✂️Free Online Tool

Trim FLAC Audio

Trim FLAC audio files online with lossless quality. Set start time and duration, adjust compression level from 1 to 12.

Drop your file here, or browseSupports MP4, MOV, AVI, MKV, WebM, MP3, WAV and more

Lightning Fast

Process files in seconds with our optimized servers

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Precise Trimming

Set exact start and end points with frame accuracy

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No Quality Loss

Maintain original quality with smart re-encoding

How to Trim FLAC Audio Online

  1. Upload Your FLAC File: Drag and drop or click "Add Files" to select a .flac file. Batch trimming is supported — queue multiple files and apply the same trim points across all of them.
  2. Set Start Time and Duration: Under Trim, enter Start Time and Duration in HH:MM:SS.sss format (millisecond precision). For example, Start 00:02:30.500 with Duration 00:03:00.000 extracts a 3-minute song starting 2 min 30.5 sec in. Use the original FLAC length minus your start point if you want the file from a marker to the end.
  3. Pick Compression Level, Channel, or Sample Rate (Optional): Leave Audio Codec at FLAC to keep the trim lossless. Compression Level defaults to 12 (smallest file, slowest encode); pick 1-4 when batch-processing dozens of files or 5-8 to match what most ripping tools produce — every level is bit-identical, only encode time and file size differ. Audio Channel defaults to Original (switch to Mono to halve size for a voice recording, or Stereo to upmix a mono source). Audio Sample Rate stays at the source rate by default; only change it if you specifically need 44.1 kHz CD-rate output from a 96 kHz hi-res master.
  4. Trim and Download: Click Convert. Files process in your browser session — no sign-up, no watermark, no MP3 conversion step. The output stays FLAC and stays lossless.

Why Trim FLAC?

FLAC (Free Lossless Audio Codec) is a royalty-free, open-source lossless format released in July 2001. Its container is frame-based: every FLAC frame carries its own header and is independently decodable, which is why a properly written trim can preserve bit-perfect audio between trim points without re-encoding the entire stream. That matters because most "online FLAC cutters" actually export the result as a high-bitrate MP3 — Flixier's own help text admits this — silently turning a lossless source into a lossy file.

  • Album rips into per-track files — A 60-minute concert recording or a single-file album rip (FLAC + CUE) split into individual track FLACs for portable players, Plex, or Roon. Keep each track at the original 16-bit/44.1 kHz or 24-bit/96 kHz fidelity instead of re-encoding to lossy AAC.
  • Removing pre-gap silence and applause — Live recordings often have 5-15 seconds of audience noise or count-in at the start. Trim those out without resampling or transcoding so the master MD5 hash of the kept audio still matches the source archive.
  • Audiophile sample preparation — Extract a reference 30-60 second clip from a 24-bit/192 kHz master for A/B testing DACs, headphones, or speakers. The trimmed file remains lossless at the original bit depth so the test is honest.
  • Podcast and broadcast editing — Many podcast workflows capture to FLAC for safety masters, then edit out long pauses, retakes, or sponsor breaks before encoding the publish copy to MP3 or AAC. Doing the cuts in FLAC keeps the editing pipeline lossless.
  • Voice memo cleanup for legal or research use — Interview, lecture, and deposition recordings stored as FLAC for chain-of-custody integrity. Trim to just the relevant exchange without altering the rest of the file's checksum.
  • Mobile uploads — iOS 11+ and Android 3.1+ have supported FLAC natively for years, but most phone-side editors still down-convert to AAC/MP3. Doing the trim in a browser keeps the round trip lossless end-to-end.

FLAC vs Other Lossless and Lossy Formats

Property FLAC WAV ALAC MP3
Lossless Yes Yes (PCM) Yes No
Typical size (3-min track @ CD quality) ~20-25 MB ~30 MB ~22-27 MB ~3-7 MB
Compression Yes (~50-60% of WAV) None Yes (~60-70% of WAV) Yes (lossy)
Bit depths 4-32 bits 8, 16, 24, 32 16, 24 N/A
Max channels 8 Many 8 2
Max sample rate 1,048,575 Hz (spec) 4 GHz (spec) 384 kHz 48 kHz
Embedded metadata Vorbis comments + cover art RIFF chunks (limited) iTunes-style ID3v1/v2
Royalty-free, open Yes Yes Yes (Apple open-sourced 2011) Yes (patents expired 2017)
Native iOS iOS 11+ All versions All versions All versions
Native Windows Explorer Win 10+ All versions Limited All versions

Sources: RFC 9639 (FLAC spec), Apple ALAC source release (2011), Wikipedia FLAC and ALAC articles.

FLAC Compression Level Reference

Level Relative Encode Time Size vs Level 8 Audio Quality Typical Use
0 ~1x (fastest) +4-6% larger Bit-identical Real-time capture
5 (libFLAC default) ~3x +1-2% larger Bit-identical General ripping (EAC, dBpoweramp)
8 (FLAC preset --best) ~10x Baseline Bit-identical Most archival rips
12 (xconvert default) ~20x -0.5 to -1.5% smaller Bit-identical Maximum-density archive

All levels decode to exactly the same PCM samples — the MD5 of the unencoded audio stays identical across levels. Higher levels just spend more CPU finding better prediction coefficients to shave a few bytes off the file. Level 8 has been the long-standing reference recommendation; levels 9-12 yield diminishing returns (often under 1%).

Frequently Asked Questions

Does trimming a FLAC file lose any audio quality?

No. FLAC is a lossless codec — the audio data between your trim points is the same PCM samples as the source. The compression algorithm doesn't approximate or discard any information; it stores the exact original waveform. The only thing that changes when you re-encode the trimmed segment is which prediction coefficients libFLAC picks (a function of compression level), and those are mathematically reversible. If you decode the trimmed FLAC to WAV, the samples inside the kept range are bit-identical to the source.

Why are some online FLAC cutters secretly converting my file to MP3?

A lot of browser-based audio editors are built on a generic Web Audio pipeline that decodes everything to a PCM AudioBuffer and then encodes the export through a built-in MP3 or AAC encoder. That's fine for a podcast cut but destroys FLAC's whole point. Flixier's own FAQ admits "Flixier exports all audio files as high-bitrate MP3s, so your files will lose some fidelity." This tool keeps the codec at FLAC end-to-end, so what you download is still a lossless FLAC.

What is the maximum sample rate and bit depth FLAC supports?

Per RFC 9639 (the current FLAC specification, published 2024), FLAC supports 4 to 32 bits per sample and sample rates from 1 Hz to 1,048,575 Hz. In practice the common values you'll see are 16-bit/44.1 kHz (CD), 24-bit/48 kHz (studio), 24-bit/96 kHz and 24-bit/192 kHz (hi-res). FLAC supports up to 8 channels (mono through 7.1).

Can I trim a 24-bit/96 kHz hi-res FLAC without downsampling it?

Yes. Leave Audio Sample Rate at "Unchanged" (the default) and Audio Channel at "Original." The output FLAC will carry the same 24-bit depth and 96 kHz sample rate as the source. Don't change the sample rate unless you specifically need CD-rate output — downsampling from 96 kHz to 44.1 kHz applies a resampling filter and is not reversible, even though both ends are still "lossless" in the codec sense.

Why does my trimmed file's size not match (duration ratio) × (original size)?

FLAC compression efficiency varies across the track. A quiet passage compresses much better than a loud, complex one because the linear-prediction residuals are smaller. If you trim out a dense orchestral climax and keep a sparse intro, the kept portion's per-second size will be lower than the file's average. The MD5 of the unencoded audio in the FLAC streaminfo header is also recomputed for the new shorter stream, so the headers differ too.

Should I trim before or after converting FLAC to MP3?

Always trim in FLAC first, then convert to MP3 if you need the lossy copy. Trimming a FLAC is lossless; trimming an MP3 either re-encodes (generation loss) or leaves you with imperfect frame-aligned cuts (since MP3 frames overlap via the bit reservoir). Cutting first in the lossless master and encoding once at the end is the cleanest workflow. See FLAC to MP3 for the encoding step or FLAC to WAV if you need uncompressed PCM.

Will the cover art, ReplayGain tags, and Vorbis comments survive the trim?

FLAC stores metadata in separate blocks (Vorbis comment block for tags, PICTURE block for embedded cover art, APPLICATION blocks for things like ReplayGain). A trim operation that just rewrites the audio frames typically preserves these metadata blocks. Track-specific tags like TRACKNUMBER, TITLE, and ARTIST stay; ReplayGain values, however, are tied to the original audio's loudness and should be recomputed after a non-trivial trim (most music players will do this on import).

How precise are the Start Time and Duration inputs?

The inputs accept millisecond precision in HH:MM:SS.sss format, which is more than enough for music editing — one millisecond at 44.1 kHz is about 44 samples. The actual cut snaps to the nearest FLAC frame boundary internally (typically every 4608 or 16384 samples at the default block sizes), so you may be off from the requested point by up to ~0.1 to 0.4 seconds in the absolute worst case. For sample-accurate edits on the order of single samples, work in WAV/AIFF instead.

Can I trim a single-file album rip (FLAC + CUE) into individual tracks?

Not in one operation — the CUE file isn't read here. Workflow: open the CUE in a text editor to read each track's INDEX 01 timestamps, then queue the same FLAC multiple times with different Start Time and Duration values (one per track). Alternatively, use a dedicated CUE-aware tool like CUETools locally and bring the per-track FLACs here only to refine trim points or change compression level. For batch audio work on other formats see Audio Cutter, Trim WAV, or Trim MP3.

My trimmed FLAC plays fine but sounds slightly different in foobar2000 vs VLC — why?

Identical lossless audio can sound different on playback if the players use different resamplers, ReplayGain settings, or output bit-depth handling. Foobar2000 by default applies ReplayGain track gain if tags are present; VLC by default does not. If you've trimmed away the loudness profile the tags were computed against, the gain values are stale. Strip or recompute ReplayGain, or disable it in your player, and the two files will sound the same.

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