For lossy formats like MP3 and AAC, the trim is keyframe-accurate (the nearest encoded frame to your chosen second, typically within 20ms). For lossless formats like WAV and FLAC the trim is sample-accurate. If you need frame-perfect MP3 cuts, the tool can re-encode to PCM first — but for most uses keyframe accuracy is invisible.

No — by default we use FFmpeg's stream-copy mode (`-c copy`), which extracts the bytes between your two timestamps without decoding or re-encoding. Quality is byte-perfect identical to the source. The resulting file is in the same container and codec as the original.

Up to 200MB. Trimming with stream-copy is extremely fast — a 2-hour podcast usually trims in under 10 seconds because no audio decoding happens. Larger trims may need slightly more time as FFmpeg seeks through the file.

Online tools that upload your file run FFmpeg natively on a server, often with parallel processing. Our browser version runs FFmpeg compiled to WebAssembly in a single thread, so it is typically 2–4× slower than a server. The trade-off is privacy: your audio never leaves your device.

This tool extracts a single contiguous range. To remove a middle section while keeping both ends, use the Audio Segment Remover tool. To split an audio file into multiple equal pieces, use the Audio Splitter.

In stream-copy mode, the container metadata (title, artist, album art) is preserved exactly. If you choose a different output format, FFmpeg copies what it can but some custom tags may be dropped — this is normal for cross-format conversion.

Server-side tools use multi-threaded native FFmpeg running on dedicated CPUs with fast disks and parallel pipelines. Our engine is FFmpeg compiled to WebAssembly, which runs single-threaded inside your browser tab and has no access to native hardware acceleration. That makes browser-based jobs typically 3–8× slower than a server. The trade-off is total privacy: your audio file is never uploaded, never logged, and never stored — closing the tab erases everything from memory immediately. For most clips up to a few minutes the wait is small, and for sensitive recordings (voice memos, drafts, confidential meetings) the privacy gain is well worth it.

No. Everything runs entirely inside your browser tab using FFmpeg compiled to WebAssembly. The file is read into local memory only, processed in the same tab, and the result is offered as a direct download. Nothing is transmitted to any server, no account is required, no analytics are tied to your file, and closing the tab discards every byte from memory.

The file picker accepts audio inputs up to about 1 GB, which is well above what mainstream "free tier" online converters allow. The real ceiling is your device — everything runs inside your browser tab, which shares memory with the rest of the page. Most podcasts, songs, and voice memos sit comfortably under that limit even on a phone. If a very large lossless WAV or FLAC ever fails, trim it first or transcode to MP3 / Opus to bring the size down before re-running the tool.

MP3, WAV, OGG (Vorbis and Opus), FLAC, M4A (AAC), AAC, Opus, AIFF, and WMA all decode reliably via FFmpeg WASM. Output formats depend on the specific tool — most editing tools default to MP3 (universal) or WAV (lossless) but expose a format picker so you can pick the one that fits your downstream player or DAW.

Recent Chrome, Edge, Firefox, Safari, and other Chromium-based browsers all work. The tool relies on WebAssembly and SharedArrayBuffer, which require the page to be served over HTTPS with the right cross-origin headers — this site is configured correctly by default. On phones the same code runs but is slower than on a desktop because mobile CPUs are weaker.

No. The tool is completely free, requires no account, attaches no watermark, applies no usage caps, and shows no popup ads on your output. Because the work happens on your own device, there is no per-user quota for us to enforce — your hardware and browser memory are the only limits. The download is the file you would get from running FFmpeg locally, nothing more, nothing less.

Trimming itself is sample-accurate and lossless — the selected range is extracted exactly. The only quality consideration is the output format. If you trim a WAV and output as WAV, the trimmed audio is byte-identical to the corresponding portion of the original. If you trim a WAV and output as MP3, the trim is lossless but the MP3 encoding step adds a generation of compression loss; pick a high MP3 bitrate (256+ kbps) to keep this minimal.

Not in a single operation — the trimmer extracts one continuous range. To remove a middle section, run the tool twice (extract the part before the cut, then the part after), then merge the two clips with Audio Merger. Each individual trim stays lossless; the merge is also lossless if both pieces are in the same format.

Sample-accurate — the trim happens at the exact sample where you marked it, which means precision down to 1/44100 of a second on a CD-quality file (and even finer on higher sample rate sources). The waveform display lets you zoom in to find specific transients (a snare hit, a consonant in speech) and cut on them.

Yes. Audio trimming preserves the timing of every sample within the selected range exactly. The only place timing matters is at the cut points — and even there, sample-accurate trimming means there is no audible gap or click at the boundaries. If you do hear a click, it is usually because the cut is in the middle of a non-zero sample value; the tool offers an optional 5-millisecond fade-in/fade-out to smooth such transitions.

WAV for archival, editing, or any further processing — it is uncompressed and lossless. MP3 for general sharing where you want maximum compatibility — virtually every device plays MP3. AAC for sharing with iPhones and modern audio players where you want better quality than MP3 at the same bitrate. OGG/Opus for the open-source web (best quality at low bitrates). FLAC for lossless compression that is roughly half the size of WAV.

Yes. The trim respects the source’s exact channel layout (mono, stereo, 5.1, etc.), bit depth (16-bit, 24-bit, 32-bit), and sample rate (44.1kHz, 48kHz, 96kHz, 192kHz). All of those carry through to the output if your chosen format supports them. WAV and FLAC support everything; MP3 caps at 48kHz stereo by spec.

Each trim is one range per operation. To extract several ranges, run the tool multiple times — the source stays loaded between runs so the second and third trims are nearly instant.

The 200 MB upload limit applies, which is plenty for hours of high-quality stereo audio. A 1-hour MP3 at 192 kbps is about 86 MB; a 1-hour WAV at CD quality is about 600 MB and exceeds the cap. For very large files, convert to FLAC or MP3 first to bring them under the limit, then trim.