GStreamer Daemon (GstD)
GStreamer Daemon (GstD) is a service that enables remote control of GStreamer pipelines via gst-client
. It allows:
- Remote pipeline management – Start, stop, modify pipelines dynamically.
- Multi-client access – Multiple users can control pipelines simultaneously.
- Embedded & network control – Use
gst-client
to manage pipelines over TCP or UNIX sockets. - Low-latency communication – Ideal for real-time media applications.
⚙️ Installing GStreamer Daemon (`gstd`)
Before installing GStreamer Daemon, ensure you have the required dependencies.
🛠️ Build System & Development Tools
sudo apt install -y meson ninja-build git
🎞️ GStreamer Core & Plugins
sudo apt install -y gstreamer1.0-tools gstreamer1.0-plugins-base
📦 Additional GStreamer Plugins
sudo apt install -y gstreamer1.0-plugins-good gstreamer1.0-plugins-bad \ gstreamer1.0-plugins-ugly gstreamer1.0-libav
🏗️ GStreamer Development Libraries
sudo apt install -y libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev
🔗 GLib & D-Bus Dependencies
sudo apt install -y libglib2.0-dev libdbus-1-dev libgirepository1.0-dev python3-gi
🌐 Daemon & Networking Libraries
sudo apt install -y libdaemon-dev libsoup2.4-dev
📄 JSON & Editing Support
sudo apt install -y libjansson-dev libedit-dev
🐛 Debugging Tools
sudo apt install -y valgrind
🏗️ Installing and Running GStreamer Daemon
To install GStreamer Daemon:
git clone https://gitlab.freedesktop.org/gstreamer/gstd.git cd gstd meson build ninja -C build sudo ninja -C build install
Verify installation:
gstd --version
To start the daemon:
gstd -e --tcp-base-port=5000
Note: The -e
flag enables execution mode, and --tcp-base-port
sets the TCP control port.
📡 Controlling GStreamer Daemon with gst-client
Each command follows this format:
gst-client COMMAND pipeline_name "PIPELINE_DESCRIPTION"
🎞️ Creating a Pipeline
gst-client pipeline_create test_pipeline "videotestsrc ! videoconvert ! autovideosink"
Parameters:
- pipeline_create – Creates a new pipeline.
- test_pipeline – Pipeline name (identifier).
- Pipeline elements:
- videotestsrc – Generates test video. - videoconvert – Ensures format compatibility. - autovideosink – Displays video output.
Use case: Used for testing video playback.
▶️ Starting a Pipeline
gst-client pipeline_play test_pipeline
Use case: Runs an existing paused pipeline.
⏹️ Stopping a Pipeline
gst-client pipeline_stop test_pipeline
Use case: Pauses pipeline execution without deleting it.
❌ Deleting a Pipeline
gst-client pipeline_delete test_pipeline
Use case: Frees up resources when a pipeline is no longer needed.
📡 Example: Transmitting an RTP Stream
gst-client pipeline_create udp_stream "videotestsrc is-live=true pattern=ball ! videoconvert ! x264enc tune=zerolatency bitrate=500 speed-preset=ultrafast ! rtph264pay ! udpsink host=127.0.0.1 port=5002"
Parameters:
- pipeline_create udp_stream – Creates a new pipeline named udp_stream.
- videotestsrc is-live=true pattern=ball – Generates a moving ball pattern as a live video source.
- is-live=true – Ensures the video source behaves like a real-time stream. - pattern=ball – Uses a bouncing ball as the test video pattern.
- videoconvert – Ensures compatibility with encoders and sinks.
- x264enc tune=zerolatency bitrate=500 speed-preset=ultrafast – Encodes video into H.264 format with low latency.
- tune=zerolatency – Optimizes the encoding for minimal delay. - bitrate=500 – Sets the bitrate to 500 kbps. - speed-preset=ultrafast – Uses the fastest encoding preset.
- rtph264pay – Converts H.264 video into RTP packets for network transmission.
- udpsink host=127.0.0.1 port=5002 – Sends the RTP stream to UDP port 5002 on localhost.
Use case: Broadcasting an RTP stream over UDP.
▶️ Playing the RTP Transmission
gst-client pipeline_play udp_stream
Use case: Starts broadcasting the RTP stream.
⏹️ Stopping the RTP Transmission
gst-client pipeline_stop udp_stream
Use case: Stops broadcasting the RTP stream without deleting it.
❌ Deleting the RTP Transmission Pipeline
gst-client pipeline_delete udp_stream
Use case: Removes the RTP transmission pipeline.
📡 Example: Receiving an RTP Stream
gst-client pipeline_create udp_receiver "udpsrc port=5002 ! application/x-rtp, encoding-name=H264 ! rtpjitterbuffer latency=100 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink sync=false"
Parameters:
- pipeline_create udp_receiver – Creates a new pipeline named udp_receiver.
- udpsrc port=5002 – Listens on UDP port 5002 for incoming RTP packets.
- application/x-rtp, encoding-name=H264 – Declares the media format as RTP with H.264 encoding.
- rtpjitterbuffer latency=100 – Buffers packets to compensate for network jitter.
- latency=100 – Sets a buffer delay of 100 milliseconds.
- rtph264depay – Extracts raw H.264 video from the RTP stream.
- avdec_h264 – Decodes the H.264 video stream.
- videoconvert – Ensures compatibility before displaying the video.
- autovideosink sync=false – Displays the video, disabling synchronization for lower latency.
Use case: Receiving an RTP video stream over UDP.
▶️ Playing the RTP Receiver
gst-client pipeline_play udp_receiver
Use case: Starts the RTP receiver pipeline.
⏹️ Stopping the RTP Receiver
gst-client pipeline_stop udp_receiver
Use case: Stops the RTP receiver pipeline without deleting it.
❌ Deleting the RTP Receiver Pipeline
gst-client pipeline_delete udp_receiver
Use case: Removes the RTP receiver pipeline.
🛑 Stopping GStreamer Daemon
To stop the GStreamer Daemon:
gstd -k
Use case: Terminates the running GStreamer Daemon (`gstd`).
✅ Checking if `gstd` is Running
To verify if GStreamer Daemon is running:
ps aux | grep gstd
Use case: Displays `gstd` processes if running.
📊 Summary
GStreamer Daemon enables remote control of pipelines via gst-client
.
Feature | Description |
---|---|
Remote Pipeline Control | Manage GStreamer pipelines via gst-client
|
Dynamic Modification | Add/remove elements in real-time |
Multi-Client Access | Multiple users can control pipelines |
Embedded System Friendly | Works efficiently with TCP and UNIX socket connections |