In this guide
Sample rate is the number of sample frames per second, measured in Hz. Choose a rate for the recording or delivery requirements; increasing an existing file's rate cannot recreate sound the source did not capture.
Audio sample rate is the number of sample frames used to represent one second of digital audio. It is measured in hertz (Hz): 44,100 Hz means 44,100 frames per second, also written as 44.1 kHz. Each frame contains one sample for each channel, so a stereo recording at 48 kHz has 48,000 frames per second, with a left and a right sample in each frame.
Sample rate is one part of a digital recording's format, alongside channel count and bit depth. Audacity's sample-rate guide explains the units and common rates. The setting does not tell you whether a recording is noisy, clipped or previously compressed.
What does sample rate affect?
In ideal sampling theory, a rate sets an upper frequency limit at half the rate, known as the Nyquist frequency. Real systems also need filtering and a transition band below that limit. A higher rate allows a higher theoretical frequency range, but microphones, converters, filters, the source and the listener all affect what is actually captured or heard. A larger number alone does not guarantee a better recording.
For uncompressed PCM, sample rate also affects data size when duration, bit depth and channel count stay fixed. For example, one minute of stereo 16-bit PCM at 48 kHz needs 48,000 × 2 channels × 16 bits × 60 seconds ÷ 8 = 11,520,000 bytes of audio payload, before headers or metadata. Audacity's digital audio fundamentals describes this calculation. Use the audio file size calculator for other settings.
Sample rate vs. bit depth vs. bitrate
| Setting | What it describes | Example |
|---|---|---|
| Sample rate | Sample frames per second. | 48 kHz means 48,000 frames per second. |
| Bit depth | Bits used per sample in a fixed-point PCM representation. | 16-bit PCM stores each channel's sample in 16 bits. |
| Bitrate | Amount of encoded data per second, commonly expressed in kbps. | A 192 kbps MP3 targets a data rate, not 192,000 samples per second. |
These numbers describe different things. For uncompressed PCM, rate, bit depth and channels determine the payload rate; for compressed audio, the codec and its settings matter too. See audio bitrate explained for compressed file examples.
Which sample rate should I use?
Start with the requirement of the project, device or recipient. CD audio uses 44.1 kHz, and 48 kHz is common in video workflows. A project or delivery may specify a different rate, bit depth or channel layout; follow that specification rather than converting simply to make the number larger.
If no change is required and your editor accepts the source, retain the original. For a practical choice between common rates and an actual listening demonstration, see 44.1 kHz vs. 48 kHz. Higher rates can increase PCM storage and processing without guaranteeing an audible improvement.
How do I check a file's sample rate?
- Open the original file in a compatible media inspector or editor and look for sample rate in its file properties. Check that you are reading the file's properties rather than only a project or playback-device preference.
- For a supported WAV, the WAV file inspector reads the stored rate, encoding, bit depth and channel count. The WAV inspection guide explains what those fields do and do not prove.
- After exporting, inspect the downloaded file again and listen in the destination project. A header reports a format setting; it cannot establish source quality or recover missing detail.
What is resampling?
Resampling changes the samples used to represent audio at a new rate. A one-second 44.1 kHz recording has 44,100 frames; a properly resampled one-second 48 kHz copy has approximately 48,000. Merely relabeling the original frames as 48 kHz makes them play faster and higher in pitch. Actual frame counts can also be affected by rounding and compressed-codec delay.
For a short supported recording, the sample rate converter accepts one mono or stereo MP3 or WAV up to five minutes and 50 MiB, and prepares 16-bit PCM WAV at 44.1 or 48 kHz locally in the browser. The output must also fit 50 MiB. It does not preserve tags or artwork; resampling and PCM rounding alter samples, and values outside PCM16 range clip. Use a compatible editor for longer files, other rates or 24-bit delivery.
Keep the source, preview the actual output, inspect the downloaded WAV and check timing in your destination. Converting an MP3 to WAV does not undo earlier lossy compression; raising the rate cannot recreate frequencies missing from the original.
Common questions
Is a higher sample rate always better?
No. It permits a higher theoretical frequency range, but a recording's actual quality depends on the whole capture and delivery chain. Higher PCM rates also use more space and may be unnecessary for the intended destination.
Is 48 kHz stereo really 96 kHz?
No. Stereo has two samples per frame, but its sample rate is still 48,000 frames per second. Channel count is a separate property.
Does upsampling improve an old recording?
Upsampling can make a file compatible with a required project rate, but it cannot recreate audio information absent from the source or remove existing noise and distortion.
Does changing sample rate change pitch?
Proper resampling aims to retain duration and pitch. Simply relabeling the same samples with a different playback rate changes both speed and pitch.