Convert TS to OGG — Free Online Tool
Convert TS broadcast stream files to OGG audio using the open Vorbis codec, stripping the video and transport stream wrapper to produce a lightweight, freely-licensed audio file. Ideal for extracting audio from broadcast recordings or IPTV captures for use in open-source and Linux-centric workflows.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
MPEG-2 Transport Stream (TS) files are designed for broadcast and streaming delivery, typically carrying multiplexed video (often H.264 or H.265), audio (often AAC or AC3), and sometimes subtitles in separate program streams. During this conversion, FFmpeg demuxes the TS container, discards all video streams, and re-encodes the audio — whatever codec it was (AAC, AC3, MP3, etc.) — into Vorbis, then wraps the result in an OGG container. This is a full audio transcode, not a remux, because TS audio codecs like AAC and AC3 are not natively supported inside OGG. The OGG/Vorbis output is a pure audio file with no video data, making it significantly smaller than the source TS file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In the browser-based tool this runs via FFmpeg.wasm (WebAssembly); the identical command can be run on any desktop with FFmpeg installed to process files larger than 1 GB. |
-i input.ts
|
Specifies the input file, an MPEG-2 Transport Stream (.ts). FFmpeg will demux the container, identifying all program streams including video, audio, and any subtitle PIDs present in the broadcast recording. |
-c:a libvorbis
|
Sets the audio encoder to libvorbis, which is the reference encoder for the Vorbis codec — the standard lossy audio format for OGG containers. Since TS audio (typically AAC or AC3) cannot be stream-copied into OGG, a full re-encode with this codec is required. |
-q:a 4
|
Sets the Vorbis variable bitrate quality level to 4 on a 0–10 scale, targeting approximately 128 kbps. This is a sensible default for broadcast-sourced audio, balancing perceptual quality against file size; it can be raised to 6 or 7 in the desktop command for higher fidelity. |
output.ogg
|
Defines the output filename and tells FFmpeg to use the OGG container format. No video streams are mapped to this output — FFmpeg automatically omits video when the output container (OGG) has no video codec defined, resulting in a pure audio file. |
Common Use Cases
- Extract the audio commentary track from a recorded over-the-air broadcast (DVR .ts file) to archive as a standalone audio file in an open format
- Pull audio from an IPTV stream recording to create a podcast or radio segment without carrying along the video payload
- Convert AC3 or AAC broadcast audio embedded in a TS file into Vorbis for playback in open-source media players or Linux desktop environments that prefer royalty-free codecs
- Archive spoken-word content (lectures, news broadcasts, public radio recordings) captured as TS files into compact OGG files for long-term storage
- Prepare audio extracted from a broadcast TS recording for upload to platforms like the Internet Archive or other open-content repositories that favor open formats
- Strip the video from a live-streaming capture saved as a TS file to produce an OGG audio file suitable for embedding in a web page using the HTML5 audio element
Frequently Asked Questions
By default, FFmpeg selects the first (or best-ranked) audio stream from the TS file and encodes only that one into the OGG output. TS files from broadcast sources often contain multiple audio tracks — for example, a main mix and a secondary language track. If you need a specific track, you would need to add a flag like '-map 0:a:1' to the command to select the second audio stream. The tool as configured processes the default audio stream only.
OGG only supports Vorbis, Opus, and FLAC as its audio codecs. Broadcast TS files almost always carry audio in AAC, AC3 (Dolby Digital), or occasionally MP2 — none of which are valid inside an OGG container. This means a full transcode is unavoidable: the source audio is decoded to raw PCM and then re-encoded to Vorbis. There is no lossless path from AAC or AC3 to Vorbis; some quality loss occurs as a result of this transcode.
Vorbis uses a variable bitrate quality scale from 0 (lowest) to 10 (highest), where -q:a 4 targets roughly 128 kbps. For most broadcast audio sources — which are typically encoded at 128–192 kbps AAC or 192–384 kbps AC3 — a Vorbis quality of 4 produces good perceptual quality with a modest file size. To match higher-quality broadcast audio more closely, you could raise the value to -q:a 6 or -q:a 7 in the desktop FFmpeg command, which targets approximately 192–224 kbps.
Replace the '4' in '-q:a 4' with a value between 0 and 10. For example, 'ffmpeg -i input.ts -c:a libvorbis -q:a 6 output.ogg' produces higher-quality audio at a larger file size, while '-q:a 2' produces a smaller file with more audible compression artifacts. The scale is not linear in bitrate terms — values 4 through 6 cover the most commonly useful range for broadcast-quality source material.
Subtitles embedded in the TS stream will not be transferred to the OGG file. Although OGG does support chapter markers in its metadata, FFmpeg does not automatically map TS subtitle streams or timing cues to OGG chapter data during this conversion. Any subtitle or DVB teletext streams present in the source TS file are simply dropped. If preserving subtitles is important, a different output format would be more appropriate.
TS files contain a full video stream alongside the audio, and the video typically accounts for 90–98% of the file size. Removing the video and encoding only the audio track as Vorbis at -q:a 4 will reduce a typical broadcast TS file to roughly 2–8% of its original size. For example, a 1 GB one-hour TS recording might produce an OGG audio file of around 50–80 MB, depending on the original audio bitrate and content.
Technical Notes
TS is a multiplexed container format designed for resilient broadcast delivery; it uses fixed-size 188-byte packets and can carry multiple programs, each with their own video, audio, and data streams. When demuxing, FFmpeg must parse the Program Specific Information (PSI) tables to identify which PIDs carry audio. The audio codec in a broadcast TS is most commonly AAC-LC or AC3, both of which require full decoding before re-encoding to Vorbis — there is no stream copy shortcut. OGG/Vorbis supports metadata tags (artist, title, album, etc.) but broadcast TS files rarely carry rich metadata, so the OGG output will typically have minimal or no tags. OGG does not support subtitles, so any DVB subtitle or teletext streams in the TS are silently discarded. Multiple audio tracks in the TS (e.g., stereo main + SAP) require explicit stream mapping to extract and are not handled automatically. The OGG format supports chapters natively via Ogg skeleton or comment headers, but FFmpeg does not populate these from TS timing data without additional configuration. For very long broadcast recordings, be aware that Vorbis encoding is CPU-intensive compared to a simple remux, so conversion time will be proportional to the audio duration.