Extract Audio from TS to AIF — Free Online Tool
Extract audio from a TS (MPEG-2 Transport Stream) broadcast file and save it as a lossless AIF file, decoding the source audio — typically AAC or AC3 — into uncompressed PCM at 16-bit big-endian resolution. This is ideal for broadcast professionals who need pristine, uncompressed audio from TV recordings or live stream captures for editing, archiving, or mastering.
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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 are broadcast containers that typically carry AAC, AC3, or MP3 audio streams alongside video. During this conversion, FFmpeg discards all video streams entirely and decodes the compressed audio track — whatever codec it uses in the source TS — into raw PCM samples, then writes them into an AIF container using the pcm_s16be codec (16-bit signed big-endian PCM). This is a full decode-and-re-encode of the audio: the compressed bitstream from the TS is fully decompressed to uncompressed audio. Because AIF is a lossless, uncompressed format, no further quality is lost after the initial decode — the output represents the full fidelity recoverable from the source compressed audio. The big-endian byte order of pcm_s16be is native to the AIF format specification, ensuring maximum compatibility with Apple and professional audio software.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles the full decode-remux pipeline — reading the TS container, demuxing the audio stream, decoding the compressed audio (AAC or AC3), and encoding it as PCM into the AIF output. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg parses the TS container, identifies all program streams inside it (video, audio, subtitles), and makes them available for mapping and processing. |
-vn
|
Disables all video output, discarding every video stream present in the TS file. Since AIF is an audio-only format and we are extracting audio, this prevents FFmpeg from attempting to encode or include the video track in the output. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — 16-bit signed big-endian PCM — which is the standard uncompressed audio encoding for the AIF format. This fully decodes the source TS audio (typically AAC or AC3) into raw uncompressed samples stored in big-endian byte order, as required by the AIF specification. |
output.aif
|
Specifies the output filename with the .aif extension, which tells FFmpeg to write an Audio Interchange File Format container. The .aif extension triggers FFmpeg's AIFF muxer, which wraps the pcm_s16be audio stream in a valid AIF file compatible with macOS, Logic Pro, Pro Tools, and other Apple-ecosystem and professional audio tools. |
Common Use Cases
- Extracting commentary or dialogue from a broadcast TV recording captured as a TS file for use in a video editing or audio mastering session on macOS using Logic Pro or Final Cut Pro
- Pulling clean stereo or surround audio from a live sports broadcast TS capture to archive as an uncompressed AIF master before distributing to editors
- Converting the AC3 or AAC audio track from a recorded DVB or satellite TS stream into a standard AIF file for import into Pro Tools or other professional DAWs that work natively with AIF
- Extracting music or sound effects from a broadcast TS file and saving as lossless AIF for re-use in post-production without introducing additional compression artifacts
- Stripping audio from a large HLS-captured TS segment for analysis or transcription, where a simple uncompressed PCM file is easier to process programmatically
- Archiving the audio from a recorded broadcast event in a lossless AIF format to ensure long-term preservation without lossy codec degradation
Frequently Asked Questions
The TS file's audio track is almost always compressed — typically using AAC or AC3 — so some quality loss occurred when that broadcast was originally encoded. This conversion fully decodes that compressed audio to uncompressed PCM and stores it in AIF, so no additional quality is lost in the conversion step itself. The AIF output is as faithful as possible to what the compressed source contains. If you need truly lossless audio end-to-end, the original broadcast recording would have needed to capture uncompressed audio in the first place.
By default, FFmpeg selects the first audio stream it finds in the TS file, which is typically the primary audio track. AIF does not support multiple audio tracks in a single file, so only one track can be extracted per conversion. If you need a specific track — for example, a secondary language or a separate surround mix — you can add the flag '-map 0:a:1' (for the second audio stream) to the FFmpeg command before the output filename to target a specific track by index.
TS files store audio in compressed formats like AAC (typically 128–192 kbps) or AC3 (typically 192–640 kbps), which are dramatically smaller than uncompressed audio. AIF with pcm_s16be stores audio as raw 16-bit PCM at the full sample rate — for example, stereo audio at 48kHz uses approximately 5.5 MB per minute. A 30-minute broadcast recording could easily expand from a few hundred MB of compressed TS audio to over 1 GB as an AIF file. This is expected behavior for any conversion to an uncompressed format.
Yes, if the source TS file contains a 5.1 AC3 or AAC surround audio track, FFmpeg will decode all six channels and write them into the AIF file as multichannel PCM. However, compatibility with multichannel AIF files varies across applications — most professional DAWs like Pro Tools and Logic Pro handle them correctly, but some consumer audio players may only play back the stereo downmix or the first two channels.
Yes. The default command uses pcm_s16be (16-bit big-endian PCM), which is the AIF standard and compatible with virtually all software. To get 24-bit resolution, change '-c:a pcm_s16be' to '-c:a pcm_s24be' in the command: 'ffmpeg -i input.ts -vn -c:a pcm_s24be output.aif'. This is worth doing if you intend to do further audio processing or mastering, as it provides headroom and avoids quantization artifacts during editing. Note that the source TS audio is typically only 16-bit, so 24-bit output will not recover information that was not in the source.
On macOS or Linux, you can run a simple shell loop: 'for f in *.ts; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.ts}.aif"; done'. On Windows Command Prompt, use: 'for %f in (*.ts) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif"'. This processes every TS file in the current directory and creates a matching AIF file for each. The browser-based tool processes one file at a time, so the FFmpeg command is particularly valuable for bulk processing large collections of broadcast recordings.
Technical Notes
TS (MPEG-2 Transport Stream) is designed for robust broadcast transmission and can carry multiple audio tracks with different codecs — most commonly AAC (the default for DVB-T2 and HLS) and AC3/Dolby Digital (common in North American ATSC broadcasts). Both are lossy compressed formats, meaning this conversion involves a lossy-to-lossless transcode: the compressed audio is fully decoded and written as uncompressed PCM into AIF. The pcm_s16be codec (16-bit signed big-endian PCM) is the canonical AIF audio codec, specified in Apple's original AIF format documentation, and is why AIF files use big-endian byte order — unlike WAV, which uses little-endian PCM. Metadata handling is limited: TS broadcast streams carry MPEG metadata and service information that does not map to AIF's AIFF-C chunk structure, so most metadata (channel name, broadcast timestamps, program info) will not transfer to the output file. Subtitle tracks embedded in the TS are dropped entirely, as AIF has no subtitle support. If the source TS contains multiple audio programs (common in multi-channel broadcast streams), only the default stream is extracted unless explicitly mapped. Sample rates from broadcast TS files are typically 48kHz, which is preserved in the AIF output and is the standard for professional audio workflows.