Extract Audio from TS to MP3 — Free Online Tool
Extract MP3 audio from MPEG-2 Transport Stream (.ts) files directly in your browser — no upload required. This tool strips the video track and re-encodes the AAC or AC3 audio from your TS container into a universally compatible MP3 file using the LAME encoder.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your TS file here
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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
TS files are broadcast containers that typically carry video (H.264 or H.265) alongside audio streams encoded in AAC, AC3, or MP3. Because MP3 is a pure audio format, the video stream is completely discarded rather than remuxed. The audio stream — most commonly AAC in broadcast and HLS recordings — cannot be stream-copied into an MP3 container, so it is fully decoded and re-encoded using the libmp3lame encoder at the specified bitrate (default 128k). This is a transcode operation: the audio goes through a decode-then-encode cycle, which introduces a small degree of generational quality loss compared to the original. If your TS file contains multiple audio tracks (common in broadcast recordings with alternate language streams), only the first audio track is extracted by default.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing tool. In the browser version, this runs via FFmpeg.wasm (a WebAssembly port of FFmpeg) entirely on your local machine — no data is sent to a server. |
-i input.ts
|
Specifies the input MPEG-2 Transport Stream file. FFmpeg will demux the TS packet stream to identify and decode the contained video, audio, and any subtitle elementary streams before processing begins. |
-vn
|
Disables video output entirely — this is the core flag that makes this an audio extraction operation. All video streams from the TS file (whether H.264, H.265, or MJPEG) are discarded, producing a pure audio output file. |
-c:a libmp3lame
|
Selects the LAME MP3 encoder to re-encode the audio stream. Since the source audio in a TS file is typically AAC or AC3, it must be fully decoded and re-encoded — it cannot be stream-copied directly into an MP3 container. |
-b:a 128k
|
Sets the MP3 output bitrate to 128 kilobits per second, the standard default that balances file size and perceptual audio quality. Increase to 192k or 320k for music content, or lower to 96k for speech-only recordings to save space. |
output.mp3
|
Defines the output filename and format. The .mp3 extension tells FFmpeg to mux the encoded audio into an MP3 file, which is compatible with virtually every media player, mobile device, car stereo, and streaming platform. |
Common Use Cases
- Pull the audio commentary track from a recorded broadcast sports event to listen on the go as a podcast-style MP3
- Extract a musical performance or concert captured from a digital TV broadcast (TS recording) into an MP3 for your music library
- Convert HLS-recorded .ts segment files or reassembled transport streams from streaming captures into shareable MP3 audio
- Strip the audio from a IPTV or DVR recording to create an MP3 of a radio show or talk program broadcast over digital TV
- Extract the audio from a surveillance or CCTV recording stored as a TS file for use as an audio-only evidence clip
- Separate the audio track from a video lecture or webinar recorded in TS format to create an MP3 study resource
Frequently Asked Questions
Yes, a small quality loss occurs. Since both AAC (common in TS broadcast streams) and MP3 are lossy formats, converting between them means decoding the compressed AAC audio back to raw PCM and then re-encoding it as MP3 — a process called transcoding. This double-lossy step introduces generational degradation. At the default 128k bitrate the result is perfectly acceptable for speech and most casual listening, but if you need the highest fidelity, increase the output bitrate to 192k or 320k, or consider extracting to a lossless format like FLAC instead if your source TS contains FLAC audio.
By default, FFmpeg selects the first audio stream found in the TS container, which is typically the primary language track. If you need a specific alternate track, you would need to modify the FFmpeg command to add a stream selector such as '-map 0:a:1' to pick the second audio track. The browser tool extracts the default (first) audio track automatically, so use the desktop FFmpeg command displayed on this page if you need fine-grained track selection.
Yes — the '-b:a 128k' flag controls the output MP3 bitrate. Replace '128k' with a lower value like '96k' or '64k' to reduce file size (at the cost of audio quality), or raise it to '192k', '256k', or '320k' for better fidelity. For broadcast speech content, 96k is often sufficient; for music extracted from a TV performance, 192k or higher is recommended. The command you copy from this page can be edited directly before running it on your local machine.
Generally no. TS containers use DVB/MPEG-2 service information metadata (PMT, SDT tables) that is incompatible with the ID3 tag system MP3 uses. FFmpeg may carry over basic stream-level metadata if it can map it, but broadcast-specific fields like channel name, program title, or EPG data will be lost. If you need to preserve metadata in the output MP3, you would need to add it manually using an ID3 tag editor after the conversion.
Dramatically smaller. A TS file combines both a video stream and audio stream — the video (H.264 or H.265) accounts for the vast majority of the file size. Stripping the video entirely and keeping only the re-encoded MP3 audio typically reduces file size by 90–98%. For example, a 1GB one-hour TS broadcast recording might yield an MP3 of roughly 55–115MB depending on the bitrate you choose.
The single-file command shown on this page processes one file at a time. To batch process an entire folder of TS files on your desktop, you can wrap the command in a shell loop. On Linux or macOS: 'for f in *.ts; do ffmpeg -i "$f" -vn -c:a libmp3lame -b:a 128k "${f%.ts}.mp3"; done'. On Windows Command Prompt: 'for %f in (*.ts) do ffmpeg -i "%f" -vn -c:a libmp3lame -b:a 128k "%~nf.mp3"'. The browser tool processes files one at a time.
Technical Notes
TS (MPEG-2 Transport Stream) is a packetized container designed for error-resilient transmission over unreliable broadcast channels — it is not a simple interleaved container like MP4. As a result, FFmpeg must demux the TS packet stream before it can access the audio elementary stream. The most common audio codec in broadcast TS files is AAC-LC (used heavily in DVB-T, HLS, and ATSC streams), though AC3 (Dolby Digital) is prevalent in North American broadcast and some European cable recordings, and MP3 itself is occasionally used in older DVB streams. All of these require full decoding before re-encoding to MP3 via libmp3lame, since MP3 containers cannot carry non-MP3 audio streams. The '-vn' flag ensures that even complex multi-program TS files with multiple video PIDs are handled cleanly by suppressing all video output. One known limitation: some TS files contain discontinuities or PTS/DTS timestamp errors from incomplete recordings or DVR glitches — these can cause audio sync drift or gaps in the output MP3. If you encounter this, the desktop FFmpeg command can be extended with '-fflags +discardcorrupt' to handle damaged packets more gracefully. MP3 does not support multi-channel audio beyond stereo, so if your TS source contains 5.1 AC3 audio, it will be automatically downmixed to stereo during encoding.