Extract Audio from TS to OGG — Free Online Tool

Extract audio from MPEG-2 Transport Stream (.ts) broadcast files and convert it to OGG Vorbis format — a free, open-source audio container ideal for web and desktop playback. This tool strips the video entirely and re-encodes the AAC or AC3 audio typically found in TS broadcast streams into Vorbis, giving you a lightweight, open-format audio file.

FFmpeg Command

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Estimated output:

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How It Works

MPEG-2 Transport Stream files carry multiplexed audio and video streams designed for broadcast transmission, and their audio tracks are most commonly encoded in AAC or AC3 (Dolby Digital). OGG is a pure audio container format from Xiph.Org that does not support video, so the video stream is completely discarded using the -vn flag — no video decoding or encoding occurs. The audio stream cannot be stream-copied into OGG because neither AAC nor AC3 is a supported codec in the OGG container; instead, the audio is fully decoded and re-encoded to Vorbis (libvorbis) using variable-bitrate quality mode. This means there is a generation of lossy transcoding involved, so quality is optimized through the -q:a parameter rather than preserving the original bitstream.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg tool. In this browser-based tool, the same command runs via FFmpeg.wasm (WebAssembly) entirely in your browser — no server involved. On your desktop, this calls your locally installed FFmpeg binary.
-i input.ts Specifies the input MPEG-2 Transport Stream file. TS files are multiplexed containers carrying video, audio, and sometimes subtitle streams packetized for broadcast transmission — FFmpeg demuxes these streams before processing.
-vn Disables video output entirely. Since OGG is a pure audio container and we only want the audio from the TS broadcast file, this flag ensures no video stream is decoded or written, significantly speeding up the conversion.
-c:a libvorbis Specifies Vorbis as the audio encoder, using the libvorbis library. This is required because the source audio in the TS file (typically AAC or AC3) is not compatible with the OGG container — the audio must be decoded and re-encoded into Vorbis, which is OGG's native and most widely supported codec.
-q:a 4 Sets the Vorbis variable-bitrate quality level to 4 on a 0–10 scale, targeting approximately 128–160 kbps average bitrate. This is the quality parameter specific to libvorbis — it is not a bitrate value but a quality index, where higher numbers produce better audio fidelity at the cost of larger file size.
output.ogg Defines the output filename with the .ogg extension. FFmpeg uses this extension to confirm the OGG container format, which will hold the re-encoded Vorbis audio stream extracted from the original TS broadcast file.

Common Use Cases

  • Extract a broadcast TV recording's audio track to create a podcast or audio archive without carrying along the large video stream
  • Convert DVR-recorded .ts files from a set-top box into OGG Vorbis for playback in open-source media players like VLC or Audacious on Linux
  • Pull the audio from a live-streamed HLS or broadcast capture saved as a .ts file for use in a video editing project as a standalone audio asset
  • Convert the commentary or narration track from a broadcast .ts sports or news recording into OGG for accessibility purposes or transcription workflows
  • Reduce file size by stripping video from large .ts transport stream recordings when you only need the audio portion for archiving or review
  • Extract multi-language audio from a multi-track .ts broadcast file into a portable OGG file for distribution on open-source or Linux-first platforms

Frequently Asked Questions

Yes, there is a degree of quality loss because this conversion involves transcoding — the original AAC or AC3 audio in the TS file is fully decoded and re-encoded as Vorbis. This is unavoidable since neither AAC nor AC3 can be stored natively in an OGG container. At the default quality setting of -q:a 4, the resulting Vorbis audio is generally transparent for most listeners at around 128–160 kbps variable bitrate, but it is not a lossless or bit-perfect copy of the source.
The OGG container format only supports three audio codecs: Vorbis, Opus, and FLAC. MPEG-2 Transport Stream files from broadcast or DVR sources almost always carry AAC or AC3 audio, neither of which is compatible with the OGG container. Because the codec must change, FFmpeg must fully decode the source audio and re-encode it — stream copying with -c:a copy is not possible in this scenario.
Replace the -q:a 4 value in the command with a number between 0 and 10, where higher numbers produce better quality and larger files. For example, using -q:a 6 targets approximately 192 kbps variable bitrate Vorbis, while -q:a 2 produces a smaller file at around 96 kbps. For archival purposes or when source audio quality is high (such as 5.1 AC3 from broadcast), -q:a 7 or -q:a 8 is a reasonable choice to minimize transcoding artifacts.
By default, FFmpeg selects the first audio stream in the TS file, which is typically the primary language track. If your broadcast recording contains multiple audio tracks (e.g., alternate language, audio description, or surround vs. stereo), you can specify a particular track by adding -map 0:a:1 (for the second audio track) before the output filename in the command. The OGG container does support multiple audio streams, so you could also map multiple tracks if needed.
Some basic metadata may carry over during conversion, such as language tags on audio streams, but broadcast-specific metadata embedded in the MPEG-2 Transport Stream (like program info, EPG data, or service IDs) is not preserved in the OGG output. OGG Vorbis supports standard tags like TITLE, ARTIST, and ALBUM via Vorbis Comments, but FFmpeg will only map metadata fields it recognizes and can translate. You may want to use a tool like EasyTag or FFmpeg's -metadata flag to manually set tags after conversion.
The command shown converts a single file, but you can batch process on the command line using a shell loop. On Linux or macOS, use: for f in *.ts; do ffmpeg -i "$f" -vn -c:a libvorbis -q:a 4 "${f%.ts}.ogg"; done. On Windows Command Prompt, use: for %f in (*.ts) do ffmpeg -i "%f" -vn -c:a libvorbis -q:a 4 "%~nf.ogg". This is especially useful for converting a full season of recorded broadcast episodes at once.

Technical Notes

MPEG-2 Transport Stream files present a unique challenge for audio extraction because their audio is almost always encoded in broadcast-centric codecs — AAC (common in digital TV and HLS), AC3/Dolby Digital (common in North American broadcast), or occasionally MP2 (legacy European broadcast). None of these codecs are natively supported inside an OGG container, making re-encoding mandatory. The Vorbis codec used here is a variable-bitrate lossy codec, and the -q:a scale is specific to libvorbis (not to be confused with -b:a bitrate targeting). At -q:a 4, expect roughly 128–160 kbps average bitrate. If the source TS contains 5.1 surround AC3 audio, libvorbis will encode all six channels into the OGG file by default — you can downmix to stereo by adding -ac 2 to the command. One known limitation: OGG Vorbis does not support sample rates above 192 kHz, but this is not a concern for broadcast TS files which are typically at 48 kHz. Chapter support is technically available in OGG, but FFmpeg does not automatically transfer TS program boundaries into OGG chapter markers. If lossless extraction is a priority, consider using FLAC as the codec instead (-c:a flac), which OGG also supports and would eliminate the transcoding quality loss.

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