Estimate audio file size in MB / GB from sample rate, bit depth, channels and duration. Compare WAV, AIFF, FLAC and lossy formats.
Built and maintained by Paul Clark, Redmoon Software · Checked against what this tool computes on
| Format | Compression | File size | Bitrate (kbps) |
|---|
Uncompressed PCM size = sample rate × bit depth × channels × duration / 8 bytes. FLAC averages ~55% of PCM (varies with material); MP3 320 and AAC 256 are constant-bitrate estimates.
Bitrate = sample rate × bit depth × channels, in bits per second.
Uncompressed audio size is fully determined: sample rate × bit depth × channels × duration ÷ 8 gives bytes, and sample rate × bit depth × channels ÷ 1000 gives the bitrate in kbps. The defaults of 48 kHz, 24-bit, stereo and 3:30 therefore produce 2304 kbps and 57.7 MB. Duration is parsed flexibly, accepting mm:ss, h:mm:ss or a plain number of seconds. The comparison table then estimates the same recording as FLAC, ALAC, MP3 320, AAC 256 and Opus 128.
All four inputs multiply together, so each one doubles or halves the file in a straight line. Sample rate and channels move the number fastest: going from the 48 kHz, 24-bit stereo default up to 192 kHz, 32-bit float and 7.1 surround makes the file roughly twenty-one times larger. Choose 44.1 kHz for CD-targeted music, 48 kHz for anything paired with video, and 24-bit over 16-bit for recording headroom rather than for any audible resolution.
The lossless rows are blunt averages — FLAC is assumed to land at 55 % of PCM and ALAC at 60 % — when real compression ranges from about a third on sparse acoustic material to nearly the full size on a loud, dense master. The lossy rows are pure bitrate arithmetic and ignore container overhead and variable-rate encoding. The PCM figure also excludes headers and embedded metadata, and it will not warn you that a classic WAV file cannot exceed 4 GB.
Around 30.3 MB, at a bitrate of 1411 kbps. That is the CD standard, and it works out at roughly 10 MB per minute of stereo audio. Enter 3:00 in the duration field to check it, or 180 as a plain number of seconds.
Exactly, at the same sample rate and channel count, since size is directly proportional to bit depth. Against 24-bit it is a third larger. The benefit of 32-bit float is recording headroom that survives clipping, not extra audible detail.
This calculator assumes 55 % of the PCM size, so roughly 45 % saved, with ALAC slightly behind at 60 %. Real results depend entirely on the material: quiet, sparse recordings compress much further, while heavily limited masters may barely shrink at all.
2304 kbps, from 48000 × 24 × 2 divided by 1000. That is over seven times the 320 kbps of a maximum-quality MP3, which is a useful reminder of how much data lossy formats discard to reach their target size.
Three formats are accepted. Use mm:ss for a song, such as 3:30, use h:mm:ss for a long recording, or type a plain number of seconds. The result line echoes the duration it parsed in seconds so you can confirm it read you correctly.
A sample rate captures frequencies up to half itself — the Nyquist limit — so 44.1 kHz covers up to about 22 kHz, comfortably past the roughly 20 kHz ceiling of human hearing, with margin for the anti-aliasing filter to roll off without affecting the audible band.
48 kHz is the video and broadcast standard for historical reasons rather than audible ones, which is why the two coexist. The higher options — 88.2, 96 and 192 kHz — do not capture anything audible that 44.1 misses.
The argument for high rates is in processing rather than in capture: some nonlinear plugins alias less at higher rates, and time-stretching or heavy pitch-shifting has more material to work with. Those are real and narrow benefits, paid for with proportionally larger files and higher CPU load.
The practical trap is mixing rates in one project. Sample-rate conversion is not free — it is a resampling operation with its own artefacts — so a session that mixes 44.1 and 48 kHz material is doing conversion somewhere, often silently. Pick the rate your delivery format needs and record everything at it.
Bit depth is the separate axis and is more consequential for recording: 24-bit gives headroom to record conservatively without approaching the noise floor, which is the practical benefit people often attribute to sample rate.
The standard or formula behind this tool, so you can check it rather than take it on trust.
Nyquist–Shannon sampling theorem.
A sample rate captures frequencies up to half itself, which is why 44.1 kHz covers the ~20 kHz limit of human hearing with margin for the anti-aliasing filter.
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