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I thought the spec lab graphs showed an almost identical representation of the source? Where would the clipping show?

 

You can see it on SpecLab, but you need to be able to perfectly overlay the 2 versions (in this case, the Panny cheapo discontinued player vs the Oppo flagship), zoom in to the area of interest and fade them back and forth.

 

I have already done this with HTTYD Red Death crash scene, but I'm not going to post it here mainly because it's the wrong tool for the job and I want this thread to stay focused on the test results from Shred's investigation.

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So check THIS out... me and Bosso were pretty surprised at how the waveform in the end scene in ETE was limited like crazy:

cada14bed1f8b737e586120f4eee1428.png

 

Well that isn't the production team putting aggressive limiters on the mix... it's the OPPO clipping the nuts out of the re-directed bass!!!

88a66a6d61da7f89153df1557f223cfb.gif

Red is el cheapo Panasonic, green is the flagship from OPPO MVL set to 60 out of 100.

 

Look at it zoomed into the transient in the end!

f1effc7d0f081eac40a93083ebb22107.gif

Red: Panny.  Green: OPPO

 

Horrific!

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Wow.

 

As I first reported in Nov 2014, the scene is encoded hot, but I was convinced that the production included a brick wall limiter to aid in making it such a hot scene.

 

To see the Panny reproduce it with all it's dynamics blew my mind. No brick wall limiter. Instead, brick wall Oppo.

 

Imagine what it does to your listening pleasure of the final transient of the ship taking off!!!  <<<SQUASH!!!>>>

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That is sad the $1300 "high end" Oppo does that and a $200 Panny doesn't. I do wonder how many AVR's and preamps are doing the same thing though? Good stuff fellas. That is an eye opener to say the least.

 

It's like Abraham said most of us were more than likely sending clipped signals. Plus it doesn't even have to be at 0dbMV if the trims are set too high. I know I was with my pos Sherbourn. I have it at the lowest setting now, which is -10, to get it closer to running flat before I even knew about this. To think I had EtE running 10db hot when that scene hit. :o  :wacko:

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This is the whole scene and the zoomed in bit is highlighted in yellow.  I don't know the exact time of the zoomed in part but it's somewhere where the dragon hits the ground or the explosion after. 

 

I tried this on jriver to see what was going on in my setup & to see what the effect of jriver's clip protection is. I used the timings from http://data-bass.ipbhost.com/index.php?/topic/12-the-low-frequency-content-thread-films-games-music-etc/page-13#entry388which says 1:22:18 - 1:23:08

 

My convolution filter has to be in place because that's what does the mixing to the SW output

 

top row: jriver with flat line overflow, manual bass mgmt (i.e. reduce mains by 15 & SW by 5 then mix then add 15dB back in)

middle row: same as top row but with clip protection on

bottom row: same as middle row but without the "add 15dB back in" (i.e. my usual setup as I add 15dB in the SW channel using a setting in my prepro) 

 

post-1440-0-99494600-1425633963_thumb.png

 

This looks to me like jriver clip protection does a pretty good job of preserving the shape of the waveform at the cost of reduced dynamic range. Adding gain back in a subsequent analogue stage seems like the easier/better choice to me though.

 

The particular filter I have in place peaks at 5Hz and trims 1dB at 10Hz so seems like it will have minimal effect on that <10Hz peak. I guess I could create another filter that is just a pass through but seems like a reasonable comparison anyway. For reference it looks like this;

 

post-1440-0-22065400-1425633971_thumb.png

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It may be worthwhile to see if reviewers who still actually measure AVR or Pre-pro & disc player performance can start doing a WCS test on the SW out. Might be time for spreading the word around about this. I suspect nearly all of us have been experiencing clipping in the SW output on some tracks either analog or digital domain.

 

 

 

 

"This exercise has been an eye opener for me. I can only now imagine what sorts of signals are being sent to subwoofers when owners routinely run +6dB, +10dB, +15dB hot by adjusting the AVR (or pre/pro or player as the case may be) trim and/or MVL."

 

No doubt. Imagine the guys using commercial subs who don't have enough firepower on hand to reproduce the content at REF without severe limiting to begin with and then schlep the waveform nastiness on top of it and add another 10dB in overall level requirement. Bleck... :o

 

 

3II3d00d,

If you don't mind my asking. What is your signal chain for your system? Obviously JRiver in the PC. What soundcard sends the signal from the PC and in what form to what equipment after that? Basically what all equipment do you have in line before the signal hits the amps?

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I tried this on jriver to see what was going on in my setup & to see what the effect of jriver's clip protection is. I used the timings from http://data-bass.ipbhost.com/index.php?/topic/12-the-low-frequency-content-thread-films-games-music-etc/page-13#entry388which says 1:22:18 - 1:23:08

 

I ran the scene in this time frame and I don't understand what you're showing here.  I am semi-familiar with jriver but I don't know if you've zoomed in on a section or showing something different from the waveform.  Is this from an analog out or internal digital loop or something like that? 

post-1247-0-27667300-1425669450_thumb.jpg

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I ran the scene in this time frame and I don't understand what you're showing here.  I am semi-familiar with jriver but I don't know if you've zoomed in on a section or showing something different from the waveform.  Is this from an analog out or internal digital loop or something like that? 

it's a digital loopback using the RME mixer

 

I found there were 2 sections that clipped but the section I'm showing is the one where it really clipped hard, it's right towards the end of that clip. I then zoomed right in to be able to see what was actually happening.

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If you don't mind my asking. What is your signal chain for your system? Obviously JRiver in the PC. What soundcard sends the signal from the PC and in what form to what equipment after that? Basically what all equipment do you have in line before the signal hits the amps?

 

 

I have a PC running Jriver & use acourate to generate filters that deal with speaker correction, room correction, delays and bass management. This goes out via firewire to an RME FireFace 800 then from there it goes via analogue to the 7.1 analogue inputs on a Marantz AV7005. This runs in pure direct and trims are set so that the amps are driven to input sensitivity at MV = 0 (though my setup means MV = -5 is as far as I can really go though I don't listen that loud anyway). The mains are powered by a cinepro 2k6 mk3 and the sub by a SpeakerPower SP1-6000-RACK. The sub itself is a pair of UXL-18s in 190L.

 

The AV7005 is in the chain to provide the dual HDMI outputs (PJ and TV) & to handle other sources. 

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I ran the scene in this time frame and I don't understand what you're showing here.  I am semi-familiar with jriver but I don't know if you've zoomed in on a section or showing something different from the waveform.  Is this from an analog out or internal digital loop or something like that? 

here's what it looks like zoomed out to the whole track which I guess is what you're showing?

 

post-1440-0-48797700-1425673170_thumb.png

 

that shows the impact on dynamic range much more clearly but zooming in (and with "show clipping" on)

 

post-1440-0-80185900-1425673613_thumb.png

 

most of that scene is -30 or so when there is no clipping, the earlier clip is maybe -10 to -15, the big peak only just gets near the top

 

in comparison the top row, which is, I think, in the ballpark of the oppo, has lots of peaks in the -20 to -10 range and then sustained spells of clipping

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I have a PC running Jriver & use acourate to generate filters that deal with speaker correction, room correction, delays and bass management. This goes out via firewire to an RME FireFace 800 then from there it goes via analogue to the 7.1 analogue inputs on a Marantz AV7005. This runs in pure direct and trims are set so that the amps are driven to input sensitivity at MV = 0 (though my setup means MV = -5 is as far as I can really go though I don't listen that loud anyway). The mains are powered by a cinepro 2k6 mk3 and the sub by a SpeakerPower SP1-6000-RACK. The sub itself is a pair of UXL-18s in 190L.

 

The AV7005 is in the chain to provide the dual HDMI outputs (PJ and TV) & to handle other sources. 

 

Thanks. That's about what I expected. I already have multiple Presonus Firestudio's I could use but you have 3 units involved counting the PC. I'm trying to go the other way and keep it at 2 before the speaker. Player or source/amp/speaker

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I tried to match your zoomed waveform pic with mine by looking at the parts you have in red on mine but I couldn't tell where it was supposed to go.  I just zoomed mine in where the highest peaks were for the sub out of the Panny player to give you an idea of the most demanding part.

 

post-1247-0-60631900-1425679725_thumb.jpg

 

This is why I think that uncompressed dynamic source is less demanding/dangerous to amplifiers and drivers:  Imagine the center line is zero position on the driver and above it is positive throw and below it is negative throw.  Now play the scene above in your head and match driver movement to the waveforms.  A dynamic source will move out and in at a little slower of a rate of speed with only a handfull of parts where xmax is reached.  A compressed/clipped waveform tries to throw the driver faster and more violently (like a square wave) and it holds near xmax before slamming itself back down to the maximum negative throw position as fast as mechanically possible.  This is going to put more physical stress on the driver.

 

In some of these peaks there that get held, the driver wouldn't even be moving much so you wouldn't even be getting much sound out of it but it would require much more power from the amplifier heating the coil much faster.  Now imagine this is happening in the end scene of Earth to Echo that lasts over a minute... :unsure:

 

I don't know what to think about your waveform here dood.  It looks like a limiter is dialed up to a low threshold level to me.  Might just be how the scale in the software is, I don't know.  Can you take your analog sub out and put it into speclab to see if the waveform looks any closer to mine for the scene?  I usually capture 1:22:58-1:23:09. 

post-1247-0-92618400-1425680533_thumb.jpg

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Thanks. That's about what I expected. I already have multiple Presonus Firestudio's I could use but you have 3 units involved counting the PC. I'm trying to go the other way and keep it at 2 before the speaker. Player or source/amp/speaker

Connecting pc straight to amp can be a bit iffy,  depending on what you do with it... 

Some applications handles volume worse than other,  especially if Windows own volume mixer is involved. 

Depending on the gainstructure of your setup,  unexpected full volume blasts for whatever reason can do damage,  to system or your hearing... 

 

I'd feel better having an external volume control, between pc and amp. 

A volume control you trust like an avr or something,  that way you can disable volume control on pc and like anything else control volume in avr instead. 

 

Something like the oppo as source/preamp is probably a lot more stable on the volume control than a pc would be,  though it would probably work,  if you pay extra attention to which volume controls are active within the pc and are sure nothing can upset that control. 

 

I'm also using jriver but only as processing,  input via wasapi loopback,  outputs to a Asus stx II 7.1 soundcard Btw. 

I play games and other things apart from movies and music,  that is a lot of different applications that each can have there own volume controls,  I feel safer having a master volume control outside pc ( analogue multichannel avr)...

Though it is tempting slimming down the signal chain...  

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I don't know what to think about your waveform here dood.  It looks like a limiter is dialed up to a low threshold level to me.  Might just be how the scale in the software is, I don't know.  Can you take your analog sub out and put it into speclab to see if the waveform looks any closer to mine for the scene?  I usually capture 1:22:58-1:23:09. 

 

I think it is at least partly the vertical range, the default seems to be 60dB whereas switching to 36dB gives this. I think the strong peak is at 1:23:04

 

 post-1440-0-47517600-1425681684_thumb.png

 

This is audacity btw. I'll try the SW out over the weekend.

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No doesn't look like it, the options are 36,48,60 (and some much wider ranges). One last comparison, I dumped the hard clipped one out to speclab along with the unclipped one

 

post-1440-0-64513800-1425684396_thumb.png

 

edit: just realised there are separate "waveform" and "waveform (dB)" views, it looks like a linear scale is what you've been showing? the dB one is more appropriate for representing how it might be heard. The below is the same track but one linear and the other logarithmic.

 

post-1440-0-24524100-1425684882_thumb.png

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Yeah, the one on top looks way better.  I think the waveform should relate to voltage if I'm not mistaken... like an O-scope.  At least that's what I'm used to.  I've used Ableton, Cubase, and Reaper and they all seem to work out the same.  I'm afraid I have no experience in Audacity. 

 

The speclab waveform looks wacky though, like there is a low shelf jacking up the low, low end or something like that.

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Connecting pc straight to amp can be a bit iffy,  depending on what you do with it... 

 

Agreed completely. It doesn't give me the warm and fuzzy.

The way I'm looking at it though if going into an AVR anyway I have no need for the pc processing really. I don't know it would be nice to have the pc connected for some things but for watching movies and TV it really wouldn't make a difference in my setup.

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We hooked up the old player that the OPPO replaced to compare waveforms of the HTTYD crash scene.  It is a Panasonic DMP-BD85 with 7.1 analog outputs. 

12751-dmpbd85img1big.jpg

 

This Panasonic has no master volume so we adjusted the fixed sub output to -12dB to avoid analog clipping.  Here is a comparison of the clipped OPPO's out and the non-clipped Panasonic's out for the HTTYD Dragon Crash scene:

605ee5b5327637d7bb60d81f43491284.gif

It is obvious that the OPPO is reaching a headroom limit and clipping the waveform's peaks in real world content.  Once again, the OPPO clips the real world content regardless of master volume or trim volume.  This clipping has nothing to do with the voltage out of the subwoofer jack. 

 

Oh and by the way, with the trim set to -12dB, the Panny also passed the worst case 7 channel re-directed bass test:

bd89a678efb83e47ddba02a93c9c8c10.gif

This is the OPPO vs Panny with the Panasonic preserving the peaks of the waveform.  This was a player that was priced under 200 dollars when it was new and it is outperforming the OPPO with the WCS and real world source material.  We ran through 9, WOTW, and OZ and for all of them there are similar parts showing clipped waveform on the OPPO for the content that is heavy on the re-directed bass. 

 

 

BUMPING THIS ^^^

 

Because the more I look at these results, the more it burns my ass.

 

Turns out that Audioholics discovered the same problem in a review of the Oppo BDP-83SE from 5 years ago. Oppo apparently coaxed Audioholics into a test methodology revision that resulted in the Oppo passing.

 

http://www.audioholics.com/audio-technologies/0dbfs-blu-ray

 

Here's an excerpt:

 

"When we initially tested the analog outputs of the Oppo BDP-83SE we concluded that it could not properly handle 0dBFS input signals since it was distorting the summed subwoofer output with all speakers set to "small".

 

"Incidentally we did measure other BD players that could properly pass a 0dBFS signal with all channels set to small..."

 

 

Oppo is blaming the source software. Oppo is claiming a fix would lower audio quality.  :rolleyes: :rolleyes:  :wacko: :wacko:

 

EDIT: BTW, we tested the Panny with Max's WCS disc with the 7 sats channels AND SW trim set to '0' and it passed the 1 Hz, 5 Hz, 10 Hz, 20 Hz and 40 Hz tests perfectly.

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