Hot answers tagged

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

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

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

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

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

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 ...


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

"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

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

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

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

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

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

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

as a general advice, i found this post very useful: http://audiocookbook.org/guidelines-for-making-a-sound-design-demo-reel/ good luck!


1

I would go even further and compress/limit everything to -1.1 dB. I'm sure a lot of fellows won't agree but I'd rather have one showreel aimed at the web with these specs, and a separate one for the professionals which includes headroom and does not generate any loudness conflicts.


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 ...


1

Whenever you have a digital device be sure that the levels won't reach 0 dBFs. On analogue devices you can go beyond 0 dB(u or v in most cases) but you will get some overdrive. This harmonic distortion can be a desired effect. So when you record something be sure that the analogue digital converter -> for example your USB interface isn't reaching 0dB. But ...


1

I still don't have 50 experience to comment on an above post, but I'll second the Mix-pre. I'd buy used for around $600 or so and spend the rest on whatever new recorder you find to your liking. You're going to get good, quiet preamps, and low noise phantom power, along with several other features. You're going to want low noise equipment when recording ...


Only top voted, non community-wiki answers of a minimum length are eligible