Official OPPO BDP-105 Owner's Thread - Page 99

 

Dunno about the USB input but you can do multichannel via ethernet although not DSD (yet).

Throw in stuff like ambient noise level of 30db and perceptual noise shaping and it gets really hard to justify needing more than 16 for reproduction.

Did what you suggested without success. I switched the stereo signal from FRONT LEFT/RIGHT to DOWN MIX then back to FRONT LEFT/RIGHT. It did not correct the issue. Leaving the multichannel disc in the player, I powered down then powered back up. I then got output from all 5 channels.

Dont get it. Will send to OPPO support.

I have a Pioneer Kuro display and some consider it "high end". i read the manual and still do not understand the benefits of using SOURCE DIRECT. What exactly is the 105 passing through to the display?

thanks

AES 69th Convention, 1981, May 12-15, Dynamic Range Requirement For Subjective Noise Free Reproduction Of Music (available as reprint, which I have) found that "A dynamic range of 118 dB is determine necessary for subjective noise free reproduction of music in a dithered audio recorder. [...] Standardization of a 16 bit linear format would limit the dynamic range capability to 96 dB"

So, CD was close, but not quite good enough, despite what Moran et al claim to have found.

I think though that lower-resolution music recorded properly can sound better than higher-resolution music that's been ProTooled to death. The care taken at the studio, and reduction of processing to "improve" the signal makes more of a difference than the bit rate/depth. So why do SACDs sound better to me than CDs? Probably simply because when authoring an SACD, the person twiddling the knobs is trying to make it sound as good as possible.

If you want to hear some minimally-processed regular CDs that sound really good, try the ones from Mapleshade. The Fabulous Sounds of Three Blind Mice XRCD is also excellent, but sadly out of print and horribly expensive.
Edited by jimshowalter - 1/18/13 at 10:08am

Can you not also do MC via the 105's HDMI inputs? Mac Minis, as one example, have HDMI ports....

If you choose source direct on the Oppo, it sends the video out without any processing on the Oppo, and the processing then depends on your display.

I have a Kuro, and it does an excellent job of processing the signal, but the Oppo also does an excellent job, and so our setup is configured to use the Oppo for processing, and the Kuro is set to PURE, Dot-by-Dot.

Which you prefer is entirely up to you. It's not clear that I can see a difference either way, but by consolidating all of the processing in one device might make it easier to deterministically control sync delays, etc.

It's analogous to the discussion about whether the DACs in the Oppo or the DACs in an external processor should be used for audio.

(At one point we also had a Lumagen, and used that to do all the processing, with the Kuro set to PURE Dot-by-Dot and the Oppo set to SOURCE DIRECT.)

thanks Jim. I'm in the middle of having the Kuro ISF calibrated to the 105. I'm thinking this might make DIRECT a better selection?

Not sure if plex has been tried by any one or not? Plex server and plex player suppose to be running on the pc with using oppo as usb dac soundcard or hdmi or what ever other way you can feed sound output of pc to the oppo, then plex client/player app on the ipad controls the pc plex palyer. Plex app has the option as to what player we want to use, so choose the pc player insad of ipad or iphone. To me plex is better than everything else, just the cover art display alone makes it look beautiful. Also it gives you ability to control audio video controls while the content is being played, including lip synch. It has never failed me on oppo.
I would request oppo to include plex as an app along with existing netflix, pandora etc. it will take oppo to whole new level. Oppo already is my most favorite interms of cable box picture quality. With great sound quality, this a keeper.
Edited by FlatRocky - 1/18/13 at 10:27am

That throws a curve into the question. Depending on your willingness to part with money, and the capabilities of the calibrator doing the work, you might consider calibrating both the Kuro and the Oppo. If you put the Kuro in Dot-by-Dot, I think that bypasses the calibration settings, but doesn't lose them. Same is probably true for the Oppo. So you could run Dot-by-Dot with the calibrated Oppo doing all processing, then compare to calibrated Kuro with the Oppo in direct mode and doing no processing.

Update: Scratch that. From the unofficial FAQ http://watershade.net/wmcclain/BDP-83-faq.html#calibration: "Do I adjust the player or the display? The standard advice is to adjust the display unless you have some specific reason not to. The display is likely to have finer adjustment controls than the player."

Putting the Kuro into dot-by-dot mode should just turn off scaling. It has nothing to do with the calibration itself.

Putting the Kuro into dot-by-dot mode should just turn off scaling. It has nothing to do with the calibration itself.

Digital music playback has ALWAYS been affected by the cable used to transmit the digital audio from one place to another. Movie video and sound playback is a "stop and start" playback system... video is essentially still images... they only stay up for a brief period of time, but the images are just a sequence of still frames. As long as the data is accurate, the images are accurate. Movie sound is interleaved with video over the HDMI cable and is synchronized with the frames (unless there is an unexpected delay in the signal path, of course, video or audio) -- both frames and audio are buffered and march-stepped along until it is "their turn" to "appear". Music playback adds the inescapable analog element of time... music playback is a continuous process that is VERY susceptible to timing errors. This is MEASURABLE and has been proven mathematically in the mid 1990s by Dr. Malcolm Omar Hawksford. For the last 15 years or so, measurement equipment sensitive enough to detect bit jitter in the 10s of picoseconds has been available and it is now well understood that low jitter sounds better and produces less distortion in the analog signal than higher distortion. Jitter arises from many sources and affects the gaps between bits... when those gaps are uniform, digital words are accurately converted to analog without added distortions. When bits are displaced forward or backward in time (most audible at 100s of picoseconds and up), distortions in the analog audio signal are measurable and can even be calculated mathematically. Jitter can be constant or cyclical depending on the source causing the jitter (power supplies, interference from other high-frequencies, etc.). HDMI does not (currently) transmit the master clock with music data and tends to run much higher in jitter because of that. Cable design can reject more noise sources and preserve more of the timing integrity of the bits.

In ALL THINGS DIGITAL... cables don't matter. If you understand what I wrote in the previous program, you will understand that music playback is never "digital" and is perhaps the ONLY thing we do with computers and digital that is NOT purely digital. Time is an analog entity... you cannot change it. Music is FULLY digital when you send a recording of yourself playing marimba to a friend in Laos... you send the bits, your friend receives exactly the same bits you sent. When your friend LISTENS to that digital music, he/she is listening to an ANALOG event, not digital. All because of TIME. When data is NOT timing critical, you're in the digital domain. As soon as you begin doing anything (like listening to digital music files) you crossover into the ANALOG domain. And from decades of experience, we know EVERYTHING matters in the Analog domain... everything. Toslink is a bad connection option for digital music playback because the interface itself is not designed to maintain the timing accuracy of the bits. There are MUCH better optical interfaces (like the AT&T glass fiber optical interface, but it is far more expensive with cables that are also expensive and susceptible to damage). Toslink exists because the interface parts are dirt cheap, not because it is a good quality interface for music. SPDIF over coax is problematic because coax terminated with RCAs does not maintain 75 ohm impedance in the cable... BNC connectors would have been a far better standard for SPDIF because they maintain the 75 ohm requirement that reduces internal signal reflections along the cable).

Right now, the best sounding transmission method for music playback in my experience is USB, particularly when the DAC receiving the digital music operates asynchronously (the DAC controls the flow of data from the computer... if the USB DAC is not asynchronous, the computer controls data flow to the DAC and that is NOT a good thing).

Of course if you are only listening to MP3s or compressed downloads from iTunes over a system with a low-to-midrange AVR, none of this talk of bit timing will matter enough to worry about. But once you cross the threshold into high-end quality audio (need not be horribly expensive, there's a ton of great performing high-end quality components for sale used here and on other audio-related online forums or ad-sites), there are a lot of details that do become audible enough to be worth optimizing.

Mebbe.  Never tried it but my apartment/house is already wired for ethernet and HDMI has length constraints.

Just having 192k (or 384k) DACs doesn't mean the USB interface has the same sample rate capability. I've reviewed equipment costing up to $10,000 that has a USB DAC input limited to 48kHz sample rates. It doesn't matter what you set the JRiver app to, the DAC will tell the app that it can't accept input at higher than 48 kHz and, in response to that information from the DAC, the JRiver app (or any other app) will limit all output to nothing higher than 48kHz. It would appear that Oppo has knowingly or unknowingly chosen a USB interface that can't accept sample rates higher than 48 kHz. I haven't tried this yet myself, but I will be doing this soon and can check it with a Mac and PC with different apps.

All that said, employing a USB interface that can accept sample rates higher than 48 kHz is not expensive, you just have to select the right parts. The AudioQuest DragonFly USB DAC sells for $250 and is asynchronous and accepts up to 96 kHz. Finding USB DAC inputs that accept higher than 96 kHz is not easy at this time... there are a few products out there, but most USB DACs operate up to 48 or up to 96 kHz. It would be disappointing if the USB DAC input on the 105 is limited to 48 kHz for those of us to appreciate the prettier sound you get from good recordings at higher sample rates.

If bits are streamed and clocking is done on the receiving end, the cable is irrelevant. The bits could be typed in by hand on a manual typewriter, scanned, OCR'd, and turned into bits on the receiving end, and provided the same bits wind up there, the transport medium is utterly irrelevant.

I agree that timing is critically important when the samples are taken, but the cable isn't taking samples, it's just transmitting samples.

I agree that timing is critically important when the samples are converted back to analog, but the cable isn't performing D/A, it's just transmitting samples.

If there are digital transport standards idiotic enough to make the timing depend on the cable, they should be abolished. Are there such standards? And, even if there are, what's to keep the receiving end from buffering and reclocking the data?

 

It should also be noted that even though these dacs have 24-bit precision, they are only accurate down to 20-bits, given their SINAD( or SNR+D) specs of -120dB or so. A truly accurate 24-bit dac would have a SINAD of -146.24dB or (6.02 *  24 + 1.76). These ultra low specs do not exist as far as we know from any 24-bit dac on the market today.

 

Also the -120dB spec is about the thermal noise level of a 10Kohm resistor, over a 20KHz bandwidth at a temperature of 20C assuming a 2Vrms full-scale output.

 

A 1K resistor with the same conditions as above has a thermal noise level of -131dB below the same 2V reference output.

 

So it doesn't make much sense then to require more resolution that would be swamped by inherent component thermal noise anyways.

 

24-b resolution, hence 20-bit accurate dacs are hear to stay with us for a long while. The extra 4 bits are just noise flapping in the wind, so to speak.

 

Anyone interested in a free resistive thermal oise calculator can use this one here at: http://www.daycounter.com/Calculators/Thermal-Noise-Calculator.phtml

Great thanks! he is working with both and leaving the Dot To Dot setting

OK, I'm no expert, but from what I've read, the problem is that if it's PCM going over the cable, it is subject to jitter just like spdif is. And jitter in HDMI is potentially a *lot* higher than spdif!
If you use bitstream on HDMI you are OK, because you are just passing the data up to decoder in your AVR and jitter between devices is not inherent in that type of error-corrected transfer.

there's a bunch of threads about hdmi or jitter in general, here's one: http://www.avforums.com/forums/blu-ray-dvd-players/897825-can-jitter-heard-esp-blu-ray-lpcm-over-hdmi.html
I think some of those guys are the same ones you see over in the CD transport threads.

In my experience I think jitter is audible but I wouldn't bet on it.

Completely agree with this - the cable either works or doesn't. Lost samples sound VERY bad.

It is very easy to stay in the digital domain using SPDIF to SPDIF, AES/EBU to AES/EBU, etc and verify bit for bit transfers with no corruption or degradation in the audio data. Studios all over the world do this regularly. The cable, if it works, works. That analog conversion is where the rubber meets the road. However, you would be surprised how well good converters and good analog gear can reproduce audio.

I did a test several years ago to help me understand what was and was not important on a basic level with digital audio workstations. I started with 16 tracks of digital audio recorded onto a pair of TASCAM DA88s - multi-track drums, bass, guitar, keys. My point was to test several DAWs including Protools, Nuendo, Samplitude, and Sonar to determine if there would any sonic differences in an identical mix done on each. I also had a small Allen & Heath 16 channel analog mixer that I used to live sound - not very high-end or expensive, but good basic quality so I decided to also do the same mix using that. I carefully calibrated the A&H inputs to provide the same channel gain as the DAWs - all faders set to 0 and all tracks panned hard left or hard right (this required because of differences in panning law from mixer to mixer). It was also imperative that the recording deck (TASCAM DAT) from the A&H be clocked to the DA88s or otherwise there would be drift between the playback and the recording. Now I had 5 mixes from the various DAWs along with the A&H all equal length - I put them on a CD and proceeded to blind compare them on random playback and making notes....

cymbals clearer on #1, piano smoother on #2, bass more definition on #4...cymbals dull on #1 - say what? My notes after a half hour of listening were completely inconsistent and I started really paying attention. Putting away expectations and being honest with myself, they all sounded the same. So back into the studio and into some digital analysis software - turns out all 4 DAWs indeed were bit for bit identical - proving that math is math and computers are quite good at math. More impressive was the frequency response from the A&H mixer. While it was of course not bit identical to the digital mixes, it was incredibly close across the board. Granted very simple "lab'ish" mixes, but it just goes to show how robust audio processing can be with decent equipment.

It also illustrated that sending the digital mixes across SPDIF into the TASCAM DAT, with the TASCAM DAT locked to the SPDIF clock no less, produced exactly the same bit-for-bit mix as an internal software bounce would create. Digital audio cable transmission is impeccably reliable as long as the specs are followed reasonably well and the equipment is working properly.

When I read the White Paper about the Oppo DACs and how they deal with re-clocking - http://www.esstech.com/PDF/sabrewp.pdf - it seems like they have it covered and should be doing a very good job with the DA conversion.

The 104 manual specs says:
When streaming from JRiver, you can press the information button.
I have up-sampling set up to 192K. Once you start and restart playback, the sampling rate changes in the display.

Here is a screen shot of an up-sampled CD received on the USB at 176.4K

Oppo176.JPG 769k .JPG file


- Rich

Bob

would like to aks you couple questiong regarding the HDMI (video) in 105;
would this connection be possible (confused about "Split AV"; HDMI sound off - using analog directly to my Krell S1200
Video:
BDP105 -->HDMI 1 --> projector JVC RS66U
BDP105-->HDMI 2 --> processor Krell S1200

will I get video in my Krell using HDMI 2? if you are using Split A/V can the HDMI sound be turned OFF? while using Split A/V in 105 is the HDMI still using the QDEO processor? What is used in HDMI 2 in SPlit A/V mode? It says in the manual page 55
"possible video in HDMI 2".

Thank you

Well this is a pretty unusual problem. Your player may need service, although I find it hard to imagine what might be broken which would cause this.

I suspect OPPO is going to ask you to do a fresh firmware install and reset just to get the player into a known state for further diagnosis:

1) Download a new copy of the Official 1220 firmware for USB install and put it on a USB stick.

2) Install the firmware from the USB stick. Accept all 3 parts of the firmware when offered

3) On power up, after the install, do a complete Reset of factory defaults. Also Erase Persistent Storage.

4) Power down the player, then pull the power plug for about 10 seconds. Do not skip this step.

5) Power up and re-enter your personal settings.

6) Power down once more -- settings are saved during power down.

Now see if the problem still exists.

If you are using some sort of "Region Free" hardware mod with the 105, see if the problem exists even without that attached.
--Bob

That paper was written 30+ years ago, before noise shaping became standard practice. The 19.5 bits I referred to is perceptually 118 db.

Not only that, if you actually are able to put 118 db on top of the typical noise floor in a quiet room, you will soon destroy your hearing.
Edited by ehlarson - 1/18/13 at 2:54pm

Yes, if using the Oppos DACS there's nothing to worry about, jitter is eliminated. This is getting OT a bit. But my point is if you are outputting PCM via HDMI (or TOS or Spdif) you have a potential jitter issue, and indeed the cable can make a difference. Not saying its audible or not, just explaining the distinction. The fact that all the bits get to the other end isn't the issue, it's the exact timing of the transfer.

^ This issue is highly overblown, as any decent HDMI sound processor will buffer and re-clock the HDMI input -- which automatically re-syncs with every video frame. (HDMI audio is not a separate signal. HDMI audio is always embedded in the "blanking intervals" of an HDMI video signal -- even if that is just a black screen.) Thus there is no time for clock drift to happen and the timing of the bits going from the input buffer to the DACs in the sound processor is controlled by the internal circuit design of the sound processor. If you have an AVR that offers adjustable audio delay, for example, it must be buffering and re-clocking the input.

AVRs certainly have OTHER ways to screw up their handling of HDMI audio input, and some may do a worse job with LPCM input than with Bitstream. But those are just bugs in the specific AVRs.
--Bob

Jitter is audible, if bad enough, although it takes surprisingly large amounts of jitter to show up in an A/B test. (Looking for that article, but haven't found it. Read it just a couple of weeks ago, and should have bookmarked it.)

To be clear, if a player sent a signal over HDMI or any other connection, and the receiving end depended on the player for its clock, then yes, they receiver would be at the mercy of the mastering studio, player, and cable.

But that's not what receivers do, because it would be stupid.

The only thing the D/A converter needs to know is the value of the original samples. It knows that the samples were supposed to be taken by an infinitely accurate clock at the recording rate (for example, 44.1 kHz). So it takes those samples and converts them to analog at the recording rate.

It doesn't matter if the samples arrive from the player separated by wildly varying time gaps. Those gaps are ignored. Only the values of the samples matter. That, and the recording rate.

If you think your way through this, you'll eventually conclude, correctly, that the only possible source of timing errors is when the recording is made, and when the samples are converted in the D/A converter at the end of the playback chain.

If a playback chain is susceptible to any other timing errors, its designer is an idiot.

Shannon's sampling theory was written 60+ years ago. Age doesn't negate validity.

Said from a man with a $,$$$ preamp
I have a 2K Preamp that buffers but HDMI LPCM is not is strong point.

I have an Marantz AV8801 on order which aspires to the performance described above.

- Rich