Extract Audio from TS to AIFC — Free Online Tool

Extract audio from MPEG-2 Transport Stream (.ts) files and convert it to AIFC format using the PCM S16BE codec — a big-endian, uncompressed 16-bit audio format favored in professional Apple-ecosystem workflows. This tool is ideal for pulling broadcast-quality audio from TS streams and saving it in a lossless, archival-grade container.

FFmpeg Command

Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg

Free — no uploads, no signups. Your files never leave your browser.

Estimated output:

Conversion Complete!

Download

How It Works

TS files are broadcast containers that commonly carry AAC, AC3, or MP3 audio alongside video. During this conversion, FFmpeg discards the video stream entirely and re-encodes the audio to PCM signed 16-bit big-endian (pcm_s16be) — the default codec for AIFC. Unlike simple remuxing, this process fully decodes the compressed audio (e.g., AAC or AC3 from the TS stream) and re-encodes it as uncompressed PCM, which eliminates any lossy codec artifacts but increases file size significantly. The result is an AIFC file containing raw, uncompressed audio data in big-endian byte order, compatible with Logic Pro, Final Cut Pro, and other Apple professional audio tools.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg program. In the browser version, this runs via FFmpeg.wasm (WebAssembly), so no installation is needed and no files leave your device.
-i input.ts Specifies the input MPEG-2 Transport Stream file. FFmpeg will parse the TS container, identify all available streams (video, audio, subtitles), and make them available for processing.
-vn Disables video output entirely, telling FFmpeg to ignore all video streams from the TS file. Since the goal is audio extraction to AIFC — a pure audio format — no video data is written to the output.
-c:a pcm_s16be Sets the audio codec to PCM Signed 16-bit Big-Endian, which is the standard uncompressed codec for AIFC files. This decodes the compressed audio from the TS stream (e.g., AAC or AC3) and re-encodes it as raw PCM in the byte order expected by the AIFC/AIFF format specification.
-b:a 128k Specifies a target audio bitrate of 128 kilobits per second. For uncompressed PCM codecs like pcm_s16be, the actual bitrate is determined by sample rate and bit depth rather than this parameter, but the flag is included for compatibility. The true bitrate of 16-bit stereo at 48kHz PCM is approximately 1536 kbps regardless of this setting.
output.aifc Defines the output filename and tells FFmpeg to wrap the PCM audio in an AIFC container. FFmpeg infers the container format from the .aifc file extension, which triggers the appropriate AIFF/AIFC muxer.

Common Use Cases

  • Extracting clean, uncompressed audio from a broadcast TV recording (.ts) for use in a professional audio editing session in Logic Pro or Pro Tools
  • Archiving the audio track from an HLS or DVB transport stream in a lossless AIFC format for long-term preservation
  • Pulling dialogue or music beds from a recorded live stream (TS capture) to use in a professional Apple-ecosystem post-production pipeline
  • Converting broadcast AC3 or AAC audio embedded in a TS file into PCM AIFC for sample-accurate editing without generational quality loss
  • Extracting audio from a transport stream recorded off a capture card to deliver a lossless master file to a sound engineer working on macOS

Frequently Asked Questions

The conversion involves one decode step: the compressed audio (typically AAC or AC3) inside the TS container is fully decoded to raw PCM before being written to AIFC. Any quality loss that occurred when the original lossy codec was applied cannot be recovered, but no additional quality is lost in this conversion. The AIFC output is an uncompressed, lossless representation of whatever audio was decoded from the TS stream, making it an ideal format for further editing or archiving.
The TS file stores audio in a compressed format like AAC (which is highly efficient), while AIFC with PCM S16BE stores raw, uncompressed audio samples. For example, AAC at 128k bitrate is roughly 10–15x smaller than 16-bit PCM at a standard sample rate like 48kHz stereo. This size increase is expected and is the trade-off for having a fully uncompressed, editable audio master.
PCM S16BE stands for Pulse Code Modulation, Signed 16-bit, Big-Endian — meaning each audio sample is stored as a 16-bit signed integer with the most significant byte first. AIFC inherits this byte order from the AIFF format, which was designed by Apple in the 1980s on Motorola 68000-based Macs that used big-endian architecture. Modern x86 and ARM systems are little-endian, but AIFC preserves this convention for compatibility with tools and hardware that expect the Apple/AIFF byte order.
Yes. TS files commonly carry multiple audio tracks — for example, different languages or commentary streams in a broadcast. To select a specific track, you can modify the FFmpeg command by adding '-map 0:a:1' (for the second audio track, zero-indexed) before the output filename. Without a map flag, FFmpeg selects the default audio stream, which is usually the first one detected in the TS file.
For PCM codecs like pcm_s16be, the '-b:a' bitrate flag has limited effect since the actual bitrate is determined by the sample rate and bit depth rather than a compression target. To get higher bit depth, change the codec flag to '-c:a pcm_s24be' for 24-bit or '-c:a pcm_s32be' for 32-bit audio — these are all valid AIFC codecs. If your source TS audio was recorded at 48kHz, 24-bit AIFC will give you a more accurate representation of the full dynamic range.
Yes. On Linux or macOS, you can wrap the command in a shell loop: 'for f in *.ts; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.ts}.aifc"; done'. On Windows PowerShell, use: 'Get-ChildItem *.ts | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a pcm_s16be -b:a 128k ($_.BaseName + ".aifc") }'. This is especially useful when processing large TS archives locally, where the 1GB browser limit might be a constraint.

Technical Notes

AIFC (Audio Interchange File Format Compressed) is a container that supports both uncompressed PCM and a range of compressed codecs, but in professional practice it is almost always used with PCM variants. The pcm_s16be codec used here produces standard CD-quality audio (when the source is 44.1kHz) or broadcast-quality audio (at 48kHz, the standard sample rate in TS streams). Notably, FFmpeg preserves the source sample rate from the TS file — it does not resample unless explicitly instructed — so if your TS stream carries 48kHz audio, the AIFC output will also be 48kHz. Metadata embedded in the TS container (such as stream language tags or program information) is generally not carried over to AIFC, since the format has limited metadata support compared to containers like MKV or MP4. AIFC does not support multiple audio tracks — it is a single-stream audio file — so if your TS source has multiple audio programs, only the default stream will be extracted unless you specify otherwise. The format is natively supported on macOS and in Apple professional applications including Final Cut Pro and Logic Pro, but may require additional codec support on Windows or Linux systems.

Related Tools