gstreamer-pipeline

name	gstreamer-pipeline
description	Build and run GStreamer pipelines with gst-launch-1.0, introspect elements with gst-inspect-1.0, discover media with gst-discoverer-1.0, enumerate devices with gst-device-monitor-1.0, play files with gst-play-1.0, detect type with gst-typefind-1.0. Pipeline syntax (element ! element, named elements, caps filter), common flows (file playback, transcoding, RTP send/receive, HLS/DASH segmenting, WebRTC via webrtcbin, RTSP server via rtspclientsink/rtsp-server library, screen capture, webcam, appsrc/appsink integration). Use when the user asks to build a GStreamer pipeline, construct a gst-launch command, do real-time video processing with GStreamer, pipe ffmpeg output into GStreamer, build an embedded/Linux media pipeline, use webrtcbin, stream RTSP via GStreamer, or author a plugin-graph pipeline.
argument-hint	[pipeline]

GStreamer Pipeline

Context: $ARGUMENTS

Quick start

• Run a pipeline: -> Step 4 (launch -- <pipeline-desc>)
• Look up an element's properties: -> Step 2 (inspect --element <name>)
• Probe a file or URL: -> Step 3 (discover --uri <uri>)
• List cameras / mics / screens: -> Step 3 (devices)
• Copy-paste a ready-made recipe: -> Read references/pipelines.md

When to use

• User wants a running pipeline: "play this", "transcode", "publish RTSP", "receive WebRTC".
• User wants to verify an element exists locally AND is linkable with a given caps.
• User wants to enumerate cameras/mics with gst-device-monitor-1.0.
• User wants to probe a file's streams/caps/duration via gst-discoverer-1.0.
• User wants a gst-launch-1.0 one-liner for bench-testing before coding the C / Rust / Python client.

For documentation lookup (which properties does hlssink2 expose? what does webrtcbin support?), use the gstreamer-docs skill first.

---

Step 1 — Check GStreamer is installed

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --element filesrc --dry-run

If this errors with "gst-inspect-1.0 not found", install:

• macOS: brew install gstreamer
• Debian/Ubuntu: apt install gstreamer1.0-tools gstreamer1.0-plugins-{base,good,bad,ugly} gstreamer1.0-libav
• Fedora: dnf install gstreamer1 gstreamer1-plugins-{base,good,bad-free,ugly-free} gstreamer1-libav
• Windows: MSI installer at gstreamer.freedesktop.org/download/

Verify plugin sets are present; many elements (e.g. webrtcbin, hlssink2, srtsink, x264enc) ship in -bad or -ugly or Rust plugins (gst-plugins-rs).

---

Step 2 — Introspect an element before using it

Never hand-assemble an element's properties from memory. Use:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --element webrtcbin
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --element hlssink2
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --element x264enc

List every factory:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py list-elements
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py list-elements --plugin coreelements

Find which elements handle a URI scheme (e.g. rtsp://):

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --uri-handlers

Filter by factory type:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --types "Codec/Encoder/Video"

Pair with the gstreamer-docs skill for the canonical URL and full property table if the CLI output is terse.

---

Step 3 — Probe devices and media

Discover streams, caps, duration, tags of a file or URL:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py discover --uri file:///path/to/file.mp4
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py discover --uri rtsp://camera.local/stream -v -t 10

List capture + output devices:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py devices
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py devices --classes "Video/Source" "Audio/Source"
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py devices --follow   # watch add/remove events

Typefind (MIME / caps) on a blob:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py typefind /tmp/unknown.bin

---

Step 4 — Run a pipeline with launch

Use -- to separate pipeline tokens from gst.py flags, then write the pipeline in GStreamer shell syntax:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch -- \
  videotestsrc num-buffers=300 \
  ! video/x-raw,width=1280,height=720,framerate=30/1 \
  ! timeoverlay \
  ! autovideosink

Enable verbose caps negotiation:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch -v -- videotestsrc ! fakesink

Set GST_DEBUG without manual env export:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch \
  --debug-level "3,webrtcbin:6" \
  --debug-file /tmp/gst.log \
  --verbose-tool -e -- \
  videotestsrc ! x264enc ! rtph264pay ! udpsink host=127.0.0.1 port=5000

Add --eos-on-shutdown when writing to a muxer so Ctrl-C finalises the file cleanly (otherwise matroskamux / mp4mux leave a truncated output).

Dry-run echoes the exact gst-launch-1.0 invocation and exits:

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch --dry-run -- filesrc location=in.mp4 ! decodebin ! autovideosink

---

Step 5 — Quick playback with play

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py play video.mp4
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py play rtsp://cam/stream --audio-only
uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py play video.mp4 --no-interactive

--no-interactive disables keyboard controls — use this in any non-interactive agent run.

---

Step 6 — Grab a recipe

references/pipelines.md has 15+ production-ready pipelines:

• File playback / transcoding (H.264 + AAC in MP4)
• Webcam + mic -> MP4
• Screen capture -> MP4 (macOS / Linux / Windows)
• RTP send + receive (H.264, Opus)
• MPEG-TS over UDP with FEC
• RTSP client consumption and RTSP server publish
• HLS segmenting (hlssink2)
• DASH segmenting (dashsink)
• WebRTC publish with webrtcsink (Rust, auto-negotiation)
• Low-level WebRTC with webrtcbin
• SRT send / receive
• appsrc / appsink bridges for custom code
• NDI / DeckLink if available

Read references/pipelines.md when the user needs a starting pipeline for any of the above.

---

Gotchas

• All CLI tools have the -1.0 suffix. gst-inspect-1.0 not gst-inspect, gst-launch-1.0 not gst-launch. The 1.0 is the major API version; GStreamer 0.10 was EOL'd in 2012 and tools without -1.0 usually don't exist.
• ! is NOT a logical-NOT in gst-launch. It's the link operator. Quote your pipeline or use -- so the shell passes it through: gst.py launch -- videotestsrc ! fakesink.
• Caps filters are video/x-raw,width=1280,... with NO space after the commas. Some shells swallow the parens / commas — single-quote the caps string when in doubt.
• queue is mandatory between threads. Whenever you have a tee fan-out, or sink/source crossing subsystems (audio<->video, CPU<->GPU), insert a queue on each branch. Without it the pipeline serialises on one thread and the second branch starves.
• videoconvert + audioconvert are cheap insurance. When two elements disagree on caps, drop videoconvert ! videoscale or audioconvert ! audioresample in between. GStreamer will no-op if the caps already match.
• decodebin3 / uridecodebin3 use GST_MESSAGE_STREAM_COLLECTION + GST_EVENT_SELECT_STREAMS, NOT the old autoplug-* signals. Port old decodebin code carefully.
• playbin3 has different bus messages from playbin. Don't copy-paste signal handlers between them.
• Rust plugins (gst-plugins-rs) ship separately. webrtcsink, awstranscriber, whipclientsink aren't in -good/-bad/-ugly — they need gst-plugins-rs (available from gstreamer.freedesktop.org or your distro). If gst-inspect-1.0 webrtcsink comes up empty, install gst-plugins-rs.
• webrtcbin vs webrtcsink: webrtcbin is the low-level C element where you drive SDP + ICE manually via signals (, , etc.). is the Rust high-level element that auto-negotiates and speaks WHIP/WHEP. Pick one.

---

Examples

Example 1: "Play an MP4 with verbose caps"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch -v -- \
  filesrc location=in.mp4 ! decodebin3 ! videoconvert ! autovideosink

Example 2: "Record my webcam + mic to MP4 for 30 seconds (macOS)"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch -e -- \
  avfvideosrc device-index=0 ! video/x-raw,width=1280,height=720 \
  ! videoconvert ! x264enc tune=zerolatency bitrate=3000 ! h264parse \
  ! qtmux name=mux ! filesink location=out.mp4 \
  avfaudiosrc ! audioconvert ! audioresample ! avenc_aac bitrate=128000 \
  ! mux. \
  --eos-on-shutdown

(Kill with Ctrl-C after 30s; -e ensures qtmux finalises.)

Example 3: "Probe an RTSP camera for its streams"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py discover \
  --uri rtsp://admin:pw@10.0.0.42/stream -v -t 10

Example 4: "What video inputs can I record from?"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py devices --classes "Video/Source"

Example 5: "Dump the full property list for webrtcbin"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py inspect --element webrtcbin

Example 6: "Publish a test pattern to MediaMTX via WHIP"

uv run ${CLAUDE_SKILL_DIR}/scripts/gst.py launch -v -e -- \
  videotestsrc is-live=true \
  ! video/x-raw,width=1280,height=720,framerate=30/1 \
  ! videoconvert \
  ! 'whipclientsink name=ws signaller::whip-endpoint=http://127.0.0.1:8889/live/whip'

(Requires gst-plugins-rs; fall back to a webrtcsink + signaller setup if absent.)

---

Troubleshooting

WARNING: erroneous pipeline: no element "webrtcbin"

Cause: Plugin not installed. Solution: webrtcbin lives in gst-plugins-bad. Install it: macOS brew install gst-plugins-bad; Debian/Ubuntu apt install gstreamer1.0-plugins-bad libnice-dev. Verify with gst-inspect-1.0 webrtcbin.

could not link filesrc0 to decodebin0

Cause: filesrc produces ANY caps, but some decoders want a typefind step. Solution: Insert decodebin3 which handles typefind internally: filesrc location=in.mp4 ! decodebin3 ! .... Or use uridecodebin3 uri=file:///....

gst-launch-1.0: command not found

Cause: Tools not on PATH, or only gst-launch installed without the -1.0 suffix (very old GStreamer). Solution: Reinstall GStreamer 1.x. On macOS homebrew: brew install gstreamer.

Output MP4 from the recording pipeline is 0 bytes or unplayable

Cause: Process killed without EOS -> qtmux / mp4mux never wrote the moov atom. Solution: Re-run with -e / --eos-on-shutdown, use Ctrl-C only once, then wait 1-2s for finalisation.

avfvideosrc opens but produces no frames on macOS

Cause: Privacy permission not granted to the terminal / IDE running gst-launch. Solution: macOS System Settings -> Privacy & Security -> Camera -> add your terminal. Restart the terminal after granting.

Pipeline runs but no audio in output

Cause: Missing queue on the audio branch of a muxer fan-in, or wrong sample rate. Solution: Insert queue ! audioconvert ! audioresample before the muxer pad. Ensure mux. syntax attaches to the named mux element.

"Delayed linking failed for: webrtcbin.sink"

Cause: webrtcbin refuses RTP caps that don't specify payload-type / encoding-name. Solution: Add a capsfilter: rtph264pay config-interval=-1 pt=96 ! application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin.

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Reference docs

• Full set of copy-paste pipeline recipes -> references/pipelines.md