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Sample Rate Converter

Compare audio sample rates — 8 kHz telephone, 44.1 kHz CD, 48 kHz video, 96/192 kHz hi-res, 384 kHz ultra hi-end. Shows Nyquist frequency, aliasing threshold, data rate, and PCM file-size estimates for any duration, bit depth, and channel count.

Input

Selected: 44,100 Hz

Result

Nyquist Frequency (max representable signal)
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SR / 2
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Sample Period
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Data Rate (decimal)
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PCM File Size (uncompressed)
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⚠ Aliasing
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Formulas
Nyquist = Sample Rate / 2
Sample period = 1 / Sample Rate
Bytes/sec = SR × (bit depth / 8) × channels
File size = Bytes/sec × duration (PCM, no compression)
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Common Audio Sample Rates Compared

Sample RateNyquist16-bit Stereo Data Rate1 min file sizeTypical Use
8 kHz4 kHz32 KB/s~1.9 MBLandline telephone
11.025 kHz5.5 kHz44.1 KB/s~2.6 MBLow-quality voice
16 kHz8 kHz64 KB/s~3.8 MBWideband VoIP, AMR-WB
22.05 kHz11.025 kHz88.2 KB/s~5.3 MBLow-fi audio, old games
32 kHz16 kHz128 KB/s~7.7 MBFM radio quality
44.1 kHz22.05 kHz176.4 KB/s~10.6 MBCD Audio standard
48 kHz24 kHz192 KB/s~11.5 MBDVD, broadcast, pro video
88.2 kHz44.1 kHz352.8 KB/s~21.2 MBHi-res audio (2× CD)
96 kHz48 kHz384 KB/s~23.0 MBDVD-Audio, professional
176.4 kHz88.2 kHz705.6 KB/s~42.3 MBHi-res audio (4× CD)
192 kHz96 kHz768 KB/s~46.1 MBBlu-ray, studio mastering
352.8 kHz176.4 kHz1.41 MB/s~84.7 MBDXD mastering format
384 kHz192 kHz1.54 MB/s~92.2 MBUltra hi-end audio

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About Sample Rates, Nyquist & Aliasing

Sample rate is how many times per second an audio signal is measured during analog-to-digital conversion (ADC). This audio sample rate calculator and sample rate converter lets you compare rates side by side, estimate PCM file sizes, and understand the Nyquist implications of any choice. Nyquist frequency is half the sample rate — by the Nyquist-Shannon sampling theorem, that's the highest signal frequency that can be perfectly reconstructed from discrete samples. Any frequency above Nyquist will "fold back" into the audible range as a different, spurious frequency — this artifact is called aliasing. Real-world converters prevent it with a steep low-pass anti-aliasing filter placed before the ADC stage. To understand how a sample period relates to frequency in general, the frequency-to-period converter illustrates the same reciprocal relationship.

Why 44.1 kHz for CDs?

Human hearing tops out around 20 kHz. A 44.1 kHz sample rate provides a 22.05 kHz Nyquist, leaving a ~2 kHz "guard band" for the anti-aliasing filter to roll off before reaching the hearing limit. The specific 44.1 number comes from early Sony PCM adapter designs that recorded audio onto video-tape — three audio samples fit per video line at NTSC (60 Hz × 245 lines × 3 = 44,100).

48 kHz vs 44.1 kHz

48 kHz is the standard for video, broadcast, and most professional audio (AES3, SMPTE). 44.1 kHz is the standard for music CDs and consumer audio downloads. Sample rate conversion (SRC) between them is lossy because the ratio 44100:48000 = 147:160 is irrational in audio terms, so professional workflows pick one and stick to it. Modern DAWs handle the conversion transparently, though quality varies between implementations.

Hi-res audio: 96 kHz / 192 kHz

Sample rates above 48 kHz are mostly used for production headroom (filters, time-stretching, pitch-shifting work better with more samples) and for oversampling in plugin processing to push aliasing artefacts above the audible band. Whether 96 kHz delivery audibly improves over 44.1 kHz remains controversial — most double-blind ABX listening tests don't find a statistically significant difference for well-recorded material. The file sizes triple or quadruple, and DAW CPU load increases proportionally.

File size impact

Uncompressed PCM file size grows linearly with sample rate, bit depth, and channel count: bytes/sec = SR × bits/8 × channels. The PCM bitrate for stereo 16-bit CD audio (44.1 kHz) is 1,411 kbps; hi-res audio at 24-bit / 96 kHz stereo runs to 4,608 kbps, making hi-res audio file sizes nearly 3.5× larger per minute (~34.5 MB vs ~10 MB). Lossless compression (FLAC, ALAC) reduces these by roughly 50–60% without any quality loss; lossy codecs (AAC, Opus, MP3) reduce them further with perceptual coding trade-offs.

Frequently Asked Questions

What is the Nyquist frequency?
Half the sample rate. For a sample rate of SR, signals above SR/2 cannot be accurately represented — they alias to lower frequencies. CD audio (44.1 kHz) has a Nyquist of 22.05 kHz. The Nyquist-Shannon theorem proves that any signal below this limit can be perfectly reconstructed from the samples.
What causes aliasing?
When you try to sample a signal whose frequency exceeds the Nyquist limit, the digital system "folds" it back into the representable range. A 22 kHz signal sampled at 44.1 kHz produces a 50 Hz artifact (44.1 − 22 = 22.1; the fold around Nyquist 22.05 gives 22.05 − (22 − 22.05) = something close to your input minus 2× the overshoot). In practice, anti-aliasing filters (low-pass below Nyquist) prevent this by removing high frequencies before sampling.
Why does CD use 44.1 kHz instead of 48 kHz?
Historical: early digital audio was recorded onto video tape (which was the cheapest reliable high-bandwidth medium in the 1970s). 44.1 kHz was the highest sample rate that fit into NTSC or PAL video frames. The Compact Disc standard adopted this rate in 1982; the format predates dedicated digital audio recorders.
Should I record at 96 kHz or 44.1 kHz?
Most production benefits from 48 or 96 kHz for the extra headroom during processing — filters work cleaner with more samples, and pitch-shifting/time-stretching have more raw material. The final delivery is usually downsampled to 44.1 (music) or 48 (video). For simple acoustic recordings with no heavy processing, 44.1 kHz is enough.
How big is a 3-minute stereo 16-bit CD-quality file?
Data rate = 44,100 × 2 × 2 = 176,400 bytes/s. For 3 minutes (180 seconds): 31,752,000 bytes ≈ 31.75 MB. With FLAC compression, typically 60% of uncompressed = ~19 MB. With AAC at 256 kbps: ~5.8 MB. With MP3 at 128 kbps: ~2.9 MB.
What's DSD or DXD?
DSD (Direct Stream Digital) is a 1-bit format at very high sample rates (2.8224 MHz for DSD64, 5.6448 MHz for DSD128). DXD (Digital eXtreme Definition) is 24-bit PCM at 352.8 kHz, commonly used for editing/mastering DSD recordings. Both are archival/audiophile formats — file sizes are enormous (DXD ~85 MB/min stereo, DSD64 ~33 MB/min).
Does bit depth affect sample rate, and which matters more for quality?
They control different dimensions of quality. Sample rate determines the highest frequency that can be represented (Nyquist = SR/2). Bit depth determines the dynamic range: 16-bit gives ~96 dB, 24-bit gives ~144 dB theoretical dynamic range. For music delivery, 16-bit / 44.1 kHz already exceeds the limits of human hearing and typical playback equipment. For recording and mixing, 24-bit is strongly preferred over 16-bit because headroom prevents clipping during loud transients — the extra sample rate matters far less than the extra bit depth during production.
What sample rate should I use for my DAW project?
Match your delivery format: 44.1 kHz for music releases (Spotify, Apple Music, CD), 48 kHz for video and broadcast (YouTube, film). Using a higher rate than your delivery target (e.g., 96 kHz for a 44.1 kHz release) adds CPU load and storage cost with marginal benefit for most genres. If you use plugins that alias at standard rates (some older saturation or synth plugins), 88.2 or 96 kHz can reduce those artefacts — check the plugin documentation. Keep the rate consistent throughout the project to avoid unnecessary sample rate conversion.
What is oversampling in audio plugins, and how does it relate to sample rate?
Oversampling is when a plugin internally upsamples the audio (e.g., to 4× or 8× the project rate), applies non-linear processing (saturation, limiting, distortion), then downsamples back. This pushes aliasing artefacts generated by the non-linearity above the Nyquist of the output rate, where they are removed by the downsample filter. A project running at 44.1 kHz with a 4× oversampling plugin effectively processes at 176.4 kHz internally. The calculator's data rate figures show why this increases plugin CPU cost proportionally to the oversampling factor.

Related Tools

Frequency to Period Converter
Hz to time period — useful for converting sample rate to sample interval
Hz to Note Converter
Find the musical note at a Nyquist frequency or sample rate
Period to Frequency Converter
Time period to frequency — inverse of the Hz/period relationship
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