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In Reply to: RE: Oppo 205 scuppered my plans... posted by Jim Pearce on May 30, 2017 at 08:55:42
My Ayre C-5xeMP plays HDCDs, but I think a used one may still be pricey. I'd suggest you try playing the HDCD discs with the 205. You may not miss the added dynamic range. The sound with the new 205 is impressive.
> > My Ayre C-5xeMP plays HDCDs < <
Yes, but it does not decode them. The Ayre DX-5 Blu-ray player does decode HDCDs via the MediaTek system chip (that also performs the video decoding). Now the patents have expired so the Ayre QX-5 DAC includes HDCD decoding as well.
To the OP here are two things to consider:
1) In my opinion the vast majority of the sonic improvement due to HDCD was actually the fact that Keith Johnson designed an audiophile-quality piece of equipment, and very little to do with the compansion scheme used by only *some* "HDCD" discs.
Just because a disc lights up the "HDCD" light does *not* mean that it requires any decoding. For more information on this, please refer to the link below.
2) While I am not aware of any DACs with both HDMI inputs and HDCD decoding, there are some HDMI-to-S/PDIF decoders available for very reasonable prices. Monoprice has a small box that will do exactly what you want for under $50:
Then you can use any DAC with HDCD decoding. If you go this route, I strongly recommend using a Toslink connection to your DAC as this will provide galvanic isolation between your audio and video systems, thereby improving the performance of both (and avoiding any possibility of creating ground loops).
Hope this helps.
As always posts here reflect solely my personal opinion, and not necessarily those of my employer or bartender.
I misremembered. The C-5xeMP plays an HDCD disc as a CD, but the Oppo 105 recognizes it as an HDCD.
I've been trying to address this question to Charles Hansen. Oppo Tech tells me I should expect the balanced and unbalanced outputs from a 205 to sound identical. Is that also true of the inputs to an Ayre K-5xeMP preamp?
"The UDP-205 player revealed major differences using the XLR analog outputs. It's not subtle - I noticed it immediately."
As far as I can tell, "The UDP-205 player revealed major differences using the XLR analog outputs. It's not subtle - I noticed it immediately." is not about XLR v RCA but about the Oppo 205 v a 10 year old Ayre disc player that also uses XLR.
I agree, the context makes the reference ambiguous at best. On re-reading he is comparing to his Theta VIII series 3 DAC/pre-amp. But he did say (above):
"The balanced connectors are a true differential design from the ESS Sabre Pro DAC to the XLR connector, and should help improve the sound quality, especially in systems that have the same balanced design goals."
Oppo claims the outputs on its player sound identical, yet a reviewer found "major sonic differences...noticeable immediately". So what's up with that?
In my experience, most (even those who should know better) don't control all the variables properly when conducting comparisons such as this. I've seen experienced reviewers writing for print magazines attempt to reach conclusions while using different brands and models of cables for each connection. And even if there is a difference, is it due to the source or the receiver?
It's a difficult experiment to control, and may not even be possible with some equipment. An example of this was Sony's original SACD player, the SCD-1. It was pretty much universally agreed that the single-ended outputs sounded better than the balanced outputs. When you look at the circuit, this makes perfect sense. At some point after the DAC chip, the signal was purely single ended. The RCA jacks connected directly to this point, but the positive (+) and negative pin (-) of the XLR went through two op-amp buffers with 100% negative feedback (to create unity gain), one wired in non-inverting mode and the other in inverting mode (to create the out-of-phase signal). It's no surprise that when connecting that output signal to a true differential balanced input on a preamplifier that the result would be a reduction in sound quality, as signal was passing through an additional amplification stage of lower performance than the discrete unbalanced circuitry used for the single-ended output.
The reason for audiophiles to use balanced has almost nothing to do with the reason balanced connections are mandatory in professional recording studios. In the pro environment they are worried about picking up external magnetic hum fields in long (100') cables carrying low-level (microphone) signals. That is not a problem at all for home playback - I've even used 20' unbalanced cables between a low-level MC phono cartridge and the preamp without hum problems.
The real sonic advantage of balanced only comes into play if the internal circuitry is truly differential from input to output. Since every circuit in the world (analog or digital, tube or solid-state, audio or video) is nothing more than a modulated power supply, it only makes sense that the quality of the power supply is important. And as there is no such thing as a "perfect" power supply, gains can be made by reducing the audio circuit's sensitivity to imperfections in the power supply. A differential balanced stage will typically exhibit an increase in power-supply rejection ratio (PSRR) between 40dB (100x) and 80dB (10,000x) over identical single-ended circuitry. Therefore using balanced differential circuitry is like making the power supply between 100x and 10,000x better.
The easiest way to hear what difference this makes is to listen to a Pono Player using a headphone with detachable cables. When comparing otherwise identical single-ended and balanced harnesses (same cable manufacturer and model), the signal path from the DAC chip to the headphone driver is fully balanced differential all the way through for a balanced harness, while a single-ended harness is fully balanced differential *except* for the final output buffer stage. The only variable in this test is whether a single stage in the chain is operated in single-ended or differential balanced mode. There are many reports on the internet about the improvement in sound quality available from using a balanced connection, but I've never seen the opposite conclusion.
The bottom line is that properly executed differential balanced circuitry will always sound better - but only if all else is held equal. The penalty is pretty much a doubling of the circuitry and parts count - hence price, size, and power consumption. There are other ways to both improve power supplies and also reduce a circuit's sensitivity to imperfections in the power supply, but those can also be applied to differential balanced circuits for further improved performance. In other words, it is also possible to make a single-ended circuit perform better than a balanced circuit - but only by improving the single-ended circuit and not the balanced one. That is why the caveat "if all else is equal" is so critical.
As usual, strictly my personal opinion and not necessarily that of my employer or hairdresser.
Oppo may claim the balanced and single-ended outputs of the 205 should sound identical, but using a balanced connection between my 205 and Ayre K-5xeMP resulted in more detail and a greater sense of transparency. The improvement over single-ended was very noticeable. I played CDs, SACDs, and Blu-rays, jazz trios, baroque, and opera. In each case, the sound was superb.
Thank you, Charles Hansen, for a post that convinced me to make the change.
> > using a balanced connection between my 205 and Ayre K-5xeMP resulted in more detail and a greater sense of transparency. The improvement over single-ended was very noticeable. < <
Thanks for sharing your experiences. Enjoy your new Blu-ray player!
Oppo actually claims that the xlr output sounds better, no matter what customer service says:
"The stereo output section offers both XLR balanced and RCA single-ended connectors. By transmitting a pair of differential signals, the balanced output provides better common-mode noise rejection and improves signal quality."
Actually their claims across their audiophile players - BDP-95, BDP-105 and now UDP-205 - remain consistent:
1. The dedicated L,R RCA outputs sound better than the L,R front 5.1
2. The xlr outputs sound better than the dedicated L,R stereo outputs.
Personally I can't attest to having an opinion even with the BDP-95 because I use the xlr outs to defeat bass management in the player without having to dive into the menu. The one thing I'm sure of is that any given stereo or mono disc will sound better in 2.1 (or 1.1) with BM or through the xlr outputs.
> > By transmitting a pair of differential signals, the balanced output provides better common-mode noise rejection and improves signal quality." < <
Yes that is true for specific situations, that are mostly found in recording studios and not home playback systems. My experience is that the common-mode noise rejection of the signal input is not nearly as important as the fact that *differential* balanced circuitry will also provide the same type of rejection to imperfections in the power supplies. The importance of power supply quality becomes apparent when one realizes that the audio circuitry (indeed *all* electronic circuits) are various types of ways to modulate the power supply. Just as with computers, garbage in = garbage out (GIGO). Hence the importance of power supply quality. Differential balanced circuits add an entire extra level of rejecting imperfections in the power supplies.
> > The dedicated L,R RCA outputs sound better than the L,R front 5.1 < <
For the Oppo players this is certainly true of the measured performance. They normally use an 8-channel DAC chip for the 7.1 analog output, whereas the dedicated stereo outputs typically parallel 4 DAC chip channels for each stereo channel. This improves the S/N ratio by 6dB, assuming all else is held constant. Oppo may also use better parts and or power supplies on the dedicated stereo channels (they may be too expensive to use for all 8 channels, just as most customers use higher quality main speakers than surround speakers).
> > I use the xlr outs to defeat bass management in the player without having to dive into the menu. < <
That's fine for all CDs, but it turns out that there are some DVDs that were authored improperly with regards to bass-management. The normal 2-channel mixdown will *not* recover the bass on these discs if you have an HT-2.0 or HT-2.1 system. Ayre added a feature on the DX-5 to correct for this, as described in the owner's manual. To the best of my knowledge, this is the only Blu-ray/DVD player in the world with that feature. Hope this helps.
I play all MC discs in MC mode through the 5.1 analog outputs. I thought it was generally accepted that most common-mode noise comes from the power supply.
If you are not mixing down to stereo, there is no problem created by improperly authored video discs.
> > I thought it was generally accepted that most common-mode noise comes from the power supply. < <
There is at least one other source of common-mode noise found in all digital products - switching noise from the DAC chip itself. With each new sample the output switches to a new state. Real world devices are imperfect and have parasitic capacitances, such that some of the electrical energy used to turn the individual DAC switches on and off is coupled into the audio output signal. Using either two single-ended DACs or one balanced output DAC per audio channel means that the switching noise will be the same in both phases. If the first analog stage is a true differential balanced design, it will reject the common-mode switching noise from the DAC chip. (This is akin to what happens in the recording studio where a low-level microphone signal picks up line frequency hum from the AC wiring in the studio walls equally in both conductors of the balanced mic cable.)
As always, only my personal opinion and not necessarily that of my employer or dog-catcher.
Will the ESS minimum phase slow filter on the Oppo 205 likely approximate what you have done with "MP"?
> > Will the ESS minimum phase slow filter on the Oppo 205 likely approximate what you have done with "MP"? < <
Good question, but hard to say with any certainty. The only real way to know is with listening tests, but there is enough information that we can say "probably not". Although both are described as "slow rolloff, minimum-phase digital filters", there are many differences we can see and likely more that we can't see. About the only visibility I have into the ESS filter comes from the datasheet, which is not publicly available. It shows graphs of both its frequency response and its impulse response. Based on that alone, there are significant differences between the filters. Ayre's single-rate slow rolloff MP filter has about 1-1/2 cycles of post-ringing, while the ESS has about 4-5 cycles of post-ringing.
We have a few data points to help us predict the audibility of this difference. Ayre DACs with a "Listen"/"Measure" filter selection allow the choice of two minimum-phase filters, one with about 20 cycles of post-ringing, the other with 1-1/2 and the difference is fairly easily noticed. In contrast when JA recently reviewed the Meridian Ultra DAC (link below), he found it virtually impossible to hear any differences between minimum-phase filters, the shortest with about 6-7 cycles of post-ringing and the longest with ~35 cycles of post-ringing. Based on these two sets of data points, I suspect that the filter length alone would create an audible difference between the filters. However I also suspect that the slow rolloff MP filter in the ESS DAC chip would be my preference of the 7 choices offered.
That is just the tip of the iceberg, however. While the digital filter is one part of the design that affects the sound, there are many other aspects to digital filter design not mentioned above, including the window shape, the dithering applied, the interpolation rate (ESS uses 8x, while Ayre uses 16x). Further there are many more factors that affect the sound of a D/A converter In my experience, digital filters are one of about a half-dozen major design aspects that will significantly affect the sound of a digital product. Also important are the analog circuitry, the power supplies, the clock implementation, the DAC chip itself, and the presence or absence of any DSP algorithms (such as Asynchronous Sample Rate Conversion - ASRC - and many others). I would be loathe to rank the importance of them, but would agree that the digital filter (or lack thereof, in the case of "non-oversampling") plays an important factor in the overall sound quality.
As always, strictly my own opinions and not necessarily those of my employer or trash collector.
> when JA recently reviewed the Meridian Ultra DAC (link below), he found it
> virtually impossible to hear any differences between minimum-phase filters,
> the shortest with about 6-7 cycles of post-ringing and the longest with ~35
> cycles of post-ringing.
My comment was made auditioning a 192kHz file, whereas my impulse response
measurements were made with 44.1kHz data. After the review was published,
I was told that the Meridian's filters behave differently at 2Fs and 4Fs rates from
how they do with baseband data. I had assumed they were the same at all rates.
You're ability to explain things in a simple and concise manner, so clear that EVEN I can begin to understand, is much appreciated!
Thanks for the kind words, and glad I could be of assistance. However many would disagree with your assessment of my posts as "concise"... :-)
Enjoy your weekend!
Thanks for the detailed explanation. I think I now understand why I should use a balanced connection between the Oppo 205 and Ayre preamp.
> Now the patents have expired so the Ayre QX-5 DAC includes HDCD decoding as
Any thoughts on how the QX-5's decoding of HDCD compares with
dbpoweramp's rips of HDCD CDs, Charley?
> > Any thoughts on how the QX-5's decoding of HDCD compares with
dbpoweramp's rips of HDCD CDs, Charley? < <
Good question, but I don't know the answer.
HDCD decoding has been available in both dBpoweramp and Foobar for many years, and for many years there have been various versions of the HDCD algorithm floating around the internet. Looking at the source code for the two versions I could find, I am clear that the most important HDCD feature (expansion curve for the "Peak Extend" function) is slightly off in both. Apparently the values for the inverse-gain curve were estimated by eyeball from the graphs published in the AES paper in one version. While the other version was alleged to have come from a decompilation of the code used in Microsoft's Windows Media Player (MS purchased Pacific Microsonics in September 2000), it also contained obvious errors in the inverse-gain curve.
I'm far more OCD than to use inaccurate values and instead took the effort to learn the exact 16-bit values for every single point in the curve (which compresses the top 9dB of signal into only 3dB on the disc). The end result is that I am confident that Ayre's implementation is bit-accurate with the original, while the other versions I've seen are not. I am unsure if the code used in dBpoweramp is based on the slightly inaccurate code circulating on the internet or if they also took the trouble to get it exact. I am also completely unsure of the audibility of the errors in inaccurate versions.
> The end result is that I am confident that Ayre's implementation is
> bit-accurate with the original, while the other versions I've seen are not.
Thanks very much for the explanation, Charley. I asked because I have
been getting some anomalous results ripping HDCD discs with dbpoweramp,
in that the true bit depth of the resultant file varies. I did wonder if this was
due to interpolated disc errors during the rip.
> > I have been getting some anomalous results ripping HDCD discs
with dbpoweramp, in that the true bit depth of the resultant file
varies. I did wonder if this was due to interpolated disc errors
during the rip. < <
I seriously doubt that any possible errors in HDCD decoding would lead to this result. Following is the curve for the Peak Extend (PE) function given by Keith Johnson in the AES preprint:
The vast majority of the possible values map in a 1:1 relationship, which results in the 45 degree line at the left part of the graph. The end point is also trivially calculated, as it simply compresses the input signal by 1 bit (= 6.02dB). The errors I've noted in all of the available versions of software HDCD decoders are simply in the shape of the curve. This would translate as a slightly non-linear transfer function and therefore introduce a small amount of harmonic distortion - but *not* a meaningful change in the amplitude of that signal.
I would guess what you are seeing is simply variations in the amount of "headroom" the mastering engineer left in the recording. Modern "loudness war" pop material would always have full-scale audio data. With Peak Extend, this would expand the 16 bits available from the Redbook format to 17 bits. But dBpoweramp puts the decoded data into a 24-bit container (for compatibility reasons), and pads the 7 LSBs with zeroes.
On the other hand I have seen some recordings, especially those of classical music and/or audiophile labels that leave more headroom. The signal rarely (if ever) even gets within a few dB of 0dBFS. Depending on the tools used to measure the bit depth, it seems possible that they might simply report the number of *active* bits and ignore the static zero bits used for padding. In that case a recording that never used the very top bits might be reported as having a lower bit depth. However, this is all just guesswork on my part as to what you are seeing.
As always my posts reflect my own opinions, and not necessarily that of my employer or cable installer.
The other thing about dBpoweramp (which I use and love) is that they have established what I believe is the best method to ensure the accuracy of a rip, which they call Accurate Rip. When you rip a CD, it creates a checksum which is then compared to the checksums of all of the other rips of the same disc from other users world-wide. When you have > 3 rips from different users with different discs and different CD mechanisms all giving identical MD5 checksums, one can be absolutely confident of having a bit-perfect rip.
While both Apple's iTunes ripper and Exact Audio Copy (EAC) use some tricks to ensure an accurate rip, the dBpoweramp ripping software also performs the same tricks. However it alone can compare your rip to those of other users with different discs and CD mechanisms. When ripping an HDCD disc you should also be able to use dBpoweramp's Accurate Rip feature to ensure that the rip itself is bit perfect and that there have been no interpolated errors. If you are getting bit-perfect rips of your HDCDs, I don't see any way that dBpoweramp's HDCD decoding could possibly be so poor as to affect the decoded bit depth.
dBpoweramp has always been at the forefront of digital audio tools. There was a period of many years where their reverse-engineered ALAC encoder out-performed Apple's own tools. Specifically if one used iTunes to transcode a high-res file to ALAC, it would play perfectly fine. But if one took that same ALAC file and transcoded back to WAV with iTunes, it would be truncated to 16 bits. It took years for Apple to correct, but dBpoweramp's version never had a similar problem.
As usual all postings strictly my own opinions, and not necessarily that of my employer or organ-grinding monkey.
Two winters ago, I ripped my whole CD collection using dBPowerAmp. Having no desktop computers in the house--indeed, no computers with built-in optical drives--I used my laptop with external drives. I found that a minority of the CDs I ripped--somewhere in the 10-20% range--were hard to rip; they were often new and perfect-looking, but they'd get caught up in dBPoweramp's error-correction algorithms. (Not a complaint about DBPoweramp--just wait and I'll get to the point.)
When I was maybe halfway through, after a particularly frustrating rip, on a whim I pulled out a Gordon-Rankin-designed AQ Jitterbug and added it between my laptop and the external drive. After that, I didn't have a single frustrating rip. I didn't do the math, but this is a convincing sample: Hundreds of trials with and without. circa 10% error rate without, 0% error rate without.
If you need to rip a ton of CDs, I highly recommend the combination: dBPowerAmp +AQ Jitterbug.
As Paul Simon said on some recording or other circa 1969--was it the Carnegie Hall one?--, "Apropo of nothing..."
> > I pulled out a Gordon-Rankin-designed AQ Jitterbug and added it between my laptop and the external drive. After that, I didn't have a single frustrating rip. < <
That is a *great* story. And I would agree that its really not worth the trouble to "scientifically" test it by re-ripping the same disc with and without the JitterBug - life's too short. The likelihood of it being a coincidence must be extremely small. Gordon has stated that he has seen USB packet errors due to noise on the connection that are corrected by the JitterBug. Your experience would seem to corroborate this.
I've never used an external drive, but I also use dBpoweramp and have had similar problems with "difficult" rips that have tried my patience. I don't want to open up my laptop and try to install a JitterBug inside, but certainly can imagine that whatever was causing the problems in your setup could also affect other external devices, such as DACs. Thanks for sharing.
Thanks for the detailed responses,Charley. I did use dBpoweramp's Accurate Rip so
am not sure why I was getting anomalous results. This was with a. DAC that indicated
bit depth on its display, BTW.
1) A decoded HDCD file will only have a maximum number of active bits of 17 in most cases (1 extra bit if Peak Extend (PE) is engaged) or possibly 18 in rare spots in some tracks with extremely quiet passages if both PE and Low Level Extension (LLE) are engaged. (Please remember that PE is always on for the entire disc, while LLE - if engaged - only auto-activates when the peak signal level falls below -45dBFS.) This audio data could be placed into word with lengths anywhere between 18 and 32 bits before being transmitted.
We don't know exactly how dBpoweramp packs the 17- (or sometimes 18-) bit data. The two obvious choices are to pad the LSBs to create either 20-bit words or 24-bit words. I would imagine that 24 is the more likely number. One way to check this would be to compare the file sizes of the undecoded and decoded tracks (taking care to avoid erroneous reporting due to the smallest cluster size created during disk formatting). Also different transport protocols (eg, S/PDIF, USB, Ethernet) may have different word lengths permitted.
2) We don't know exactly how the DAC calculates the reported bit depth. One way would be to count the number of bit clocks per word clock. It is conceivable that this could vary at different points in the DAC's circuitry due to the way that different transmission protocols are decoded. Another way is to simply read the Status Bytes in incoming S/PDIF data. If the professional format is used (as opposed to the consumer format), the source can optionally send data on bit depths for 16, 18,19, 20, 22, 23, and 24 bits - but *not* 17 or 21 bits.
The only thing of which I am confident is that once we understand what is happening, it will all make perfect sense in hindsight... :-)
My problem is finding high quality HDCD compatible DACs with xlr outs at all, forget about the HDMI in part. And yes, I'm aware that not all HDCD discs benefit from decoding.
Understood. While every Ayre product ever made includes balanced inputs/outputs, the DX-5 Blu-ray player was the first Ayre product able to decode HDCD files. (That function was built into the MediaTek chipset used.) As a general principal it doesn't seem to me to be a good idea to lose functionality as new products are introduced, so the recent QX-5 Twenty DAC also includes HDCD decoding - this time implemented in an FPGA. (As an added bonus, it differentiates between discs made with the Pacific Microsonics D/A converter that do *not* require HDCD decoding - so-called 'fake HDCDs' - and those that *do* require HDCD decoding, and display that information to the user.)
As always, my posts reflect my own opinion and not necessarily those of my employer or favorite singer-songwriter.
The QX-5 looks like a great machine Charles, but I'd hate to own something like this and use so little of its capability. For the most part I just spin my outrageously large collection of Hi Rez discs. The Bryston - with its SACD capability - is a much better fit for me, as far as I can see.
The more I look at this disappearance of HDCD capability issue the more I'm inclined to temporize by leaving the BDP-95 in the system (RCA stereo only) along with the UDP-205 until I either rip my crucial HDCDs with dBpoweramp or decide that I don't like this outcome (or it really isn't necessary in most cases).
A great discussion thanks to you and JA.
I went without HDCD decoding for several years with my Esoteric DV-50. The Oppo BDP-95 was a big step up playing HDCD CDs. Otherwise - of course - not so much. Actually I considered a C-5xeMP, but it would likely cost about the same as a Berkeley DAC 2. I'm pretty sure a Bryston bda-3 (at $3200 Canadian) using my BDP-95 as transport would perform as well as the C-5xeMP on stereo material. Maybe not on the HDCD where I'd be using the Oppo on 5.1, but the HDCD sound is very good.
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