Comprehensive Guide
Audio Telemetry: The Nyquist-Shannon Theorem
Learn the principles of digital sampling, bit depth headroom, and the mathematical reconstruction of sine waves.
What makes uncompressed Audio so large?
Digital audio translates analog sound waves into numbers that computers can store. A high-quality uncompressed file (like a standard `.WAV`) does this phenomenally fast—recording tens of thousands of "snapshots" every single second, tracking both the frequency pitch and strictly mapped volume amplitude. Because it never compresses or guesses redundant data like an MP3 does, it requires massive amounts of exact Bitrate math. This Audio Spec Calculator allows producers and engineers to calculate storage footprints for session masters.
The PCM Data Equation
$\text{Bitrate (bps)} = \text{Sample Rate} \times \text{Bit Depth} \times \text{Channels}$
A standard CD uses $44,100 \times 16 \times 2 = 1,411,200$ bps (1,411 Kbps).
Key Technical Components
- Sample Rate (Hz): How many snapshots are taken per second. Due to the Nyquist Theorem, you must sample at exactly *double* the highest frequency you wish to capture. Because human hearing caps around 20,000 Hz, CDs use 44,100 Hz to capture the entire spectrum cleanly.
- Bit Depth (Bits): Defines the volume dynamic range. 16-bit audio provides roughly 96 Decibels of volume between absolute silence and maximum loudness. 24-bit pushes this to 144 dB, capturing pin-drops right next to jet engines without clipping.
- Lossless (FLAC) vs Lossy (MP3): The calculations above represent *uncompressed PCM* algorithms. Lossless `.FLAC` formats cut the footprint in half using algorithm zipping, while Lossy `.MP3` formats brutally throw away quiet frequencies to achieve file sizes 10x smaller.