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5

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 ...


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

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

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

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, ...


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

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

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

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 ...


2

While recording ambience sounds in Zoom h6 I have found that I generally have to keep the gain near 8-9 , for situations like general traffic and cityscape 6-7 usually works considering that one is recording in a generally noisy environment. Again when I have tried recording in villages, I really have to push up till 10 sometimes as these places are very ...


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

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

Low or somewhere inbetween. Unless you have acoustic treatment like baffles and obsorbers, defusers. If you have the level too high you will 'excite' the room causing potentially standing waves; peaks and troughs that are stationary. If you are sat in either a node or anti node (technical term for aforementioned) you'll end up with a thin sounding mix or ...


2

The international consensus seems now established that loudness measure should be done according to ITU R BS.1770. Algorithms to measure audio programme loudness and true-peak audio level It is widely used for broadcast delivery specs. There seems to be two main target levels : -23/-24 LUFS for broadcast, -16 LUFS for streaming. The EBU R128 is based ...


2

There is no formula you can use to make this conversion. It is unlikely your webcam would be suitable as a sound sensor due to the processing involved in detecting and multiplexing the audio into the visual data stream. Also, the 'dB' values you are quoting are meaningless. In order to be able to detect something useful, you would need a raw feed from the ...


1

You need to be able to edit the audio in an audio editor. Assuming that is so... 1: check there aren't strong levels below 100Hz (low bass rumble). If there are, remove them with an EQ or High Pass Filter. 2: normalise the file, which is usually a one-click operation in any editor. It means 'make the file as loud as it can be without clipping'. 3: if ...


1

With a limited dynamic range - which you will often find with recording backgrounds and atmos - you can increase the gain on the recorder, but you will simply find yourself increasing the level of microphone and preamp-self-noise along with the limited signal that you are recording. There is no recommended setting - other than to recommend you use your ...


1

After reading and watching videos on YouTube I came to the realization that is obvious to any mediocre+ sound engineer. I come from field of biology so please excuse my ignorance on the matter. Behringer Xenyx QX1202USB holds four microphone inputs. In each channel, one is an XLR input and one is regular 1/4" line in. It turns out that XLR input (with three ...


1

You have a few different options. You can use some gates on the lapel mics and set the threshold high enough to cut the TV bleedover, but not the character dialogue. This has the downside of potentially sounding unnatural, and you'll have to tune the attack and decay of the gates to get the speech to not sound choppy as it cuts in and out. If the TV sound ...


1

A balanced line-level output usually has TRS (tip-ring-shield) 1/4" connectors. Connecting them to an unbalanced input requires the use of mono cables which short R with S (ring/return with shield) in the TRS socket. This balanced->imbalanced conversion works only at the jack level, not at the socket level: a mono socket has only "T" (tip) and "S" (shield) ...


1

A computer mic input is monaural, so you must "mix" two or more microphones together to make a monaural signal. It may be possible to simply parallel two microphones with a "Y-adapter". But even if that works, it does not provide independent control of each microphone loudness. The conventional solution would be to use an audio mixer which will take several ...


1

Personally, I don't ride the faders too much between single words. Only if 'longer' passages are quieter/louder I turn up/down the gain. Moving the faders within words should be avoided. Until now, I've never asked the boom operator to pull away the microphone as it colors the sound. Since I'm recording at 24 bit, I can leave quite some headroom for ...


1

Hopefully the boom operator knows what they are doing and are trying to capture the best sound and levels, letting the director know when there are noises that will spoil the shot, and coordinating on shooting locations and blocking challenges with respect to the sounds. Along with that, there should be a dialog between the location sound person/people/team ...


1

In this situation, two of your proposals are commonly used in conjunction : ride the levels appropriately have the boom operator pull the microphone at right moments.


1

MP3 metadata has a sort of dynamics compression hack where an individual track can say to the player “play me at 110% volume” and another track can say “play me at 90% volume” and the result is that those 2 MP3’s seem to be at the same perceived volume. With lossless audio, you don’t have this hack. You have to use actual dynamics compression. So the way ...


1

Replaygain is a metadata based system - it analyses the absolute peak value of your audio, then writes a scaling value which the player reads and uses to amplify the audio data when you play it back. The wave format doesn't natively support the metadata used for it so you have to either alter the actual audio data to normalise it, or use a hack to the ...


1

First, you should consider forgetting about those on-board pickup/preamp systems. Acoustic instruments sound best when recorded with an external condenser microphone, full stop. If the pickup signal is no good, there's not much you can do in mix. If the acoustic guitar has decent quality, then with a standard small-diaphragm condenser mic and no processing ...


1

You want to avoid having anything too low. If there is a unity mark, then as long as impedance match, you should be able to use that. If not, adjust things so that you can have moderate settings that are not too low on anything. The lower the volume level gets, the harder it is for the signal to be differentiated from the noise and the lower the quality ...


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