Convert TS to AIF — Free Online Tool
Convert TS broadcast stream files to AIF (Audio Interchange File Format), extracting the audio track as uncompressed 16-bit big-endian PCM — the lossless format native to Apple and professional Mac audio workflows. Ideal for pulling clean audio from broadcast recordings, live stream captures, or HLS segments without any quality degradation.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
A TS file is a multiplexed container carrying video, audio (commonly AAC or AC3), and sometimes subtitle streams bundled together for broadcast transmission. Converting to AIF discards the video stream entirely and decodes the compressed audio codec (such as AAC or AC3) into raw, uncompressed PCM audio encoded as 16-bit signed big-endian samples — the standard PCM variant used by the AIF format. Because AIF is a purely audio container with no video support, FFmpeg demuxes the TS, selects the first audio stream, decodes it from its compressed form, and re-encodes it as pcm_s16be. This is a full audio transcode (decode then re-encode as PCM), not a remux, meaning the compressed audio data is decompressed to raw waveform data. The result is a lossless AIF file whose quality ceiling is bounded by the original compressed audio source.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no data leaves your machine. When running locally on your desktop for files over 1GB, this calls your system-installed FFmpeg executable. |
-i input.ts
|
Specifies the input TS (MPEG-2 Transport Stream) file. FFmpeg demuxes the container to access its individual streams — video, audio (typically AAC or AC3 in broadcast TS files), and any subtitle or data streams. |
-c:a pcm_s16be
|
Decodes the compressed audio stream from the TS (such as AAC or AC3) and re-encodes it as 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used by the AIF format developed by Apple. |
output.aif
|
Defines the output file as an AIF container. FFmpeg infers from the .aif extension that the output must be an Audio Interchange File Format container, which accepts big-endian PCM codecs and no video streams. |
Common Use Cases
- Extracting clean audio from a recorded broadcast TV segment to use in a professional Mac-based audio editing session in Logic Pro or Pro Tools
- Pulling audio from a captured live stream or HLS recording (stored as .ts) for archival in a lossless format that preserves every detail of the original decoded signal
- Converting AC3 or AAC audio from a broadcast TS file into an uncompressed AIF file for use in Apple-ecosystem workflows where AIF is the native lossless interchange format
- Preparing broadcast audio from a .ts recording for mastering or post-production, where downstream tools require uncompressed PCM rather than a lossy compressed codec
- Archiving the audio component of a digitized broadcast recording in a format that won't introduce additional generation loss if re-encoded later
- Delivering audio from a broadcast source to a client or collaborator using Mac-based tools who expects AIF as the standard exchange format
Frequently Asked Questions
The conversion involves decoding the compressed audio in your TS file (typically AAC or AC3) into uncompressed PCM, which is then stored in the AIF container. This decode step preserves all the audio information that existed in the compressed stream — no additional quality is lost during the TS-to-AIF conversion itself. However, any quality loss that occurred when the original broadcast audio was compressed into AAC or AC3 is already baked in and cannot be recovered. AIF simply stores the decoded result without further degradation.
The audio in a TS file is typically compressed using a lossy codec like AAC (which might use 128–256 kbps) or AC3 (384–640 kbps). AIF stores audio as raw uncompressed PCM, which for stereo 16-bit audio at 48 kHz — a common broadcast sample rate — requires approximately 1.5 MB per second of audio. A one-hour TS file with 192 kbps AAC audio might produce an AIF file ten to fifteen times larger, purely because PCM carries the full uncompressed waveform data.
They are discarded entirely. AIF is a pure audio format with no capacity to store video, subtitle, or chapter data. FFmpeg automatically selects only the first audio stream from the TS file and ignores all other streams. If your TS file contains multiple audio tracks (a common broadcast feature), only the default or first audio track will be extracted unless you modify the command to select a specific track using the -map flag.
By default, the command extracts the first audio stream. To target a specific audio track, add a -map flag between the input and output arguments, such as: ffmpeg -i input.ts -map 0:a:1 -c:a pcm_s16be output.aif — where 0:a:1 selects the second audio stream (zero-indexed). You can identify available audio streams and their indices by running ffmpeg -i input.ts and reading the stream listing in the output.
Yes. The command uses pcm_s16be (16-bit signed big-endian PCM) by default, which is the AIF standard. If your source audio was recorded or broadcast at higher fidelity and you want to preserve more headroom, you can switch to 24-bit by changing the codec flag: ffmpeg -i input.ts -c:a pcm_s24be output.aif. The pcm_s24be codec produces a 24-bit AIF file, which is compatible with professional DAWs on Mac and results in a proportionally larger file.
On macOS or Linux, you can loop over all TS files in a directory with a shell one-liner: for f in *.ts; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.ts}.aif"; done. On Windows Command Prompt, use: for %f in (*.ts) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". Each TS file will be converted independently, extracting its first audio track as a separate AIF file. This is particularly useful when archiving a series of recorded broadcast segments.
Technical Notes
AIF uses big-endian byte ordering for its PCM data, which reflects its origins in the Motorola 68000 architecture of early Apple hardware — this is why the codec is pcm_s16be (signed 16-bit big-endian) rather than the little-endian pcm_s16le used by WAV. TS files sourced from broadcast or HLS streams frequently carry audio at 48 kHz sample rate rather than the 44.1 kHz common in consumer music, and FFmpeg will preserve the source sample rate in the AIF output without resampling unless you explicitly add a -ar flag. Metadata from the TS file (program names, stream identifiers, broadcast timestamps) is not carried over to AIF, as the format has very limited metadata support compared to modern containers. If your TS file carries AC3 (Dolby Digital) surround audio, FFmpeg will decode it fully and write all channels into the AIF file — a 5.1 AC3 track will produce a six-channel AIF, which not all applications handle gracefully, so you may want to add -ac 2 to downmix to stereo. FLAC, an alternative lossless format, would produce smaller files than AIF for the same audio content, but AIF is specifically preferred in Apple-native and legacy Mac professional audio environments where it serves as the standard interchange format.