Audio Bench

No, the tool re-encodes the segment with FFmpeg WASM but maintains the original format's bitrate as closely as possible.

Yes, as long as your browser has enough memory, you can load and trim large files.

You can trim MP3, WAV, OGG, AAC, FLAC, and other common audio formats.

We use an advanced FFmpeg WASM filter called rubberband (or similar stretching algorithms) that adjusts frequencies independently of time.

If you shift the pitch significantly higher, yes! Small shifts are better for natural key changes.

While the slider gives a specific range, extreme values will degrade audio quality and create artifacts.

No, the recording happens entirely locally using the browser's MediaRecorder API. Your voice never leaves your device.

It typically saves in a standard web audio format like WebM or OGG, depending on your browser, which you can later convert to MP3.

Ensure you've granted microphone permissions in your browser settings and that the correct input device is selected.

If you increase the volume too much, the audio peaks will hit the maximum limit and clip, causing distortion. Increase in small increments.

This tool applies a static volume change to the entire file. For uneven audio, a compressor or normalizer is better.

FFmpeg WASM applies a volume filter multiplier directly to the audio stream.

An equalizer cuts or boosts the volume of specific frequency ranges (like bass, mids, or treble) without affecting the rest of the sound.

Yes, we provide several common presets like Bass Boost, Treble Boost, and Voice Enhancement.

This tool acts as a graphic equalizer, using FFmpeg WASM's multi-band EQ filter.

Our FFmpeg WASM engine handles different formats and sample rates, converting them to match during the joining process.

No, the files are appended back-to-back with zero gap. If you want a gap, you'll need to insert a silent audio file.

You can join as many as you want, but processing time will increase with total duration and file count.

No, this tool uses a time-stretching algorithm in FFmpeg WASM (atempo filter) to change the duration while keeping the original pitch intact.

Yes, you can select a 2.0x speed multiplier to double the tempo.

Extreme slow-downs can introduce stuttering artifacts because the algorithm has to invent audio data to fill the stretched time.

FFmpeg WASM loads the entire audio stream into memory and reverses the sequence of the audio samples.

Yes, but reversing requires loading the entire file into memory at once, so very large files might take longer or hit browser memory limits.

No, the audio samples are simply played in reverse order, preserving the original fidelity.

It depends on the original bitrate and your target. Converting a 320kbps MP3 to 128kbps will reduce the file size by roughly 60%.

Yes, compression is a 'lossy' process. However, 128kbps is generally considered near CD-quality for casual listening.

Yes! Converting a large uncompressed WAV file to a compressed MP3 or AAC will result in massive space savings.

This tool uses phase cancellation (inverting one stereo channel and summing it with the other). If the vocals have stereo reverb or aren't perfectly centered, remnants will remain.

No, this technique relies entirely on the differences between the left and right stereo channels. It will silence a mono track completely.

No, this uses traditional FFmpeg WASM phase cancellation techniques. It is fast and runs locally, but isn't as perfect as deep learning source separation.

No. This tool isolates the frequency range where vocals typically live. You will still hear instruments that share those same frequencies (like snare drums and guitars).

It uses FFmpeg WASM to apply bandpass filters, cutting away extreme highs (cymbals) and deep lows (bass) to leave the vocal range intact.

Yes, isolating vocal frequencies can sometimes improve the accuracy of transcription software by reducing low-end rumble.

No, this tool uses FFT-based noise reduction which is designed to remove constant, steady-state background noise (like air conditioning hum or tape hiss).

Applying heavy noise reduction can cause 'watery' or robotic artifacts. It's often better to leave a tiny bit of noise than to over-process the audio.

Yes, the FFmpeg WASM afftdn (audio fast Fourier transform denoise) filter runs entirely in your browser.

Yes, simply set the fade-in duration to 0 seconds and only specify a fade-out duration.

No, it merely applies a mathematical volume envelope to the specified sections via FFmpeg WASM.

By default, this tool applies a linear fade, which provides a steady, predictable decrease or increase in volume.

Volume Change blindly multiplies the volume by a set amount. Normalize scans the file first to find the absolute loudest part, and then turns the volume up exactly as much as possible without distorting.

No, peak normalization turns the entire song up by the same amount. To fix dynamic range (quiet and loud parts), you need a compressor.

The tool normalizes the peak volume to 0dBFS (the maximum digital limit) or slightly below it.

It removes ID3 tags (Artist, Album, Title), embedded album artwork, copyright info, and software encoder tags.

Not at all. The actual audio stream is simply copied over into a new file without the metadata header using FFmpeg WASM.

Slightly. If the file had a large embedded high-resolution album cover, removing it will noticeably shrink the file size.

No, it utilizes a client-side recognition process. While more limited than massive cloud AI, it ensures complete privacy.

Currently, the primary focus is on English transcription, depending on the browser's built-in Web Speech API or local fallback engines.

Client-side transcription works best with clear, slow speech and minimal background noise. Accents and overlapping talkers can reduce accuracy.

This specific tool focuses on text-based ID3 metadata fields like Title and Artist. Cover art injection is not currently supported in this module.

WAV files don't handle metadata as robustly as MP3s. This tool is optimized specifically for writing ID3 tags to MP3 files.

No, the FFmpeg WASM engine strictly updates the header tags and copies the original audio stream, preserving 100% of the original quality.