Audio Sample Rate, Bit Depth & File Size Calculator

Calculate raw PCM, WAV, AIFF, FLAC, and MP3 file sizes from sample rate, bit depth, and channels. Analyze Nyquist frequency, dynamic range, and studio multitrack requirements.

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Samples captured per second per audio channel.
Amplitude quantization resolution per sample.
Discrete audio tracks recorded or mixed.
Hours
Minutes
Seconds
Output encoding format for storage estimation.
Set > 1 to size an entire recording session (e.g., 24-track drum/vocal session).

Audio Signal & Storage Sizing Breakdown

Output File Size 35.33 MB 33.69 MiB (Uncompressed WAV)
Audio Bitrate 1,411.2 kbps 176.4 KB/sec transfer
Nyquist Frequency 22,050 Hz Max audible frequency captured
Theoretical Dynamic Range 96.3 dB 65,536 quantization levels

File Size Comparison Across Common Encodings

Format / Codec Data Rate Resulting File Size Compression Savings

DAW Multitrack Session Projections (Current Sample Rate / Bit Depth)

Session Scale 1 Minute Tracking 1 Hour Tracking 4 Hour Studio Session
1 Track (Mono) 5.04 MB 302.7 MB 1.21 GB
8 Tracks (Drum Kit / Band) 40.3 MB 2.42 GB 9.69 GB
24 Tracks (Standard Production) 121.0 MB 7.26 GB 29.06 GB
48 Tracks (Orchestral / Film Score) 241.9 MB 14.53 GB 58.12 GB
Quick Overview & Direct Answer

Audio Sample Rate, Bit Depth & File Size Calculator: Free Audio Sample Rate, Bit Depth & File Size Calculator tool: calculate uncompressed PCM, WAV, AIFF, FLAC, and MP3 audio file size from sample rate, bit.

Primary Use Audio format conversion, waveform inspection, sound trimming
Input Formats Audio files (MP3, WAV, AAC, OGG)
Output Format Converted audio files and playback waveform data
Processing Mode In-browser Web Audio API & localized processing
Access Tier Free, unlimited, no sign-up needed

The Mathematics of Digital Audio: Sampling and Quantization

In acoustic engineering and digital signal processing (DSP), converting continuous analog sound waves into binary digital representations requires two discrete processes: sampling (discretizing time) and quantization (discretizing amplitude).

The sampling rate determines the highest acoustic frequency that can be represented, governed by the Nyquist-Shannon Sampling Theorem. The bit depth determines the precision of each sample measurement, dictating the dynamic range (the ratio between the loudest undistorted signal and the quietest noise floor) and signal-to-quantization-noise ratio (SQNR).

Mathematical Formulas for Bitrate, Dynamic Range & Storage

Uncompressed Bitrate (bps) = $text{Sample Rate (Hz)} times text{Bit Depth (bits)} times text{Channels}$

Nyquist Frequency ($f_{text{Nyquist}}$) = $frac{text{Sample Rate (Hz)}}{2}$

Theoretical Dynamic Range (dB) = $(6.02 times text{Bit Depth}) + 1.76text{ dB}$

File Size (Bytes) = $frac{text{Bitrate (bps)} times text{Duration (seconds)}}{8}$

File Size (MB) = $frac{text{Bitrate (bps)} times text{Duration (s)}}{8 times 1,024 times 1,024}$

Step-by-Step Worked Numerical Example

Consider a sound engineer recording a 4-minute acoustic track in a modern Digital Audio Workstation (DAW) at professional cinema quality:

  • Sample Rate: 48,000 Hz (48 kHz)
  • Bit Depth: 24-bit Integer
  • Channels: 2 (Stereo)
  • Duration: 4 minutes = 240 seconds

Step 1: Compute the uncompressed bit stream rate:

$text{Bitrate} = 48,000text{ samples/s} times 24text{ bits/sample} times 2text{ channels} = mathbf{2,304,000text{ bits per second}}; (2,304text{ kbps})$

Step 2: Determine Nyquist limit and dynamic range:

$f_{text{Nyquist}} = frac{48,000}{2} = mathbf{24,000text{ Hz}}; (24text{ kHz})$ — comfortably above the human hearing limit of ~20 kHz.

$text{Dynamic Range} = (6.02 times 24) + 1.76 = mathbf{146.24text{ dB}}$

Step 3: Calculate the uncompressed stereo WAV file size:

$text{Bytes} = frac{2,304,000text{ bps} times 240text{ s}}{8} = 69,120,000text{ bytes}$

$text{File Size (MB)} = frac{69,120,000}{1,048,576} approx mathbf{65.92text{ MB}}; (69.12text{ MB decimal})$

If converted to lossless FLAC (averaging 60% ratio), the file compresses to approximately 39.55 MB without discarding a single bit of audio information.

Why 32-Bit Floating Point is Transforming Recording

Traditional 16-bit and 24-bit digital formats use signed two's-complement integers. If a loud transient (such as an unexpected drum rimshot or vocal scream) exceeds $0text{ dBFS}$ (Full Scale), the digital converter runs out of bits and truncates the waveform into a harsh horizontal flatline. This produces irreparable clipping distortion.

Modern professional field recorders (such as Sound Devices and Zoom F-series) and DAWs record in 32-bit floating point. Under IEEE 754 float representation:

  • 1 bit is allocated to the sign ($pm$)
  • 8 bits represent the exponent ($2^E$, providing scale)
  • 23 bits represent the fractional mantissa (providing resolution)

This architecture delivers an astronomical 1,528 dB of dynamic range. In practice, a recording that appears severely clipped and distorted at $+30text{ dBFS}$ can simply have its track fader turned down by $30text{ dB}$ in post-production to recover the completely pristine, unclipped analog waveform.

Audio Standards & Delivery Reference Matrix

Standard / Application Sample Rate Bit Depth Stereo Bitrate Storage / Minute
VoIP / Whisper Speech Recognition 16 kHz 16-bit 512 kbps 3.66 MB/min
Red Book Audio CD 44.1 kHz 16-bit 1,411.2 kbps 10.09 MB/min
Broadcast Video / Cinema (AES3) 48 kHz 24-bit 2,304 kbps 16.48 MB/min
Pro Studio High-Res Master 96 kHz 24-bit 4,608 kbps 32.96 MB/min
Ultra High-End Audiophile Master 192 kHz 24-bit 9,216 kbps 65.92 MB/min
DAW Float Mixing Master 48 kHz 32-bit Float 3,072 kbps 21.97 MB/min

Frequently Asked Questions

The Nyquist-Shannon theorem states that to faithfully record and reconstruct an analog audio signal without aliasing distortion, the sampling rate must be strictly greater than twice the highest frequency present in the signal (f_sample > 2 * f_max). Because human hearing tops out around 20 kHz, a minimum sampling rate of 40 kHz is required. CD Audio chose 44.1 kHz to allow steep anti-aliasing low-pass reconstruction filters between 20 kHz and 22.05 kHz.

Fixed-point audio (16-bit or 24-bit) has an absolute ceiling at 0 dBFS; any signal exceeding this value is instantaneously squared off, causing harsh harmonic distortion. In contrast, 32-bit floating point audio reserves 1 sign bit, 8 exponent bits, and 23 mantissa bits, delivering over 1,528 dB of dynamic range. A 32-bit float signal can exceed 0 dBFS by over +770 dB without clipping; normal gain staging can bring it back into nominal range with zero fidelity loss.

Red Book Audio CD uses a 44,100 Hz sampling rate, 16-bit resolution, and 2 channels (stereo). Data rate = 44,100 samples/sec × 16 bits/sample × 2 channels = 1,411,200 bits per second (1,411.2 kbps). One minute of uncompressed CD audio consumes roughly 10.09 MB of disk space.

44.1 kHz originated from early digital audio recordings stored on NTSC and PAL video tape cassettes (Sony PCM-F1). When digital audio for professional video was standardized in the 1980s, 48 kHz was chosen because it divides evenly into standard video frame rates: exactly 2,000 samples per frame at 24fps, 1,600 samples at 30fps, and 1,920 samples at 25fps. This prevents sample drift and frame synchronization jitter.