Convert TS to AIFC — Free Online Tool
Convert MPEG-2 Transport Stream (TS) broadcast files to AIFC audio by extracting and encoding the audio track as big-endian PCM using the pcm_s16be codec. This is ideal for pulling broadcast-quality audio from TS recordings into a professional audio format compatible with legacy Apple and pro audio workflows.
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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
During this conversion, FFmpeg demuxes the TS container — which may contain multiplexed video, audio, and subtitle streams — and extracts only the audio track. The audio (commonly AAC or AC-3 in broadcast TS files) is decoded from its original compressed format and re-encoded as uncompressed 16-bit big-endian PCM (pcm_s16be), then wrapped in the AIFC container. This is a full decode-and-re-encode of the audio stream, not a remux. The video stream and any subtitle or secondary audio tracks are discarded, since AIFC supports only a single audio track and no video. The result is a lossless PCM representation of the audio content, though any quality lost in the original TS audio compression (e.g., AAC encoding) cannot be recovered.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the underlying engine that performs all demuxing, decoding, encoding, and muxing. This is the same engine running in your browser via WebAssembly. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg will parse the TS container, identify all streams (video, audio, subtitles), and make them available for mapping into the output. |
-c:a pcm_s16be
|
Sets the audio encoder to 16-bit signed big-endian PCM, which is the standard uncompressed audio codec for the AIFC format. The decoded TS audio (e.g., AAC or AC-3) is re-encoded into this raw PCM format, producing lossless output from this point forward. |
-b:a 128k
|
Nominally sets the target audio bitrate to 128 kbps. For a fixed-rate lossless codec like pcm_s16be, the actual bitrate is determined by sample rate, bit depth, and channel count rather than this parameter — it is effectively ignored by the PCM encoder but included for consistency. |
output.aifc
|
Defines the output filename and tells FFmpeg to write an AIFC container. FFmpeg infers the container format from the '.aifc' extension and automatically discards the video and subtitle streams from the TS input, since AIFC supports audio only. |
Common Use Cases
- Extracting the audio from a recorded broadcast TV program (captured as a TS file) for archiving or editing in a DAW that requires AIFC or AIFF input
- Pulling a clean audio track from a live-streamed or DVR-recorded TS file for use in post-production workflows on older macOS or ProTools setups that expect big-endian PCM
- Converting HLS-sourced TS segments into a single uncompressed AIFC file for audio analysis or quality inspection
- Archiving the audio commentary or dialogue track from a broadcast TS recording in an uncompressed, durable format for long-term storage
- Preparing broadcast audio from a TS file for ingestion into legacy Apple Final Cut Pro 7 or Logic Pro projects that natively handle AIFC
- Decoding AC-3 or AAC surround audio from a TS file to a flat stereo PCM AIFC for delivery to a client who requires uncompressed audio without a video container
Frequently Asked Questions
The conversion decodes whatever compressed audio codec is in your TS file (typically AAC or AC-3) and re-encodes it as uncompressed 16-bit PCM. The PCM output itself is lossless at 16-bit depth, but any quality degradation introduced during the original lossy compression of the TS audio is permanent. No additional loss occurs in this conversion step, but the output quality ceiling is bounded by the original TS audio quality.
TS files store audio in compressed formats like AAC or AC-3, which dramatically reduce file size. AIFC with pcm_s16be stores audio as raw uncompressed PCM, which requires approximately 10 MB per minute for stereo 44.1 kHz audio. A TS file with AAC audio might use 128–192 kbps for audio, while the equivalent pcm_s16be AIFC will use around 1411 kbps for CD-quality stereo — roughly a 7–11x size increase for the audio portion alone.
Both are discarded entirely. AIFC is an audio-only container and cannot hold video, chapter markers, or subtitle tracks. FFmpeg automatically drops the video and subtitle streams when writing to AIFC. If you need to preserve video or subtitles, AIFC is not an appropriate target format — consider MP4 or MKV instead.
By default, FFmpeg selects the first audio stream detected in the TS file, which is typically the primary audio track. AIFC does not support multiple audio tracks, so only one track can be written to the output. If you need a specific audio track other than the default, you can modify the FFmpeg command by adding '-map 0:a:1' (for the second audio track) before the output filename to select it explicitly.
Replace '-c:a pcm_s16be' with '-c:a pcm_s24be' in the command to output 24-bit big-endian PCM, which preserves more dynamic range and is preferred for professional audio mastering. Other valid options for AIFC include 'pcm_s32be' for 32-bit integer, 'pcm_f32be' for 32-bit float, and 'pcm_alaw' or 'pcm_mulaw' for telephony-grade compressed PCM. The full modified command would be: ffmpeg -i input.ts -c:a pcm_s24be output.aifc
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.ts; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.ts}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.ts) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc"'. This processes each TS file in the current directory and outputs a corresponding AIFC file with the same base name.
Technical Notes
AIFC (Audio Interchange File Format Compressed) uses big-endian byte ordering inherited from its AIFF roots and the Motorola 68000 architecture of early Macs, which is why the codec is named pcm_s16be — signed 16-bit big-endian PCM. This is distinct from WAV's little-endian PCM. When extracting audio from a TS file, the source audio codec matters: AAC is the most common in HLS-derived TS files, while AC-3 (Dolby Digital) is typical in broadcast DVB or ATSC TS recordings. Both will be fully decoded before re-encoding to PCM, so channel layout and sample rate from the source are preserved in the output by default. Note that if the source TS contains AC-3 5.1 surround audio, the AIFC output will retain all 6 channels in PCM — file sizes will be proportionally larger. The '-b:a 128k' flag has no practical effect on lossless PCM codecs like pcm_s16be (bitrate is determined by sample rate, bit depth, and channel count), but it is harmless. Metadata such as program name or broadcast timestamps embedded in the TS container will not be carried over to AIFC, as the format has limited metadata support compared to modern containers.