WAV Bit Depth & Sample Rate
Explained: Which Settings to Use

Which WAV Bit Depth Should You Use?

Short answer: 16-bit for finished files, 24-bit for recording and editing, 32-bit float for files that will be processed further and may go above 0 dBFS. Bit depth sets the noise floor, not the ceiling: every integer WAV clips at 0 dBFS.

What is Sample Rate?

Sample rate (measured in Hz or kHz) defines how many times per second the audio waveform is captured. Think of it like frames in a video — more samples per second means a more accurate representation of the original sound.

According to the Nyquist-Shannon theorem, a sample rate must be at least twice the highest frequency you want to capture. Human hearing tops out at roughly 20 kHz, so 44.1 kHz (used by CDs) captures everything we can hear.

What is Bit Depth?

Bit depth determines the number of possible amplitude values for each audio sample. Higher bit depth means a larger dynamic range — the difference between the quietest and loudest sound the file can represent without distortion.

The formula is simple: dynamic range ≈ 6 dB × bit depth. So 16-bit audio has ~96 dB of dynamic range, 24-bit has ~144 dB, and 32-bit float has over 1500 dB (effectively unlimited).

44.1 kHz vs 48 kHz: Which Sample Rate?

This is the most common question when choosing WAV settings, and the answer depends entirely on your destination format.

Bottom line: Use 44.1 kHz for music-only projects. Use 48 kHz if the audio will be synced with video. Avoid mixing the two in one project — sample rate conversion introduces tiny (though usually inaudible) artifacts. CleverUtils uses the SoXr audiophile-grade resampler to minimize these artifacts.

16-bit vs 24-bit: Which Bit Depth?

For playback and distribution, 16-bit is all you need. For recording and editing, 24-bit gives you a much lower noise floor.

Bottom line: If you're converting audio for listening, sharing, or burning CDs — 16-bit is perfect. If you're editing in a DAW, recording live audio, or running effects chains — choose 24-bit to keep the noise floor low, or 32-bit float if levels may go above 0 dBFS.

Stereo vs Mono: When to Use Each

Stereo (2 channels) is the standard for music and most audio. Mono (1 channel) cuts file size in half and is ideal for voice-only content.

WAV File Size Reference

WAV files are uncompressed, so size is exactly predictable. The formula is:

Here is how common settings compare for a 3-minute file (1 MB = 1,000,000 bytes):

For comparison, the same 3-minute song as MP3 (VBR V2) would be about 4.3 MB — about 7x smaller than CD-quality WAV.

Which Settings Should You Use?

Here's a quick-reference guide based on what you're doing with the audio:

Converting MP3 to WAV Doesn't Improve Quality

This is the most common misconception. When you convert a 128 kbps MP3 to a 44.1 kHz / 24-bit WAV, the file gets much larger but the audio quality stays exactly the same.

MP3 compression permanently removes audio data. Converting to WAV unpacks the remaining data into an uncompressed container, but it cannot restore what was already discarded. Think of it like unzipping a photo that was already resized — you get more pixels, but they're interpolated, not recovered.

So why convert at all? Because WAV is a better working format:

• Audio editors (Audacity, Pro Tools, Logic) work natively with WAV — no decoding overhead
• Re-saving a WAV doesn't degrade quality (unlike re-encoding MP3)
• CD burning requires PCM/WAV input
• Some hardware and broadcast systems only accept uncompressed audio

If you need the highest possible quality, always start from the original lossless source (CD, FLAC, studio master) rather than a lossy MP3.

WAV Bitrate: Sample Rate × Bit Depth × Channels

WAV has no bitrate setting of its own. Its bitrate follows from the format: for example 44,100 × 16 × 2 = 1,411,200 bits per second.

How to Change WAV Bit Depth

• CleverUtils: in the converter above, pick 16-bit, 24-bit or 32-bit float, plus 22.05, 44.1, 48 or 96 kHz and stereo or mono. When bits are reduced, Shibata dither is added.
• Audacity: File → Export Audio, choose WAV, then set Encoding to Signed 16-bit PCM, Signed 24-bit PCM or 32-bit float.
• FFmpeg: -c:a pcm_s16le, pcm_s24le or pcm_f32le, for example ffmpeg -i input.mp3 -c:a pcm_s24le output.wav.