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* '''Low-latency communication''' – Ideal for '''real-time media applications'''. |
* '''Low-latency communication''' – Ideal for '''real-time media applications'''. |
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== |
== 📡 Controlling GStreamer Daemon with <code>gst-client</code> == |
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Before installing GStreamer Daemon, ensure you have the required dependencies. |
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Each command follows this format: |
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=== 🛠️ Build System & Development Tools === |
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<pre> |
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sudo apt install -y meson ninja-build git |
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</pre> |
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=== 🎞️ GStreamer Core & Plugins === |
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<pre> |
<pre> |
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gst-client COMMAND pipeline_name "PIPELINE_DESCRIPTION" |
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sudo apt install -y gstreamer1.0-tools gstreamer1.0-plugins-base |
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</pre> |
</pre> |
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=== |
=== 🎞️ Creating a Pipeline === |
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<pre> |
<pre> |
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gst-client pipeline_create test_pipeline "videotestsrc ! videoconvert ! autovideosink" |
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sudo apt install -y gstreamer1.0-plugins-good gstreamer1.0-plugins-bad \ |
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gstreamer1.0-plugins-ugly gstreamer1.0-libav |
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</pre> |
</pre> |
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'''Parameters:''' |
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=== 🏗️ GStreamer Development Libraries === |
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* '''pipeline_create''' – Creates a new pipeline. |
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<pre> |
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* '''test_pipeline''' – Pipeline name (identifier). |
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sudo apt install -y libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev |
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* '''Pipeline elements:''' |
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</pre> |
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- '''videotestsrc''' – Generates test video. |
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- '''videoconvert''' – Ensures format compatibility. |
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- '''autovideosink''' – Displays video output. |
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'''Use case:''' Used for '''testing video playback'''. |
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=== 🔗 GLib & D-Bus Dependencies === |
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<pre> |
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sudo apt install -y libglib2.0-dev libdbus-1-dev libgirepository1.0-dev python3-gi |
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</pre> |
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=== |
=== ▶️ Starting a Pipeline === |
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<pre> |
<pre> |
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gst-client pipeline_play test_pipeline |
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sudo apt install -y libdaemon-dev libsoup2.4-dev |
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</pre> |
</pre> |
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'''Use case:''' Runs an existing '''paused''' pipeline. |
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=== 📄 JSON & Editing Support === |
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<pre> |
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sudo apt install -y libjansson-dev libedit-dev |
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</pre> |
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=== |
=== ⏹️ Stopping a Pipeline === |
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<pre> |
<pre> |
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gst-client pipeline_stop test_pipeline |
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sudo apt install -y valgrind |
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</pre> |
</pre> |
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'''Use case:''' Pauses '''pipeline execution''' without deleting it. |
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== 🏗️ Installing and Running GStreamer Daemon == |
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To install GStreamer Daemon: |
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<pre> |
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git clone https://gitlab.freedesktop.org/gstreamer/gstd.git |
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cd gstd |
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meson build |
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ninja -C build |
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sudo ninja -C build install |
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</pre> |
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=== ❌ Deleting a Pipeline === |
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Verify installation: |
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<pre> |
<pre> |
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gst-client pipeline_delete test_pipeline |
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gstd --version |
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</pre> |
</pre> |
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'''Use case:''' Frees up '''resources''' when a pipeline is no longer needed. |
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To start the daemon: |
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== 📡 Example: Transmitting an RTP Stream == |
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<pre> |
<pre> |
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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" |
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gstd -e --tcp-base-port=5000 |
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</pre> |
</pre> |
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'''Parameters:''' |
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'''Note:''' The <code>-e</code> flag enables '''execution mode''', and <code>--tcp-base-port</code> sets the '''TCP control port'''. |
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* '''pipeline_create udp_stream''' – Creates a new pipeline named ''udp_stream''. |
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* '''videotestsrc is-live=true pattern=ball''' – Generates a moving '''ball pattern''' as a live video source. |
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- '''''is-live=true''''' – Ensures the video source behaves like a real-time stream. |
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- '''''pattern=ball''''' – Uses a bouncing ball as the test video pattern. |
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* '''videoconvert''' – Ensures compatibility with encoders and sinks. |
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* '''x264enc tune=zerolatency bitrate=500 speed-preset=ultrafast''' – Encodes video into '''H.264 format''' with low latency. |
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- '''''tune=zerolatency''''' – Optimizes the encoding for minimal delay. |
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- '''''bitrate=500''''' – Sets the '''bitrate''' to 500 kbps. |
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- '''''speed-preset=ultrafast''''' – Uses the '''fastest''' encoding preset. |
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* '''rtph264pay''' – Converts H.264 video into RTP packets for network transmission. |
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* '''udpsink host=127.0.0.1 port=5002''' – Sends the RTP stream to '''UDP port 5002''' on '''localhost'''. |
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'''Use case:''' '''Broadcasting an RTP stream''' over UDP. |
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== 🎛️ <code>gst-client</code> Options == |
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<code>gst-client</code> is used to interact with <code>gstd</code>. It provides multiple options: |
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=== |
=== ▶️ Playing the RTP Transmission === |
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<pre> |
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* '''<code>gst-client -h, --help</code>''' → Show help options. |
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gst-client pipeline_play udp_stream |
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* '''<code>gst-client -v, --version</code>''' → Print the current version of <code>gstd-client</code>. |
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</pre> |
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* '''<code>gst-client -i, --interactive</code>''' → Enter interactive mode after executing commands. |
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* '''<code>gst-client -f script</code>''' → Execute commands from a file. |
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'''Use case:''' Starts broadcasting the RTP stream. |
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=== 🔗 Connection Options === |
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* '''<code>gst-client -p, --tcp-port=PORT</code>''' → Connect via TCP (default: <code>5000</code>). |
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* '''<code>gst-client -a, --tcp-address=ADDRESS</code>''' → Specify the IP address (default: <code>localhost</code>). |
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* '''<code>gst-client -u, --unix</code>''' → Use UNIX socket instead of TCP. |
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* '''<code>gst-client -b, --unix-base-path=PATH</code>''' → Set UNIX socket path (default: <code>gstd_unix_socket</code>). |
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=== ⏹️ Stopping the RTP Transmission === |
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== 📡 Controlling GStreamer Daemon with <code>gst-client</code> == |
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<pre> |
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gst-client pipeline_stop udp_stream |
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</pre> |
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'''Use case:''' Stops broadcasting the RTP stream without deleting it. |
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Each command follows this format: |
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=== ❌ Deleting the RTP Transmission Pipeline === |
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<pre> |
<pre> |
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gst-client pipeline_delete udp_stream |
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gst-client COMMAND pipeline_name "PIPELINE_DESCRIPTION" |
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</pre> |
</pre> |
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'''Use case:''' Removes the RTP transmission pipeline. |
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=== 🎞️ Creating a Pipeline === |
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== 📡 Example: Receiving an RTP Stream == |
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<pre> |
<pre> |
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gst-client pipeline_create |
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" |
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</pre> |
</pre> |
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'''Parameters:''' |
'''Parameters:''' |
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* '''pipeline_create''' |
* '''pipeline_create udp_receiver''' – Creates a new pipeline named ''udp_receiver''. |
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* '''udpsrc port=5002''' – Listens on '''UDP port 5002''' for incoming RTP packets. |
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* '''test_pipeline''' → Pipeline name (identifier). |
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* '''application/x-rtp, encoding-name=H264''' – Declares the '''media format''' as RTP with '''H.264 encoding'''. |
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* '''Pipeline elements:''' |
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* '''rtpjitterbuffer latency=100''' – Buffers packets to compensate for '''network jitter'''. |
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- '''videotestsrc''' → Generates test video. |
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- '''''latency=100''''' – Sets a '''buffer delay''' of 100 milliseconds. |
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- '''videoconvert''' → Ensures format compatibility. |
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* '''rtph264depay''' – Extracts raw H.264 video from the RTP stream. |
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* '''avdec_h264''' – Decodes the H.264 video stream. |
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* '''videoconvert''' – Ensures compatibility before displaying the video. |
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* '''autovideosink sync=false''' – Displays the video, '''disabling synchronization''' for lower latency. |
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'''Use case:''' |
'''Use case:''' '''Receiving an RTP video stream''' over UDP. |
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=== ▶️ |
=== ▶️ Playing the RTP Receiver === |
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<pre> |
<pre> |
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gst-client pipeline_play |
gst-client pipeline_play udp_receiver |
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</pre> |
</pre> |
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'''Use case:''' |
'''Use case:''' Starts the RTP receiver pipeline. |
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=== ⏹️ Stopping |
=== ⏹️ Stopping the RTP Receiver === |
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<pre> |
<pre> |
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gst-client pipeline_stop |
gst-client pipeline_stop udp_receiver |
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</pre> |
</pre> |
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'''Use case:''' |
'''Use case:''' Stops the RTP receiver pipeline without deleting it. |
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=== ❌ Deleting |
=== ❌ Deleting the RTP Receiver Pipeline === |
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<pre> |
<pre> |
||
gst-client pipeline_delete |
gst-client pipeline_delete udp_receiver |
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</pre> |
</pre> |
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'''Use case:''' |
'''Use case:''' Removes the RTP receiver pipeline. |
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== 🛑 Stopping GStreamer Daemon == |
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To stop the GStreamer Daemon: |
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<pre> |
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gstd -k |
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</pre> |
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'''Use case:''' Terminates the running GStreamer Daemon (`gstd`). |
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== ✅ Checking if `gstd` is Running == |
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To verify if GStreamer Daemon is running: |
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<pre> |
<pre> |
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ps aux | grep gstd |
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gst-client pipeline_create udp_receiver "udpsrc port=5004 ! application/x-rtp, encoding-name=H264 ! rtpjitterbuffer latency=100 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink sync=false" |
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</pre> |
</pre> |
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'''Use case:''' |
'''Use case:''' Displays `gstd` processes if running. |
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== 📊 Summary == |
== 📊 Summary == |
Revision as of 12:59, 24 February 2025
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.
📡 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 |