Convert TS to WAV — Free Online Tool
Extract and convert audio from MPEG-2 Transport Stream (.ts) broadcast files into uncompressed WAV format using the PCM signed 16-bit little-endian codec. Ideal for archiving, editing, or broadcasting audio sourced from digital TV recordings, HLS streams, or live capture files.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your TS file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
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
A TS file is a broadcast-oriented container that multiplexes video, audio (commonly AAC, AC3, or MP3), and sometimes subtitles and multiple audio tracks into a single stream. During this conversion, FFmpeg demuxes the TS container, discards all video and subtitle streams, and transcodes the audio — whatever codec it was encoded in — into raw PCM signed 16-bit little-endian audio, which is the standard uncompressed encoding used in WAV files. Because TS audio is typically compressed (AAC or AC3), this is a full decode-and-re-encode pass, not a lossless remux. The result is an uncompressed WAV file that is bit-perfect in the PCM domain, though the original compression's artifacts from the source TS are baked in.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary, the open-source multimedia processing engine that handles demuxing the TS container, decoding the compressed broadcast audio stream, and encoding the output as uncompressed PCM WAV. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg reads and parses the TS packet structure, identifies the Program Map Table, and locates the audio elementary streams embedded in the broadcast container. |
-c:a pcm_s16le
|
Sets the audio codec for the output to PCM signed 16-bit little-endian, which is the standard uncompressed audio encoding used in WAV files. This decodes whatever compressed audio codec was in the TS (such as AAC or AC3) into raw PCM samples, making the output suitable for editing and broadcast workflows. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to wrap the PCM audio in a RIFF WAV container, producing a widely compatible uncompressed audio file that can be opened in virtually any audio or video editing application. |
Common Use Cases
- Extracting the audio track from a recorded over-the-air or cable TV broadcast (.ts file) for archiving or transcription purposes
- Pulling clean audio from an HLS-captured transport stream for use in video editing software like Premiere Pro or DaVinci Resolve, which natively handles WAV
- Converting AC3 or AAC audio embedded in a DVR recording to uncompressed WAV for ingestion into a digital audio workstation (DAW) for post-production
- Preparing broadcast-captured dialogue or commentary audio in WAV format for submission to a studio or broadcaster that requires uncompressed deliverables
- Stripping the audio from a live streaming capture (.ts) to create a standalone podcast episode or radio segment in a universally compatible format
- Archiving the audio component of a transport stream recording in a lossless PCM format to prevent further generational quality loss during repeated editing
Frequently Asked Questions
The conversion does involve a decode step, since the audio in a TS file is almost always compressed — typically as AAC or AC3. Decoding that compressed audio and re-encoding it as PCM in the WAV file will not introduce additional compression artifacts, but any artifacts from the original TS encoding are already present and cannot be reversed. The resulting WAV is uncompressed PCM, so it will not degrade further from this point forward regardless of how many times you edit or export it.
TS files store audio in compressed formats like AAC or AC3, which can be 10–20x smaller than uncompressed audio. WAV with PCM 16-bit audio stores every sample as raw data — a stereo 48kHz stream at 16-bit depth uses roughly 11 MB per minute. This size increase is expected and is the nature of uncompressed audio. If file size is a concern, consider converting to FLAC instead, which is losslessly compressed.
By default, FFmpeg selects the first or highest-priority audio stream in the TS file, which is typically the default language track as signaled in the stream metadata. If your broadcast recording has multiple audio tracks (e.g., English and Spanish, or a stereo mix alongside a 5.1 surround track), only one will be included in the WAV output since WAV does not support multiple audio tracks. To select a specific track, you can add '-map 0:a:1' (for the second audio stream) to the FFmpeg command before the output filename.
WAV has very limited metadata support compared to TS. Standard TS metadata such as program names, channel identifiers, and broadcast timestamps will not carry over into the WAV file. Basic tags like title or artist may be written if they were embedded as ID3 or similar metadata in the TS audio stream, but this is uncommon in broadcast recordings. If metadata preservation is important, consider using a format like FLAC or MP4 for the output.
FFmpeg will decode the AC3 5.1 stream and output it as a 6-channel (5.1) PCM WAV file by default, preserving the channel layout. WAV fully supports multi-channel PCM audio, so your surround information will be retained. However, some playback software may not correctly interpret a 6-channel WAV, so if you need stereo compatibility you can add '-ac 2' to the command to downmix to stereo during the conversion.
The default output using '-c:a pcm_s16le' produces 16-bit audio at whatever sample rate the source TS uses (commonly 48kHz for broadcast content). To change the sample rate, add '-ar 44100' before the output filename to resample to 44.1kHz (standard for CD audio). To use 24-bit depth instead, replace 'pcm_s16le' with 'pcm_s24le' in the command: 'ffmpeg -i input.ts -c:a pcm_s24le output.wav'. Higher bit depths are useful for professional audio workflows that require more headroom for editing.
Technical Notes
Transport Stream is a robust broadcast container designed for lossy transmission environments — it uses fixed-size 188-byte packets and includes error correction structures not found in typical media containers. When extracting audio, FFmpeg must parse the PAT (Program Association Table) and PMT (Program Map Table) embedded in the TS to identify audio elementary streams. The default audio codec in broadcast TS files is often AAC (used in ATSC 3.0 and many HLS streams) or AC3/E-AC3 (common in ATSC 1.0 over-the-air broadcasts), both of which must be fully decoded before being written as PCM to the WAV container. The output codec 'pcm_s16le' means signed 16-bit PCM in little-endian byte order, which is the WAV standard defined in the original Microsoft RIFF specification. WAV files have a maximum file size limit of 4GB due to the 32-bit size field in the RIFF header — for very long recordings (typically over 6 hours at stereo 48kHz 16-bit), this limit may be reached, and FFmpeg will warn you. In such cases, consider splitting the source or using the RF64 WAV extension. Subtitle streams and chapters present in the TS file are dropped entirely since WAV has no mechanism to carry them.