16 bit files almost unlistenable now...

[Blake]

6 hours ago, rando said:

When did this place suddenly become populated by 30 something females and their accompanying tastes in music?   

 

6 hours ago, rando said:

I was plumbing the deep vein of female interest in audiophilia for starters.  Much of the music discussed in here is prime repertoire for the age group and sex I mentioned. 

 

Rando, your expertise with respect to the electronic music and hip hop genres is most informative.....  keep going please, tell me more.  

 

God wiling, with your informed advice, I can graduate from the electronic music and hip hop that I listen to now, to whatever music it is that you enjoy.

 

Whatever music it is, I am quite certain I am not yet ready or worthy to receive it, but with your guidance Sensei Rando, all things are possible!  

[gmgraves]

12 hours ago, firedog said:

And your point is? I didn't say anything about one medium being intrinsically superior to another. 

But I've compared such transfers of tape to both DSD and PCM - I like the DSD better and think it sounds more like the tape.  It's my perception. Why do you feel a need to chime in? I don't think you have the ability to tell me what I'm hearing. 

I'm not trying to tell you what you you're hearing! I wouldn't dream of it. You said that you will continue to buy high-res and DSD because you felt that they might sound better in the original format. I was merely commenting that my experience says t'ain't necessarily so! That buying in the original format is no guarantee that the recording will sound better. Awfully sorry for the confusion. And for what it's worth, I agree that DSD, done right, sounds very analog tape-like - only without the tape hiss, scape flutter, and low-level noise modulation.  

[gmgraves]

9 hours ago, semente said:

 

It's not just the mastering engineer that is to be blamed:

 

 

I agree and I didn't even have to listen to your example, to do so! I can imagine that it sounds just wretched - just from looking at the still frame picture, above.

[tmtomh]

1 hour ago, jabbr said:

 

What are you thinking about “rather aggressive” noise shaping? The implication is that this is a bad thing. Wouldn’t you want to send quantization noise out into the MHz range?

 

Also SACD/DSD/SDM isn’t the same as PWM: pulse width modulation, the pulses are constant width whose width depends on the carrier frequency (50% duty cycle typically)

 

I like to think of this with a nice relatively gentle and high corner slope filter which doesn’t damage/change the audible spectrum. Noise shaping “pushes” the digital quantization noise out further above the audible spectrum and pushes this noise into a spectrum where the filter most effectively removes it.

 

My understanding is that DSD has to use aggressive noise shaping because of its 1-bit sample rate. I didn't say (and IMHO didn't even imply) that this is a bad thing - in fact, I made the comment in the context of saying that I generally find SACD to sound slightly better than redbook. So yes of course, quantization noise should indeed be sent up into the ultrasonic region (although doesn't 1x DSD end up with a big noise spike around 30k, and not just up in the mHz region? Or am I mistaken about that?). I have no issue whatsoever with how SACD/DSD uses noise shaping. If you feel "aggressive" is a negative-sounding word regardless of the context, or that "aggressive" does not accurately describe the actual noise-shaping behavior of SACD/DSD, fine. Personally I'd consider that a semantic argument and not worth disputing one way or the other, so have at it. ?

 

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?

[AudioDoctor]

11 hours ago, PeterSt said:

 

Do yourself a favor and start with the early days of it. And then really the early days. Take Rapture from Blondie to get in a (kind of flawed) mood and next jump to Rapper's Delight (Sugar Hill Gang) - the 14+ minute version - which could well have been the first (Maybe the Blondie was a first single).

https://en.wikipedia.org/wiki/Rapper's_Delight

You see Debby Harry (Blondie) mentioned there as well.

 

 

This was my very first "rap" album. Btw one which at first was quite unplayable because of being too tinny on computer playback (not CDP), but these days 100% normal (go figure - Al Stewart's Year of The Cat is also such an example).

 

 

I will do that, thanks.

[jabbr]

1 hour ago, tmtomh said:

My understanding is that DSD has to use aggressive noise shaping because of its 1-bit sample rate. I didn't say (and IMHO didn't even imply) that this is a bad thing - in fact, I made the comment in the context of saying that I generally find SACD to sound slightly better than redbook. So yes of course, quantization noise should indeed be sent up into the ultrasonic region (although doesn't 1x DSD end up with a big noise spike around 30k, and not just up in the mHz region? Or am I mistaken about that?). I have no issue whatsoever with how SACD/DSD uses noise shaping. If you feel "aggressive" is a negative-sounding word regardless of the context, or that "aggressive" does not accurately describe the actual noise-shaping behavior of SACD/DSD, fine. Personally I'd consider that a semantic argument and not worth disputing one way or the other, so have at it. ?

 

Yeah there's a lot written out there. Some of the confusion and misinformation, I think arises from the misapplication of recording issues to playback. Certainly for recording the use of noise shaping is necessary and universal. In playback the upsampling can be done either in the DAC chip or CPU. If you mean by "aggressive" as "processor intensive" then yes, indeed processor intensive. 

 

In any case whether we start with PCM44 or DSD64 and in both cases upsample using a processor intensive algorithm run on a CPU, the sound can be terrific. Indeed many/most recordings don't have any information above 22kHz and so CD is capable of containing. There are some recordings with information above 20kHz but not really above 50kHz (unless special and uncommon microphone and recording techniques are being used) so in those cases 24/96 or DSD64 are able to capture. One could debate whether an increased noise floor above 30 kHz will make a noticeable difference -- well in that rare case a DSD128 recording would suffice (and these recorders are readily available today).

[jabbr]

12 hours ago, PeterSt said:

Do yourself a favor and start with the early days of it. And then really the early days. Take Rapture from Blondie to get in a (kind of flawed) mood and next jump to Rapper's Delight (Sugar Hill Gang) - the 14+ minute version - which could well have been the first (Maybe the Blondie was a first single).

https://en.wikipedia.org/wiki/Rapper's_Delight

You see Debby Harry (Blondie) mentioned there as well.

 

 

Nice. Grandmaster Flash nor DJ Kool Herc unfortunately have not been given sufficient credit, nor seen commercial success.

Blondie's "Rapture" was the first to hit #1 -- typical

 

Origins of "hip hop" -- best we can tell was invented by DJ Kool Herc between 1972-1973

https://en.wikipedia.org/wiki/DJ_Kool_Herc

 

 

[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.

[Audiophile Neuroscience]

5 hours ago, kumakuma said:

 

Couldn't agree more. The music, performance, and recording/mastering quality are so much more important than the resolution.

 

You might say if its crap to begin with, increasing the resolution results in highly resolved crap! ?

[sandyk]

11 hours ago, numlog said:

Lossless volume control with UDA38Pro DAC

 

No, I dislike the grain/ loss of detail of 44.1kHz to DSD64 conversion.

 I prefer native DSD64 over all PCM so it's not the DAC or a preference for so called PCM ''sharpness''.

 

Dont have the equipment for ripping so only downloading and torrenting rips of CDs I already own from reputable site.

All the 24bit 44.1kHz albums would have been digital versions, most of 16 bit albums were CD rips.

Are you suggesting these rips could be 'bad'?

 

Let's just say they could be mediocre.

Are you able to try recording an album from a CD player via Coax SPDIF ?

[Sonicularity]

How is this for a low bar?  I am happy with Google Music. (they use Lame 3.98 CBR 320) I'm unable to hear differences in the ABX tests I have performed between any hi res file and a well-encoded mp3 on my humble system.  The only significantly noticeable differences I have observed was with speakers; either from placement or different speakers altogether.

 

I use a cheap (<$500 USD) Asus Windows 10 laptop with Google Music Desktop Player and APO EQ as a dedicated source.   My sound output is set at 16/44.1, since I am only streaming Google mp3 files.   I have a calibrated mic and used Room EQ and Math Audio Room EQ to dial in my listening position.   

 

This won't shock anyone, but I am connecting my laptop to my DAC, a Teac UD-301, with a good quality, generic 2 meter USB cable.   I use the DAC's balanced outputs, again using good quality cables that work as expected.  The DAC is connected to a Marantz MM7025 2-channel amp, and I use 12 AWG speaker wire terminated with banana plugs to a pair of Klipsch RP-280F floor speakers.

 

Simple and efficient.  The only thing that matters is that I feel that it sounds wonderful.  I am in bliss and see no need to upgrade anything at this time.  If I did choose to make any update, I would get different speakers, as this would make the most significant difference in the sound.  The rest of my gear, including the streaming mp3 files, would see almost no change to my ears, unless that other gear was not functioning properly.

 

[sandyk]

16 minutes ago, Sonicularity said:

How is this for a low bar? 

 

 If you are so easily pleased, it begs the question as to why you are a member of Computer Audiophile Forum ?

[Sonicularity]

9 minutes ago, sandyk said:

 

 If you are so easily pleased, it begs the question as to why you are a member of Computer Audiophile Forum ?

 

I play music via a computer and enjoy excellent sound quality.

[sandyk]

30 minutes ago, Sonicularity said:

 

I play music via a computer and enjoy excellent sound quality.

 

.mp3 is excellent sound quality ?

Many people wouldn't consider 320kilobits audio excellent sound quality either.

 Still, if it floats your boat ….

[Ralf11]

pick a favorite song or album and try listening in Redbook etc.

[Ralf11]

alternatively, invite GUTB over for a beer...

[Sonicularity]

7 minutes ago, sandyk said:

 

.mp3 is excellent sound quality ?

Many people wouldn't consider 320kilobits audio excellent sound quality either.

 Still, if it floats your boat ….

 

I completely understand what you are saying.   I simply can't hear any significant difference when I ABX on headphones or speakers.  So, to me, my boat is extremely buoyant using mp3.  Enjoy the music.

[Ralf11]

Gadzooks!!   Scramble the A.I.T. immediately!

[lucretius]

On 7/13/2018 at 7:20 PM, Hugo9000 said:

Well, I for one am really disappointed that you really aren't listening with Barbie headphones.  I wanted to ask your opinion on whether that was the optimal transducer for this particular treasure:

 

 

Here is the MQA version, for comparison:

 

 

That's a hoot!

[rn701]

Good grief, y'all. April 1st was almost four months ago.

[GUTB]

1 hour ago, Sonicularity said:

How is this for a low bar?  I am happy with Google Music. (they use Lame 3.98 CBR 320) I'm unable to hear differences in the ABX tests I have performed between any hi res file and a well-encoded mp3 on my humble system.  The only significantly noticeable differences I have observed was with speakers; either from placement or different speakers altogether.

 

I use a cheap (<$500 USD) Asus Windows 10 laptop with Google Music Desktop Player and APO EQ as a dedicated source.   My sound output is set at 16/44.1, since I am only streaming Google mp3 files.   I have a calibrated mic and used Room EQ and Math Audio Room EQ to dial in my listening position.   

 

This won't shock anyone, but I am connecting my laptop to my DAC, a Teac UD-301, with a good quality, generic 2 meter USB cable.   I use the DAC's balanced outputs, again using good quality cables that work as expected.  The DAC is connected to a Marantz MM7025 2-channel amp, and I use 12 AWG speaker wire terminated with banana plugs to a pair of Klipsch RP-280F floor speakers.

 

Simple and efficient.  The only thing that matters is that I feel that it sounds wonderful.  I am in bliss and see no need to upgrade anything at this time.  If I did choose to make any update, I would get different speakers, as this would make the most significant difference in the sound.  The rest of my gear, including the streaming mp3 files, would see almost no change to my ears, unless that other gear was not functioning properly.

 

 

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.

 

[sandyk]

4 minutes ago, GUTB said:

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.

 

That's not surprising,  as you keep claiming that Vinyl sounds way better than your digital playback !

Your cartridge Plus TT and Low noise MC RIAA Preamp is probably worth 10 x what your virtually ENTRY LEVEL (by your standards)  digital gear is worth though.

[Don Hills]

9 hours ago, gmgraves said:

I agree and I didn't even have to listen to your example, to do so! I can imagine that it sounds just wretched - just from looking at the still frame picture, above.

 

You may have missed the point of the video. Take the time to watch it right through.

[Sonicularity]

1 hour 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.

 

 

Thank you for the advice.

[buonassi]

As is the case with some others who've already posted:  The better I dial in my system, the harder it is to tell 16 bit from 24 bit.  It's to the point where I just don't care about high res anymore.  

 

A big revelation was USB input on my DAC.  Not even using a regen could get it sounding as good as coax.  For my DAC at least.

 

I guess my point is - you've got to know every inch of your system, play and tweak, before you can make assertions of this is better than that (let alone "unlistenable").