Extract Audio from TS to AIFF — Free Online Tool
Extract audio from a TS (MPEG-2 Transport Stream) broadcast file and save it as a lossless AIFF file. The audio is decoded from its original compressed format (typically AAC or AC3) and re-encoded to uncompressed PCM — ideal for archiving broadcast audio at full quality.
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FFmpeg Command
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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
TS files used in broadcast and streaming typically carry compressed audio streams — most commonly AAC or AC3. AIFF is an uncompressed, lossless format that stores raw PCM audio data. This means the conversion is not a simple remux: the compressed audio in the TS container must be fully decoded and then re-encoded as uncompressed 16-bit big-endian PCM (pcm_s16be), which is the standard PCM encoding AIFF uses. The video stream is discarded entirely. The result is a larger file than the source audio, but one that contains no further lossy compression — making it suitable for editing, archiving, or professional audio workflows on macOS.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In this browser-based tool, it runs via FFmpeg.wasm compiled to WebAssembly — the same command works identically on a desktop installation of FFmpeg for files over 1GB. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg reads the TS container and demuxes its elementary streams — typically one or more video PIDs and one or more audio PIDs encoded in AAC, AC3, or similar broadcast codecs. |
-vn
|
Disables video output entirely. Since AIFF is an audio-only format, this flag drops the video stream from the TS file and prevents FFmpeg from attempting to encode it — without this, the command would fail because AIFF has no video codec support. |
-c:a pcm_s16be
|
Decodes the compressed broadcast audio from the TS (typically AAC or AC3) and re-encodes it as signed 16-bit big-endian PCM — the standard uncompressed audio encoding used inside AIFF files. Big-endian byte order is required by the AIFF specification, distinguishing it from WAV which uses little-endian PCM. |
output.aiff
|
Defines the output filename and tells FFmpeg to write an AIFF container. The .aiff extension triggers the correct muxer, producing a file natively compatible with macOS audio applications like Logic Pro, GarageBand, and Final Cut Pro. |
Common Use Cases
- Archiving the audio from a broadcast television recording (DVR or IPTV capture) into a lossless format for long-term preservation
- Extracting a music performance or live concert captured from a broadcast TS file into AIFF for import into a DAW like Logic Pro or Pro Tools
- Converting HLS-segmented TS audio content into a single lossless AIFF file for offline editing
- Pulling the audio track from a broadcast news clip into AIFF for transcription or audio mastering
- Preparing broadcast audio for professional sound design work on macOS, where AIFF is natively supported in apps like GarageBand and Final Cut Pro
- Extracting and decompressing AC3 5.1 surround audio from a TS file into a lossless PCM AIFF for further processing or downmixing
Frequently Asked Questions
No — if the original TS file contains compressed audio (such as AAC or AC3), that lossy compression has already permanently removed some audio data. Converting to AIFF decodes and stores what remains as uncompressed PCM, so no additional quality is lost in this step, but the quality ceiling is set by the original compressed source. AIFF is best thought of as a lossless container for whatever quality already exists in the TS file.
Broadcast TS files compress audio using codecs like AAC (typically 128–256 kbps) or AC3 (384–640 kbps), which achieve significant file size reduction. AIFF stores audio as raw uncompressed PCM at 16-bit depth, which requires roughly 10 MB per minute per channel at 44.1 kHz. A 1-hour broadcast audio track that was only a few hundred MB as AAC could expand to several GB as AIFF — this is normal and expected for lossless uncompressed audio.
By default, FFmpeg selects the first (or best-ranked) audio stream from the TS file. TS files commonly carry multiple audio tracks for different languages or accessibility purposes. If you need a specific track, you can add -map 0:a:1 to the command to select the second audio stream (zero-indexed), replacing the default stream selection. AIFF only supports a single audio track, so only one stream can be output.
Yes. The default command uses pcm_s16be (16-bit big-endian PCM), but AIFF also supports higher bit depths. Replace -c:a pcm_s16be with -c:a pcm_s24be for 24-bit output, or pcm_s32be for 32-bit. For broadcast sources that were originally 16-bit AAC or AC3, 24-bit output offers no real benefit — however, if the source was recorded or mastered at higher bit depth before encoding into the TS, using 24-bit AIFF preserves more headroom during further editing.
Standard metadata such as title or artist tags may be carried over if present in the TS stream, but broadcast-specific metadata like program clock references, PID timing, or EPG data in the TS container is not preserved in AIFF. Surround channel layout information (e.g., from an AC3 5.1 track) may or may not be retained depending on how FFmpeg maps the channels — for critical surround work, inspect the output with ffprobe to verify the channel layout.
On macOS or Linux you can run a shell loop: for f in *.ts; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.ts}.aiff"; done. On Windows Command Prompt, use: for %f in (*.ts) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". This applies the same extraction and PCM conversion to every TS file in the current directory, which is especially useful when processing multiple broadcast recordings or HLS segments.
Technical Notes
AIFF uses big-endian byte ordering, which is why FFmpeg uses the pcm_s16be codec (signed 16-bit big-endian PCM) rather than the little-endian pcm_s16le used in WAV files. TS containers most commonly carry AAC audio at 128–192 kbps or AC3 at 384 kbps — both are lossy codecs, so the conversion to AIFF is a lossy-to-lossless transcode, not a lossless-to-lossless operation. The -vn flag is critical here: without it, FFmpeg would attempt to encode the video stream into AIFF, which is an audio-only format and would cause an error. TS files may also contain DVB or SCTE subtitles and multiple audio PIDs — these are all ignored during this extraction. One known limitation is that TS files with variable or corrupted PCR timestamps (common in live stream captures or partial recordings) may cause FFmpeg to produce audio with sync drift or gaps; using -fflags +discardcorrupt can help in such cases. The output sample rate will match the source audio's sample rate (typically 44.1 kHz or 48 kHz for broadcast), and no sample rate conversion is applied unless explicitly requested.