Old 01-16-2019, 11:29 AM   #1
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Default Lufs limiter ?

Does anyone make a LUFs limiter. Most limiters deal with peak signals but not loudness so if you have an uneven bass guitar for example your limiter might limit the Peaks but still not the loudness. Are there any limiters out there the limit based on loudness or did I just invent something and I need to hurry and patent it?
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Old 01-16-2019, 11:52 AM   #2
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Wouldn't an auto loudness matcher that is fed a signal of constant loudness on the side chain do the trick?

There is the TBProAudio AB Level Matching JSFX, and I made a JSFX of my own (post #6 in this thread https://forum.cockos.com/showthread.php?t=188958). None of those do LUFS though, but I'm working on it... slowly
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Old 01-16-2019, 12:00 PM   #3
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I guess in theory but since loudness is frequency-dependent the constant loudness would be relative. Somehow it would have to take into account The LUFs algorithm
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Old 01-16-2019, 12:01 PM   #4
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Originally Posted by Coachz View Post
Does anyone make a LUFs limiter. Most limiters deal with peak signals but not loudness so if you have an uneven bass guitar for example your limiter might limit the Peaks but still not the loudness. Are there any limiters out there the limit based on loudness or did I just invent something and I need to hurry and patent it?
That would be a compressor triggering how you want with rms detection.
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Old 01-16-2019, 12:03 PM   #5
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Which LUFs? Momentary, short term or integrated? Integrated requires calculating the entire song which is why LUFs valued scripts....often analyze the entire song then make the adjustment to the entire song and, is the one many of us use the most because we want the songs overall loudness. If short term, I'd hazard it's not 'that' much different than RMS depending, and momentary is somewhat a fancy name for peaks.
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Old 01-16-2019, 12:15 PM   #6
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Short term and from the bit I've read it's different than rms because it is very frequency dependent
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Old 01-16-2019, 12:31 PM   #7
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Short term and from the bit I've read it's different than rms because it is very frequency dependent
I get you, I just don't see much practical/meaningful difference and I think these 'small' discrepancies below depend on which plugin is calculating/displaying what, meaning they may not all reflect the exact same window/instant but could be wrong... but still less than 1 dB...

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Old 01-16-2019, 12:38 PM   #8
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Short term and from the bit I've read it's different than rms because it is very frequency dependent
I wouldn't go that far. It's just K-weighted RMS with a 3 second window. Considering the window time is so long, I doubt it would be of much use for evening out an instrument. ReaComp gives you control over RMS size and high pass on the detector. This would be just as useful, if not more so.

K-Weighting:

Last edited by ErBird; 01-16-2019 at 12:43 PM.
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Old 01-16-2019, 12:41 PM   #9
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It can be done, I think, but there are issues to deal with. For instance, short term LUFS is calculated for 3 seconds, which seems a bit longish too me. I would consider monmentary LUFS, which is over 400 ms, better.

There is also the issue with in-between silence. I had to deal with that when coding a script that evens out uneven drum hits. That script is available in post #9 here https://forum.cockos.com/showthread.php?t=215222

But the question got me thinking, so right now I am tweaking my StageGainer JSFX to not take a sidechain but a fixed RMS. If I get this working, I think adjusting it for LUFS is rather straightforward. TBProAudio's EBUR128 JSFX implements a library for LUFS calculations that works fine.
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Old 01-16-2019, 12:59 PM   #10
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I would consider monmentary LUFS, which is over 400 ms, better.
Good to know, I didn't realize momentary was that long.
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Old 01-16-2019, 01:00 PM   #11
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Originally Posted by ErBird View Post
This would be just as useful, if not more so.
Agreed.
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Old 01-16-2019, 01:26 PM   #12
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OK, try this... It is not LUFS, just simple RMS. Using LUFS is left as an exercise for the user

Code:
 desc:MF/StageGainerFixedRMS
 /*******************************************************************
 * Tries to adjust the gain of the main channel (1/2) so that the RMS
 * matches the target RMS
 *
 * M Fabian Jan 2019
 */

slider1:-23<-60,0,1> RMS target (dB)
slider2:400<50,500,10> RMS Window (ms)
slider3:0<0,1,1{Mono,Stereo}> Mode

slider5:0<0,10,0.01> Gain (read only)
slider6:0<0,1,0.01> RMS main (read only)
slider7:0<0,2,1{maxSquare,sumSquareAvg,squareSumAvg}> Stereo RMS

slider9:-128<-128,5,0.1> Avg Gain dB (read only)

@init
// ext_noinit = 1;     // Do not exe this init part on playback start

// Constants for Play State, note that play_state < 0 means error
/* const */ PS_STOPPED = 0;
/* const */ PS_PLAYING = 1;
/* const */ PS_PAUSED = 2;
/* const */ PS_REC = 5;
/* const */ PS_RECPAUSED = 6;

/* const */ MONO_MODE = 0;
/* const */ STEREO_MODE = 1;

/* const */ MAX_SQUARE = 0;
/* const */ SUM_SQUARE_AVG = 1;
/* const */ SQUARE_SUM_AVG = 2;

init_all = -1;     // First time

mainGain = 1;
mainGainSum = 0;
numCalc = 1;
floorGain = 10^(-60 /20);     // RMS below -60 dB is regarded as 0

mainSumSquare = 0;

/*
 * These are three different ways to handle RMS of a stereo signal, x1 being one channel and x2 the other
 ' For mono signals, where x1 == x2, all three give the same result, namely x1*x1.
 */
function maxSquare(x1, x2)
(
	max(x1*x1, x2*x2);
);
function sumSquareAvg(x1, x2)
(
	((x1 * x1) + (x2 * x2)) / 2; 
);
function squareSumAvg(x1, x2)
(
	(x1 + x2) * (x1 + x2) / 4;
);
//ENDBLOCK

@slider
targetRMS = 10^(slider1/20);

tmpX = slider2 * (srate / 1000);
(tmpX != RMS_SAMPLES) ?
(
	/* const */ RMS_SAMPLES = tmpX; // This many samples for the ms window defined by slider2
	samplesCount = RMS_SAMPLES;

	buf1 = 128; // The first buffer starts here
	buf2 = buf1 + 2 * RMS_SAMPLES;
	memset(buf2, 0, 2 * RMS_SAMPLES);       // clear the initial output buffer

	left = 0;
	right = 1;
	curr_in = buf1;
	curr_out = buf2;

	pdc_delay = RMS_SAMPLES;
	pdc_bot_ch = 0; // tells REAPER we delay the first two channels (spl0/spl1).
	pdc_top_ch = 2; 
);

mode = slider3;
stereoRMS = slider7;

// ENDBLOCK

@block
samplesDone = RMS_SAMPLES - samplesCount;
(samplesDone == 0) ? samplesDone = 1;     // This should happen very seldom

slider5 = mainGain;
slider6 = sqrt(mainSumSquare / samplesDone);

// The Avg Gain dB gives the average in dB over all the gain settings
// It is meant to give an indicator as to how to set the output of the last FX so you can set-it-and-forget-it
slider9 = 20 * log10(mainGainSum/numCalc);

//ENDBLOCK

@sample

(play_state == PS_PLAYING || play_state == PS_REC) ?
(
	(mode == MONO_MODE) ?
	(
		// In MONO_MODE, use only the left channel
		mainSumSquare += maxSquare(spl0, spl0);
	):
	(mode == STEREO_MODE) ?
	(
		// In STEREO_MODE, we use both channels, but have three different ways to combine stereo data to single RMS streams
		(stereoRMS == MAX_SQUARE) ?
		(
			mainSumSquare += maxSquare(spl0, spl1); 
		):
		(stereoRMS == SUM_SQUARE_AVG) ?
		(
			mainSumSquare += sumSquareAvg(spl0, spl1); 
			
		):
		(stereoRMS == SQUARE_SUM_AVG) ?
		(
			mainSumSquare += squareSumAvg(spl0, spl1); 
		);
	);
	
	curr_in[left] = spl0;
	curr_in[right] = spl1;
	
	spl0 = mainGain * curr_out[left];
	spl1 = mainGain * curr_out[right];     
	
	left += 2;     // The buffers are interleaved; even indices are left, odd indices right
	right += 2;
	samplesCount -= 1;     // One down...
	
	(samplesCount == 0) ?     // Is it time to update mainGain?
	(
		currentRMS = sqrt(mainSumSquare / RMS_SAMPLES);
		currentRMS <= floorGain ? 
		(
			mainGain = 0;
		):(
			mainGain = targetRMS / currentRMS;
		);

		samplesCount = RMS_SAMPLES;
		mainSumSquare = 0;
		
		// samplesCount == 0, always happens exactly when left == 2 * RMS_SAMPLES. It does, doesn't it...? It should!
		// Note that this relies on that RMS_SAMPLES can never be 0 (compare BUFLEN of PDC_canceller)
		// (left != 2 * RMS_SAMPLES) ?
		
		// swap(curr_in, curr_out);
		temp = curr_in;
		curr_in = curr_out;
		curr_out = temp;

		left = 0;
		right = 1;
		
		// update the calc of main gain avg
		mainGainSum += mainGain;
		numCalc += 1;
	);
	

);
//ENDBLOCK
If it seems to be something like what was requested, I can take some time to fix it to handle LUFS.
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Old 01-16-2019, 02:04 PM   #13
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Interesting

https://www.gearslutz.com/board/show...49&postcount=6
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Old 01-16-2019, 02:07 PM   #14
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Yea, that's why I was trying to be careful when saying it isn't 'that' much different from a practical perspective.
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Old 01-16-2019, 02:14 PM   #15
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I guess they do exist. But I don't know if you can truly limit it to a number like don't go over -14 lufs.

https://www.fabfilter.com/products/p...imiter-plug-in

I could use a very simple streamlined one to put on every channel.
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Old 01-16-2019, 02:16 PM   #16
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A little more detail here https://ask.audio/articles/demystify...etering-levels but saying the same thing, of course. And gating is done by the JSFX code I put together in the post above, only I gate at -60 dB, while LUFS does it at -70 dB.

And yes, you gan "limit" to a specific RMS, teh JSFX above pretty accurately sets the output RMS to whatever the target setting is. If I use the same window as Reaper's master meter does (300 ms by default) the output lands almost exactly at the set value.
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Old 01-16-2019, 02:18 PM   #17
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It's odd that when I Google for Lufs limiter I come up with my own post that I posted 3 hours ago and Google says it was posted 13 hours ago. Weird

https://www.google.com/search?q=lufs...&ie=UTF-8#ip=1
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Old 01-16-2019, 02:18 PM   #18
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It isn't really FYI - LUFS is just a scale, any limiter or compressor can be used to manipulate the end result on that scale.. I have L2 - I started to bring that up earlier but didn't for the same reasons, couple points of interest...

1. It has LUFs metering but that is decoupled from the limiting itself aka it's showing the result of limiting on a LUFs scale (which makes sense).

2. It does have one other LUFs related feature which is true-peak limiting which tries to calculate intersample overs and pull them down.

That said, I love that limiter, due to the fact LUFs metering is built in, along with other features and FabFilter's usual great interfaces and useability. Love it.
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Old 01-16-2019, 02:21 PM   #19
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Quote:
Originally Posted by karbomusic View Post
It isn't really FYI. I have L2 - I started to bring that up earlier but didn't for the same reasons, couple points of interest...

1. It has LUFs metering but that is decoupled from the limiting itself aka it's showing the result of limiting on a LUFs scale (which makes sense).

2. It does have one LUFs related feature which is true-peak limiting which tries to calculate intersample overs and pull them down.

That said, I love that limiter, due to the fact LUFs metering is built in, along with other features and FabFilter's usual great interfaces and useability. Love it.
Yeah it's a nice meter I like the feature of setting the output to 1 to 1 ratio for Unity gain however you have to remember to turn it off when you render. It would be nice if Reaper rendering had a way to hook into that feature to disable it when you do a render.
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Old 01-16-2019, 02:25 PM   #20
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Originally Posted by Coachz View Post
Yeah it's a nice meter I like the feature of setting the output to 1 to 1 ratio for Unity gain however you have to remember to turn it off when you render. It would be nice if Reaper rendering had a way to hook into that feature to disable it when you do a render.
If I'm using L2, it's because I'm using the limiter and I want to see the resulting LUFs at the end of my chain, including any limiting L2 is providing. If I literally just want to see LUFs metering, I use YouLean (real time) or Orban for offline. Orban is really my go to for the final, final because by then it's a rendered master and I just drag/drop it into orban to see what the actual final LUFs of the master is. Since orban has a queue, it's nice to just drop the entire album in there to compare all the tracks in one go.
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Old 01-16-2019, 02:37 PM   #21
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Originally Posted by Coachz View Post
Interesting
https://www.gearslutz.com/board/show...49&postcount=6:

"Another relevant difference comes into play in certain longer term EBU modes (Short Term, Integrated and Loudness Range), they use special mechanisms to ignore program pauses otherwise distorting the RMS value."
From the BS.1771-1 Standard:
"The short-term loudness is generated by using the measurement algorithm described in Recommendation ITU-R BS.1770, but without the use of a gate, and setting the integration period to be 3 seconds."

https://www.gearslutz.com/board/show...08&postcount=9:
(also by FabienTDR)

"Note that the EBU documents don't mention RMS, but they undeniably do it in their algorithm with the square root operation "hidden" as a division of the dB signal (in the dB scale, a division by 2 is equivalent to the square root in the linear domain)."

I think Fabien is mistaken about short-term being gated, but please correct me if I'm wrong. As I understand it, momentary and short-term loudness are both ungated, pre-filtered RMS measurements.
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Old 01-16-2019, 02:40 PM   #22
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It seems I naturally mix at about -23 Lufs. Odd but when I mix and finally get around to putting a Lufs meter on the master I'm usually in that range even though my room is rocking. When I go to push the mix to -14 Lufs I have to pull the master down or I blow myself out of the room. Then to render I have to put master at 0dB and pray I don't hit play or instead manually turn down my monitors.

I wish I could figure a way to mix into the -14. Maybe leave the meter up at all time and mix towards it?
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Old 01-16-2019, 02:46 PM   #23
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Quote:
Originally Posted by Coachz View Post
It seems I naturally mix at about -23 Lufs.
With peaks near zero? I ask because an important point of LUFS.... Loudness Units Full Scale so if you show -23 LUFS but peaking at -10 (as an example) then it's really sort of -13 LUFS from a loudness perception standpoint. If that's the case, your monitors may be too loud. Have you ever calibrated you monitors to a reference SPL between reaper and the outside world?
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Old 01-16-2019, 02:52 PM   #24
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Originally Posted by Coachz View Post
I wish I could figure a way to mix into the -14. Maybe leave the meter up at all time and mix towards it?
Why don't you make your life easier and use the SWS loudness functions? That way you can just get the mix sounding good without worrying about the master level as long as it's not clipping? After that, export to 32-bit, normalize by loudness and apply your limiter or saturation after that.

I have a script that normalizes by short-term loudness if you're interested.
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Old 01-16-2019, 03:15 PM   #25
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Quote:
Originally Posted by karbomusic View Post
With peaks near zero? I ask because an important point of LUFS.... Loudness Units Full Scale so if you show -23 LUFS but peaking at -10 (as an example) then it's really sort of -13 LUFS from a loudness perception standpoint. If that's the case, your monitors may be too loud. Have you ever calibrated you monitors to a reference SPL between reaper and the outside world?
Peaks not near zero I'm always down about 13 lufs peaking at like -10. My monitors volumes are right where I like them. I had an spl meter before Armageddon here. Now I wing it.
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Old 01-16-2019, 03:20 PM   #26
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Originally Posted by ErBird View Post
Why don't you make your life easier and use the SWS loudness functions? That way you can just get the mix sounding good without worrying about the master level as long as it's not clipping? After that, export to 32-bit, normalize by loudness and apply your limiter or saturation after that.

I have a script that normalizes by short-term loudness if you're interested.
I'm rendering out to 16 bit 44.1khz. Can SWS render that and set the output to -14 LUFS ? If so, how please ?
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Old 01-16-2019, 03:42 PM   #27
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I'm rendering out to 16 bit 44.1khz. Can SWS render that and set the output to -14 LUFS ? If so, how please ?
The 32-bit is just to make an intermediate master file. This is because SWS loudness doesn't work in real-time, you need an item to apply it to. By rendering your project to 32-bit, that gives you a high-quality file you can then import into a new session and boost up to the necessary loudness without increasing the quantization errors (digital noise) that would be there if you rendered to 16-bit. This is basically equivalent to putting all of your tracks in a folder and rendering the folder, assuming you don't have anything on your master bus.

After you set the desired loudness and put your limiter/saturator on the track, you can render to whatever you want. But the intermediate "master" should be 32-bit.
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Old 01-16-2019, 03:47 PM   #28
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The 32-bit is just to make an intermediate master file. This is because SWS loudness doesn't work in real-time, you need an item to apply it to. By rendering your project to 32-bit, that gives you a high-quality file you can then import into a new session and boost up to the necessary loudness without increasing the quantization errors (digital noise) that would be there if you rendered to 16-bit. This is basically equivalent to putting all of your tracks in a folder and rendering the folder, assuming you don't have anything on your master bus.

After you set the desired loudness and put your limiter/saturator on the track, you can render to whatever you want. But the intermediate "master" should be 32-bit.
Thanks but I'm too lazy to do that.
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Old 01-16-2019, 03:50 PM   #29
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Thanks but I'm too lazy to do that.
I understand. Not everyone has an extra 20 seconds to spare.
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Old 01-16-2019, 04:06 PM   #30
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Rendering here takes a lot longer than that
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Old 01-16-2019, 04:13 PM   #31
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Quote:
I understand. Not everyone has an extra 20 seconds to spare.

lolol- i do!
glad your a fan of da 32! =less worriezzz+better qaulities.
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Old 01-16-2019, 04:15 PM   #32
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Rendering here takes a lot longer than that

tried recording in at 32bit?
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Old 01-16-2019, 04:20 PM   #33
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Nah, 24 bit is fine
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Old 01-16-2019, 04:23 PM   #34
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Quote:
Nah, 24 bit is fine

yeah- i used to think so to-until i learned the advantages..
how's your clipping going these days?
even using a 16bit interface-mixing and itb processing at 32bit (+higher sampling rates) is superior in every way..
bouncing out your finals...is another thing..
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Old 01-16-2019, 04:31 PM   #35
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mixing and itb processing at 32bit is superior in every way..
bouncing out your finals...is another thing..
Which is why I suggested what I did. I mean there's no other way to implement the SWS loudness functions, because they're not real-time. You have to make an intermediate render. Rendering the mix to 32-bit is only there to preserve quality.
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Old 01-16-2019, 04:34 PM   #36
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yeah- i used to think so to-until i learned the advantages..
how's your clipping going these days?
even using a 16bit interface-mixing and itb processing at 32bit (+higher sampling rates) is superior in every way..
bouncing out your finals...is another thing..
No clipping here. Tons of headroom
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Old 01-16-2019, 04:40 PM   #37
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Tons of headroom

^that says to me -"more hiss in your recordings,or from your amplifier"
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Old 01-16-2019, 04:40 PM   #38
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Peaks not near zero I'm always down about 13 lufs peaking at like -10. My monitors volumes are right where I like them. I had an spl meter before Armageddon here. Now I wing it.
No, I mean it should show peaks near zero AND your monitors right where you like them. That's what proper calibration means - let's assume reaper shows -18 dBFS RMS with peaks near zero with pink noise, when calibrated that should be at the proper monitoring volume such as 83 dB or whatever SPL you like. Without that, those LUFs numbers mean nothing without constantly converting them.

Nothing wrong with winging it but if so, no use even caring about LUFs values to begin with, just use your ears and forget about meters and especially forget about LUFs. As far as integrated there is no shortcut, you either render it, analyze it which is essentially the same as a render, play the entire song real-time or don't use integrated.

You come seem me, I'll give you damn SPL meter.

Last edited by karbomusic; 01-16-2019 at 04:47 PM.
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Old 01-16-2019, 04:45 PM   #39
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No, I mean it should show peaks near zero AND your monitors right where you like them. That's what proper calibration means - let's assume reaper shows -18 dBFS RMS with peaks near zero with pink noise, when calibrated that should be at the proper monitoring volume such as 83 dB or whatever SPL you like. Without that, those LUFs numbers mean nothing without constantly converting them.

Nothing wrong with winging it but if so, no use even caring about LUFs values to begin with, just use your ears and forget about meters.
With speakers turned off, if Reaper output says -14 lufs, won't I get a render at -14 lufs? No calibration performed.
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Old 01-16-2019, 04:48 PM   #40
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With speakers turned off, if Reaper output says -14 lufs, won't I get a render at -14 lufs? No calibration performed.
Sure, if the peaks are near zero in the final render but only if they are near zero.
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