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6

The X-axis is obviously frequency, and the Y-axis is power. Each curve on the graph is a "perceived loudness" level. The curves dip lower towards the middle-high end of the spectrum, because we hear those frequencies more easily. Each curve is labelled with a number equivalent to its perceived loudness level. If you look at each of those curves on the 1,000 ...


5

To me, it's one of those subjective things in the mix. You're dialogue level sounds appropriate (equal to -27dB on a DOlby LM100 for LEQ(a) ), but for music, in my opinion you sort of have to ride it against dialogue to what feels right, especially since the dynamics of music are always changed compared to dialogue. It should probably be nice and full, but ...


4

Wiki explains normalization here. I'm not a big fan of normalizing while mastering sfx. I find a lot of commercial sfx libraries are mastered way too hot, especially when I am previewing them in a room calibrated at 79db or 85 db reference level (standard listening levels for TV and Film mixing). There's this belief amongst some that louder sfx are more ...


4

Heh. Welcome to the rest of your career. :) I fall more into the "client is always right" camp. I think it's my job to speak up when I disagree and educate the clients just as you have done but at the end of the day, they're the ones who have sweat blood over this film for months, if not years on end, and the director always always gets the last word. "...


4

This is not a sound level. It's a pressure level, or voltage level, or whatever physical sound transmission you're imagining. DC offset is the absolute sample value of the average of all audio samples. Plain sample values are also used to calculate peak levels, except that you don't average anything but take the maximum of the absolute values. Because the ...


4

This is a very old thread, but I just saw it. I am the technical manager of London Short Film Festival, and the author of the "how to make a DCP" guide the OP referred to. First off, I'm really sorry to hear that the film did not screen as intended. And why indeed did the DCP-making guide I wrote suggest adjusting peak audio levels to -10dB in DCP-o-matic?...


4

If it was recorded with one microphone, there is no good way to rectify this problem. Vocals and guitars share the same frequencies, so EQ is out of the picture. Sorry.


3

I'm no expert, but I expect that is what you pay for. If you've not looked already, it might be worth checking the B&K microphone page, just to see what is already out there. They are pretty much the standard for professional measurement equipment like this, and if you need to ask how much they cost, then you can't afford one! But the fact that only a ...


3

I suggest going with the TV-style mix. In which, I usually take the 85 SPL theatric mix, boost it by 6dB, slight remixing where necessary, and have a brickwall at -3dB for good measure to ensure nothing clips (TV often brickwalls dialogue around -10dB, but since it's for web-based display, I allow myself up to about -3dB. Going about 6dB hotter than ...


3

If you have enough time in advance you could pick up a cable tester that has a tone generator in it for about $20. Here's one. Pull it into the studio, run it through your mixer, and mark the point at which its output = -20 dbfs. It'll probably be close to unity gain. Then when you're out in the field you can just plug the tester into the mixer, set the ...


3

Assuming you are using the IO2 with the guitar input, with Mic/Line / Guitar switch set to guitar, I'd start by bringing down the gain knob then turning it up slowly. Also make sure that you aren't by some quirk of monitoring hearing the guitar coming straight through the interface AND having it come through Cubase at the same time? If none of that works, ...


3

There are several ways to do this. Simple answer: You can edit each channel separately. When the wave-form is selected, you can hit the up arrow key or down arrow key to change the selection to left only or right only. You can choose to Normalize each channel to 100% or adjust them by ear. Or you can play it totally safe and mix either channel separately ...


2

If you are going to use normalization, only use it for searching/indexing and locating a particular sound. Do not normalize the original or raw material. The main reason for this is that once you normalize the audio, you burn in all the sampling artifacts, noise and harmonic distortion that goes on at the low end of the bit-resolution spectrum. Adding ...


2

I'm not a fan of normalization at all. In theory, it could be a useful tool, but I've yet to find a situation where it was truly useful. When you normalize files, you merely take the loudest sound and adjust the overall amplitude of the file to make that loudest sound match the dB level (or percentage) you specified. The result is a ton of files that have ...


2

Sounds like your problem is the signal to noise ratio in your field recordings. Normalization is the process of scanning your file for the highest value (peak or RMS, depending on your selection) and assigning it to the new value that you have determined. In turn, the same increase is applied to the rest of your file. So for example, if you have a door slam ...


2

I agree with Stavro, it's totally subjective. Get your dialogue to sit at the right average level, then mix the music around it. Numbers won't help you much, as different instruments will interact differently with voice. A lower pitched instrument like a bass drum might need to be boosted where a trumpet, or other mid horn, would need to be cut. Also, bear ...


2

Loudness levels NOT being equal across the frequency spectrum. The Fletcher-Munson curve demonstrates that our perception of frequencies is altered depending on the sound's loudness.


2

I think you're talking about the mixing process. This is a very wide ranging topic for which there are no 'quick fix' tools or tricks. A decent starting point may be to find an area in your video that you feel best represents the overall tone and content and work on that for a while. Make sure you're getting the dialog (or narration, or whatever is supposed ...


2

If you want to go hardcore... http://www.itu.int/rec/R-REC-BS.1770/en This is the spec that most North American Broadcasters adhere to when measuring. http://www.atsc.org/cms/index.php/standards/recommended-practices/185-a85-techniques-for-establishing-and-maintaining-audio-loudness-for-digital-television ATSC's guidelines


2

Busy day for me right now, but I'll point you to a blog post I did a little while ago that covers some of the basic of loudness metering: http://www.dynamicinterference.com/blog/2012/04/what-the-hell-do-all-these-letters-and-numbers-mean/ It'll get you started at least.


2

How does it sound? Is there no reason how it could actually be "too quiet"? If everything is audible and the play with dynamics sounds good and is reasonable, then I don't see a problem. How about asking them to specify, what exactly is too quiet. Everything or specific parts? I find that that's the only way to understand what they're really hearing. It ...


2

"I export the Fx, Dialogue and Music stems as .wavs and deliver them to the editor"... sadly THAT could be your problem right there... Editors have been known to forget things like panning the stereo stems left and right, which can result in a 3db gain increase. Or they think it's "too quiet" in their suite, and bump everything up... the NLE doesn't ...


2

tl;dr: digital has no headroom. 0dB is the max. this is the European kind. USA digital is 2dB hotter. give it a bit of time, it will start to make sense. from here


2

Maybe you're trying to do the wrong thing with them. Just crushing them may not help at all. Start with EQing them, Unless they're bass synths you should be shelving the bass quite a bit. If they are bass, then you should be looking at carving out frequencies for the kick /snare and/or sidechain compressing them. Tetsujin is correct here. Listen with ...


2

Is this requirement as ridiculous as it seems to be to me? Yes. No serious theater mix delivery specifications includes such limit. Nearly all theater mix will have sample peaks above -10 dB FS. Secondly, is this kind of level change actually normal in DCP creation? No. The mix should be delivered in conformance with whatever specifications apply. The ...


2

There usually is a "native" setting called Unity. At that position, the signal is neither attenuated nor amplified artificially. Basically you need to see what is the Unity for your particular device, and try to keep it at that level. With 24bit recordings, softer sounds can always be successfully amplified/normalized in post. Of course, you don't want to ...


2

There's Way too much to go into here, about leveling audio, and as I haven't heard the audio clip, I will keep it short. The speech leveller is probably causing the problem and you may not be using the hard limiter properly. I would suggest scrapping the speech leveller and instead using the single band compresser with 'instant' attack(0ms) ,a short ...


2

You are talking about perceived loudness. This is quite unlike peak level. What you can do is measure the perceived loudness for all your songs and then adjust the gain accordingly. The best way currently available to measure perceived loudness is using the R128 standard. I hope you're on a Mac, because then you can use this free commandline tool: https://...


2

Basically, you're right with all your points. The level between different albums can indeed be dramatically different, depending on how agressively it was mastered. Records from the early days of CDs are often very quiet (engineers celebrated not having to worry about vinyl's noise floor anymore, and also, early CD players didn't really work reliable with ...


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