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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). Because AAC encodes audio in fixed blocks of 1024 samples, the smallest clean cut at 44.1 kHz is roughly 23 ms — XConvert rounds your start/end to the nearest frame boundary so playback stays in sync.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 in-place keeps you in the raw stream so the result still plays in any AAC decoder without a container rewrap.
--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 without a re-encode..aac ADTS file you can splice on frame boundaries 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 |
| Lossless cut accuracy at 44.1 kHz | ~23 ms (frame-aligned) | ~23 ms (frame-aligned) | ~26 ms (frame-aligned) |
| 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 |
If you keep the output as AAC at the same bitrate, sample rate, and channel count as the source, the cut is performed on ADTS frame boundaries — each AAC frame is a self-contained 1024-sample block that doesn't depend on its neighbours, so frames can be sliced out without decoding. At worst you lose up to one frame (~23 ms at 44.1 kHz) of accuracy at each cut. If you re-encode to a lower bitrate or change the sample rate, that's a separate lossy step on top of the cut. 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.
Because AAC is a block-based codec: each frame contains exactly 1024 PCM samples (or 960 in the LD/ELD profiles), which is about 23 ms at 44.1 kHz or 21 ms at 48 kHz. Lossless cuts have to land on a frame boundary — XConvert snaps your requested start and end to the nearest frame so the output stays in sync. If you need sample-accurate cuts, re-encode the AAC to PCM via AAC to WAV, trim, and re-encode — but that introduces a generation loss.
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 base frames are still 1024-sample AAC blocks, so frame-aligned 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. The output profile matches the source unless you explicitly re-encode.
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, cut first to preserve quality, then see AAC to MP3 — converting once at the final length avoids encoding twice.