Free Online Audio Toolkit — Trim, Compress & Convert
Edit your audio files directly in your browser — no account, no upload, no waiting. Alfreto's Audio Toolkit uses FFmpeg.wasm to trim clips with a visual waveform editor, compress file size by targeting a lower bitrate, and convert between popular formats including MP3, WAV, AAC, OGG, FLAC, OPUS, and M4A. Everything stays on your device. Nothing is ever sent to a server.
Ready.
How to Use the Audio Toolkit
Alfreto's Audio Toolkit is a free, browser-based tool that handles three common audio tasks in one place: trimming, compressing, and converting. It runs entirely on your device using FFmpeg.wasm — a full version of the professional FFmpeg engine compiled for the browser — so your files are never uploaded anywhere.
Step 1 — Load Your Audio File
Click Choose Audio or drag and drop a file into the upload area. Supported input formats are MP3, WAV, AAC, OGG, FLAC, OPUS, and M4A. Once loaded, the waveform appears in the timeline, and the file's duration and size are shown in the stats panel below.
Step 2 — Choose an Operation
Select one of the three tabs based on what you need to do:
- ✂️ Trim — Cut your audio to a specific time range. Drag the blue handles on the waveform to set start and end points visually, or type exact timestamps in the fields below (format: hh:mm:ss). Use the Play button or press Spacebar to preview the selection before processing.
- 📦 Compress — Reduce the file size by lowering the audio bitrate. Use the slider to select a target bitrate (32–320 kbps). You can also switch from stereo to mono to cut the file size roughly in half. The output keeps the same format as the input.
- 🔄 Convert — Change the audio format. Select an output format card (MP3, AAC, WAV, OGG, FLAC, or OPUS) and choose a bitrate for lossy formats. WAV and FLAC ignore the bitrate setting since they are lossless.
Step 3 — Process and Download
Click Process when your settings are ready. On first use, the browser downloads the FFmpeg engine (~10 MB) from a CDN — a one-time step that is cached for future use. A progress bar shows real-time processing status. When finished, the output audio appears in the preview panel. Compare it with the source, then click Download to save the result to your device.
Understanding Audio Formats
Choosing the right output format depends on your use case:
- MP3 — The most universally compatible format. Works on every device, platform, and media player. A good default for music and podcasts.
- AAC — Better quality than MP3 at the same bitrate. Preferred on Apple devices and widely supported on Android and web browsers.
- WAV — Uncompressed lossless audio. Ideal for professional editing or archiving, but produces large file sizes.
- FLAC — Lossless with compression. Preserves full quality while producing smaller files than WAV. Great for archiving music.
- OGG (Vorbis) — Open-source format with good quality at low bitrates. Common in web games and Linux environments.
- OPUS — A modern, highly efficient codec designed for voice and music. Excellent for streaming and messaging apps at very low bitrates.
When Should You Use This Tool?
- Trim a ringtone — Cut a specific part of a song to the exact length you need.
- Reduce a podcast file — Lower the bitrate to compress a large recording before uploading or sharing.
- Convert for compatibility — Change a WAV or FLAC recording to MP3 so it works on every device.
- Extract a clip — Isolate a specific moment from a long recording, interview, or lecture.
- Convert for web — Use OPUS or OGG for web audio that loads fast and sounds good even at low bitrates.
- Process sensitive audio — Since nothing is uploaded, this tool is safe to use with private recordings, voice memos, or confidential interviews.
Frequently Asked Questions
Is my audio file uploaded to a server?
No — never. All audio processing happens locally inside your browser using FFmpeg.wasm, a version of the professional FFmpeg engine compiled to run entirely in JavaScript. Your audio file is read directly from your device's memory, processed there, and the result is written back to your device. At no point does any file leave your computer, phone, or tablet.
Which audio formats are supported?
The toolkit accepts MP3, WAV, AAC, OGG, FLAC, OPUS, and M4A as input. For output, you can export to MP3, AAC, WAV, OGG, FLAC, and OPUS. This covers the vast majority of audio files you are likely to encounter — from music and podcasts to voice recordings and web audio.
How do I trim an audio file?
Select the Trim tab after loading your file. The waveform of your audio will be displayed in the timeline. Drag the blue start and end handles to define the clip you want to keep, or type the exact start and end times in the fields below (format: hh:mm:ss). Press the Play button or hit Spacebar to preview the trimmed selection before processing. The output keeps the same format as the input file.
What bitrate should I use to compress audio?
It depends on the type of content. For music, 128–192 kbps is a balanced range that sounds good on most speakers and earphones. For voice-only recordings like podcasts, interviews, or lectures, 64–96 kbps is perfectly sufficient and keeps file sizes very small. Use 256–320 kbps if you need near-lossless quality for professional or archival purposes. As a general rule: lower bitrate = smaller file = lower quality.
Can I convert MP3 to WAV, or WAV to MP3?
Yes. Open the Convert tab and select your desired output format from the format cards. You can convert from any supported input — MP3, WAV, AAC, OGG, FLAC, OPUS, or M4A — to any of the six output formats. For lossy outputs like MP3 and AAC, select the bitrate from the dropdown. WAV and FLAC are lossless, so the bitrate setting has no effect on those formats.
What is the difference between WAV and FLAC?
Both WAV and FLAC are lossless formats, meaning they preserve 100% of the original audio quality with no degradation. The key difference is file size: WAV is uncompressed and produces the largest files, while FLAC uses lossless compression that typically reduces file size by 40–60% compared to WAV — with no loss in quality. For editing and archiving, both are excellent choices. FLAC is generally preferred when storage space matters.
What is OPUS and when should I use it?
OPUS is a modern, open audio codec developed by the IETF. It is exceptionally efficient — it can deliver very good audio quality at bitrates as low as 32–64 kbps, making it ideal for voice calls, streaming, and web audio. If you are preparing audio for a web application or messaging service, OPUS is often the best choice. It is supported natively in all modern browsers and many media players.
Why does it take time on the first use?
On first use, your browser downloads the FFmpeg.wasm engine files (approximately 10 MB) from a CDN. This is a one-time step. Once downloaded, the engine is cached by your browser and subsequent uses within the same session start immediately. If you close the browser and return later, the files are loaded from cache and the wait is significantly shorter.
Is there a file size limit?
There is no server-imposed file size limit since everything runs locally in your browser. In practice, very large files (above 500 MB) may run into your browser's available memory limit, which varies by device. For best results, process files under 200 MB. If you need to work with very large files, consider using a desktop audio editor like Audacity alongside this tool.
Which browsers are supported?
The toolkit works in all modern desktop and mobile browsers: Google Chrome, Microsoft Edge, Mozilla Firefox, and Safari. The FFmpeg engine runs in single-threaded mode for maximum compatibility — no special browser flags or server headers are required. For the best experience, use the latest version of Chrome or Edge.
Can I trim and convert at the same time?
Yes. The tool applies all active settings in a single FFmpeg pass. Set your trim points in the Trim tab and your output format in the Convert tab, then click Process. Both operations will be applied simultaneously, saving time compared to running two separate steps.
Does converting to a lossless format improve quality?
No. Converting a lossy file (like MP3 or AAC) to a lossless format (like WAV or FLAC) does not restore quality that was already lost during the original compression. The lossless container simply stores the already-degraded audio without further compression. To avoid quality loss, always work from the original uncompressed source when possible.
Why Use Alfreto's Audio Toolkit?
Most online audio tools work by uploading your file to a remote server, processing it there, and sending it back. That approach introduces privacy risks, file size limits, and dependency on server availability. Alfreto takes a different approach: every operation runs inside your own browser, using WebAssembly technology that brings professional-grade processing power to the client side.
Your audio never reaches any server. There is no account to create, no terms to agree to for data processing, and no risk of your files being stored, analyzed, or accessed by anyone else.
Because processing happens on your device, there is no server-imposed upload limit. Large files that other tools refuse are handled without restriction.
The tool is powered by FFmpeg — the same open-source engine used by professional video editors, streaming platforms, and broadcasters worldwide. You get production-quality results in a simple browser interface.
Once the FFmpeg engine is cached after the first use, the tool works without an active internet connection. Process files anywhere, even without Wi-Fi.