mirror of
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Update DOCS.md
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Alexandre
@@ -60,7 +60,7 @@ sudo apt-get dist-upgrade -y
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# Install RTSP server
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# Install RTSP server
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sudo apt-get install -y micro ffmpeg lsof
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sudo apt-get install -y micro ffmpeg lsof
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sudo -s cd /root && wget -c https://github.com/bluenviron/mediamtx/releases/download/v1.9.1/mediamtx_v1.9.1_linux_arm64v8.tar.gz -O - | sudo tar -xz
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sudo -s cd /root && wget -c https://github.com/bluenviron/mediamtx/releases/download/v1.18.1/mediamtx_v1.18.1_linux_arm64v8.tar.gz -O - | sudo tar -xz
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```
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```
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</details>
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</details>
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@@ -125,126 +125,6 @@ fi
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```
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```
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</details>
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</details>
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<details>
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<summary>Optional: use gstreamer instead of ffmpeg</summary>
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```bash
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# Install gstreamer
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sudo apt-get update
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#sudo apt-get install -y \
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# gstreamer1.0-rtsp \
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# gstreamer1.0-tools \
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# gstreamer1.0-alsa \
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# gstreamer1.0-plugins-base \
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# gstreamer1.0-plugins-good \
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# gstreamer1.0-plugins-bad \
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# gstreamer1.0-plugins-ugly \
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# gstreamer1.0-libav
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apt-get install libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev libgstreamer-plugins-bad1.0-dev gstreamer1.0-plugins-base gstreamer1.0-plugins-good gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-libav gstreamer1.0-tools gstreamer1.0-x gstreamer1.0-alsa gstreamer1.0-gl gstreamer1.0-gtk3 gstreamer1.0-qt5 gstreamer1.0-pulseaudio -y
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```
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Create a script named `rtsp_audio_server.py`:
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```python
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#!/usr/bin/env python3
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import gi
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import sys
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import logging
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import os
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import signal
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gi.require_version('Gst', '1.0')
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gi.require_version('GstRtspServer', '1.0')
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from gi.repository import Gst, GstRtspServer, GLib
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# Initialize GStreamer
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Gst.init(None)
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# Configure Logging
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LOG_FILE = "gst_rtsp_server.log"
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logging.basicConfig(
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filename=LOG_FILE,
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filemode='a',
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format='%(asctime)s %(levelname)s: %(message)s',
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level=logging.DEBUG # Set to DEBUG for comprehensive logging
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)
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logger = logging.getLogger(__name__)
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class AudioFactory(GstRtspServer.RTSPMediaFactory):
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def __init__(self):
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super(AudioFactory, self).__init__()
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self.set_shared(True)
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self.set_latency(500)
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self.set_suspend_mode(GstRtspServer.RTSPSuspendMode.NONE)
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logger.debug("AudioFactory initialized: shared=True, latency=500ms, suspend_mode=NONE.")
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def do_create_element(self, url):
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pipeline_str = (
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"alsasrc device=plughw:0,0 do-timestamp=true buffer-time=2000000 latency-time=1000000 ! "
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"queue max-size-buffers=0 max-size-bytes=0 max-size-time=0 ! "
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"audioconvert ! "
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"audioresample ! "
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"audio/x-raw,format=S16BE,channels=2,rate=48000 ! "
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"rtpL16pay name=pay0 pt=96"
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)
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logger.debug(f"Creating GStreamer pipeline: {pipeline_str}")
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try:
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pipeline = Gst.parse_launch(pipeline_str)
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if not pipeline:
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logger.error("Failed to parse GStreamer pipeline.")
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return None
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return pipeline
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except Exception as e:
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logger.error(f"Exception while creating pipeline: {e}")
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return None
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class GstServer:
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def __init__(self):
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self.server = GstRtspServer.RTSPServer()
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self.server.set_service("8554")
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self.server.set_address("0.0.0.0")
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logger.debug("RTSP server configured: address=0.0.0.0, port=8554.")
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factory = AudioFactory()
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mount_points = self.server.get_mount_points()
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mount_points.add_factory("/birdmic", factory)
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logger.debug("Factory mounted at /birdmic.")
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self.server.attach(None)
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logger.info("RTSP server attached and running.")
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def main():
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server = GstServer()
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print("RTSP server is running at rtsp://localhost:8554/birdmic")
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logger.info("RTSP server is running at rtsp://localhost:8554/birdmic")
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loop = GLib.MainLoop()
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def shutdown(signum, frame):
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logger.info(f"Shutting down RTSP server due to signal {signum}.")
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print("\nShutting down RTSP server.")
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loop.quit()
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signal.signal(signal.SIGINT, shutdown)
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signal.signal(signal.SIGTERM, shutdown)
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try:
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loop.run()
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except Exception as e:
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logger.error(f"Main loop encountered an exception: {e}")
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finally:
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logger.info("RTSP server has been shut down.")
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if __name__ == "__main__":
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if not os.path.exists(LOG_FILE):
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open(LOG_FILE, 'w').close()
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main()
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```
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</details>
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<details>
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<details>
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<summary>Optional: Startup automatically</summary>
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<summary>Optional: Startup automatically</summary>
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