or a playback chain that is poor at handling 16 bit format.

Sounds about right.

[mansr]

1 minute ago, Hugo9000 said:

 

Please note the very advanced clock.  And the fact that it's a TOP MOUNT CD player!!  How audiophile is that?

But is it a Tice clock?

[mansr]

1 minute ago, AudioDoctor said:

I misspoke, it is both a tape and CD player.  Also, an integrated amplifier.

The bass is a bit squishy.

[mansr]

4 minutes ago, AudioDoctor said:

Haha, well that happens in a pineapple under the sea...

I'd like to be under the sea

In an octopus' garden in the shade.

[mansr]

6 minutes ago, cjf said:

The tone deaf millennial artists will ensure the death of anything that resembles real HiFi in the future. Do they even use real instruments anymore?

I know a few local musicians. They all play very real instruments. Quite well, too.

[mansr]

Just now, sandyk said:

With a 1,000W PMPO output too ?

PMPO, Purposefully Misleading Power Overstatement.

[mansr]

Quote

As for PWM, perhaps I made an error and it's Pulse Density Modulation that DSD uses? My point there was that DSD/SACD does not, to the best of my knowledge, use conventional PCM. That's true, yes?

DSD can be seen as either:

• 1-bit PCM with values of 1 and -1
• PWM with pulse widths of 0 and 1
• PDM

Mathematically it is all the same.

[mansr]

1 minute ago, gmgraves said:

No thanks, Don't want to hear that kind of music. But thanks for the correction.

It's clearly a parody, a joke. Mute the sound if you don't want to hear it. It's still funny.

[mansr]

7 hours ago, diecaster said:

MP3 bit depth is always 16 bits.

That's not quite true. The output of the inverse transform generally has a precision exceeding 16 bits even if the input does not. Truncating it back to 16 bits increases the distortion somewhat. The standard doesn't specify how an encoder should handle the input resolution. Best results are obtained by doing the transform in 32-bit (or better) precision, then encoding the output as closely as the format permits. The end to end performance depends on the complexity of the signal. Simple signals are coded more accurately than complex ones. That said, don't expect mp3 to exceed 16-bit precision with anything resembling music.

[mansr]

4 hours ago, esldude said:

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

@cookiemarenco uses both tape and DSD. I'm not sure whether she uses that particular flow, though.

[mansr]

28 minutes ago, cookiemarenco said:

Thank you for the mention, mansr.  I'm not sure what the 'flow' is you've mentioned, but yes, we use both DSD and tape.  Here's why...

I enjoy working on 2" tape for multi-track recording, with more than 8 tracks  but tape is very expensive.  

My guess is 90% of our basic recording starts on DSD256 where we can keep more takes (meaning multiple performances of the same song).

 

I have owned a commercial recording studio since the early 80's and was part of the development / beta testing for the first digital audio recorders on the computer.  By the mid 90's I was tired of digital sound from PCM and returned to working on tape.  In the early 2000's we were introduced to DSD recording and developed techniques we use today (running through the analog console to mix from DSD and output the stereo mix back to DSD).  We did (and still do) maintained the digital gear necessary to run mastering sessions for all formats.  

 

We were encouraged to try DSD256 using the Pyramix system (the only system that handles DSD256 recording).  I'm not a fan of going to DXD to mix.  I don't enjoy the sound after recording to DSD (I'm sure if you're reading this you'd have the ears to hear the difference in the studio as we do).    I do understand that many engineers don't have access to a large analog console with all the outboard gear necessary to mix and for them it is necessary to mix inside the computer in DXD.'

 

About editing in DSD...  95% of our edits stay in DSD without conversion to DXD.  Because we are mixing through an analog console and back to DSD, if the edit sounds good, we have found it's not necessary to convert to DXD.  The style of recording we now specialize in (acoustic / live) also lends itself to this.

 

If we have a session that is more complex or pop oriented, we will assemble the basic performance on DSD, transfer to tape and do the overdubs on tape.  It's faster and easier.

I was referring to going from DSD to tape and back to DSD, as mentioned by esldude. Sounds like you sometimes do that, and sometimes you skip the tape and record the analogue mixer output directly to DSD.

[mansr]

31 minutes ago, gmgraves said:

Of course I know the word pedantic, I consider myself an educated man.  I brought up the pedanticism because my point was that it's extremely difficult to edit in the DSD format, I didn't think that the point needed expanding. But you are right, I didn't know that some record companies convert their DSD to tape to edit it and I thank you (for what it's worth) for the information. My reaction to that is why bother? Why not capture the performance on analog tape, edit it with a razor blade, and then transfer the edited tape to DSD? That certainly makes more sense than doing a double conversion. Of course DSD recorders are a lot smaller than a pro analog tape deck, and that might be a legitimate reason for capturing a performance in DSD. I don't think it's anything I'd ever do, but then, I record live performances and don't do much in the way of editing, so that point is moot with me.:)

If tape can be avoided entirely, you won't have to deal with flutter and hiss. The degradation caused by encoding to DSD twice is minor in comparison.

[mansr]

18 minutes ago, GUTB said:

In regards to tape transfers, I'll point out that in my experience -- without exception -- digital music from a tape master source ALWAYS sounds better.

Better than what?

[mansr]

15 minutes ago, numlog said:

For conversion foobar2000 was used,no dithering, 24 bit FLAC to 16 bit WAV.

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

[mansr]

Just now, adamdea said:

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.

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.

[mansr]

10 minutes ago, adamdea said:

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.

At 44.1 kHz (and 48 kHz) there's really no safe place to put noise. Dither shaping at these sample rates relies on our hearing being less sensitive at higher frequencies. Allowing the noise level to rise a bit beyond, say, 12 kHz gives a little more dynamic range in the lower region. The trouble is that the 10 dB or so difference in sensitivity between 5 kHz and 10 kHz is barely enough to allow any useful shaping before the increased noise level at high frequencies becomes audible. With higher sample rates, you can readily raise the noise level by 20 dB or more in the 40 kHz region without anything bad happening. Even there, though, if the level is too high, IMD will shift the noise back into the audible range.

[mansr]

1 minute ago, danadam said:

AFAICT shaped filter for >48kHz is not available in sox. Do you know why?

Nobody made a suitable one.

[mansr]

2 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.

Masking simply means one sound becoming indiscernible in the presence of another dominating sound. In the present context, we're talking about low-level noise being masked by louder sounds, either in the recording or from the surroundings. While 16-bit TPDF dither noise is generally not audible at normal playback levels, older recordings sourced from tape often have a background hiss easily discernible until the music starts, at which point it is no longer noticeable.

[mansr]

Just now, PeterSt said:

Nice theory.

What's that supposed to mean?

[mansr]

6 minutes ago, esldude said:

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?

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.

[mansr]

18 minutes ago, adamdea said:

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

I did a quick test. Using the iFi Nano at maximum volume and headphones (low-end Sennheiser), I can readily hear a 3 kHz tone at -105 dBFS and with difficulty a bit lower. With added 16-bit TPDF dither, the tone is still easily audible alongside the noise. Shaped dither at 48 kHz sample rate is difficult or impossible to hear. A 0 dBFS signal would be painfully loud, of course.

[mansr]

18 minutes ago, Jud said:

This is interesting, especially @mansr's explanation of how noise level and signal level are measured, which I now recognize as what @esldude was alluding to earlier.

 

I want to move from the more theoretical (I'm never going to intentionally listen to music at 0dB attenuation) to the everyday.  Can someone tell me what a ballpark difference in peak vs. average dB level would be for a track that gets a rating of, say, 12 on the DR Database ( http://dr.loudness-war.info/ )?

A random sampling of a few DR12 tracks suggests a typical peak to RMS difference of around 15 dB.