Audio Recording
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Record at a quality as close to what you want for a final mix as possible (spend time getting mic placement, room acoustics, preamp gains, etc. perfect!). Don’t ever say you’ll wait to make something better at a later stage. You can only do so much with bad raw material. Use reference quality (flat frequency response) earphones and monitor speakers in your editing.
have -6dB as the ceiling for recording (peak below -6dB … not near 0dB)
average the recording around -18dB
mono (single) mic individual instruments:
instruments need a certain distance from mic to create character and timbre
stereo (two) mic ensemble or solo performance = experiment with different stereo mic techniques!
closer for less of the room’s natural reverberation sound
further away for more of the room’s natural reverberation sound
the further your mic is away from your source, the more preamp gain you need, and thus you end up raising your noise floor
classical music stereo mic placement = 4-20 feet away and 7-10 feet above performers’ floor level
lossless audio files (WAV, AIFF, FLAC, CAF) = highest quality
use through entire recording/editing process (note different maximum size for each … and make sure your DAW is setup to record unlimited time)
lossy audio files (MP3, AAC, OGG) = these discard data in order to compress and save space
save as lossy ONLY when you are completely done with all editing and are ready to stream online (in which case use CBR) or to save space on a device (in which case VBR is better)
sample rate = number of times per second that the signal voltage is measured
use 44.1kHz or 96kHz for recording audio alone
use 48kHz for recording audio to sync with video
bit depth = use 24 bit = this is the resolution at which the amplitude of the signal voltage is measure
bit rate = amount of data (bits) required to represent one second of audio (higher bit rate = better quality)
bit rate = bits per sample (bit depth) x samples per second (sample rate) x number of channels
bit rate of CD quality ==> 1411.2kbps = 16 x 44,100 x 2
(4min song would be 40MB)
good practices:
turn off all notifications and wifi, close any unnecessary programs, disable auto updates, use caffeine app
use a backup recorder: dedicated audio recorders like those made by Zoom or Sound Devices make great backup recorders (format media cards in the device)
mic level = a few thousandths of a volt; 1-50mV (requires a preamp to boost to line level)
instrument level = between mic and line level in strength; 70-150mV (the level that comes out of guitar is instrument level and must be amplified by a preamp)
line level = about 1000 times the level of mic level
consumer line level = -10dB (320mV)
professional line level = +4dB (1-1.5V)
speaker level = the level that comes out of an amplifier (no max voltage limit)
Amplitude
gain plugin should be the first plugin for every mix (get each track down to around -18dB and make -6dB the ceiling)
Volume and panning automation (also envelops such as fade in, fade out)
Spectral Filter
Equalization (high pass, low pass, low shelf, high shelf, peak filter, notch filter)
Time = combine the original audio signal with delayed and modified copies of itself
Modulation effects: Chorus, Tremolo, Flanger and Phaser
Fine tune each completed song to fit best with all other songs in an album; many times, those that master are only given stereo mix downs of tracks (rather than multitrack DAW files or stems)
Slight adjustments to EQ, stereo image, and reverb
dynamic processing to deliver to a service:
overall gain changes, compression, and limiting
Spacing and fades are added to the beginning and ending of each song
Make each song able to sound well played on any device
Aim for -14LUFS (integrated) and -1TP to prepare your tracks for streaming services (Youtube = -13LUFS, Spotify = -14LUFS, iTunes = -16LUFS, etc.)
LUFS stands for Loudness Units Full Scale, and is a way of measuring the overall volume of a song in a way that’s much closer to the way the human ear detects volume changes.
TP in this case stands for True Peak, and is used to measure the actual peak loudness of the file when played back in the analog realm (ie, a speaker).
Instead, loudness normalisation positively encourages the use of dynamics and transients. Tracks are made punchy by being dynamic rather than just loud, and compression and limiting become musical effects rather than essential competitive processing. Headroom is restored, inter-sample clipping is banished, and the digital environment finally achieves the sonic quality and dynamic range of which it is capable.
In theory, two pieces of music that register identical LUFS readings should sound like they’re at the same level, and in practice, they do indeed sound like they’re at the same level, regardless of whatever the peak or RMS readings say. So we have an immediate, practical benefit — if you’re mastering and want consistent levels among tracks, check their LUFS readings.
about stereo microphone techniques ... allowing you to hear how they differ (note, they accidentally *showed* the NOS when speaking about ORTF and vis versa in the very beginning of the video ... but I think when they perform, the label is correct) - ... download and print this protractor to set up ORTF:
Put all the tracks of a single song together ()
Dynamics Processing: Compressor (, , ), Limiter, Expander (ex. noise gate), Normalize, Distortion, Envelop
Delay = , Delay, Echo
"Both of these songs have the same apparent loudness, but the second track has been turned down considerably to make this happen." (pic above and quote from "")
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