5.1/7.1 PCM, HDMI, and DSP - An Explaination of the Future-Proof receiver - Page 28 - AVS Forum
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post #811 of 3041 Old 12-14-2006, 04:31 PM
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Does the Onkyo sr604 apply the 10 DB boost to HDMI PCM for LFE or not?
I have this receiver and a HDMI HD DVD player, but I don't how to test the damn thing.. Its eating away at me that I may be listening to the bass all wrong !!
Help!
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post #812 of 3041 Old 12-14-2006, 04:34 PM - Thread Starter
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Sorry for updating this so late (call me lazy ), but I've now added a notation under Words of Caution describing the LFE issue, including a link to KMO's stickied thread, great comprehensive coverage, by the way. Hopefully, this will make the thread one-stop shopping.
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post #813 of 3041 Old 12-14-2006, 04:42 PM
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Quote:
Originally Posted by arib0nd View Post

Does the Onkyo sr604 apply the 10 DB boost to HDMI PCM for LFE or not?
I have this receiver and a HDMI HD DVD player, but I don't how to test the damn thing.. Its eating away at me that I may be listening to the bass all wrong !!
Help!

It should be very easy for you to tell. It's obvious when LFE is lacking by so much. To test this, just play an SD DVD disc on your HD DVD player. Have the player send a bitstream for one scene with lots of LFE, and then have the player decode the track into PCM for that same scene. If overall sub output seems to be significantly lacking with the PCM option, then the LFE problem likely exists. If it is a problem, then play around in the receiver's menu and look for an LFE boost option.
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post #814 of 3041 Old 12-14-2006, 04:44 PM
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Quote:
Originally Posted by Lindahl View Post

Sorry for updating this so late (call me lazy ), but I've now added a notation under Words of Caution describing the LFE issue, including a link to KMO's stickied thread, great comprehensive coverage, by the way. Hopefully, this will make the thread one-stop shopping.

Understood
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post #815 of 3041 Old 12-14-2006, 07:38 PM
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Well, audiophile recordings could take advantage of 192 kHz sampling. Plus the PS3 may convert DSD to LPCM at 192 kHz in the future.

Many times when a surround processor adds any kind of DSP manipulation to the LPCM signal, it only works at a certain sampling rate. Some processors can't even accept 96 kHz and have to downsample it to 48 kHz. If they can't do that, then no DSP functions will be allowed.

Having DSP engines and surround processors that can actually manipulate 8 channel, 24 bit/192 kHz signals without loss or downconversion will then be in compliance with Blu-Ray specs.

Dan

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post #816 of 3041 Old 12-14-2006, 07:44 PM
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Quote:
Originally Posted by J y E 4Ever View Post

Crap, as soon as I begin to think i'm getting a basic grasp of this stuff out comes this fella with the following quote, "It's possible that true 192 kHz processing via HDMI may be added to the new models"

Sir, are you saying that my new Onkyo 674 can't handle "true" 192 kHz processing? BaHumBug I tell ya....

Sheesh, first my $575.00 receiver can't correct the LFE -10dB thingy and now this.

Oh well, at least the receiver looks nice.

By the way, what soundtrack would ever possess something as high quality as a 192 kHz sampling rate?

Can you even hear the difference from 48 to 96 all the way up to 192?

Funny thing is, the PS3 tells me that it supports 192 but only with 2 channels, the highest the 5.1 gets is 96.

In all seriousness, what do you mean by "true" 192 processing? Is what my Onkyo performs "fake" processing, like its immitating a 192 sampling rate.

Could you boys slow down for me, i'm trying to catch up.

You haven't been reading before you post... The answers are in the thread.
However, this might clear up the processing rate issue.

44KHz or 48KHz is the sample rate at which measurements of the (original) analogue sound vibration of air (or any molecules & not counting synthesized sounds), having been converted by a microphone into an electrICAL signal and then converted into a digital electrONIC signal that closely approximates the analogue. The human ear hears up to roughly 20KHz. A sample rate of more than roughly twice in the shortest duration cycle (highest frequency) does not yield enhanced results, while less than 1/2 cycle measurements lose accuracy (for any frequency range). 176KHz and 192KHz are 2X twice of 44KHz and 48KHz respectively. It is in this domain that the fancy spacial sound field effects, EQ, et al, take place before going either directly to a Digital to Analogue Converter (DAC) or divided in four and then to a 4xKHz DAC to create an imperfect impression on an electrICAL signal for amplification sufficient for a loudspeaker to move air that carries sound.

The sample rate of an Analog to Digital Converter (ADC) can be likened to sand paper. The bigger the grains of sand, the coarser the feel, the lower the number. 60 grit is way rough, 800 grit feels almost perfectly smooth. The lower the number of samples/cycle the bigger the square steps that contour the analoge wave form, the less closely the infinite variations of the wave are followed. Sandpaper, like music does not feel smoother if you use 1000 grit. Smooth is smooth.

As for true processing of a 192KHz signal that would refer to processing an incoming 192KHz signal as recorded on the disc and output by the player, as opposed to multiplying a 4xKHz from the player internally. Either way, the original sampling of analogue sound (not synths or Sound Effects (SFX)) was still at 44KHz or 48KHz (for 'best' quality). Most likely the speakers can't handle anything with higher resolution anyway, as the cones can only respond so accurately.

Boys (and gals), will that suffice? Bob? KMO? Anyone?

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post #817 of 3041 Old 12-14-2006, 07:52 PM
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Quote:
Originally Posted by Dan Hitchman View Post

Well, audiophile recordings could take advantage of 192 kHz sampling. Plus the PS3 may convert DSD to LPCM at 192 kHz in the future.

Many times when a surround processor adds any kind of DSP manipulation to the LPCM signal, it only works at a certain sampling rate. Some processors can't even accept 96 kHz and have to downsample it to 48 kHz. If they can't do that, then no DSP functions will be allowed.

Having DSP engines and surround processors that can actually manipulate 8 channel, 24 bit/192 kHz signals without loss or downconversion will then be in compliance with Blu-Ray specs.

Dan

Using the PS3 on my Pioneer Elite receiver, I hear no difference between 88.2 kHz and 176.4 kHz conversion of my stereo SACD tracks, and the receiver does full room correction at both sampling rates.

In my opinion, there's absolutely no need to get worried over 192 kHz handling. Very few sources will be, or are recorded at this rate, and I'm sure the difference is negligible at best when compared to 96 kHz.
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post #818 of 3041 Old 12-14-2006, 07:56 PM
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Quote:
Originally Posted by sdurani View Post

Here's why:

The .1/LFE channel was intended to overcome a limitation of commercial movie theatres: they don't use bass management. However, their speakers still rolled off in the low frequencies. So it was up to a separate Low Frequency Effects channel to feed the subwoofers that filled in the bass not being reproduced from the main channels.

But that meant that the .1/LFE channel needed to have as much bass energy as the front three channels combined, which would require the LFE channel to be 10dB louder than the other channels. If the other channels in the soundtrack are recorded at the highest level available (in order to have the audio signal as far above the noise floor as possible), then recording the LFE channel 10dB higher will send it into clipping/distortion.

So the LFE channel is brought down by 10dB during encoding at the studio, with the understanding that it will automatically be boosted by 10dB when decoded at home. This way the LFE content is played back at the originally intended level. Now I know what you're thinking: isn't that volume boost also going to raise the noise in the LFE channel by 10dB? Yeah OK it is. But, since our human hearing isn't all that sensitive in the low frequencies, you'll never notice it.

Sanjay


This should be a sticky post
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post #819 of 3041 Old 12-14-2006, 07:58 PM
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original recordings aren't made at 44Khz are they?

also, i'm trying to reconcile these two seemingly contradictory statements

The bigger the grains of sand, the coarser the feel, the lower the number. 60 grit is way rough, 800 almost feels smooth.
and
Sandpaper, like music does not feel smoother if you use 1000 grit. Smooth is smooth.

Boo!
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post #820 of 3041 Old 12-14-2006, 08:02 PM
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Quote:
Originally Posted by krabapple View Post

This should be a sticky post

Sanjay's .1 LFE post is so gooey it is sticky enough to be recognized as such.

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post #821 of 3041 Old 12-14-2006, 08:09 PM
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Quote:
Originally Posted by SiriuslyCold View Post

original recordings aren't made at 44Khz are they?

also, i'm trying to reconcile these two seemingly contradictory statements

The bigger the grains of sand, the coarser the feel, the lower the number. 60 grit is way rough, 800 almost feels smooth.
and
Sandpaper, like music does not feel smoother if you use 1000 grit. Smooth is smooth.

If I am correct the grit fineness number is the quantity of grains that fit in a 1/4" square, and like wire gauge numbering is inversely proportional to size. The higher the frequency the smaller (shorter, actually) the wavelength.

The statement would probably read easier... [with sandpaper] the lower the grit number, the bigger the grains of sand, the coarser it feels. 60 grit is way rough, 800 almost feels smooth.

A fool and his money soon has a pile of last year's state-of-the-art equipment
Me, I wait to jump when I see previous-state-of-the-art pricing
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post #822 of 3041 Old 12-14-2006, 09:47 PM
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Quote:
Originally Posted by VeniVideoVici View Post


The sample rate of an Analog to Digital Converter (ADC) can be likened to sand paper. The bigger the grains of sand, the coarser the feel, the lower the number. 60 grit is way rough, 800 grit feels almost perfectly smooth. The lower the number of samples/cycle the bigger the square steps that contour the analoge wave form, the less closely the infinite variations of the wave are followed. Sandpaper, like music does not feel smoother if you use 1000 grit. Smooth is smooth.

JeffLL

I think you're on the right track, but the analogy is not quite right. It's the bit depth (usually 16 bit, 20 bit, or 24 bit) of the ADC that determines the "coarseness" of the digital representation of the input analog waveform. At 16 bits, the amplitude of the waveform at each sampling point is represented by a whole number between +32767 and -32767 (and that fine a division already looks like "really fine sandpaper" to me ). At 20 bits, the amplitude at each sampling point is represented by a number between +524,287 and -524,287. At 24 bits, the amplitude at each sampling point is represented by a number between +8,388,607 and -8,388,607.

The sampling rate (the rates commonly used for digital audio are 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, or 192 Hz) determines what frequencies are captured by the digitization, and can be accurately represented in the "reconstructed" analog waveform. According to Nyquist's theorem, 44.1 kHz is more than enough to reconstruct a waveform with frequency components up to 20 kHz. 192 kHz would allow us to reconstruct a waveform with frequencies up to 90 kHz (maybe needed to reproduce musical compositions of some non-human sentient species, that I often suspect are lurking on these boards ).

To illustrate, consider what would happen if we "downsized" either the bit depth or sampling rate to a much smaller (worse) value than is actually used in digital audio.

Imagine digitizing an audio waveform with 24 bit depth, and sampling rate of 11 kHz. Then we could only reconstruct frequency components up to about 5 kHz, so we would lose the top two octaves of human hearing - "no highs". But the audio information at frequencies below 5 kHz would be undistorted - so the result would be similar to using a high quality digital recording and playback scheme, and then tacking on a 5 kHz lowpass filter at the output.

Now imagine digitizing the waveform with 8 bit depth (levels between +127 and -127) and sampling rate of 192 kHz. We would have information about frequency components extending well above the audible range - but because of the "choppy" digitization caused by the insufficient bit depth, we would get weird sounding distortion and artifacts in the output waveform, and the problems would be present at all audible (and higher) frequencies.
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post #823 of 3041 Old 12-14-2006, 10:12 PM
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OK...................
I read through this threa and if someone can answer a question, please.

I am looking for a reciever with preouts for an amp to drive Magnepans (so Onkyo is out, right since I saw they don't have preouts) that will tide me over for at least two years with my PS3 for audio through HDMI.

Lastly, I am cheap so $250-$500 would be great!!!
IS THERE ANYTHING OUT THERE LIKE THIS (NO LFE problems in the receiver and I am only going 5.1)


HEEEEEELP!!!!!!!

_________________________________________________
My Gear:
JVC RS4810, Prismasonic HD6000M anamorphic lens, Screen Excellence 4K 115 inch wide
Lumagen Radiance 2021
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B&W CT700 Series Speakers, (2) PowerSound Audio XS30s
Custom HTPC running Mediabrowser, 12TB Server for storage
...
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post #824 of 3041 Old 12-14-2006, 11:37 PM
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A quick note about Denon and -10DB LFE issues.

Seems that Denon is aware of this "trouble" - there's a setting in the menu to quickly change LFE from 0db to -10db if needed.

The official line in the Denon 4306 manual states this is for DTS. Apparently, DTS music and DTS movies use different levels of LFE!

Yves
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post #825 of 3041 Old 12-15-2006, 01:27 AM
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The manual's not accurate there. If everything's working correctly, you normally never need to adjust the LFE setting. Manual intervention should not be required. 0dB is correct in almost all instances. The problem is that some receivers only offer -10dB on the PCM input, and you can't get to the 0dB level.

Early DTS music releases on CD need -10dB LFE. That was effectively an error on their part, and they went to normal LFE level later. That's what the option is for - wrongly mixed discs, or to compensate for wrongly-behaving attached devices. Don't think you need to fiddle with it routinely.
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post #826 of 3041 Old 12-15-2006, 05:07 AM
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Quote:
Originally Posted by KMO View Post

The manual's not accurate there. If everything's working correctly, you normally never need to adjust the LFE setting. Manual intervention should not be required. 0dB is correct in almost all instances. The problem is that some receivers only offer -10dB on the PCM input, and you can't get to the 0dB level.

Early DTS music releases on CD need -10dB LFE. That was effectively an error on their part, and they went to normal LFE level later. That's what the option is for - wrongly mixed discs, or to compensate for wrongly-behaving attached devices. Don't think you need to fiddle with it routinely.

Exactly.
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post #827 of 3041 Old 12-15-2006, 05:12 AM
 
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Quote:
Originally Posted by sdurani View Post

If the other channels in the soundtrack are recorded at the highest level available (in order to have the audio signal as far above the noise floor as possible), then recording the LFE channel 10dB higher will send it into clipping/distortion.

With all due respect, this logic is wrong!!

If a mixing engineer is raising the relative amplitude of the program level to push the "noise floor down" they are incompetent. The noise floor is already -90db down, or greater with digital signals, so if you are using PROGRAM VOLUME adjustments for better quality...STOP

All you are doing is raising the program volume level and screwing up the experience for anyone who has a calibrated system! SMPTE RP155 (-20dBFS = 0VU = +4dBu) or for non-professional levels insert correct dBu or dBV.
Quote:


So the LFE channel is brought down by 10dB during encoding at the studio, with the understanding that it will automatically be boosted by 10dB when decoded at home.

FYI the mixing (.1) LFE monitor is calibrated for +10dB or (95 dBSPL) at -20dBFS, this combined with the +10dB emphasis applied by the DD decoder creates a composite 115 dBSPL peak potential at 0 dBFS. All this is to accommodate the potential acoustical sum of the surrounds.
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post #828 of 3041 Old 12-15-2006, 05:52 AM
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Quote:
Originally Posted by tbrunet View Post

With all due respect, this logic is wrong!!

If a mixing engineer is raising the relative amplitude of the program level to push the "noise floor down" they are incompetent. The noise floor is already -90db down, or greater with digital signals, so if you are using PROGRAM VOLUME adjustments for better quality...STOP

All you are doing is raising the program volume level and screwing up the experience for anyone who has a calibrated system! SMPTE RP155 (-20dBFS = 0VU = +4dBu) or for non-professional levels insert correct dBu or dBV.

Calm down. Not quite sure what you're saying here - I think you've misunderstood what he's saying. The basic point is that the main channels are calibrated for peaks of 105dB SPL. If they were calibrated for max 115dB SPL the same as the LFE (so -30dBFS=0VU), then either (a) you'd never use the full range of the channel, meaning a lower effective signal-to-noise ratio, or (b) you'd be in danger of blowing up listeners' tweeters.
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post #829 of 3041 Old 12-15-2006, 06:30 AM
 
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Quote:
Originally Posted by sdurani View Post

If the other channels in the soundtrack are recorded at the highest level available (in order to have the audio signal as far above the noise floor as possible),

I'll let Sanjay respond to this

There are many incompetent mixing engineers that believe a signal is valid and non-distorsted if it does not exceed (0 dBFS). Thus SQ is improved when the headroom is used e.g.

"in order to have the audio signal as far above the noise floor as possible"

When in reality this is DR compression aka 'distortion', and it makes calibration pointless!

Have a nice day
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post #830 of 3041 Old 12-15-2006, 07:15 AM
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I really think you're talking at cross-purposes here, tbrunet. Nobody is talking about compression, or attempting to artificially boost loudness.

We're just talking about what the 0dBFS level is. At the moment, it's 105dB SPL for normal channels and 115dB SPL for LFE.

If you wanted the LFE to be equal level with the mains, that would mean raising the 0dBFS level of the mains to 115dB SPL. If you did that, you'd use even less of the available range, thus increasing the effective noise by 10dB. (Unless of course you entered a new loudness war to compress the music up into the 105dB-115dB range... )

Understand?
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post #831 of 3041 Old 12-15-2006, 07:37 AM
 
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Quote:
Originally Posted by KMO View Post

Nobody is talking about compression, or attempting to artificially boost loudness.

Originally Posted by sdurani
Quote:


If the other channels in the soundtrack are recorded at the highest level available (in order to have the audio signal as far above the noise floor as possible),

Understand?
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post #832 of 3041 Old 12-15-2006, 07:59 AM
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Quote:
Originally Posted by tbrunet View Post

When in reality this is DR compression aka 'distortion', and it makes calibration pointless!

Not true at all. The dynamic range is not being squeezed (difference between loudest and softest sounds is not being changed).

The main channels are recorded so that peaks are right below clipping. At 10dB higher, the LFE channel would clip. So during encode the LFE is dropped 10dB and during decoding it is raised 10dB to restore its volume relative to the main channels.

This doesn't change the dynamic range because the entire contents of the LFE channel (including the softest sounds) are subject to the same drop/boost.

Sanjay

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post #833 of 3041 Old 12-15-2006, 08:20 AM
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Okay, tbrunet, I can see where the misunderstanding's come from.

No-one meant to suggest compression during a track. But sdurani is suggesting deliberately tailoring the volume of a track and/or album so its peak is at 0dBFS. No compression involved, but certainly losing any calibration of a natural dialogue level.

Sanjay, you wouldn't do that, if you're interested in reference calibration - you'd mix everything with the levels set so that 0dBFS = 105dB SPL, and not perform any further adjustment to to "fill" the space. So on playback, with equipment set to "0dB" master volume, you'd get the original intended level. This is quite important on film/TV mixing - you need dialogue level to be constant. If you did want to fill the space, you can do that with Dolby Digital, as long as you put in a "dialnorm" parameter so the receiver knows to automatically nudge the master volume down appropriately to put the dialogue back at normal level.

But none of this is anything to do with the original point Sanjay was trying to make. Which is that the 0dBFS = 105dB SPL calibration for mains is appropriate to make best use of the available space to minimise noise. If you calibrated it so 0dBFS = 115dB SPL, you'd have lots of headroom you'd never use, and 10dB of extra noise relative to dialogue levels.
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post #834 of 3041 Old 12-15-2006, 08:39 AM
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Quote:
Originally Posted by KMO View Post

But sdurani is suggesting deliberately tailoring the volume of a track and/or album so its peak is at 0dBFS.

That's not what I meant. In my attempt to keep the description as non-technical as possible, I'm apparently not communicating it clearly. The point is to not drop the main channels by 10dB during encode just to accomodate the LFE channel's additional headroom. So the main channels are encoded as high as possible (calibration level, not calibration minus 10db).

Sanjay

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post #835 of 3041 Old 12-15-2006, 08:42 AM
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Ah, okay. That's the way I originally understood you, but your last post seemed to suggest you were actually thinking in terms of always using all available space.

tbrunet's attempting to confuse everyone by getting far more technical than either of us originally intended...

I suggest we drop it now, as we've taken up almost a page on this off-topic nit-pick.
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post #836 of 3041 Old 12-15-2006, 09:11 AM
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LFE reference is 115 dBSPL
Mains Reference is 105 dBSPL

The 10 dB bump is applied in the LFE amp feed at the mix studio.
The 10 dB bump is applied in the processor at the consumer level.

Done all the time, no problems with calibration, just two different 0 dBfs references. End of story.

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post #837 of 3041 Old 12-15-2006, 09:33 AM
 
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There nothing I said thats confusing, and I'm just correcting the miss understanding of the 'technical' correlation of said 'noise floor' and the coding of relative volume

Originally Posted by sdurani
Quote:


If the other channels in the soundtrack are recorded at the highest level available (in order to have the audio signal as far above the noise floor as possible)

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post #838 of 3041 Old 12-15-2006, 09:34 AM
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Quote:
Originally Posted by KMO View Post

tbrunet's attempting to confuse everyone by getting far more technical than either of us originally intended...

Correct. If you look at his posting history, you'll see he has a special place in his heart for me.

My original post was a very deliberate attempt to keep the explanation non-technical (which I try to do as much as possible). Folks will naturally ask why not simply encode the LFE channel 10dB higher than the main channels. Without mentioning the noise floor, how does one explain why you can't lower the other channels by 10dB during encoding?

Sanjay

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post #839 of 3041 Old 12-15-2006, 09:40 AM
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Quite. As it is, dialogue is 65dB above the noise floor. If the mains were calibrated the same way as the LFE channel, dialogue would be 55dB above the noise floor.
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post #840 of 3041 Old 12-15-2006, 09:47 AM
 
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Quote:
Originally Posted by sdurani View Post

The main channels are recorded so that peaks are right below clipping.

FWIW, anyone who mixes content with this simple rule of thumb, is professionally lacking.. Sorry but the facts are what they are Sanjay!
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