Audio Trimmer Online
Trim audio files with start and end time controls. Free, in your browser, no file uploads.
.mp3, .wav, .ogg, .flac, .aac · up to 100 MB
What you can do
Trim any audio to the exact second
No quality loss
Non-destructive trim: copies the segment without re-encoding. Quality identical to the original.
100% private
Processing happens in your browser. Your file is never uploaded to any server.
All formats
Compatible with MP3, WAV, AAC, OGG, FLAC, AIFF, and more.
Instant
No queues or waiting. Trimming completes in seconds directly on your device.
How it works
Three steps, no hassle
Upload your audio file
Drag or select your MP3, WAV, AAC, OGG, FLAC, or other compatible format file. Up to 200 MB, no signup.
Set the trim points
Enter start and end time in mm:ss or hh:mm:ss format. Use the built-in player to listen and adjust with precision.
Download the trimmed audio
The resulting file preserves the original format and exact quality of the selected segment. Download with one click.
FAQ
Got questions?
It depends on the trimming method used. Non-destructive trimming (stream copy) copies the audio data without re-encoding it, preserving exactly the original quality with no additional loss. FFmpeg implements this with the -ss (start seek) and -to (end time) flags combined with -c copy, which extracts the segment by copying compressed packets directly without going through the decode-re-encode cycle. This method is extremely fast (limited only by disk read/write speed) and produces a file bit-for-bit identical to the original in the selected segment. The only caveat is precision in formats with temporal compression like MP3: the cut occurs at the keyframe nearest to the specified point, which in MP3 can be up to 26 ms before the exact time (each MP3 frame contains 26 ms of audio at 44.1 kHz). For trims where exact millisecond precision is critical, destructive trimming (with re-encoding) allows cutting at any point, but introduces an additional generation of lossy compression.
The most common audio formats are fully supported for trimming: MP3 (MPEG-1 Audio Layer III), WAV (Waveform Audio File Format), AAC (Advanced Audio Coding, including .m4a files), OGG Vorbis, FLAC (Free Lossless Audio Codec), AIFF (Audio Interchange File Format, common on macOS and Logic Pro), WMA (Windows Media Audio), and Opus. Specific support depends on the browser: Chrome 120+ and Edge 120+ support the largest number of formats through the Web Audio API and MediaRecorder. Firefox has native support for OGG/Vorbis and Opus. Safari (macOS/iOS) natively supports AAC/M4A, MP3, WAV, and AIFF. For less common formats like FLAC in Safari or WMA in Firefox, prior conversion to MP3 or WAV ensures universal compatibility.
For precision trimming, the most effective approach is to combine listening with exact numeric values. The accepted time format is hh:mm:ss.mmm where milliseconds are optional (00:01:30.500 = 1 minute, 30 seconds, and 500 milliseconds). To find the exact point in a podcast or recording: use the built-in player, pause at the desired moment, and note the displayed time. For ringtone production, the standard is that the most memorable part of a song generally occurs in the chorus, which in pop music typically starts between 45 and 60 seconds from the beginning. For sample extraction, look for attack transients (drum hits, piano notes) that have a clear onset; a 50 ms error at the cut point can cause the sample to sound with a click or audible artifact at the start.
The practical limit depends on the RAM available on the device. The browser's Web Audio API loads the entire file into memory for processing. A 100 MB MP3 file represents approximately 8–10 hours of audio at 32 kbps, or 1–2 hours at 128 kbps, which is more than sufficient for most use cases. On modern devices with 8 GB RAM or more, files up to 500 MB process without issue. On mobile devices with limited memory, files larger than 100–150 MB may cause the browser to close the tab due to lack of memory (OOM kill). If you need to trim large files (long interviews, studio session recordings), the recommendation is to use FFmpeg from the command line directly, which processes audio in streaming without loading it entirely into memory.
Fade in and fade out are audio filters that apply a gain ramp to the start and/or end of the trimmed segment. In FFmpeg, they are implemented with the afade filter: -af 'afade=t=in:st=0:d=2,afade=t=out:st=28:d=2' applies a 2-second fade in at the start and a 2-second fade out starting at second 28 of a 30-second clip. Typical duration for music is 1–3 seconds for fade in (depending on tempo and style) and 2–5 seconds for fade out. For ringtones, a 0.5–1 second fade in avoids the abrupt jump when the phone rings, and a 2–3 second fade out sounds more natural than an abrupt cut. Fades require audio re-encoding (they cannot be applied with stream copy), which introduces an additional generation of lossy compression in formats like MP3 or AAC, imperceptible at bitrates of 128 kbps or higher.
Audio trimming has multiple professional and everyday use cases. Ringtone creation: trimming a song's chorus to 30–40 seconds for use as a ringtone on smartphones (M4R format is the AAC container that iOS uses for ringtones; Android accepts MP3 directly). Podcast editing: removing prolonged silences, presenter errors, or irrelevant segments before publishing. Sample extraction: taking 1–8 bar fragments from a song to use as a sample in music production, DJing, or composition. Creating audio clips for social media: platforms like Instagram Reels, TikTok, and YouTube Shorts accept audio up to 60–90 seconds. Preparing material for e-learning: trimming fragments from conferences or interviews to include in training modules. Audiobook editing: separating chapters from a long recording into individual files.
Trim audio online: technical guide to precision and formats
Audio trimming is one of the most frequent editing operations in sound production, podcasting, and digital content creation. At a technical level, there are two fundamental approaches with very different implications for output quality. Non-destructive trimming, also known as stream copy or muxing without re-encoding, extracts the audio segment by copying existing compressed packets without going through the decode-re-encode cycle. This method preserves the original quality bit for bit and is what FFmpeg implements when using the -ss (seek to start position) and -to (end time) flags together with -c copy. The limitation of this approach in formats with temporal compression like MP3 (which organizes audio into 26 ms frames at 44.1 kHz with inter-frame dependencies for the stereo channel) is that the cut point must coincide with a frame boundary, introducing a maximum error of approximately 26 ms relative to the specified time. Destructive trimming, on the other hand, fully decodes the audio to PCM (Pulse Code Modulation, the uncompressed representation of the sound wave as a sequence of numeric samples) and then re-encodes only the selected segment to the target format. This allows exact precision to the sample (an individual sample, equivalent to 1/44100 of a second at 44.1 kHz) but implies an additional generation of lossy compression in formats like MP3 or AAC. At bitrates of 128 kbps or higher, degradation from a single additional compression generation is imperceptible in critical listening; at 64 kbps or less, it may be audible in complex content like music with high spectral density.
FFmpeg, the world's most widely used open source audio and video processing engine (originally developed by Fabrice Bellard in 2000 and actively maintained by the FFmpeg community), is the technical reference for audio trimming operations. The basic syntax for lossless trimming is: ffmpeg -ss 00:01:30 -to 00:03:00 -i input.mp3 -c copy output.mp3. Placing -ss before -i (input seeking) is faster than after -i (output seeking) for long files because it leverages container indexing. For trimming with re-encoding and fade: ffmpeg -ss 00:01:30 -to 00:03:00 -i input.mp3 -af 'afade=t=in:st=0:d=1,afade=t=out:st=88:d=2' -b:a 192k output.mp3. In the context of ringtone creation, the technical format varies by mobile operating system. iOS (iPhone) requires the M4R format, which is simply an AAC-LC file in an MPEG-4 container with the .m4r extension (the same structure as a .m4a). The maximum duration for an iOS ringtone is 30 seconds for a call tone and 25 seconds for a message tone. Android accepts MP3, AAC, OGG, and WAV directly, without technical duration restriction (though users typically prefer 20–40 seconds). The recommended sample rate for ringtones is 44.1 kHz (CD standard), and the recommended bitrate is 128 kbps for an optimal balance between quality and file size.
For podcast editing, trimming is just one of the basic operations in a complete post-production workflow. Professional podcast editors use DAWs (Digital Audio Workstations) like Adobe Audition, Reaper, or GarageBand for complex edits, but for simple trimming and segment extraction operations, online tools offer the advantage of not requiring software installation. The standard delivery format for podcasts distributed via RSS is MP3 (for universal compatibility with all podcast players from iPodder in 2003 to Spotify, Apple Podcasts, and Google Podcasts) at 64–128 kbps for voice content, with mono instead of stereo to halve the file size without perceptible loss in spoken voice. Sample extraction for music production requires additional copyright considerations. In the United States, the fair use doctrine (17 U.S.C. § 107) does not provide an automatic exception for music samples; the case Grand Upright Music Ltd. v. Warner Bros. Records Inc. (1991) established that unlicensed sampling constitutes copyright infringement even for very short fragments. In modern music production practice, producers work with sample packs under Creative Commons or royalty-free licenses, or obtain clearance from record labels. Technically, a usable sample in production generally has between 2 and 8 bars (4–16 seconds at 120 BPM), enough time to establish a recognizable rhythmic or melodic groove.