A field team faces an unresolved physical question: Where does a 30 kHz tone go at 48 kHz? They must answer it before changing input audio tone frequency on the real device. Predict the direction first.
See the relationship before changing it
The figure reads from left to right. The blue card is input audio tone frequency. The middle card applies this page's relationship. The green card is observed alias. Walk the arrows once: set the input, apply the rule, then read the result with its unit.
The retained audit below checks several chapter fixtures. This added model holds every other chapter fixture fixed, so the numeric fixture does not switch without explanation.
Derive the baseline in four named moves
- 1
Name the input. The chapter baseline for input audio tone frequency is 30000.
- 2
Name the relationship. fN=fs/2; falias=|f-fs·round(f/fs)|; L=20log10(A/AFS); SNR=6.02N+1.76
- 3
Substitute the chapter fixture. Set input audio tone frequency to 30000. The page ledger gives observed alias as 18000 Hz.
- 4
Read the result. Keep Hz beside the value. Use it only inside the technical boundary on this page.
Predict, then change input audio tone frequency
Try Predict the direction of observed alias. Move one control, calculate, then check your prediction.
Observe The sample rate fixes where a tone folds. dBFS describes its digital amplitude, while anti-alias rejection and real noise belong to the device path and must be tested. Reset the control to 30000 and compare observed alias.
Explain Only input audio tone frequency moves here. The other chapter fixtures remain fixed.
Check yourself
What should you do before trusting a moved-control result?
What does this small model leave out?
1. Sampling repeats the spectrum
A sampler can represent frequencies only inside one Nyquist band. Frequencies outside it are mirrored into that band unless the analogue input path attenuates them first.
2. Fold to the nearest sample-rate copy
Find the ceiling48,000/2 = 24,000 Hz.
Choose the nearest copyk = round(f/fs).
Take the absolute differencefalias = |f − kfs|.
3. Keep level scales separate
dBFS is a relative digital ratio. It becomes absolute sound pressure level only after microphone-path calibration. The ideal converter result SNR ≈ 6.02N + 1.76 is another bounded ratio, not a measured phone noise floor.
4. Try the input tone
TryMove an ideal input tone above the chapter's 24 kHz ceiling while the 48 kHz rate stays fixed. The half-scale amplitude is labelled interaction-only.
ObserveAt 30 kHz the sampled record shows 18 kHz; at 26 kHz it shows 22 kHz. Both values are below 24 kHz and can look like genuine in-band tones.
ExplainThe sample rate fixes where a tone folds. dBFS describes its digital amplitude, while anti-alias rejection and real noise belong to the device path and must be tested.
The folding model assumes an ideal unfiltered sampler.
- measured evidence
- Needs separate evidence
Use field evidence or a deeper model before release.
5. Reproduce the chapter checks
The 98.1 dB figure is total ideal quantisation-noise power under a full-scale-sine assumption, not a spectral-bin floor.
6. Write an honest lab claim
Report sample rate, FFT size, window, gain, processing, and relative dBFS. Use a controlled quiet-input or reference measurement for the actual floor, and avoid claiming SPL without calibration.
7. Check yourself
Why does 30 kHz appear at 18 kHz?
What does −6.02 dBFS mean in the interaction?
Does 98.1 dB prove the phone's noise floor?
These are the chapter inputs, worked results, and named teaching assumptions.
- 48 kHz rate
- Frequency, sample rate, or event rate
- 24 kHz ceiling
- Frequency, sample rate, or event rate
- 30→18 kHz
- Frequency, sample rate, or event rate
- 26→22 kHz folds
- Frequency, sample rate, or event rate
- ideal 16-bit 98.1 dB result reproduce the chapter
- Gain, loss, margin, or level ratio
- The half-scale −6.02 dBFS readout is labelled as an interaction value
- Named teaching assumption
This page does not certify a phone's anti-alias filter, noise floor, or SPL.
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