Extract Audio from TS to WAV — Free Online Tool
Extract the audio track from a TS (MPEG-2 Transport Stream) broadcast file and save it as an uncompressed PCM WAV file. This tool discards the video stream entirely and decodes the source audio — whether AAC, AC-3, MP3, or another codec — into raw 16-bit PCM, giving you a lossless-quality WAV ready for audio editing, archiving, or broadcast workflows.
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
MPEG-2 Transport Streams typically carry audio encoded as AAC, AC-3, or MP3 alongside video. During this conversion, FFmpeg demuxes the TS container to isolate the primary audio stream, then decodes that compressed audio codec into raw PCM samples. The decoded PCM data is then written into a WAV container using the pcm_s16le codec — 16-bit signed little-endian samples at the source file's original sample rate and channel count. This is a full decode-and-re-encode step (from lossy compressed to uncompressed PCM), so the resulting WAV captures the best fidelity possible from the source, though any quality loss from the original lossy encoding in the TS file cannot be recovered. The video stream is explicitly dropped using the -vn flag, so none of the TS video data is processed or written.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all demuxing, decoding, and encoding. In the browser-based version of this tool, FFmpeg.wasm runs this same engine compiled to WebAssembly, so the command is identical to what you would run locally. |
-i input.ts
|
Specifies the input file — an MPEG-2 Transport Stream. FFmpeg will demux the TS container to access its individual streams, which typically include one or more video tracks and one or more audio tracks encoded in AAC, AC-3, or MP3. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore all video streams in the TS file. Since the goal is audio-only extraction, this prevents any video processing and ensures the output WAV file contains no video data. |
-c:a pcm_s16le
|
Decodes the TS audio stream (AAC, AC-3, MP3, or whatever codec the source uses) and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio format for WAV files. This produces fully lossless WAV output from the decoded audio data. |
output.wav
|
Specifies the output file as a WAV container. FFmpeg infers the WAV format from the .wav extension and writes the raw PCM audio samples with a standard WAV header, making the file immediately compatible with audio editors, DAWs, and broadcast tools. |
Common Use Cases
- Extracting the audio commentary track from a recorded broadcast TV program (.ts file from a PVR or DVR) to edit in a DAW like Audacity or Adobe Audition
- Pulling the audio from an HLS-recorded live stream segment stored as a TS file for transcription or captioning workflows
- Converting broadcast AC-3 or AAC audio from a TS recording into an uncompressed WAV for submission to a post-production house that requires PCM deliverables
- Archiving the audio from a television capture to WAV for long-term storage, since WAV is universally supported and requires no codec to decode
- Isolating a music performance or live event audio from a broadcast TS recording for mastering or remixing
- Extracting dialogue or voiceover audio from a broadcast segment to use as source material for a podcast or radio production
Frequently Asked Questions
The WAV output is uncompressed PCM, so no additional quality is lost during the WAV encoding step itself. However, the audio in a TS file is almost always stored in a lossy format like AAC or AC-3, meaning some quality was lost when that original recording was made. The conversion faithfully decodes that compressed audio to PCM — you get the best quality the TS file contains — but the losses baked into the source codec cannot be undone.
By default, FFmpeg selects the first audio stream in the TS file, which is typically the primary language track. If you need a specific track, you can modify the command by adding -map 0:a:1 (for the second audio track, zero-indexed) before the output filename. TS files from broadcast sources often carry multiple audio programs, so checking the stream list with ffmpeg -i input.ts first is recommended if you're unsure.
WAV with pcm_s16le stores raw, uncompressed audio samples — there is no compression applied at all. A one-hour stereo WAV at 44.1kHz and 16-bit depth is approximately 600MB regardless of how short or quiet the audio is. The TS file's audio was compressed using AAC or AC-3 at a much lower bitrate (often 128–384 kbps), so the WAV output can be 5–15 times larger than the audio portion of the source TS file.
Yes. The default command uses pcm_s16le (16-bit signed little-endian), which is standard for most applications. To get 24-bit PCM, change -c:a pcm_s16le to -c:a pcm_s24le in the command. For 32-bit float PCM (common in professional DAW workflows), use -c:a pcm_f32le. Note that since the TS source audio is typically 16-bit or lossy, going above 16-bit adds no real fidelity — it only increases file size.
On Linux or macOS, you can run: for f in *.ts; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.ts}.wav"; done — this loops over every .ts file in the current directory and produces a matching .wav file. On Windows Command Prompt, use: for %f in (*.ts) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". The browser-based tool processes one file at a time, so the command-line approach is recommended for batch jobs or files over 1GB.
WAV has limited metadata support compared to TS. Basic tags like title or artist may be carried over by FFmpeg if the WAV file supports them via INFO chunks, but broadcast-specific TS metadata such as program IDs, service names, and DVB stream descriptors will not be preserved. Channel layout (stereo, 5.1 surround) is preserved in the WAV header as long as the source audio is stereo or a standard layout that pcm_s16le supports.
Technical Notes
TS (MPEG-2 Transport Stream) is a multiplexed container commonly carrying H.264 or H.265 video alongside AAC, AC-3, or MP3 audio at broadcast bitrates. The audio extraction process requires FFmpeg to fully decode the compressed audio codec before writing PCM output — unlike a container-to-container remux, this is always a transcode step. The output uses pcm_s16le: 16-bit signed little-endian PCM, the most universally compatible WAV subformat, readable by virtually every audio editor, NLE, and broadcast tool. If the TS file contains AC-3 5.1 surround audio, FFmpeg will decode all six channels and write them into the WAV file; however, WAV's channel mapping for surround formats should be verified in your target application. TS files from DVRs or capture cards occasionally have slight timestamp discontinuities or broken PTS values in the audio stream, which can cause clicks or sync drift in the output — adding -fflags +discardcorrupt to the command can help mitigate this. The WAV format does not support chapters or subtitle streams, so any subtitle or chapter data in the TS file is silently discarded during conversion.