Free Audio Sample Rate Converter

What is sample rate?

Sample rate is how many times per second an audio signal is measured when it's digitized. 44.1kHz means 44,100 samples per second, 48kHz means 48,000, and 96kHz means 96,000. A higher number means the signal is measured more frequently — it doesn't by itself mean the recording sounds better, only that it's represented with more data points per second.

44.1kHz vs. 48kHz

44.1kHz is the long-standing standard for music — it's the CD format, and most sample libraries, DAWs, and music distribution pipelines default to it. 48kHz is the standard for video and broadcast audio instead. Neither is simply "better" than the other; they're different conventions for different destinations. The right choice comes down to what your project or platform actually requires — matching a video editor's timeline calls for 48kHz, while a music-focused project usually calls for 44.1kHz.

What is 96kHz used for?

96kHz shows up in some high-resolution audio releases and production workflows, where the extra samples per second can give more headroom during intensive processing before that processing starts introducing artifacts. Converting an existing 44.1kHz file up to 96kHz afterward doesn't retroactively give it that benefit — the extra headroom only matters when the original recording and processing chain were actually done at 96kHz from the start.

Does converting to a higher sample rate improve quality?

No. Upsampling doesn't restore detail that wasn't captured in the original recording — it represents the same underlying information with more samples, rather than pulling in new information that was never there. If a recording was made at 44.1kHz, that rate set the ceiling on what was captured; changing the file's stored rate afterward doesn't move that ceiling.

What is bit depth?

Bit depth is a separate setting from sample rate — it controls how finely each individual sample's amplitude is measured. 16-bit is the CD standard; 24-bit and 32-bit are common in production for the extra headroom they give during mixing and processing. Bit depth conversion here only applies to uncompressed WAV and AIFF files, since compressed formats like MP3 or AAC don't store audio as raw PCM samples and so don't expose a user-facing bit depth to convert.

Sample rate vs. bit depth

They're independent settings that happen to get adjusted together in a lot of audio software, which is why they're easy to conflate — but changing one doesn't change the other.

When should you convert to each rate?

How to convert sample rate

1. Upload an MP3, WAV, FLAC, M4A, AAC, OGG, or AIFF file.
2. Choose a target sample rate, and optionally a bit depth for WAV/AIFF.
3. Download the converted file.

Common uses

Matching a video editor's expected 48kHz audio, converting samples down to 44.1kHz for a music project, preparing audio for a platform with a specific sample-rate requirement, and standardizing a batch of files recorded at inconsistent rates before combining them.

Want the fuller breakdown of why upsampling and downsampling behave so differently, and what bit depth is actually doing under the hood? Read Sample Rate and Bit Depth: What They Actually Change. Combining files afterward? The Audio Joiner already normalizes sample rate automatically when it merges files, but resampling first is useful if you need a specific rate for a reason beyond just joining.

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