Audio Basics
These are the audio terms you need before reading analysis or mastering pages. They are grouped here because they explain the same signal path: how sound is represented, measured, clipped, delayed, and played back.
Sample Rate
Sample rate is how many audio samples are stored per second. 44.1 kHz and 48 kHz are common delivery rates. Higher sample rates can represent higher frequencies, but they also use more CPU and memory.
For most browser analysis and mastering work, the best first choice is the file's original sample rate. Avoid unnecessary conversion unless a tool requires it.
Bit Depth
Bit depth describes the resolution of stored sample values. More bits do not raise the ceiling — full scale is 0 dBFS at any word length — they lower the quantization noise floor underneath it, roughly 6 dB per bit. That is what lets quiet detail survive and what makes it cheap to work well below full scale. Final consumer files are often 16-bit or compressed formats, while production files are commonly 24-bit or floating point.
Inside Web Audio and libsonare processing, samples are represented as floating-point values. Floating point is the one case where the format itself has room above full scale, so an intermediate stage can overshoot 1.0 without clipping; the final export back to fixed-point still needs sensible peak safety.
Mono and Stereo
Mono has one channel. Stereo has left and right channels. Stereo can create width and spatial placement, but it also introduces mono-compatibility questions.
When a stereo signal is summed to mono, some side information can weaken or cancel. This is why the mastering demo shows correlation (a -1 to +1 reading of how in-phase the left and right channels are) and stereo image.
Amplitude and dB
Amplitude is the instantaneous sample value. dB is a logarithmic way to describe level. Audio work uses dB because human loudness perception is closer to logarithmic than linear.
Typical digital sample values are normalized to the range -1.0 to 1.0. When level is measured against that digital ceiling, it is written in dBFS (decibels relative to full scale): 0 dBFS is the loudest value the format can represent, and every real signal sits below it as a negative number.
Clipping and Headroom
Clipping happens when a signal asks for more level than the system can represent. Digital clipping can sound harsh because waveform peaks are cut flat.
Headroom is the margin between the current peak level and the ceiling. Mastering needs enough headroom before final limiting, and enough true-peak safety after final rendering. True peak counts the higher peaks that appear between the stored samples once playback or lossy encoding reconstructs the waveform, so it can sit above the plain sample peak.
Latency
Latency is delay. Some processing, especially lookahead limiting and spectral repair, needs future samples before it can decide what to do. Offline rendering can tolerate latency. Live monitoring and interactive tools need it kept low.
Implementation notes
The browser demo decodes files through Web Audio, processes floating-point buffers through libsonare WASM, and exports stereo 16-bit PCM WAV. The mastering worker keeps heavy rendering away from the UI thread. Object URLs for source, rendered audio, reports, and settings are revoked when replaced to avoid accumulating browser memory.
Related: Mono Compatibility, True Peak Safety, Reading Mastering Meters