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.aac file. Raw ADTS streams from YouTube extraction tools, AAC pulled from a TS/M2TS broadcast capture, FFmpeg dumps, and .aac exports from audio editors all work. Batch is supported — drop in a folder of episode rips and apply the same cut to all of them.42.5) or HH:MM:SS.sss format (e.g. 00:00:42.500). Cuts are sample-accurate: the audio is decoded at your timestamps, so the cut lands on the exact time you enter rather than rounding to a 1024-sample AAC frame boundary (~23 ms at 44.1 kHz).AAC (Advanced Audio Coding) is the audio codec defined in MPEG-2 Part 7 (ISO/IEC 13818-7, standardised April 1997) and extended in MPEG-4 Part 3. A bare .aac file is the raw ADTS bitstream — frames concatenated with a small sync header on each one and nothing wrapping them. That makes .aac files the natural drop from streaming captures, TS broadcasts, and tools like FFmpeg's aac muxer, but they lack the metadata atoms that .m4a (the MPEG-4 container) provides. Cutting here keeps you in the raw ADTS format, so the result still plays in any AAC decoder without needing a container.
--extract-audio --audio-format aac output is raw ADTS. Trimming the dead first 10 seconds of an intro before sharing keeps the file in its original codec at close to the source bitrate..aac ADTS file you can cut to the exact span you need for an archival clip.| Property | AAC (.aac, raw ADTS) | M4A (AAC-in-MP4) | MP3 |
|---|---|---|---|
| Container | None — raw ADTS frames | MPEG-4 Part 14 (ISO/IEC 14496-14) | MPEG-1/2 Audio Layer III |
| Codec inside | AAC-LC / HE-AAC / HE-AAC v2 | Usually AAC, sometimes ALAC (lossless) | MP3 |
| Metadata / tags | Limited (ID3 sometimes works, not standard) | Full iTunes-style tags, cover art, chapters | ID3v1 / ID3v2 |
| Standardised | AAC: 1997 (MPEG-2) / 1999 (MPEG-4) | MP4 container: 2003 | 1993 |
| Typical use | Stream rips, broadcast TS demuxes, FFmpeg output | iTunes, Apple Music, Voice Memos, audiobooks | Universal distribution, podcasts, legacy players |
| Frame size | 1024 samples (or 960) per AAC block | Same — AAC inside | 1152 samples per MP3 frame |
| Cut behavior | Sample-accurate; re-encoded near source bitrate | Sample-accurate; re-encoded near source bitrate | Sample-accurate; re-encoded near source bitrate |
| Compression efficiency vs MP3 | ~20% smaller at equal perceived quality | Same — AAC is the codec | Baseline |
| Best when | You need a portable raw stream | You want metadata, art, chapters, Apple compatibility | You need playback everywhere, including a 2008 car stereo |
| Profile / bitrate | Typical use | 1-minute size | Notes |
|---|---|---|---|
| HE-AAC v2 @ 24-48 kbps | Streaming radio, voice | ~0.2-0.4 MB | Adds SBR + Parametric Stereo for low-bitrate efficiency |
| HE-AAC @ 64 kbps | Low-bandwidth music streams | ~0.5 MB | AAC-LC + Spectral Band Replication |
| AAC-LC @ 96 kbps stereo | Podcasts, lectures | ~0.7 MB | Light high-frequency softness |
| AAC-LC @ 128 kbps stereo | Default for many ripping apps, broadcast | ~0.9 MB | Roughly equal in quality to a 160-192 kbps MP3 |
| AAC-LC @ 256 kbps stereo | iTunes Plus / Apple Music delivery (since 2007) | ~1.9 MB | Effectively transparent for most listeners |
| AAC-LC @ 320+ kbps | Archival / mastering reference | ~2.3 MB | Diminishing returns above 256 kbps |
The cut is sample-accurate, and the kept audio is re-encoded as AAC at close to the source bitrate — one high-bitrate generation is inaudible in practice, but it is a re-encode, not a frame-for-frame copy of the ADTS stream. Re-encoding to a lower bitrate or a different sample rate adds real loss on top of that, so leave the codec on "Unchanged" if quality matters.
.aac is the raw ADTS bitstream — frames with a small sync header and nothing else. .m4a is the same AAC audio wrapped inside an MPEG-4 Part 14 container (the same container as .mp4, just audio-only), which adds atoms for metadata, cover art, chapter markers, and gapless playback hints. iTunes, Apple Music, the iPhone Files app, and most consumer players prefer .m4a. .aac shows up when AAC is the demuxed stream from a TS/MPEG transport, when streaming tools dump raw audio, or when a developer needs a container-free file. If you want tags and album art, see AAC to M4A.
It shouldn't be — the audio is decoded at your requested timestamps, so the cut is sample-accurate rather than snapping to a 1024-sample AAC frame boundary (~23 ms at 44.1 kHz, ~21 ms at 48 kHz). If a cut sounds offset, double-check the timestamp format you entered (plain seconds vs HH:MM:SS.sss), and note that a few milliseconds of encoder priming silence at the head of any AAC file is normal.
Yes, but be aware that HE-AAC (also called AAC+ or aacPlus) layers Spectral Band Replication on top of AAC-LC, and HE-AAC v2 adds Parametric Stereo on top of that. The cutter decodes both profiles fine, so cutting works the same. Some legacy players that only understand AAC-LC will play HE-AAC content as half-rate mono — that's a player limitation, not a cut artefact. Because the cut re-encodes, the output profile may not match a low-bitrate HE-AAC source exactly — for critical streaming use, check the result before distributing.
iPhone, iPad, and macOS natively decode AAC, since Apple adopted AAC as the iTunes default in April 2003. A raw .aac file will play in the Files app and QuickTime, but it won't carry artwork or appear nicely in Music. If you want the clip to land in the Music library with metadata, wrap it as M4A via AAC to M4A before importing. For ringtones, you also need to rename to .m4r after wrapping (Apple's ringtone container is M4A under a different extension, capped at 30 seconds).
Yes, once it's demuxed to a .aac file. AAC audio inside MPEG-2 transport streams (broadcast captures, Blu-ray rips) and MP4/MOV video usually demuxes cleanly to an ADTS .aac stream with FFmpeg's -c:a copy. Drop that file in and trim it like any other AAC. If you have the video file and want to extract+trim the audio in one step, use Audio Cutter, which accepts video inputs and outputs the audio-only segment.
A cut already shortens the duration proportionally — a 60-second clip cut to 20 seconds is roughly one-third the size at the same bitrate. To shrink further: drop the bitrate (256 kbps → 128 kbps roughly halves the file with little audible loss for casual listening), switch to mono for voice (cuts size by ~40-50%), or use HE-AAC at 64 kbps for a near-broadcast quality stream at a quarter of AAC-LC 256. For aggressive reduction without re-cutting, see compress AAC.
Yes — drop in the whole set and apply one start/duration to every file. This is the usual workflow for stripping a uniform intro (say, the first 8 seconds) off every episode of a podcast feed or removing a station ID from a stack of broadcast captures. Files download individually or as a ZIP archive. If your sources have different intro lengths, process them in smaller batches with the appropriate offsets.
Only if your playback target genuinely lacks AAC support. AAC is decoded natively by every modern browser, every iPhone since 2007, every Android since 1.0, CarPlay and Android Auto, every modern smart speaker, and the AAC profile is part of the Bluetooth A2DP spec. The honest case for MP3 is a pre-2005 hardware MP3 player, an older car stereo, or a podcast host that demands MP3 ingest. In that case, set the output codec to MP3 while cutting so the audio is encoded once at its final length — or cut first and then use AAC to MP3.