Trim AAC — Free Online Tool
Trim an AAC audio file to a precise start and end point, outputting a new AAC file with the same codec and bitrate. Because both input and output use the AAC format, this tool uses stream copying (-c copy) to cut the file instantly without re-encoding — preserving your original audio quality exactly.
to
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 AAC file to AAC, FFmpeg uses the -c copy flag to perform a stream copy rather than a full re-encode. This means the AAC audio data is not decoded and re-compressed — it is simply extracted between the specified timestamps and written to a new container. The process is nearly instantaneous regardless of file length. One technical nuance specific to AAC trimming: because AAC uses variable-length audio frames, the actual cut point may be aligned to the nearest AAC frame boundary rather than the exact millisecond you specify. For most use cases this is imperceptible, but if sample-accurate cuts are critical, a full re-encode (removing -c copy) would be needed.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. This is the open-source multimedia processing engine running under the hood — in the browser via FFmpeg.wasm (WebAssembly), or locally on your desktop if you have FFmpeg installed. |
-i input.aac
|
Specifies the input AAC audio file. FFmpeg reads the AAC stream from this file, detecting the codec profile (e.g., AAC-LC), sample rate, bitrate, and channel layout automatically. |
-ss 00:00:00
|
Sets the start point of the trim. In this example it is the very beginning of the file, but you can set any timestamp in HH:MM:SS format (e.g., 00:01:30 to start at one minute and thirty seconds). When placed after -i, FFmpeg decodes up to this point before beginning the output — essential for accurate cutting with -c copy. |
-to 00:00:10
|
Sets the absolute end timestamp for the output. Combined with -ss 00:00:00, this example produces a 10-second AAC clip. This is an absolute position in the file, not a duration — adjust it to the exact end time you want in the output. |
-c copy
|
Instructs FFmpeg to copy the AAC audio stream directly without decoding or re-encoding. For AAC-to-AAC trimming, this preserves the original bitrate and audio quality exactly while making the operation nearly instantaneous. No quality is lost because no lossy compression step is applied. |
output.aac
|
The filename for the trimmed AAC output file. FFmpeg writes the copied AAC frames between the specified timestamps into this new file, which will be a valid AAC audio file compatible with iTunes, iOS, and all major AAC-capable players. |
Common Use Cases
- Cut a long podcast episode recorded in AAC to remove dead air at the start or end before publishing to iTunes or Apple Podcasts
- Trim an AAC audio clip downloaded from an iPhone voice memo or screen recording to isolate a specific quote or segment
- Extract a chorus or hook from an AAC music file to use as a preview or ringtone on iOS or Android
- Shorten an AAC audiobook chapter to clip a specific passage for sharing, while keeping the original AAC encoding intact for Apple Books compatibility
- Trim an AAC audio file used as a background track in a video project to match a specific scene duration without quality loss from re-encoding
- Cut a lengthy AAC recording from a streaming capture to isolate a single song or segment before further processing
Frequently Asked Questions
No — using -c copy means the AAC audio data is never decoded or re-encoded. The audio frames between your chosen timestamps are copied byte-for-byte into the new file, so there is zero generation loss. This is one of the key advantages of trimming within the same format: you get a lossless cut of a lossy file.
AAC audio is encoded in discrete frames, typically 1024 samples per frame (about 21–23ms at 44.1kHz or 48kHz). When using -c copy, FFmpeg must start and end on a valid frame boundary, so it snaps the cut to the nearest AAC frame. The deviation is usually less than 50ms. If you need a frame-accurate cut to the exact millisecond, you must re-encode by removing -c copy and adding -c:a aac, which decodes and re-encodes the audio at the trim point.
Because the command uses -c copy, no re-encoding occurs and the bitrate flag (-b:a) has no effect. The output will always match the source file's original bitrate. To change the bitrate — for example, to produce a smaller 96k clip from a 256k source — replace -c copy with -c:a aac -b:a 96k. Note that this will introduce a re-encode step and may take longer.
Yes. The output is a standard AAC file using the same codec as the input, which is natively supported by all Apple platforms. As long as the original file played correctly in iTunes or on iOS, the trimmed version will too. AAC is Apple's preferred audio format, so compatibility is not a concern for this format-to-format operation.
FFmpeg does not natively batch process multiple files in a single command, but you can loop over files using a shell script. On Linux or macOS, use: for f in *.aac; do ffmpeg -i "$f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_$f"; done. On Windows Command Prompt, use a for loop: for %f in (*.aac) do ffmpeg -i "%f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_%f". Adjust the -ss and -to values as needed.
The -to flag specifies the absolute end timestamp in the output file (e.g., -to 00:00:10 means stop at the 10-second mark). The -t flag specifies a duration relative to the start point (e.g., -t 00:00:10 means cut 10 seconds of audio starting from -ss). If your -ss is 00:00:05 and you want to end at the 30-second mark of the original file, use -to 00:00:30. If you want exactly 25 seconds of audio from that start point, use -t 00:00:25.
Technical Notes
Trimming AAC to AAC with stream copy is one of the most efficient operations FFmpeg can perform — the process is I/O bound rather than CPU bound, making it nearly instant even for large files. The output file will preserve the original AAC codec profile (LC, HE-AACv1, HE-AACv2) and channel layout from the source. Metadata tags (artist, title, album, etc.) stored in the AAC container are typically preserved by -c copy, though chapter markers are not supported in raw AAC files. If your source file is wrapped in an M4A container (MPEG-4 with AAC audio), this tool handles it identically since M4A is simply a renamed .mp4 container carrying AAC streams. One known limitation: AAC files with seek tables or gapless playback metadata (common in iTunes-encoded files) may lose the gapless padding information after trimming with -c copy, which is generally acceptable for trimmed clips.