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[yank2channel]

I have a Cambridge Azur 851c Digital Pre-Amp/CD player with upgraded firmware to support USB 2.0 streaming that I feed .wav and .flac files off a WD My Book external hard drive that is controlled by J River MC20 installed onto an HP Pavilion laptop running Windows 8, that runs into the Cambridge via an Audio Quest USB 2.0 cable. The files range from 44.1 .flac files to 96 and 192 .wav files. I replaced Windows' crappy Direct Sound and started using ASIO instead. While the files in 44.1 sound good, and  the 96 sound better, the 192 are really thrilling, knock your socks off stuff. The My Book has 32,000 songs loaded onto it, with the file quality distribution being pyramidal in shape, mostly 44.1, less 96, and a handful of glorious 192. I want to rip some of my favorite CDs to either 96 or 192 files so I can enjoy them without having to pop discs in and out of my Cambridge. So my question is, how can I use my recently purchased copy of J River MC20 to either rip, or rip and then convert these files? Or is there a MC20 compatible plug-in or download that will do this?
Thank you in advance for helpful posts. And if you want to convince me that 192 is inferior and a waste of time, save my time and yours and don't try to 'enlighten' me of this 'fact'. Speakers are B&W 802s, Audio Research LS25MkII Preamp and Bryston 4BST amp. Yank.

[Vocalpoint]

And if you want to convince me that 192 is inferior and a waste of time, save my time and yours and don't try to 'enlighten' me of this 'fact'.

Well - I won't attempt to enlighten you (I stand clear and let pure mathematics do that) - but if you really think taking a standard 44.1 file and padding it with a bunch of digital zeros - actually makes the file sound better (not to mention grow by 4-5 times the size) - rock on bud. I wish I could somehow sell you hard drives after you finish your conversions

Cheers,

VP

[kensn]

Agreed.. Taking a normal 44.1 rip and then turning it into a 192 does nothing for the sound. You still only have the original bits with a bunch of zeros padding the file to create the 192. They will sound the same.

Ken

[dtc]

If you go from 44.1K to 176 you are actually adding new data points between the original data points. That is usually done with a curve based on the surrounding points. In fact many DACs have done that for years. Some people like the sound of the additional points and some do not. But there is more to upsampling than simply adding zeros.

If you are going to upsample 44K files, you should probably use 176 rather than 192, since you maintain the original data points. The math is easier and put less strain on the CPU, which can be important if you are upsampling in real time. And upsample 96K to 192.

You can do the conversion on the fly by simple settings in DSP studio. Just set 44K to upscale to 176 and 96 to 192. No need to convert the files, at least when you are evaluating the sound difference. It is a pretty easy experiment. Give it a try.

I just looked at your pre-amp/DAC and it is already upscaling it to 24bit/384KHz, so doing any upsampling in MC in unnecessary.  According the the Cambridge Azur webpage on your 851c,

"All incoming audio data, such as CD quality 16-bit/44.1kHz, is upsampled to 24-bit/384kHz using a high-end 32-bit Analog Devices DSP (Digital Signal Processor). "Polynomial Curve Fitting Interpolation" provides a much more accurate upsampled audio soundwave than traditional upsampling processes. And a unique timing system buffers audio data to almost eradicate digital jitter. Vital – as jitter, common in digital audio devices, loses vital detail, often leaving digital audio sounding flat and lifeless."

The increase in quality you hear in your 96KHz and 192KHz files is probably due both the re-mastering and the increased sample rate.

Seems like with your 851c, upsampling the data stream is probably no going to add anything, but, hey, give it a try.

[~OHM~]

I want to rip some of my favorite CDs to either 96 or 192 files so I can enjoy them without having to pop discs in and out of my Cambridge. So my question is, how can I use my recently purchased copy of J River MC20 to either rip, or rip and then convert these files? Or is there a MC20 compatible plug-in or download that will do this?
Thank you in advance for helpful posts. And if you want to convince me that 192 is inferior and a waste of time, save my time and yours and don't try to 'enlighten' me of this 'fact'. Speakers are B&W 802s, Audio Research LS25MkII Preamp and Bryston 4BST amp. Yank.

looks as the responders to your ACTUAL QUESTION was missed....oh well.... have a look at the wiki for the info you truly asked for....and good luck

http://wiki.jriver.com/index.php/Category:Frequently_Asked_Questions

[kensn]

If you go from 44.1K to 176 you are actually adding new data points between the original data points.

I don't think in MC conversion there is anything added to the audio.
From the Wiki:

 Conversion

Converting from less bits to more bits is perfectly lossless. Conceptually, imagine adding bits like adding zeroes at the end of a decimal. For example, the number "10" might become "10.0" or "10.00" if you add more bits, but all three representations are perfectly identical.

When Media Center inputs data, all audio is first converted to 64bit. This ensures that any processing like digital volume, Replay Gain, or any other DSP (if any is enabled) is done with as much precision as possible. It also puts the data into a format that is efficient for a computer to handle, and makes it so that tracks of varying bitdepths can seamlessly transition.

When outputting data to a soundcard or DAC, the 64bit data is converted back to the format required by hardware. This is often 24bit for high-end DACs.

The transition from 16bit to the output bitdepth (often 24bit) is bit-perfect. Again, it's like "10" vs "10.0".

[dean70]

I found upsampling 44.1k to 176/192 to sound somewhat "wooden" in quality (sounds kind of like a timber speaker cabinet with insufficient damping) . Upsampling to DSD128 on the other hand can make a notable improvement.

[dtc]

kensn - you are missing the point that more data points are added by the interpolation. You are talking about the internal bit depth that MC is using. The issue at hand is  upsampling to a higher sample rate that produces more data points. The MC DSP provides an upsampling option. That option is available for a reason. The OP's DAC does it for a reason.

[astromo]

So my question is, how can I use my recently purchased copy of J River MC20 to either rip, or rip and then convert these files? Or is there a MC20 compatible plug-in or download that will do this?
Thank you in advance for helpful posts. And if you want to convince me that 192 is inferior and a waste of time, save my time and yours and don't try to 'enlighten' me of this 'fact'. Speakers are B&W 802s, Audio Research LS25MkII Preamp and Bryston 4BST amp. Yank.

looks as the responders to your ACTUAL QUESTION was missed....oh well.... have a look at the wiki for the info you truly asked for....and good luck

http://wiki.jriver.com/index.php/Category:Frequently_Asked_Questions

Following on from TAO1857, look up the entry on CDs and DVDs and follow the sub-threads. Convert Format is also relevant and recommended.

In short, no, you don't need anything other than MC to rip and convert file type format.

[kensn]

The MC DSP provides an upsampling option. That option is available for a reason.

The reason is to support hardware that will only accept certain bit rates. Does nothing to improve the sound. I felt the poster was wanting to get Sampled 192k sound from a Sampled 44.1 file. It won't happen.

ken

[dtc]

The reason is to support hardware that will only accept certain bit rates. Does nothing to improve the sound. I felt the poster was wanting to get Sampled 192k sound from a Sampled 44.1 file. It won't happen.

ken

There is more reason to support upsampling that just to support hardware. I know of no DAC that accepts 192 KHz and not 44.1 KHz. Certainly there is reason to support sampling changes for some hardware, but that is fair from the only reason to support it.

Whether upsamplng increases the sound quality is certainly debatable, but, again, a lot of DAC manufactures do it because they think it does. I agree that a file mastered at 192 KHz should sound better than one upsampled from 44 KHz to 192 KHz (or 176 KHz). That's why I suggested he give it a try with on the fly upscaling.

[kstuart]

The only reason to upsample files that originate at 44.1 to 192, is that a few DACs actually do that internally, and some people find that JRiver does it better.

But that is a tiny fraction of DACs.

Otherwise, upsampling does not help anything.

Here is why:

* A real 24-bit/192khz digital recorder, samples more than 4 times more often than a 44khz recorder.  The advantage of this is to get correct samples more often.

* Upsampling guesses at those missing samples.  Doing that nullifies the whole advantage of recording those missing samples.   By recording at 192khz, you get the real samples.  That's the whole advantage.   Recording at the higher frequency is better, because of the way that reality is different from guesses.

* If Upsampling worked, then why not just record at 4-bit/12khz and upsample 16 times.  It should sound the same as 192khz, eh ?

[dtc]

kstuart - some people think that upsampling from 44K to 192K (or 176K) sounds better than 44KHz. You may not, but some do. 

I totally agree that a properly done 192K sampling is generally better than a upsampled 192 KHz.  But, if you have a 44KHz file, then upsampling to 176K or 384K (like the OP's DAC) can, in many people's experience, sound better. 

The guesses you talk about are informed guesses. Not as good as actual samples, but maybe better than not interpolating. That is why many DAC manufactures elect to do it.

Maybe we should all stop the theoretical discussions and just let the OP experiment for himself.

[Vincent Kars]

"All incoming audio data, such as CD quality 16-bit/44.1kHz, is upsampled to 24-bit/384kHz using a high-end 32-bit Analog Devices DSP (Digital Signal Processor). "Polynomial Curve Fitting Interpolation" provides a much more accurate upsampled audio soundwave than traditional upsampling processes.

Yank

Open DSP studio > Output format
Set JRiver to convert everything to 384.

This will tell you who does the better job, the DSP in the DAC or the DSP as done by JRiver
Enjoy

[theoctavist]

experimentation is worthless. upsampling from redbook to 96 or 192 DSD ....will not offer any benefit.  it just wont.  that is a fact, well known to anyone who has knowledge of sampling theory/digital audio.

wooden quality? yikes.

JRiver is such an amazing program, I really hope the forum remains a place for discourse that is rooted in fact rather than of the sort that fills fantasy books.

[JimH]

JRiver is such an amazing program, I really hope the forum remains a place for discourse that is rooted in fact rather than of the sort that fills fantasy books.

Thanks.   You can depend on it.  But we also have to allow reasonably free expression.

[Vocalpoint]

I really hope the forum remains a place for discourse that is rooted in fact rather than of the sort that fills fantasy books.

+1

While I can certainly appreciate the fact that people's ears are their own and that they will hear (or convince themselves to hear - via expectation bias) whatever they need to in order to satisfy some specific internal requirement - please let's not start thinking that pressing a button here or activating some option there in MC is somehow "magically transforming" a rigidly defined set of (16/44.1) bits (that mathematically speaking) - cannot have anymore "goodness" added to it.

VP

[tyler69]

But what if a DAC itself decodes 192/24 audio with a higher quality, than it does with 44/16 audio? Is this something considerable or is this a stupid question due to my limited technical knowledge?

[dtc]

experimentation is worthless. upsampling from redbook to 96 or 192 DSD ....will not offer any benefit.  it just wont.  that is a fact, well known to anyone who has knowledge of sampling theory/digital audio.
 

I would not dismiss the possibility of upsampling adding quality to the sound so quickly. It is not going to produce the quality of a file actually sampled at 192K (at least not currently) , but many people do believe it is possible for the upsampling to improve the sound somewhat. I would say that the large number of DAC designers who choose to upsample within the DAC probably do know something about sampling theory/digital audio.

Remember that no DAC can perfectly reproduce the analog from the Digital signal. The implementation of the transform is an infinite series and never exactly reproduces the original analog unless you have an infinite number of terms. Shannon/Nyquist is a great tmathematical formula but actual implementations do not reach the ideal of the formula. There is also the effect of the 22 KHz filter on the analog. Some you tell you that simply removing the effects of that filter increase the sound quality. For many that is why 88KHz or 96 KHz upsampling is good. It can remove the 22 KHz filter.

You might want to read some of the observations of Rob Watts, the designer of the Hugo DAC for Chord. He believes, that by greatly extending the number of taps in his D to A  44KHz sounds now much closer to 192KHz, because the extra taps bring the extra resolution closer to Shannon/Nyquist. That just shows that implementation is different than theory. Note that the Hugo does not use a commercial DAC chip. Watts writes the code himself and he will tell you that different decisions he makes in the code produce different results, as does the number of taps.

I understand this topic can be controversial. But simply declaring that upsampling can have no effect on sound because you understand sampling theory/digital audio is not enough. Others who understand sampling theory disagree with you and the various approaches to DAC design speak to that.

We do not need to have a long drawn out discussion about this here. It has been time time and time again in gory detail. There are proponents of NOS DACs and of upsampling DACs. Let's just say that their are different opinions on the topic.

[JimH]

... some of the observations of Rob Watts, the designer of the Hugo DAC for Chord. He believes, that by greatly extending the number of taps in his D to A  44KHz sounds now much closer to 192KHz, because the extra taps bring the extra resolution closer to Shannon/Nyquist. That just shows that implementation is different than theory.

Your logic is flawed.  It only shows that Watts had an opinion on the subject.

[Vincent Kars]

a rigidly defined set of (16/44.1) bits (that mathematically speaking) - cannot have anymore "goodness" added to it.

If we would live in a perfect world this might be true.
But we don't, we have to deal with implementations.
Just an example: http://www.analog.com/static/imported-files/tutorials/MT-017.pdf

[InflatableMouse]

It's not uncommon for certain DAC's to have a sweet spot. This might be 96 kHz, 192 khz, who knows.

However, I would personally never convert the files and save them in a higher sampling rate. This wastes space and actually alters the files. I believe its best to leave the files in their original state and as suggested by others, use JRiver's real time upsampling. 44.1 and 88.2 -> 176.4, 48 and 96 -> 192 for instance.

[dtc]

Your logic is flawed.  It only shows that Watts had an opinion on the subject.

If you think my logic is flawed, please point out the specific flaws. There are implementation differences. There are issues with 22 KHz filters.

Watts is one example. Yes, he has an opinion. You have an opinion.  The point is that there are different opinions on the topic. Really, why do so many DAC designers upscale the signal?

[Vocalpoint]

If we would live in a perfect world this might be true.
But we don't, we have to deal with implementations. Just an example: http://www.analog.com/static/imported-files/tutorials/MT-017.pdf

Won't disagree with you - however (for most in this thread) - the OP specifically stated:

"How can I use my recently purchased copy of J River MC20 to either rip, or rip and then convert these files?"

Notice the word "convert"...not up-sample, not have MC "interpret" and then play back at 384 or whatever. He said 'convert" - which to me - meant taking the exact bits of a 16/44.1 file and placing them into a 24/192 container.  Doing this does nothing but take those original bits and pad the remainder of the container - with digital zeros. Meaning no change whatsoever in the possible content of the converted file.

Now if you guys want to spiral out into Rob Watts, oversampling interpolating DACs and Nyquist theorems - have at it - but my take on this thread is that this angle is NOT what the OP intended.  He just wants to rip a bunch of standard CDs into 192 FLAC files...

VP

[kensn]

I agree VP. I was under the impression that he thought he could make the 44.1 sound like the 192 files he had by converting them. I have no doubt his 192k files sound fantastic. The upsampling conversation really has nothing to do with the OPs post.

Yank... I hope you are not disappointed with your first post on the forum, truly a great bunch of folks here.

Ken

[dtc]

Won't disagree with you - however (for most in this thread) - the OP specifically stated:

"How can I use my recently purchased copy of J River MC20 to either rip, or rip and then convert these files?"

Notice the word "convert"...not up-sample, not have MC "interpret" and then play back at 384 or whatever. He said 'convert" - which to me - meant taking the exact bits of a 16/44.1 file and placing them into a 24/192 container.  Doing this does nothing but take those original bits and pad the remainder of the container - with digital zeros. Meaning no change whatsoever in the possible content of the converted file.

I think the OP wanted to upsample the file, not simply pad it with a bunch of zeros. It makes not sense to create a much larger file with a bunch of zeros tacked onto the end of the original file.  To get the correct tempo in your padded file, you would also have to keep the timing cell at 44.1KHz, which would just create a very long silence at the end of the track. I do not  believe that that is what the OP is trying to do.

[theoctavist]

oh yeah.

https://xiph.org/~xiphmont/demo/neil-young.html

http://www.drewdaniels.com/audible.pdf

[dtc]

oh yeah.

https://xiph.org/~xiphmont/demo/neil-young.html

http://www.drewdaniels.com/audible.pdf

In many circles referencing the Boston Audio Society will get more eye rolls than affirmative nods. At best, these articles simply show that different people have different views on this topic.

As I said before, a long point/counterpoint is not going to reach a definitive conclusion here. There is by no means a consensus on this topic.

Once again, I suggest the OP do some experimenting for himself.

Then maybe we can move on to power cords

[kstuart]

The Internet is an endless cycle of:

    Newbie posts Xiph article

    Member posts audio engineer's rebuttal of Xiph article in terms of its many misunderstandings of bit depth, signal-to-noise, etc.

    Rinse and Repeat and Repeat and ...

[Mark Powell]

I think the OP has vanished. Wise of him.

Let him do what he wants. He will find out what he likes best. Leave the rest to Computer Audiophile  

I'm an 'expert'. Only experts have a mix-up of 'carefully balanced' stuff like this:

Samsung laptop-W7-JRiver19-JRemote-Cambridge Audio Stream Magic 6-dCS Debussy-cheap Yamaha AV receiver (used as preamp only)-Naim NAP250-Tannoy Kensington-Celine Dion

[Mark Powell]

I have a Cambridge Azur 851c Digital Pre-Amp/CD player...

So why the nasty personal message about  small cashews, mammaries, and only being able to afford to go on a bus?

I was actually AGREEING with you. "I think the OP has vanished. Wise of him. Let him do what he wants", etc.

And my note about my 'balanced' system' was being ironic. Which is why I put in the 'cheap' Yamaha used as preamp only, the $6000 Naim amplifier,the $12,000 dCS DAC, and the $5000 Tannoy speakers.

In your PM to me you make a DIRECT ATTACK on everyone who replied to you. Cheap systems so no wonder NONE of you can tell the difference between 44.1 and anything of higher  resolution.  You said "The Yamaha is not high fi, not even mid fi. more lie crap fi". (Rather more expensive cheap crap than your Cambridge, which is a 'masterpiece' of the Chinese electronics industry.)

Presumably all because you didn't like that upsampling a low res file won't turn it into a high res one. IF YOU HAD A CLUE YOU WOULD KNOW THAT ALREADY.

Your first and only post I see. You are off to a good start.

[Mark Powell]

I have a Cambridge Azur 851c Digital Pre-Amp/CD player

For the elucidation of others. Here's the PM I got from yank2channel. I have reported it to the moderators.
 
"I have other things to do. Since the responses ran from bullchit to fantasy, and the site owner said all the worthless posts(most of them) are because you guys have small cashews, I decided there is no support. Besides, after careful consideration and a cursory look at the "systems" most members here listen with, I realized you couldn't hear a difference in 44.1, 96 or 192 if your mothers' mammeries depended upon it. Yours for instance...yamaha?. that isn't even hi fi, not even mid fi, more like crap fi. So its no wonder the standard reply is "Nothing is better than 44.1" because you are failures, make no money and have pieces of cobbled together crap as systems. Kinda like eating all your meals out of a can and then nay saying gourmet foods. Or traveling everywhere by bus and downing first class air travel. Or being a really no talent artist and comparing yourself to Gerhard Richter. Reminds me of the guy that can tell you how to make love to a woman 30 different ways, but doesn't personally know any women."

[6233638]

Conversion during playback:

1.  Player → DSP Studio
2.  Select sample rate conversion in Output Format:
 



Offline conversion:

1.  Select tracks
2.  Tools → Library Tools → Convert Format
3.  Options → Audio → Apply DSP
4.  DSP Settings → Select sample rate conversion in Output Format, as above.

[Vocalpoint]

In your PM to me you make a DIRECT ATTACK on everyone who replied to you.

Mark,

I also received a very nasty PM for no apparent reason. The OP came unglued with my comments regarding zero pad etc.

But instead of talking it out in here - in public - like a man - he chose stealth mode like a child. His PM was immediately vaporized so i did not keep the copy.

But it was easily as bad as yours.

VP

[JimH]

For the elucidation of others. Here's the PM I got from yank2channel. I have reported it to the moderators.
 
"I have other things to do. Since the responses ran from bullchit to fantasy, and the site owner said all the worthless posts(most of them) are because you guys have small cashews, I decided there is no support. Besides, after careful consideration and a cursory look at the "systems" most members here listen with, I realized you couldn't hear a difference in 44.1, 96 or 192 if your mothers' mammeries depended upon it. Yours for instance...yamaha?. that isn't even hi fi, not even mid fi, more like crap fi. So its no wonder the standard reply is "Nothing is better than 44.1" because you are failures, make no money and have pieces of cobbled together crap as systems. Kinda like eating all your meals out of a can and then nay saying gourmet foods. Or traveling everywhere by bus and downing first class air travel. Or being a really no talent artist and comparing yourself to Gerhard Richter. Reminds me of the guy that can tell you how to make love to a woman 30 different ways, but doesn't personally know any women."

I'm sometimes in awe of what people come up with.  I received something similar by private message.

[JimH]

Yank,
Give it another try.  Nobody will think worse of you for doing it.

Jim

[mojave]

Well - I won't attempt to enlighten you (I stand clear and let pure mathematics do that) - but if you really think taking a standard 44.1 file and padding it with a bunch of digital zeros - actually makes the file sound better (not to mention grow by 4-5 times the size) - rock on bud. I wish I could somehow sell you hard drives after you finish your conversions

Cheers,

VP

Agreed.. Taking a normal 44.1 rip and then turning it into a 192 does nothing for the sound. You still only have the original bits with a bunch of zeros padding the file to create the 192. They will sound the same.

Ken

I don't think in MC conversion there is anything added to the audio.
From the Wiki:

Conversion
Converting from less bits to more bits is perfectly lossless. Conceptually, imagine adding bits like adding zeroes at the end of a decimal. For example, the number "10" might become "10.0" or "10.00" if you add more bits, but all three representations are perfectly identical.

I didn't see that this was addressed later in the thread. Bit depth conversion (16 to 24 bit, for example) and sample rate conversion (44.1 to 192 kHz, for example) are not the same thing. You are entirely correct about bit depth conversion. However, the OP was asking about sample rate conversion. Bit depth conversion is padding and sample rate conversion is interpolation. Higher sample rates do measurable things such as change the latency and lower the dithered noise. It also changes the way the DAC has to do the analog reconstruction. Did you know that to get the correct Peak Level (R128), Audio Analysis in JRiver needs to resample to 192 kHz during the analysis process?