Update DOCS.md

This commit is contained in:
Alexandre
2026-05-15 13:22:18 +02:00
committed by GitHub
parent 211074f609
commit 3c2df45ece

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