Dolby Elevates the Quality of Lossless Audio on Blu-ray - Page 4

Different = Distortion of the Original Source

Yes, it sounds different and not better. Agreed.

It is detrimental.

Dithering is detrimental:

This line from your post only supports my statements:

"Dither, contrary to what has been said before, is not detrimental to audio quality. Lack of dither can be severely detrimental."

It is clear the process of dithering is trying to correct issues while processing the signal. Saying adding more "empty filler" to what is already "empty filler"does not there makes it better.

The only argument is subjective ... if you like the "distortion" it add because you perceive it as better, that is harder to argue.

However, if you listen to the original recoded source ... and it sounds "different" ... then you have adulterated the signal and added distortion. With digital it is a bad thing. I hear thinner, grittier, less organic, unnatural sound when I listen to overly processed digital signals.

SACD encoding is another culprit. To me, it sounds awful ... not natural or organic and with hellacious artifacts due to the ultra high sampling rate. However, even many audiophiles seemed to like it because it sounded more "open with more detail & imaging" .... but at what cost. Loss of organic sound quality and a grit and hash that makes it unlistenable to me.

I can get better imaging, detail, and resolution with much better organic sound quality from a properly recorded CD. Again, just listen to any Mapleshade Audio or Wildchild CD. Most sound better than any SACD Discs I have heard.

The only time I actually hear this work to any level of satisfaction is with tubes. My tube equipments sounds great ... a pleasing form of analog distortion ... yet organic and true to life sound ... not processed in a harsh stair-stepping PCM or Lossy Compression sort of way.

I believe some of it has to do with Accounting. I am at odds with accounting and many cases marketing in most companies. Saving $2.00 on a DSP Chip in a $4,000 HDTV and getting poor results because of it is one reason why engineering sometime choose to lay down in defeat.

Believe it or not, I have spoken to engineers that work for big companies who actually want to do the right thing. However, they are overruled by Accounting who wants a cheaper DSP and Marketing who wants the HDTV Panel to look more "red" because it will look & sell better on a retail floor.

That is why I own so many Pioneer Elite and Kuro 9G Monitors. Nothing made to date can beat them.

Such is life..

I do ... all the time...

And that is why I choose my gear very carefully.

Not all gear is created equal and it depends on how it is hooked up.

Are you saying it is OK for Dolby to trash the signal because the "average consumer" does not know the difference or has gear that yo already know is is crap? This is exactly the issue I have with engineers who think their approach is acceptable because "it does not matter down the line."

Well, it does matter. Why do you think it is OK to "pre-determine" what a consumer has or wants?

The Cambridge Azur 750 which is based on the Oppo BD-95 handles audio differently than your average BD Player as does the Oppo.

I take the 7.1 outputs directly from the Cambridge Azur 750 and feed them to the Analog input of my AVR using high quality interconnects from Mapleshade Audio. The sound quality is remarkably improved.

Aren't we getting a little hyperbolic here?

Your best arguments have always been against your own strawmen. Set them up. Knock them down.


If the only argument is subjective, tell us how well non-dithered PCM sounds.

Oh, by the way, dither does not have to be added if the source has sufficient noise (hiss) to self-dither the audio. It is also unnecessary for signals that are not small enough or at the right frequencies to evoke significant quantization errors.


Those recordings presumably meet the above dither criteria.


What a hypocrite. How does that square with:

Some forms of "distortion" are good? Amazing!

What it says is that preringing is an integral and necessary part of accurate reconstruction, and that maybe people shouldn't worry about it in that application. For other applications (EQ, speaker xovers) where the preringing is in the audible bandwidth, maybe there is something to worry about.

You can't really avoid preringing with LP filters --- the math prevents it.


That's not dither nor is that what it does. The role of dither is to decorrelate the truncation error from the signal itself. Any time you truncate a digital word, you must dither before the truncation to prevent really audible artifacts. The DTS encoders and decoders and any other competent DSP do this as a matter of course.

There are dithers that do this perfectly and we know that they are perfect not only because the math says so but there is a very simple listening test that demonstrates this. With a correctly dithered signal, you can hear signals as deep into the noise as you care by using narrower and narrower bandpass filters.

And please stop using that old saw about engineers not listening. Not only is it false and hackneyed, but insulting.

I don't understand. I thought we were talking about pre-ringing in the source, from the anti-alias filter.


Exactly. That's why it's not a good idea to use LP filters at low sample rates.

Not a hypocrite at all. I am saying Analog Distortion with Tubes is far less offensive to me than Solid State Digital Distortion. I stated that analog tubes distort the signal. That is a fact. It just does not sound as bad to me ... and digital distortion sounds very bad to me. Tell me how that make me a hypocrite. You are reaching for straws here.

There is nothing "Straw Men" about my comments. I have been very clear about what the facts are and kept the discussion factual. The only "subjective" part is whether or not you like digital distortion and artifacts. I do not. There really is no argument that this type of digital degradation or alteration of the signal exists.

Dithering is detrimental...on all levels. The fact you have to dither even more to distort the signal when you process it even further ... only proves that math is not a prefect representation of the real world or analog sound.

To get back on point. It is not in the best interest of the artist or the original content producer to Up-Convert the original source. That is my point.

You counter-points really are not addressing what I am actually trying to say.

Depends on your level of concern for quality I guess...

I am picky...what can I say. This type of thing would not be allowed on my watch...which is one of the reasons why I work for myself.

To me, an up-conversion trashes the signal. It is no longer in its native format which I find desirable. I believe in letting the consumer gear do the processing ... some do it better than others. It is the better approach all around.

Also, plenty of enthusiasts ... myself included ... know how to get the most out of a native file format. By avoiding AVR processing and using a quality source player with superior design, architecture and electronic components, I can get the most of a clean original source that has not been screwed with.

I do not believe that anyone should be "pre-processing" the signal before it goes out to the end user. I know that most folks in this forum don't agree, don;t get it, don't care or all of the above. That does not change the facts.

Unfortunately, all of my complaining will not rid us of the problem(s). The responses in this forum clearly show that I am a minority when it comes to voicing these concerns.

.

You have given a lot of your opinions under the banner of fact, but that does not make them fact.

You refute all sorts of issues not raised by anyone but yourself, issues we probably agree with since they are rather obvious in many cases. Nonetheless, those are strawman arguments.


The problem is your assumption that anything done in the digital domain is automatically a distortion to be feared and loathed, even if it sounds good, but if it is done in the analog domain, and happens to sound good, that's fine. That's hypocrisy, is it not?


Well, um, digital (and math) is not the real world. You think EMM Labs converters are perfect? Nope. They are merely "good enough."


Perhaps it was not in their best interest to record at 48 kHz. Such is life in the real world.


How so? Perhaps I am sidetracked by all the extraneous editorializing.

Do you want consumers to do the 2D to 3D conversions at home, or should Cameron do it upstream?


Example of extraneous comment. How does it matter what you do at home? This is a home theater forum where 99.99% of consumers use AVRs and surround processors based on DSPs.


Which fact? The fact that you have an opinion? I agree that is a fact, and I'm sure no one would dispute it.


If you had some evidence to support your opinions on Dolby's process and how it affects home theater users in the aggregate (not the 0.01% fringe), that might help your case.

I understand what dithering is the entire process and how digital works ... clearly you are not getting what I am trying to say.

Most, but not all, Engineers do not UNDERSTAND how to listen is what I should have said. If they did understand, I would not complain about the CODECS, Consumer Gear, Professional Gear and Cables on the market or the processes most professionals use to create an end product.

Have you ever "listened" to ProTools for example? It absolutely alters the signal and has a signature sound .. and NOT in a good way. Almost all engineers agree that it does not sound good ... but it is the accepted standard, so they have no choice but to use it. This is clearly one a MANY examples where the engineers did not "understand how to listen" ....

I could give you numerous other examples ...

Again, my opinion is not invalid ... as you state it is. Sorry you don't understand what I obviously know from experience and can prove in a demonstration.

I will leave you and your thread so DOlBY can promote and defend the "Up-Converting" of native sources prior to delivery to the consumer. However, I go on record as stating this is a very bad thing to do ....

If anyone cares to put their technology where their mouth is, then please allow me to prove my point ... or ... as some might like to believe ... fail to prove my point.

I would be happy if someone sent me a Dolby "Up-Converting" Box so I can put together a demo. We can run the entire process real time with all the engineers they want in the room ... with my engineers there as well ... no scopes, no meters, no mathematical or theoretical arguments ... just ears. Then try and tell me engineers understand how to listen.

My biggest issue is the denial. Just admit the process is flawed and that in many cases processed signals do NOT sound better or at the very least they sound different and could easily be perceived as not sounding better by experienced listeners with good ears and that it is more about marketing than it is about quality .. and I will shut up and go away.

* 96khz upsampling is less airy than. 192khz. 192khz and up is definitely where you want to be in terms of musical depth. Not only that...but music that is upsampled to 96khz-192khz utilizing regular software...has more depth + subtle detail that emerges and stations itself across the entire sound stage and back...more importantly music passages are better observed and have lovely center stage focus when utlizing 5.1 192khz music...mp3 tracks that portrayed themselves as mono - singular type recordings can actually expand themselves wonderfully upsampled to 192khz and then squeezed through a nice Burr Brown Dack such as on oppo's 93 bluray> I've been sound testing software that upsamples the musical track to 5.1 192khz> and it definatlyey sounds more airy than 96khz> which in camparison sounds a bit dryer...and not as Humid /airy as 192khz audio.

Most receivers will actually take 2 channel recordings and output them as 5.1 using Plx II> which I havn't heard but there are other neat tricks as well such as Neo 9 etc...all create and take musical cue's from the front left and right channels and send them to the rear where they can be heard as the recording was intented before the sound cue's didn't exist because of how Dolby First worked...now it's capable of those sound cue's...the newer audio processing also allows you to tune in to how much information is going to the front left right and the rears etc...it supposed to enhance the soundstage Plxx II for music listening has a set front stage width which should slightly get wider compared to the standard 5.1...but also keep the center output withougt being over obvious ( I assume)....and this can all be done without upgrading to a high end amp that has those ceiling boundaries (unlimited). Neo- is more user adjustable...and Nad also has their own 5.1 mixinf teqnique for a more pleasureable music listening experience without making it sound processed.

* Another important note is dynamic range of the music...it Cannot be anything less than -3db....-5-9db is where the music Blooms and engulfs the front stage...although it will not make a dull music file sound bright and enhanced...it will make it sound more seperated with obvious timbre, better sound placement and expanded stage> so remember dynamic range is what your speakers where created for. Attack pause> attack shape stop...if you understand how dynamic range works...properly done...it basically lets your speakers come to a slight stop or resting position during playback of the song...without this slight pause your speakers are constantly in the extended position, introducing what I call high order harmonic distortion..or even worse high frequency aliasing ...that is really distortion!!...your tweeters need dynamic range to come to a full stop (passage of a song comes to a -10db threshold on average) ...this lower volume thresh-hold that dynamic compression utilizes (most artists in the 60-70-80 did this 3-4 times for music tracks beatles included)...creates the sense of dynamics and fullness of the music passage...it greatly reduces distortion from the tweeters...and provides smoother fuller and more blooming sound attack...you have to understand that most music on the market today is not intended to be played back with high end equipment...let alone tweeters...most music listeners utilize cheap desktop speakers that only play up to 12-khz at best...the music industry creates harmonics (distortion) artificially by using less dynamic range so they can be heard as full range sound through your radio or cheap speakers...it basically adds treble artificialy through distortion (distortion being your speaker never comes to a full rest and is always in the extended position thus creating the harmonic anomally...) anyone with high end equipment must realize this...and understand your not hearing the best music possible even with your $2000 pair of front end speakers...I'm not sure how HD audio Downloads sound or are prepared...but without dynamic range compression all your getting is a pressed music passage that is finely distrorted with a loss in bloom and seperation.
Edited by shingdaz - 6/8/12 at 6:22am

Wow!!! Its a great initiative of Dolby. Now we can enjoy musics with Blu-ray. Now i want to use it to enjoy true musics and sounds.

Sort of off the topic of the thread subject but I stumbled upon this thread and was captivated by the talk of quantization, dithering, etc.., and was wondering if any of you have done any SACD (DSD) conversions to PCM for streaming files as a WAV or FLAC. There is a new technique with a PS3 player that I have used and have been successful with the actual process by getting a playable flac file; However, I was not happy with the sound quality. I have suspected it was something in the DSD to PCM conversion that got lost and may have been due to the settings I used in the Weiss Saracon and Pyramix software. The options were overwhelming when it came to setting the options for dithering, quantization, apodizing filters and so forth. I had no idea what I was doing. If anyone has any suggestions I would greatly appreciate a reply post or PM. I am beginning to think I may have done everything right but perhaps the 1 bit DSD to many-bit PCM may be the issue - What I may be hearing is simply the difference in the DSD to PCM conversion?

Thanks.

I agree, when I switched from the already acceptable soundtrack to the 192KHz version we could hear the improvement right away. Better seperation of the all channels and when you listen closely you can hear the improvement of the music tracks specifically the congas that are recorded in the mix can hear the skin of the drum being tapped!

hey Buddy
Nice Work

I beg to differ, for all the obvious reasons.

Ringing is not the proper term in this case. Ringing occurs in a under damped LCR circuit. In this case the ripple in the impulse signal is caused by phase shift in the audio pass band. Not sure why after all these years people are still pointing this out in regards to digital audio.

Apparently no one has ever looked at the impulse response from a phono cartridge, analog tape machines, most dynamic microphones or loudspeakers. If anyone measures the impulse response from their loudspeakers and do not like the results, just move your measuring microphone a few centimeters and the waveform will completely change and if you move it around enough you can probably find a spot that will please you . Before retirement, I was fortunate to have Brüel & Kjær microphones for measuring.

DISCLAIMER: I am a Dolby stockholder but this reeks of marketing hype. If there is credible ABX testing that shows there is a benefit then I will sign on, otherwise it is just more audio BS.

It does indeed. And in many more. Any underdamped oscillation can be called ringing.
Not every audible effect is best determined by traditional ABX (switch among short excerpts). It may take rather longer to dial in to the subtleties. The absence of ABX evidence does not prove there is no effect.

For more insight on these topics, may I suggest this Stereophile article.

Cheers.

That was the point, in this instance it is most likely nothing more than phase rotation of the low pass filter, not energy stored in an inductor. Just pass a impulse (i.e. square wave) through a single pole (RC) all pass filter and the output waveform will look nothing like the input at various frequencies. As I said earlier, pass that same impulse through most loudspeakers and the waveform will be nothing like the input waveform.


AFAIK, there is no time limit one listens to A or B or X. If people believe device X is better than device Y then they will believe X sounds better than Y regardless that the two may be identical. It is easy to prove this.

Inductor not required. That was my reason for suggesting to read the article. Was it not enlightening?
But how many of the responses are acausal? I'll bet none.
Correct. That is why I said "typical."
Yes. And if people believe that something is marketing hype then they will believe that regardless that X and Y are not identical. Proof is in the eye of the beholder.
Edited by Roger Dressler - 7/7/12 at 5:18pm

Here is the point, Dolby Labs has claimed that folks can not here the difference between a original source and that source processed through Dolby Digital (or +). We know there is lots of info discarded by a perceptual encoder so it follows then the input waveform will be markedly different than the output waveform under many conditions.

Now they are proclaiming that a small ripple caused by a phase shift of a lowpass filter is really bad thing. I find the whole thing insulting and absurd. Not the Dolby I once knew.

If anyone does not think this is absurd then I invite you to post the square wave response from your loudspeakers (phono cartridges, analog tape machines).

There is one exception that I know of to the above post of mine, the Ampex AVR-1 video recorder. The equalizer in the audio section used delay lines (linear phase) rather than RC circuits. It would record and playback a fair likeness of a square wave.

Can you cite a reference? I'm curious to see how they stated that.
Something objectionable about the relentless pursuit of perfection? Seems ok for Lexus. Why not Dolby?

BTW, based on this reaction, am I to conclude that you did not read the Stereophile article?

And as an avid Dolby follower, are you not aware of their long relationship with Meridian? And that Rhonda has moved to Dolby to help with the project?
Since you apparently have such plots, would you mind posting (or linking to) some examples? I want to see how much pre-ringing they exhibit. Thanks.
Edited by Roger Dressler - 7/12/12 at 11:54am

Smells like snake oil in order for Dolby to reclaim relevance in the lossless audio realm and not worth the thousands required in upgrades to hear the difference. Claiming that these differences could manifest over time in a subtle fashion is reminiscent of psychobabble. Sorry, that's my opinion, but I have seen double blind tests smashing the subjective claims of laymen listeners too many times to fall for this.

Where did anyone say anything about spending thousands on upgrades to hear the difference? That's the point -- you don't have to spend anything.
You are right that some of us can hear the difference in a matter of seconds, with the right material of course. But not all material evokes the difference, and not everyone is as keen eared as you and me. For some of us it takes more time. Just look how long people can listen to MP3's and never even notice the gurgles and grit.

I think it pays to consider differentiating lossy codecs and lossless codecs here.. What Ive always gone on Wendell is from the dolby site ;it is pretty unequivocal and gives some leeway
I have generally preferred dd @ 640 kbps from a bluray than off a dvd @ 448kbps ; curious what do you find depending on well authored discs ? Little known tidbit I discovered- the optical implementation of dd in a cinema is lower than 448kbps

http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/20_Dolby_E._Standards.P.pdf

Heres a good read that takes into account your concerns but shows some insight about bitrates and perceptual encoding hopefully

http://www.practical-home-theater-guide.com/dolby-vs-dts.html