Patryk_Rebisz
Veteran
When feeding the 1khz tone from MixPre to Zoom H4 (NOT H4n) what should i set the input levels to (using Rode NTG2 with internal battery and Sennheiser G2)?
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-20db I believe.
-20db I believe.
The original H4 could take -10dB line-level via the 1/4" inputs. The best thing to do is to use a 1/8" TRS>dual 1/4" TS Y-cable to feed the MixPre's tape out to the H4's line ins.
So i understand correctly instead of XLT-to-1/4" i should use 1/8"-to-1/4" Y spliter from tape out, yes?
In that case when i feed the 1khz tone to the Zoom H4 my levels on the recorder should hit -10 on its scale, yes?
Just a word of caution - the H4n does not have true line ins, as you cannot bypass the internal preamps. The way mentioned above is still the best way be (the gain would probably be 0 on the recorder), but the H4n is well known for being very sensitive on the input and is prone to distortion. Some claim that it works fine line in for them, some don't. Try it out and if you're content then fine. If you want to be shure though the best way would be to use a pad-cable between them.
No. -10dB in this case is the voltage level of the signal and refers to unbalanced, line-level audio. As for the sound levels on the H4's meters, you still want to set your tone level to -20dB. That's the reference mark, and then your dialog should peak at -10dB with sudden peaks (sudden and loud noises like shouts and claps and coughs) hitting as high is -6dB.
There are several different kinds of dB scale, so it can get a little confusing.
Back to the interconnect, I used an H4 with a MixPre for several projects, and using the 1/8" tape out to the two 1/4" TS ins worked very well.
why is it preferable for peaking to be at -6db rather than 0db?
That's why in bold letters in original post - NOT H4n!!
The G2 isn't affected by phantom power.
It's all about headroom. This is dBFS, or Full Scale. In the analog world, 0dB is the reference point (optimal signal:noise), and there's room to go past 0dB with no adverse effects. In the world of digital audio, 0dB is the top and there's nowhere to go from there.
With digital, everything is recorded in words, and the length of that word is determined by the bit-rate. 16-bit = 16 bits (or digits) per word. 24-bit = 24 bits per word. (Sample rate = the number of words per second.) So, in the 16-bit world, 0000000000000000 = silence. 1111111111111111 = maximum amplitude. At that point, all the bits are full and the signal is at 0dB. If you try to pack any more information in there, which you can't, it goes into digital clipping. That's a nasty sound, since it's literally sheering the tops off the waveforms and resulting in square waves.
This is why we aim for -10dB as a standard peak in digital sound, so that there's a good deal of headroom in case of sudden peaks.
Ahh, so if I KNOW I'm not going to get anything unexpected but I'm working with less than stellar equipment and need the extra gain, I can push it a little bit so my peaking is closer to 0dB as long as I know I'm definitely not going over?