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[Barrie]

I'm a big fan of Software that does'nt try to do too much but instead is vary good at what it does.  That being said DVD-Audio was a big step forward in digital music recording and play back.  Why is'nt MC 11 supporting it, and do you intend to support it in the future?  It turns out that the battle between SACD and DVD-Audio was won by computers especsally since Minnetonka Audio's DiscWelder Bronze, which allows one to convert all your cd's to DVD-Audio format and burn them to DVD's.  You can even up sample your CD's first to 24bit, 96kHz or 192kHz using Sounds Logical's Resamle 1.1, then convert to DVD-Audio format for a very nice sound improvement. All I need now is for MC to support DVD-Audio Playback.  The big thing for me with MC 11 is high quality Audio playback using advanced drivers like Asio which bypass Windows altogether and work directly with your professional sound card allowing for far better fedelity.

[Mr ChriZ]

Upsampling isn't going to improve the sound quality...
It's like starting with a 1 mega pixel picture changing the
res to 4 mega pixels and ending up with a higher quality picture...
all that happens is it's streched.  Making it better quality would be magic I'd like

I'd also imagine that if the 'battle' has been won, it's more down
to the fact that DVD is becoming a more commonly used medium,
and it's down to economics.

[Barrie]

I am a studio engineer and more importantly an audiophile for 40 years, with [a lot] invested in audio electronics and many inventions and discoveries under my belt.  What you just said is complete rubbish.

If you have tried and couldn't find a difference it is because you didn't own the equipment to play back the difference. 

[edited by JimH]

[Mr ChriZ]

Well I stand corrected, and stand back for some one knows better than me.
I'm a software engineer and I understand how the audio is stored, both digitally
and in analog form, and find it difficult to understand how the quality could improve, and would like
to learn how it works.  Could you provide more info?

My Audio kits enought to have fun with and relax when listening to

Cheers
ChriZ

[jgreen]

Don't worry, ChriZ, you can't improve your tracks by slicing them into more bits.  But it's an interesting question whether a DSP or similar downstream process would perfrom more accurately on the same file if it's sliced into more bits.  I'm inclined to think it would.  If Mister engineer-inventor-audiophile can tell the difference (in a blind test), then there is some process downstream from the decoding that is benefiting from the additional bits.
Is it possible the Mister Inventor invented Minnetonka Audio's DiscWelder Bronze?  Or just Cold Fusion?

[JimH]

Barrie probably knows something we don't.  The sound is almost certainly being "enhanced".  The question is maybe, "how", or "why?"  Something could be added. 

[jgreen]

Well, I for one am more inclined to prefer the sound of something I just purchased and/or invented.

[cncb]

If you have tried and couldn't find a difference it is because you didn't own the equipment to play back the difference. 

[edited by JimH]

Right.  He/she probably needs that aftermarket $500 power cord for their amp to really hear the difference. 

Please explain to us how this is different than the picture analogy.  You are still creating something that wasn't there to begin with and it can only be a "guess"...

[Mastiff]

It's like starting with a 1 mega pixel picture changing the
res to 4 mega pixels and ending up with a higher quality picture...

But that's what FFDShow does with the DVD image, right? And that works, I can see that on my high end Barco Graphics 808 projector with HD-144 lenses and BG808S RGB amps, and a power cord for 600 dollars! (OK, I lied about the last part! It's a computer power cord for 10 dollars...but the signal cable did cost 200 dollars!)

[Mr ChriZ]

But that's what FFDShow does with the DVD image, right? And that works, I can see that on my high end Barco Graphics 808 projector with HD-144 lenses and BG808S RGB amps, and a power cord for 600 dollars! (OK, I lied about the last part! It's a computer power cord for 10 dollars...but the signal cable did cost 200 dollars!)

Yeh I guess it could do some sort of interpolation type thing, although with that it's possible it could get it wrong, where it's doing averages, and it's no more detail... just smoothed.
Done a bit of research myself.  Seems a lot of different opinions on the subject, including AudioPhiles.  I'm wondering if it does work, why it couldn't just be done at the DAC, either seperate or they could start putting it on all Audio CD Players etc, save putting it on to DVD, like you get with Oversampling, maybe they will....

Don't listen to me tho, I know Nuthing! 

[jlange]

There are some external DACs that do both upsampling and interpolation (http://www.av123.com/products_product.php?section=processors&product=1.1).

I have never listened to one of these devices, but it does make sense to me that, if implemented properly, it would enhance the signal.  When you consider a 20kHz sine wave sampled at 44.1 kHz, the digital information is only about two samples per cycle.  When reconstructed, it starts out as a square wave and is then filtered during D-to-A conversion to round the edges into something that "resembles" the original waveform.  If the sample timing doesn't correspond to the maximum peak and valley of the original sine wave, the reconstructed wave won't look much like the original at all.  Lower frequencies fare much better since 44.1kHz will sample 44 times during one 1000Khz cycle.  Our ears are most sensitive to distortion in high-frequencies (audiophiles often describe CDs as "harsh"). Also, high-frequency harmonics contribute to our sense of "air", "transparency", and "three-dimensionality" of music, which is why people describe DVD-A and SACD in these terms.

So to get a "better" reconstructed waveform, a DSP must perform some kind of curve fitting on the digital sample points and then plot new points on that curve when upsampling to 96 or 192kHz.  Drawing straight lines between the points (linear interpolation) clearly wouldn't be enough.

[Alex B]

Our ears are not sensitive to anything at 20 kHz. When listening to a sine wave very few people can hear it without significantly increasing volume. Many people don't hear it any volume.

When listening to standard music a lowpass of 16-18 kHz is usually good enough for everybody. Even with excellent hearing it is difficult to hear any difference between a straight CD audio sample and a 16 kHz lowpassed sample in a blind test.

The 2-channel stereo DVD-A and SACD recordings are often mastered differently without the current loudness and compression race that has happened to standard CDs. That makes most of the perceived difference.

[Alex B]

By the way, MC's audio resampling uses high-quality Shibatch SSRC filters. If you like to try what it does to 44.1 kHz source material just set the output to 96 or 192 kHz. My Terratec DMX 6fire 24/96 soundcard can handle up to 96 kHz, but I can't hear the difference. I can only experience it in a form of increased CPU usage.

I use resampling from 44.1 to 48 kHz only on my laptop because it's internal soundcard resamples everything to 48 kHz with mediocre quality. I rather use MC's HQ software.

[jlange]

I know that most people can't hear 20kHz (I think my own hearing rolls off at 15kHz), I was just using that as an extreme example of how sampling rate impacts waveform reconstruction.  A 10kHz wave is only sampled 4.4 times at 44kHz, which seems (to me) just barely adequate to rebuild the waveform.  Of course real music is layer upon layer of many different sounds, harmonics, transients, and so on that make for a very complex waveform.

My main point is simply that good algorithms CAN improve the sound when upsampling CD audio.

[Barrie]

Why does Upsampling improve sound quality?

From the explanation above, there is apparently no extra information in the upsampled signal that was not present in the initial signal. With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20 kHz.

This conventional analysis starts from the viewpoint that the behavior of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good – it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.

For those working with audio, it is also apparent that theories based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here – the ear may have more than one mechanism at work, and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localization mechanism ("where is it") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modeling. The problem here is that sine wave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet math is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting.

If one gets the frequency response of some equipment right, but the provision of transient information wrong, one or more of the ear’s mechanisms cannot work properly, and so we are unable to separate out echoes and cues about where a sound is coming from the rest of the "what is it anyway" signal. dCS’ upsampling filters are designed to help sort this problem out. They are best analyzed not in sine wave terms, but using wavelets.

[Barrie]

Data-word length

Some background. Here’s what a random 16-bit data word looks like for CD audio:

0011011000101110

There are 65,536 different values represented by the 16 digital bits (2 raised to the 16th power). Each of these values represents a voltage in the analog output signal. If the DAC IC outputs two volts maximum, then each different value for the 16 bits represents .0000305 of a volt (this isn’t a precise analysis, just a general conceptual overview).

0000000000000000 = zero volts
0000000000000001 = .0000305 volts
0000000000000010 = .0000610 volts
0000000000000011 = .0000915 volts
1111111111111111 = 2 volts

If you increase the word length from 16 bits to 24 bits, the number of different voltages you can represent increases from 65,536 to 16,777,216 (2 raised to the 24th power). Each step in a 24-bit word would represent .00000006 volts, still using our 2-volt output model, compared to the .0000305 volts for each step in a 16-bit word (using the same 2-volt output example). You can see that the granularity in voltages representing the musical signal is incredibly fine when the data words are 24 bits long. Take an example where the audio signal wants to be .0000455 volts (still in our 2-volt example). With 16 bits available, software has to decide whether to make this voltage become .0000305 or .0000610 volts. There is no way with 16 bits to make a voltage that is .0000455 volts. But in the 24 bit world, you have an extra 256 different voltages available between .0000305 and .0000610 volts. One of them will be almost exactly .0000455 volts.

Twenty-four bits is really higher in resolution than any consumer analog or digital audio products can achieve due to limitations in current electronic technology. Most digital products with claimed 24-bit performance lose probably three to four bits in the noise floor. But nevertheless, it is advantageous and comforting to have a digital standard that actually exceeds the capabilities of performance in consumer audio components. I’m not sure you would want to listen to music that requires all 24 bits to reproduce anyway; the loudest sounds would be as loud or louder than the loudest noise you ever heard in your life, well beyond the threshold of pain. The quietest sounds would require the total silence of an anechoic chamber to be able to hear them. What 24 bits brings to the table is headroom and footroom, which make digital audio more forgiving and easier to work with and offers an improvement in resolution even if limited to 20 or 21 bits of effective resolution.

The circuitry in the D2D-1 creates a higher resolution digital audio bitstream by analyzing sequential groups of 16-bit data words and generating interpolated (best guess) 24-bit data words to replace the original 16-bit words. The process does have some margin for error, but it is vanishingly low. Keep in mind that in converting 16-bit audio to 24-bit audio does not increase the resolution of the reproduced audio. You can’t have more than 16 bits of resolution when you start with 16 bits of data, but the higher resolution digital bitstream can be more optimally transmitted and processed to make small improvements in sound quality

[Barrie]

I have tried the DSP upsampler on the JRiver MC and it does not improve sound quality, in fact I found that there was loss.  To get the best sound from MC set everything to souce bit and sample rates.  Uncheck everything else. That doesn't mean if you use a good unsampling software tool it dosen't make an improvement, in fact it does.  Buy the way I am also an electrical engineer and worked in low level systems develpment and project management for 10 years.  You definetly need a very good system to fully appreciate the difference.  If you have the system then it is worth the time and money.  Use Sounds Logical Resampler 1.1  to do the upsampling.  Use JRiver MC for play back as I belive it is the best playback software on the market.  It is best to have a pro sound card  and use the Asio driver in J.R. MC.  I recomend E-MU 1212M or higher.   The reason I am interested in DVD-Audio as you can convert all your audio tracks that you have upsampled and convert them all to DVD-Audio format (use Diswelder Bronze to do the conversion to DVD-Audio), so they can be played back on any conventional DVD-Audio player and sill be able to play the same tracks on my computer as well.  This can also be done without the Upsampling for those who are interested in transfering all your CD audio tracks to DVD.  This is not something I've read or heard about but do regularly.  So, any possiblity J.R. MC will support DVD-Audio in the near future.  If not I will have to swith to Nero 7 Ultra which is due out on the 26th of October.  It supports 32bit 192kHz and DVD-Audio.  I am sure the reason for this is because it also supports Blu-Ray which is going to use DVD-Audio format on the new HD-DVD's.  High res audio is hear to stay and as soon as sales of i-Pods and MP3 players drops I'm sure it will be the next big thing as it is far better sounding.

[danrien]

hey i agree with this guy.  barrie i'm in school as a computer engineer (sophomore) and so it's cool to be on these boards and read tech stuff  from guys who've been in the industry for a while.

but ryan adam's cd "gold" was released in dvd-a and i was really tempted to buy it until i thought about jrmc and its inability to play it.  it would be super awesome if in 11.1 dvd-a (and sacd too i suppose) would be added.  the only hurdle i can see is licensing fees, but i'd be willing to pay 5-10 more bucks for a DVD-A enhanced version of MC.

another thing that would be really cool (but might require some innovation) is if you could do things like rip it on to your computer in 5.1 ogg format (compression generally doesn't bother my below average ears) but i'm guessing that those mean RIAA companies have protections against that.  but it would be cool, and that's all that matters in my book. haha

anyways, point is DVD-A support would rock, but you guys do what you need to to make money.

[JimH]

As someone else said, I have a hard time believing that changing the bitrate could add anything but very small. random change.  A high resolution copy of a low resolution picture isn't going to look any better.

I think a lot of what people hear is completely subjective, so if it sounds better, go for it.

[Mr ChriZ]

Cheers for the info, but it might be nice to reference
the copyright holders, keep everyone out of trouble etc.
->
http://www.aslgroup.com/dcs/dacsandupsamplers.htm

http://www.soundstage.com/revequip/assemblage_d2d1.htm

[Mr ChriZ]

I am sure the reason for this is because it also supports Blu-Ray which is going to use DVD-Audio format on the new HD-DVD's. 

Blu-Ray and HD-DVD's are seperate mediums are they not?

[Alex B]

Data-word length

Some background. Here’s what a random 16-bit data word looks like for CD audio:

0011011000101110

There are 65,536 different values represented by the 16 digital bits (2 raised to the 16th power). Each of these values represents a voltage in the analog output signal. If the DAC IC outputs two volts maximum, then each different value for the 16 bits represents .0000305 of a volt (this isn’t a precise analysis, just a general conceptual overview).

0000000000000000 = zero volts
0000000000000001 = .0000305 volts
0000000000000010 = .0000610 volts
0000000000000011 = .0000915 volts
1111111111111111 = 2 volts

If you increase the word length from 16 bits to 24 bits, the number of different voltages you can represent increases from 65,536 to 16,777,216 (2 raised to the 24th power). Each step in a 24-bit word would represent .00000006 volts, still using our 2-volt output model, compared to the .0000305 volts for each step in a 16-bit word (using the same 2-volt output example). You can see that the granularity in voltages representing the musical signal is incredibly fine when the data words are 24 bits long. Take an example where the audio signal wants to be .0000455 volts (still in our 2-volt example). With 16 bits available, software has to decide whether to make this voltage become .0000305 or .0000610 volts. There is no way with 16 bits to make a voltage that is .0000455 volts. But in the 24 bit world, you have an extra 256 different voltages available between .0000305 and .0000610 volts. One of them will be almost exactly .0000455 volts.

Twenty-four bits is really higher in resolution than any consumer analog or digital audio products can achieve due to limitations in current electronic technology. Most digital products with claimed 24-bit performance lose probably three to four bits in the noise floor. But nevertheless, it is advantageous and comforting to have a digital standard that actually exceeds the capabilities of performance in consumer audio components. I’m not sure you would want to listen to music that requires all 24 bits to reproduce anyway; the loudest sounds would be as loud or louder than the loudest noise you ever heard in your life, well beyond the threshold of pain. The quietest sounds would require the total silence of an anechoic chamber to be able to hear them. What 24 bits brings to the table is headroom and footroom, which make digital audio more forgiving and easier to work with and offers an improvement in resolution even if limited to 20 or 21 bits of effective resolution.

The circuitry in the D2D-1 creates a higher resolution digital audio bitstream by analyzing sequential groups of 16-bit data words and generating interpolated (best guess) 24-bit data words to replace the original 16-bit words. The process does have some margin for error, but it is vanishingly low. Keep in mind that in converting 16-bit audio to 24-bit audio does not increase the resolution of the reproduced audio. You can’t have more than 16 bits of resolution when you start with 16 bits of data, but the higher resolution digital bitstream can be more optimally transmitted and processed to make small improvements in sound quality

The source link: http://www.soundstage.com/revequip/assemblage_d2d1.htm.

[Alex B]

In general I would love to see more support for multichannel and other HQ audio formats, but I'm not interested of playing optical audio discs on PC. The main reason I bought Media Jukebox a few years ago was handling of HD media. I'm not going to buy any audio-only discs if the format is not allowed to be transferred to HD.

I have accepted to keep the commercial DVD videos on the original optical media, but I have experimented with ripping DVD video discs also.

Inter Video removed the DVD-Audio support from the latest WinDVD a couple of months ago because someone made a program that can grap the multichannel audio signal using WinDVD's components. Here's a thread at HA where ripping DVD-Audio is discussed: http://www.hydrogenaudio.org/forums/index.php?s=124f2a36e8b8b880fc9c883a9c2499dd&showtopic=34368