Sampling rate controversy: simple and conclusive test methods

Testing the limits of 44.1 kHz sampling rate with high-pass filtered audio samples, impulses and tones. The difference between minimum phase and linear phase resampling filters is also investigated. Poor quality resampling can lead to fake detection of different sample rates, especially in tests with high sensitivity.

Feb. 2, 2024 (Last edited: 2026.08.23.)

Probably the most popular question regarding sampling rates and audio formats is the audibility of filter ringing. There is no discussion of formats without debating about how the resampling filter changes the signal (especially the pulse)... However, the test procedure of filter ringing is just another quite simple audibility test. Filter ringing is a resonance, you either hear it or you don't, and if you don't, you don't need to worry about it...

Understanding instrument peak levels around 20 kHz is also crucial for testing the limits of 44.1 kHz. Thinking in relative levels, -20 dBFS (dBFS: decibel below full scale; 0 dBFS is the level of a full scale sine) is a good estimation of the highest "critical band level" that may occur in music in the 16 kHz - 25 kHz frequency range during a loud cymbal hit. However, such a high level is a very rare event, more of an extreme moment in music. Without a loud cymbal hit the peak level is approx. -35 dBFS in this range.

Therefore a crucial test for 44.1 kHz sampling rate is a listening test at 22 kHz with a single-frequency tone. The level should be between -30 dBFS and -20 dBFS (definitely shouldn't be more than -20 dBFS and shouldn't be lower than -30 dBFS). The rest is just demonstration. Since the top of human hearing range is 20 kHz, the answer is pretty obvious...

Furthermore, the devil lies in the details: the quality of resampling depends on the resampling process, more precisely the simplifications applied during resampling.

The test is organized as follows:

Single-frequency test tones.
Audibility of filter ringing with a bipolar pulse.
Loud crash cymbal from an orchestral recording.
Minimum phase vs. linear phase resampling filter comparison.

Test requirements:

A loudspeaker or headphone that can reproduce the test frequencies. No need for a "Hi-Res Audio" capable system!
Quiet room, relaxed ears.
Sample rate conversion, nonlinear distortion is not audible (no false positive).

Possible reasons for false detection of different sampling rates

For single-frequency tones, two versions are included (except for the 22 kHz test). By this way a poor quality resampling can be identified and false positive results can be excluded, though it's very unlikely that resampling causes any trouble nowadays. However, it's worth keeping in mind since a poor quality resampling can lead to false detection of different sample rates.

How can we detect poor quality resampling? The result of bad quality resampling is distortion that increases with frequency. Thanks to the non-harmonic nature of the distortion poor quality resampling can be easily identified by listening to a pure tone with a frequency higher than 10 kHz. Ideally pure tones at any frequency should sound the same at 44.1 kHz and 48 kHz sampling rates. If high-frequency tones sound different at different sample rates and one version has a distorted sound, then a poor quality sample rate conversion is used on this version. The other sample rate is likely free from software resampling.

Different false positive detection may occur when the frequency of the test tone is close to the half of the sampling rate. When a DAC's reconstruction filter has slow roll-off and the image component is not attenuated well, the image component together with the test tone can generate difference tone distortion in amplifiers or speakers (usually this doesn't affect the fidelity of normal playback). A simple solution is to add a bit of masking noise between 1 kHz and 10 kHz.

Clipping in pre-amp or DAC is another source of false positive results. In the following test clipping can be ruled out.

This is an online test. The validity of the test depends on your system, your system's settings and your expertise to identify possible errors.

Advice for loudspeaker listening: the volume level should be adjusted with music and should not be turned up when listening to these test files.

You can adjust the volume with the following video and check whether the browser doesn't mute the sound. Use the "master volume control", and not the volume control in the video player.

Single-frequency tone (20 kHz)

Top of the human hearing range and a practical limit of 44.1 kHz sampling rate. 320kbps MP3 files and YouTube audio tracks are also limited at 20 kHz. It's also easy to test the quality of resampling with a 20 kHz test tone.

Level: -20 dBFS. This value is a good estimation of the highest critical band level that may occur in music in this frequency region (during a cymbal crash).

Sampling rate: 44.1 kHz

Sampling rate: 48 kHz

Meaning of the test results (resampling, 20 kHz tone audibility):

Test result		Meaning
44.1 kHz	48 kHz	Resampling	Audibility of the 20 kHz tone
not audible	not audible	no resampling error	not audible
audible	not audible	resampling error when playing 44.1 kHz audio files	not audible
not audible	audible	resampling error when playing 48 kHz audio files	not audible
audible	audible	?	?

Meaning of "?": further tests are needed. Probably both sample rates have low fidelity resampling.

Single-frequency tone (22 kHz)

A practical limit for 48 kHz sampling rate. Level is -20 dBFS and sampling frequency is 48 kHz.

Sampling rate: 48 kHz

An alternative sampling rate test method: high-pass filtered audio samples

Traditional song-based discrimination tests provide low sensitivity and rely heavily on the auditory memory. Detection of sounds in silence is much easier task than comparing "wideband" sounds with complex harmonic and temporal structure. Sensitivity of a test can be greatly increased by applying different filters and thus "format discrimination tests" can be converted to "sound detection tests". Selection of audio samples (instrument sounds, signals) is still a critical step since audio samples have a great influence on the sensitivity of the tests. Furthermore, though no test is free from false positive and false negative results, identifying false positive results is easier in sound detection type tests.

The simplest and fastest way to do a sample rate discrimination test is to remove frequencies below 16 kHz or 20 kHz with a high-pass filter. This method has many advantages over a null test: definitely faster and gives more freedom. We can test sampling rates that normally can't be tested with a null test: 32 kHz vs. 44.1 kHz, 44.1 kHz vs. 96 kHz (null testing 44.1k and 82k is simple, but null testing 44.1k and 96k is a nightmare as it requires intermediate steps: conversion to a common frequency (14.112 MHz), manual sample shifts).

Note that this method assumes that below the cut-off frequency of the test filter there are no audible differences. This assumption is entirely acceptable, since downsampling from, say, 96 kHz to 44.1 kHz and reconstructing the downsampled signal with an anti-image filter is equivalent to applying a low-pass filter at 20 kHz.

Bipolar pulse - audibility of filter ringing...?

The "worst" kind of transient. Sampling rate is 44.1 kHz for this test.

Unfiltered:

High pass filtered at 16 kHz:

High pass filtered at 19.5 kHz:

A high pass filter creates the same "ringing" in a pulse as a low pass filter with the same cut-off frequency and filter length.

Why is pre-ringing in the impulse response not an issue at 44.1 kHz sampling rate?

The pre- and post-ringing in the impulse response can be considered a burst signal (a few periods of sine signal with a smooth fade-in and fade out). The frequency of the ringing is usually between 21 kHz and 22 kHz. So one answer to the question is that we can't hear the ringing in the impulse because we can't hear a short burst signal created from a 20 kHz or 21 kHz sine signal.

A different answer: in order for a resonance or burst signal to be heard, the time-integrated level of the resonance must be higher than the hearing threshold at the resonance frequency (both masked and absolute threshold, but now we can ignore masking). The level of the ringing in the impulse response of a resampling filter is about -50 dBFS (Audacity, Reaper, CoolEdit test; 100 msec integration time). Converting to SPL, this is 55 dBSPL at most - well below the absolute threshold at 20 kHz.

Crash cymbal audio sample (44.1 kHz)

Let's listen to something different: a famous cymbal crash. Sampling rate is 44.1 kHz for this test.

High pass filtered at 14 kHz:

High pass filtered at 16 kHz:

Original sample:
WARNING: LOUD! - Start with low volume.

Notes:

Original sample is attenuated with 6 dB. High-pass filtered versions are not attenuated.
Album: Seiji Ozawa - Strauss: Also sprach Zarathustra, CD Philips No. 400 072-2, Track 1. Also available on the EBU SQAM CD (Sound Quality Assessment Material).

Minimum phase vs. linear phase filter

All filters introduce some delay. In a linear phase filter's passband the delay is constant. In a minimum phase filter's passband the delay is not constant, but changes as a function of frequency. But can we hear the introduced excess delay?

The following sample contains eight pulses ordered in two groups. In the second group the pulses were filtered with a minimum phase low-pass filter. This filter has a cutoff frequency of 20.5 kHz and an attenuation of 50 dB at 22.05 kHz. The low-pass filter was applied at a 96 kHz sampling rate, so the original sample rate is 96 kHz. The audio file was converted from 96 kHz to 44.1 kHz using a linear phase resampling filter.

The point is that the addition of a "steep" minimum phase low-pass filter with a cut-off frequency higher than 20 kHz is not audible. This means that there is no audible difference between a minimum phase reconstruction filter and a linear phase reconstruction filter if low-pass filtering is done right... (And it's hard to do wrong...) (At a cut-off frequency of 20.5 kHz, the introduced excess group delay in the minimum phase version is about 100 µsec. It is too low to be heard. )

Csaba Horváth