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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: checht on August 14, 2018, 12:10:04 AM
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So, starting out on matrixing AUD + SBD. My workflow is to line 'em up, sync length, then mix them by ear at their recorded levels, render to 1 stereo track, and bring it up to a bit below zero. The recordings are within a few db of each other, so I can get the sound I want, usually applying negative gain to the SBD.
Sometimes I wonder if it would be better to bring them both to -3db then mix, render, and normalize to close to zero. I am not sure if this additional DSP is a bad idea. I realize I'm applying DSP when I apply negative gain to the SBD, and when rendering, so does the extra normalizing make a difference?
TIA
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Depends on the source material. Given you mentioned mixing in an SBD I’ll assume the source is amplified music - probably pop, rock, or electronica, please correct me if I am wrong.
If that assumption is true, then the difference you’d get in post processing from DSP is pretty minimal. Your sound card (depending on the quality of your amp and converters) will likely inject more noise overall into the system.
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So, starting out on matrixing AUD + SBD. My workflow is to line 'em up, sync length, then mix them by ear at their recorded levels, render to 1 stereo track, and bring it up to a bit below zero. The recordings are within a few db of each other, so I can get the sound I want, usually applying negative gain to the SBD.
Sometimes I wonder if it would be better to bring them both to -3db then mix, render, and normalize to close to zero. I am not sure if this additional DSP is a bad idea. I realize I'm applying DSP when I apply negative gain to the SBD, and when rendering, so does the extra normalizing make a difference?
I would suggest mixing them at whatever levels get you close to 0 dB full-scale as possible, without going over.
Negative gain (if necessary) at mix time is the right time to do that, in my opinion. Less DSP is logically better, based on Occam's Razor if nothing else.
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With modern DAW editing software you needn't worry about this much, especially in the case you describe where each source has decent levels to begin with and the overall level change is not dramatic. The mathematical calculation space in which the summing and level adjustment is taking place is significantly larger than the bit-depth of the recorded input files as well as the target output format. The data being manipulated fits entirely within that mathematical calculation space with room to spare at the both the low-level and high-level ends of the range. So regardless of whether you sum first and then tweak level for output, or tweak levels first and then sum to the target output level, the result should be mathematically equivalent.
What you don't want to do is reduce the level of the sources by an arbitrarily large amount, sum them, then increase the level of the summed output by a dramatic amount to get it back up to the desired output level. That can cause the loss of low level information and increase the noise floor.