As we start to wrap up this long series on frequency, we work our way towards the natural conclusion, export settings. While sample rate and bit depth can be and are recording settings and are decided at the beginning of the session, we also have options when bouncing down (see also: exporting) our sessions. The same goes for what bitrate mp3 to convert our audio to. Each of these settings affects frequency either super directly or mildly indirectly so we're going to focus a bit on HOW our voices are converted and encoded into a digital signal, as well as how it's converted back into an analog signal.
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Here's some things we are defining in this episode.
Sample Rate
When an analog signal (your voice) is converted into a digital signal, our ADAC (analog to digital/ digital to analog converter) is taking snapshots of our analog signal at regular intervals. The common numbers that come to mind are 44.1k and 48k. What this means is 44,100 times per second, our converters are taking a "picture" of our amplitude and assigning that a value in either positive or negative. The higher the sample rate, the more accurately we can recreate this sine wave above.
Nyquist Theorem
The Nyquist Theorem, sometimes also called the Nyquist-Shannon sampling theorem states that in order to accurately reproduce an analog sound digitally, we must have a sample rate that's equal to or greater than double the highest frequency we are trying to sample. The audible range of human hearing is (theoretically) 20Hz-20kHz. To accurately convert and reproduce frequencies up to 20kHz, we must have a sample rate of AT LEAST 40kHz. Why 44.1? Adding a little extra for error and the slope of a low pass filter.
Aliasing
If we use a sample rate that's too small to accurately capture and reproduce the frequencies present in the recording, we get a phenomenal called aliasing. This is an inaccurate capturing and recreation of a complex waveform due to too few samples. The photo above will do a better job of explaining this than I could ever do in words. It's also cray helpful to HEAR aliasing. Just listen to the episode. Trust me...
Bit Depth
When we are capturing amplitude readings at our chosen sample rate, it is rounded to the nearest available amplitude value. The higher the bit rate, the more available values to be rounded to, resulting in a more accurate recreation of the analog signal. I like to think of this as pixels in a photograph. The more available pixels, the sharper the detail of each. Our number of available values goes up EXPONENTIALLY as we raise our bit rate.
16 bit has 65,536 possible integer values per sample
24 bit has 16,777,216 possible integer values per sample
32 bit has 4,294,967,296 possible integer values per sample
And just for fun, here's a visual of the listening exercise!
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My Signal Chain
Hardware:
Audio Interface: Apogee Ensemble
Microphone: Shure SM7b
Headphones: Audio-Technia ATH-M50x
Earbuds: Klipsch R6i II
Studio Monitors: Yamaha HS7
Mic Stand: Rode PS1A Boom Arm
Software:
IzoTope RX6 Mouth De-Click
IzoTope RX6 Voice De-Noise
FabFilter ProQ3
Waves Vocal
Rider Waves CLA2A
Waves L2 Limiter
Waves WLM Meter
Waves Durrough Meter
*most of these links are affiliate links
Midroll Song: Road Trip by Joakim Karud
Closing Song: Great Days by Joakim Karud
For more info, or to ask any questions, check out my website and reach out to hello@cleancutaudio.com
Hmm, that's an interesting article, maybe I'll throw it to a friend of mine who's into writing music. By the way, since he just started, he takes music from different soundtracks, while using https://www.movavi.com/learning-portal/best-mp3-converter-for-mac.html on his mac.
Can you please tell me if Waves Vocal is suitable for a novice musician?