Trim AIFF — Free Online Tool
Trim an AIFF audio file to an exact time range while preserving the original uncompressed PCM audio stream with zero quality loss. Since both input and output are identical lossless AIFF containers, the audio data is copied directly without re-encoding — making trimming instant and bit-perfect.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
When trimming an AIFF file to another AIFF, FFmpeg uses stream copying (-c copy) to extract the specified time range directly from the raw PCM audio data without any decoding or re-encoding. AIFF stores uncompressed audio (typically pcm_s16be, pcm_s24be, or pcm_s32be big-endian PCM), which means the trim operation is essentially a byte-level cut: FFmpeg seeks to the start point, copies the raw audio frames through to the end point, and writes them into a new AIFF container. Because no codec conversion occurs, the operation is extremely fast, uses minimal CPU, and the output is mathematically identical to the corresponding section of the original file. The -ss and -to flags define the in and out points respectively, and because AIFF is an uncompressed format with no inter-frame dependencies (unlike video codecs with keyframes), sample-accurate trimming is straightforward.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In the browser-based version of this tool, this runs via FFmpeg.wasm (WebAssembly), so no installation is required and no files leave your device. |
-i input.aiff
|
Specifies the input AIFF file containing the uncompressed PCM audio to be trimmed. FFmpeg reads the AIFF container header to determine the codec variant (e.g., pcm_s24be), sample rate, and channel layout before processing. |
-ss 00:00:00
|
Sets the trim start point. Placed after -i (as an output option), this tells FFmpeg to begin reading from the given timestamp in the AIFF audio stream. Because AIFF is uncompressed PCM with no keyframe dependencies, seeking is sample-accurate rather than constrained to keyframe boundaries. |
-to 00:00:10
|
Sets the trim end point as an absolute timestamp in the output. FFmpeg will stop copying audio samples at this position, producing a trimmed AIFF file containing only the audio between the -ss and -to timestamps — in this default case, the first 10 seconds. |
-c copy
|
Instructs FFmpeg to copy the audio stream directly without decoding or re-encoding. For AIFF-to-AIFF trimming, this means the raw big-endian PCM samples are passed through unchanged, preserving the original bit depth, sample rate, and every sample value exactly — the fastest and highest-quality approach possible. |
output.aiff
|
Specifies the output filename and container format. The .aiff extension tells FFmpeg to write an AIFF container, and combined with -c copy, the output will contain the same PCM codec variant (e.g., pcm_s16be or pcm_s24be) as the source file. |
Common Use Cases
- Isolate a specific musical phrase or section from a full studio recording for use as a sample in a DAW like Logic Pro or Pro Tools
- Trim silence or dead air from the beginning or end of a field recording captured as AIFF on a portable recorder
- Extract a clean loop region from a longer AIFF audio file to use as a seamless background music loop in video production
- Cut down a long voiceover take to the best reading without touching the original uncompressed 24-bit audio quality required for broadcast delivery
- Prepare a short AIFF clip from a larger archival recording for use in a macOS application or Apple-platform project that expects AIFF input
- Quickly preview-trim a large uncompressed AIFF session export to share a specific segment with a client before committing to a full re-export from the DAW
Frequently Asked Questions
No — because -c copy bypasses encoding entirely and copies the raw PCM audio samples directly from the source file. AIFF uses uncompressed big-endian PCM (such as pcm_s16be or pcm_s24be), so there is no compression artifact to introduce and no decoding/re-encoding step that could alter sample values. The trimmed output is bit-for-bit identical to the corresponding portion of the original file.
Because AIFF stores uncompressed PCM audio with no inter-frame dependencies or keyframe restrictions (unlike compressed audio or video codecs), FFmpeg can seek and cut at the exact sample boundary closest to your specified timestamp. You will not experience the coarse keyframe-alignment issue that affects trimming compressed formats like AAC or MP3. For most practical purposes, the trim is sample-accurate.
Yes. When using -c copy, no transcoding occurs and the stream parameters — including bit depth (e.g., 16-bit, 24-bit, 32-bit) and sample rate (e.g., 44.1 kHz, 48 kHz, 96 kHz) — are carried over unchanged into the output file. The output AIFF will have the same PCM codec variant (such as pcm_s24be) as the source.
AIFF supports a limited set of metadata through its NAME, AUTH, ANNO, and ID3-compatible chunks. When using -c copy, FFmpeg will attempt to carry over any metadata present in the source AIFF to the output. However, AIFF metadata support in FFmpeg is not as comprehensive as formats like FLAC or MP4, so some embedded tags may not round-trip perfectly. For critical metadata preservation, verify the output with a tool like MediaInfo after trimming.
Modify the values after -ss (start time) and -to (end time) using the format HH:MM:SS or HH:MM:SS.mmm for millisecond precision. For example, to trim from 1 minute 30 seconds to 2 minutes 45.5 seconds, use: ffmpeg -i input.aiff -ss 00:01:30 -to 00:02:45.500 -c copy output.aiff. Alternatively, replace -to with -t to specify a duration rather than an end timestamp — for example, -t 00:01:15.500 would copy 1 minute and 15.5 seconds starting from the -ss point.
Yes, on the command line you can loop over multiple files using a shell script. On macOS or Linux: for f in *.aiff; do ffmpeg -i "$f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_$f"; done. On Windows Command Prompt: for %f in (*.aiff) do ffmpeg -i "%f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_%f". Each file will be trimmed to the same time range and saved with a prefixed filename.
Technical Notes
AIFF (Audio Interchange File Format) is a chunk-based container format developed by Apple, storing uncompressed PCM audio in big-endian byte order — the primary distinction from WAV, which uses little-endian PCM. Supported codec variants include pcm_s16be (16-bit CD quality), pcm_s24be (24-bit professional standard), pcm_s32be (32-bit high resolution), and the floating-point variants pcm_f32be and pcm_f64be used in some scientific and high-end audio workflows. Because stream copying is used, the output inherits whichever PCM variant the source uses — no downconversion or upconversion occurs. One known limitation of AIFF trimming with -c copy is that the AIFF SSND (Sound Data) chunk header will be rewritten to reflect the new file length, but FFmpeg handles this correctly. File size scales linearly with duration since the data is uncompressed: a 24-bit stereo 48kHz AIFF produces approximately 17 MB per minute, so a 10-second trim of such a file yields roughly 2.8 MB. AIFF does not support subtitles, chapters, or multiple audio tracks, so no data of those types will be lost during trimming.