Convert TS to AIFF — Free Online Tool
Convert TS broadcast stream files to AIFF by extracting the audio track and re-encoding it as uncompressed PCM — perfect for pulling high-quality audio from broadcast recordings into a lossless format compatible with macOS and professional audio software. The conversion decodes whatever audio codec is embedded in the TS file (AAC, AC-3, MP3, or FLAC) and outputs a raw 16-bit big-endian PCM AIFF file with no further compression.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Settings
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
MPEG-2 Transport Streams typically carry audio encoded in a lossy format such as AAC or AC-3 — common in broadcast television and HLS streaming. During this conversion, FFmpeg demuxes the TS container to extract the audio elementary stream, decodes it to raw PCM samples in memory, and then re-encodes those samples as 16-bit signed big-endian PCM wrapped in an AIFF container. The video stream, subtitle tracks, and any secondary audio tracks present in the TS file are all discarded, since AIFF supports only a single uncompressed audio track. Because the source audio in a TS file is almost always lossy (AAC or AC-3), this process involves a decode-then-encode step rather than a lossless remux — the output AIFF will be uncompressed, but it cannot recover quality that was already lost during the original broadcast encoding.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool — in this browser-based tool, it runs as a WebAssembly binary entirely within your browser with no server upload. When running locally on your desktop, this calls your system-installed FFmpeg executable. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg will demux the TS container and make all embedded streams — video, audio tracks, subtitles — available for mapping or discard in subsequent steps. |
-c:a pcm_s16be
|
Decodes the audio stream found in the TS file (typically AAC or AC-3) and re-encodes it as 16-bit signed big-endian PCM — the standard uncompressed audio codec for AIFF files on macOS. No video codec flag is needed because AIFF cannot contain video, so FFmpeg automatically discards all video streams. |
output.aiff
|
Sets the output filename and tells FFmpeg to wrap the PCM audio in an AIFF container. FFmpeg infers the AIFF format from the .aiff file extension, applying the correct big-endian container structure expected by macOS and Apple audio applications. |
Common Use Cases
- Extracting dialogue or music from a recorded broadcast TV program to import into Logic Pro or Pro Tools on macOS, which natively prefer AIFF for project assets
- Pulling the audio track from an HLS-recorded TS segment for forensic audio analysis, where an uncompressed AIFF is required as a working copy
- Archiving the audio portion of a broadcast news recording into AIFF for long-term storage alongside a video-only file, avoiding any additional lossy compression cycle
- Preparing audio from a TS broadcast recording for mastering or post-production, where DAWs require uncompressed 16-bit PCM input rather than AAC or AC-3
- Converting a TS file from a digital TV capture card into AIFF so the audio can be edited frame-accurately in a macOS video editing workflow that handles audio separately
- Extracting a live concert broadcast recorded as a TS stream into AIFF to serve as a high-quality reference file before further processing or distribution
Frequently Asked Questions
No — TS files used in broadcast and streaming almost always carry lossy audio codecs such as AAC or AC-3. Converting to AIFF decodes that lossy stream into uncompressed PCM, which stops any further quality degradation from additional encode/decode cycles, but it cannot restore detail that was discarded during the original broadcast compression. The AIFF output will be a lossless representation of whatever quality the TS audio already had.
AIFF stores audio as raw, uncompressed PCM samples with no compression whatsoever. A broadcast TS file might carry AAC or AC-3 audio at 128–256 kbps, while the equivalent 16-bit stereo PCM in AIFF at 44.1 kHz runs at approximately 1,411 kbps — roughly 5 to 10 times larger for the audio alone. This size increase is expected and is inherent to the uncompressed nature of the AIFF format.
By default, FFmpeg selects the first audio stream it finds in the TS file, which is usually the primary language track. If your broadcast recording contains multiple audio tracks — for example, an original language track and a dubbed version — you can specify a particular stream by adding '-map 0:a:1' (for the second audio track) to the FFmpeg command before the output filename. This tool extracts the default first audio track automatically.
Yes. The default command uses '-c:a pcm_s16be' which produces 16-bit signed big-endian PCM — standard CD-quality depth. If you need higher bit depth for professional post-production, you can substitute 'pcm_s24be' for 24-bit or 'pcm_s32be' for 32-bit PCM by running the command locally on your desktop. Note that since broadcast TS audio is typically sourced from 16-bit or lossy-encoded streams, increasing bit depth will add headroom but not genuine additional audio resolution.
AIFF has limited metadata support compared to formats like MP4 or MKV, and TS broadcast metadata such as program information, channel names, or DVB timestamps is not carried over during this conversion. Basic ID3-style tags embedded in the TS audio stream may be partially mapped, but broadcast-specific metadata will be lost. If preserving program metadata is important, consider keeping the original TS file alongside the AIFF export.
FFmpeg processes one file at a time with the command shown, but you can batch-process files using a shell loop. On Linux or macOS, run: 'for f in *.ts; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.ts}.aiff"; done'. On Windows Command Prompt, use: 'for %f in (*.ts) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff"'. This is especially useful for processing large collections of recorded broadcast segments that exceed the 1GB browser limit.
Technical Notes
AIFF uses big-endian byte ordering for its PCM data, which is why the codec flag specifies 'pcm_s16be' (signed 16-bit big-endian) rather than the little-endian PCM used in WAV files. This big-endian convention is a legacy of AIFF's origins on Motorola 68000-based Macintosh hardware. TS files originating from DVB, ATSC, or HLS sources may contain audio in AAC-LC, HE-AAC, AC-3, E-AC-3, or occasionally MPEG-1 Layer 2 — all of which FFmpeg will decode automatically before encoding to PCM. One known limitation is that TS files with variable or discontinuous timestamps (common in imperfect broadcast recordings or concatenated HLS segments) can occasionally cause sync drift in the output; if this occurs, adding '-fflags +discardcorrupt' to the command can help. The output AIFF does not support chapters or subtitle data, so any DVB subtitle streams or teletext in the TS file are silently dropped. Sample rate is preserved from the source stream — if your TS carries 48 kHz audio (standard for broadcast), the AIFF will also be 48 kHz, not resampled to 44.1 kHz unless explicitly requested.