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Sample Rate Converter - free online audio resampler & sample rate changer

Change the sample rate of any audio file - WAV, MP3, FLAC, M4A, AAC, OPUS, AIFF or OGG - to any standard rate (44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 192 kHz). Use this online resampler for quick jobs (FFmpeg server-side), or download the free Windows app for studio-grade soxr VHQ resampling.

Audio resampler Sample rate changer Convert 44.1 to 48 kHz Up to 192 kHz 16 / 24 / 32-bit output MP3 / WAV / FLAC / M4A / AAC / OPUS input
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FFmpeg server

Click to select an audio file or drag it here

WAV, FLAC, AIFF, MP3, M4A, AAC, OPUS, OGG - any sample rate, any bit depth, output is always WAV

Source rate
-
select a file above
Target rate
ffmpeg command ffmpeg -i input.wav -ar 48000 -c:a pcm_s16le output_48000hz.wav

Your file is uploaded over HTTPS, resampled on our server with FFmpeg, streamed back and deleted immediately.

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How it works

Resample audio in three steps

01

Select your audio file

Drop any audio file - WAV, FLAC, AIFF, MP3, M4A, AAC, OPUS or OGG - up to 50 MB. The sample rate converter reads the file header and shows your source sample rate, channels, bit depth and duration instantly. The Windows desktop app accepts the same formats plus WMA, APE and WavPack, with no file size limit.

02

Choose target rate and bit depth

Pick your target sample rate - 48,000 Hz is the standard for video and professional audio production. 44,100 Hz is the CD standard. Choose 16-bit for compatibility, 24-bit for studio work, or 32-bit float for further processing in a DAW.

03

Download resampled WAV

Click Resample. The file is uploaded over HTTPS, processed with FFmpeg on our server, and immediately streamed back to you. Your file is deleted from the server as soon as the download starts. For offline, higher-quality resampling use the Windows app.

Sample rate guide

Which sample rate should you use?

44,100 Hz
CD standard

The standard for music distribution, streaming platforms (Spotify, Apple Music, Tidal) and general audio. Most DAWs default to this. Use this for music projects with no video component.

48,000 Hz
Video and broadcast

The standard for video production. Required by most video editing software (Premiere Pro, DaVinci Resolve, Final Cut), broadcast delivery specs, and professional audio interfaces. If your audio goes with video, use 48 kHz.

96,000 Hz
High-resolution audio

Used in studio recording and high-resolution audio releases. Captures more headroom above 20 kHz for processing flexibility. Downsampled to 44.1 or 48 kHz for final delivery. Files are roughly twice the size of 48 kHz.

Common conversions

Convert between standard sample rates

The most common sample rate conversions, with the reason you'd run each one. Pick a target rate above and drop your file in - any source format is fine.

Convert 44.1 kHz to 48 kHz

Music produced at the CD standard (44.1 kHz) needs to move to 48 kHz before it can be embedded in video, sent to a video editor, or delivered to broadcast. This is the single most common sample rate conversion in production work.

Convert 48 kHz to 44.1 kHz

Field-recorded or video-project audio at 48 kHz needs to come down to 44.1 kHz before delivery to Spotify, Apple Music, Tidal, Bandcamp or CD pressing. Use the highest quality available for this downsample - non-integer ratios benefit most from soxr VHQ.

Convert anything to 96 kHz

Upsampling to 96 kHz is common when archiving, when running aggressive plugin chains that benefit from extra headroom above 20 kHz, or when distributing as high-resolution audio. Upsampling does not add information, but it does help downstream processing.

Convert to 22.05 kHz or 11.025 kHz

Legacy formats (older games, low-bandwidth voice, retro-style chiptune work) sometimes need to go below the CD rate. Both rates are integer divisors of 44.1 kHz so the conversion is mathematically clean.

Convert to 176.4 or 192 kHz

Studio-grade high-resolution audio rates. 176.4 is a clean integer multiple of 44.1, 192 is a clean integer multiple of 48 - pick the one that matches your original source family to avoid one extra non-integer resample.

Change MP3 sample rate (or M4A, AAC, OPUS)

Drop a compressed audio file into the converter and it will be decoded with FFmpeg, resampled to your target rate, and returned as a WAV. For sample rate work this is the right approach - resampling compressed audio in place would require a second lossy round-trip.

Windows desktop app

X Sample Rate Converter for Windows

X Sample Rate Converter Windows desktop app v1.2.1 - soxr VHQ resampler with drag and drop, recent files, MP3 M4A AAC OPUS input and one-click open output folder
What's new - v1.2.1
  • Recent files - last 10 inputs remembered, one-click reopen
  • Quality tooltips - clear explanations of QQ / LQ / MQ / HQ / VHQ
  • Open output folder - one-click reveal after each conversion
  • Auto-rename on conflict - no more accidental overwrites
  • Remembers output folder + optional completion sound

Previously in v1.1: drag & drop, MP3 / M4A / AAC / OPUS / WMA / APE input via bundled FFmpeg.

The Windows desktop app uses soxr - the SoX resampler library - which is a professional-grade resampling engine in the same quality class as Voxengo r8brain. Open source, MIT licensed, and produces audibly better results than FFmpeg's SWR resampler at high quality settings.

soxr VHQ engine - comparable to r8brain, open source and free
Drag & drop from Windows Explorer - no dialog needed
Input: WAV, FLAC, AIFF, OGG, MP3, M4A, AAC, OPUS, WMA, APE
Auto-detects source sample rate, bit depth, channels and duration
16-bit PCM, 24-bit PCM and 32-bit float output, 8 kHz to 192 kHz
Works fully offline, no file size limits, single portable .exe
Download X Sample Rate Converter Free - Windows 7/8/10/11 - v1.2.1 - .exe

No extra downloads needed

soxr + soundfile + FFmpeg
The soxr resampler, soundfile decoder and FFmpeg (for MP3/M4A/AAC/OPUS input) are all bundled inside the .exe. Single portable file, no installer, no extra downloads required - just run it.
FAQ

Frequently asked questions

What is sample rate conversion and why do I need it?
Sample rate is the number of audio samples captured per second. 44,100 Hz (CD standard) and 48,000 Hz (video standard) are the most common. When your audio file does not match your project's sample rate - for example a 44.1 kHz music track in a 48 kHz video project - the audio will play back at the wrong speed or pitch unless it is resampled. Sample rate conversion (also called resampling, or sample rate changing) recomputes the waveform at the new rate without changing pitch or duration.
How do I convert sample rate to 44.1 kHz?
Drop your file into the converter above, pick 44,100 Hz from the target rate dropdown, choose a bit depth (16-bit PCM is standard for music delivery to streaming or CD), and click Resample. The output is a WAV file at the new rate. 44.1 kHz is the right target when delivering music to Spotify, Apple Music, Tidal, Bandcamp, or for CD pressing. If your source is at 48 kHz (typical for video-project audio), downsampling to 44.1 is a non-integer ratio - use the desktop app's soxr VHQ mode for the cleanest result.
How do I convert sample rate to 48 kHz?
Same workflow: upload your file, set the target rate to 48,000 Hz, pick a bit depth (24-bit if the audio will be processed further in a DAW, 16-bit if it is going straight into a final video edit), and click Resample. 48 kHz is the standard for video production - Premiere Pro, DaVinci Resolve, Final Cut Pro, OBS, broadcast delivery specs and most professional audio interfaces all expect 48 kHz. If you are pulling music in from a streaming source at 44.1 kHz, this is the conversion you want.
Can I change the sample rate of an MP3 online?
Yes. Drop your MP3, M4A, AAC, OPUS or OGG file into the converter and pick a target rate. FFmpeg decodes the compressed audio on the server, resamples to the new rate, and returns a WAV file (always lossless on output). This is the correct approach for sample rate work - resampling an MP3 back into an MP3 would require a second lossy encode, which audibly degrades the audio. If you need to re-encode to MP3 after resampling, use our X Audio Converter as a second step.
Is "sample rate changer" the same as "sample rate converter" or "audio resampler"?
Yes, those terms refer to the same operation. "Sample rate converter" is the most common name, "resampler" is the term used in DSP literature (the algorithm is called resampling), and "sample rate changer" is the same thing in plain English. This tool does all three - it is a sample rate converter, an audio resampler, and a sample rate changer.
What is the difference between the online tool and the Windows app?
The online tool uses FFmpeg's SWR resampler on our server - good quality and convenient for occasional use. The Windows desktop app uses the soxr (SoX resampler) library at Very High Quality (VHQ) mode, which uses a much steeper anti-aliasing filter and produces audibly superior results, especially when downsampling from high sample rates. For professional or archival use, the desktop app is recommended. The desktop app also has no file size limit, drag and drop support, a recent files list, and works offline.
Does upsampling improve audio quality?
No. Upsampling - converting from a lower sample rate to a higher one - does not add any audio information that was not in the original recording. A 44.1 kHz file upsampled to 96 kHz is still a 44.1 kHz recording in terms of frequency content. Upsampling is useful for matching project requirements or for giving your DAW or digital-to-analog converter more headroom when processing, but it does not increase perceived quality.
Is my file kept on the server?
No. Your file is uploaded over HTTPS, resampled with FFmpeg, and the output is streamed directly back to your browser. Both the input and output files are deleted from the server immediately after the download completes. Any orphaned temp files from dropped connections are automatically removed after 4 hours.
Which bit depth should I choose?
16-bit PCM is the standard for final delivery - CD quality, compatible everywhere, and produces files half the size of 24-bit. 24-bit PCM is recommended for studio work, mixing, and mastering since it provides more headroom and reduces quantization noise during processing. 32-bit float is ideal if you plan to do further processing in a DAW, as it cannot clip regardless of gain changes. For straightforward resampling to deliver a file for video or streaming, 16-bit is almost always the right choice.
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