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fix lots of errors

main
Hendrik Langer 4 years ago
parent
commit
b935b9ee13
  1. 1
      raspberry/requirements.py
  2. 64
      raspberry/roberto/camera/camera_gstreamer_webrtc.py

1
raspberry/requirements.py

@ -3,6 +3,7 @@ python3-flask-sqlalchemy
python3-flask-login
python3-flask-socketio
python3-socketio
python3-websocket # python3-socketio-client
python3-serial
python3-opencv
python3-picamera

64
raspberry/roberto/camera/camera_gstreamer_webrtc.py

@ -1,5 +1,7 @@
import os
import sys
import time
import traceback
from threading import Thread
from queue import Queue
@ -82,9 +84,9 @@ except (ImportError, RuntimeError):
#############
PIPELINE_DESC = '''
v4l2src device=/dev/video0 ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay pt=96 ! tee name=video-tee !
v4l2src device=/dev/video0 ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! tee name=video-tee !
queue ! fakesink sync=true
audiotestsrc wave=red-noise is-live=true ! opusenc ! rtpopuspay pt=97 ! tee name=audio-tee !
audiotestsrc wave=red-noise is-live=true ! opusenc ! rtpopuspay ! queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! tee name=audio-tee !
queue ! fakesink sync=true
'''
@ -101,7 +103,7 @@ PIPELINE_DESC_RASPBERRY = '''
PEER_BIN_DESC = '''
queue name=video-queue ! webrtcbin.
queue name=audio-queue ! webrtcbin.
webrtcbin name=webrtcbin
webrtcbin name=webrtcbin bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302 latency=40
'''
@ -111,7 +113,7 @@ class WebRTCCamera(Thread):
self.sid = None
self._num_clients = 0
self.connected = False
self.server = 'ws://localhost:5000'
self.server = 'ws://127.0.0.1:5000'
self._queue = Queue(maxsize=10)
self._peers = {}
self.video_tee = None
@ -128,7 +130,10 @@ class WebRTCCamera(Thread):
self.connect()
while True:
try:
item = self._queue.get()
if item['sid'] == self.sid:
continue
if item['job'] == "connect_client":
if not self.pipe:
self.start_global_pipeline()
@ -140,16 +145,26 @@ class WebRTCCamera(Thread):
if self._num_clients == 0:
print("last client left. stopping global pipeline")
self.close_global_pipeline()
else:
print("%d clients left" % self._num_clients)
else:
print("unknown job: %s" % item['job'])
self._queue.task_done()
except:
e = sys.exc_info()[0]
print("Exception in camera task: %s" % e )
#print(sys.exc_info()[2])
print(traceback.format_exc())
self.disconnect()
def connect(self):
while not sio.sid:
print("connecting camera websocket..")
sio.connect(self.server, namespaces=['/webrtc'], transports=['websocket'])
print('my sid is', sio.sid)
time.sleep(1)
print('camera sid is', sio.sid)
self.sid = sio.sid
self.connected = True
@ -160,8 +175,6 @@ class WebRTCCamera(Thread):
def connect_client(self, sid, room):
if not self.is_alive():
self.start()
if sid == self.sid:
return
self._queue.put({'job':'connect_client', 'sid':sid})
def disconnect_client(self, sid, room):
@ -187,26 +200,35 @@ class WebRTCCamera(Thread):
#video_mixer = self.pipe.get_by_name('video-mixer')
#audio_mixer = self.pipe.get_by_name('audio-mixer')
bus = self.pipe.get_bus()
#bus = self.pipe.get_bus()
#bus.add_signal_watch()
#bus.connect('message', self.bus_message)
##pollfd = self.bus.get_pollfd()
##asyncio.get_event_loop().add_reader(pollfd.fd, self.poll_cb)
self.pipe.set_state(Gst.State.PLAYING)
print("GLOBAL PIPELINE RUNNING")
def close_global_pipeline(self):
print("CLOSING PIPELINE")
#bus = self.pipe.get_bus()
##bus.disconnect('message')
#bus.remove_signal_watch()
self.pipe.set_state(Gst.State.NULL)
self.pipe = None
def start_client_pipeline(self, client_sid):
print("starting client pipeline")
print("starting client pipeline for client [%s]" % client_sid)
self._peers[client_sid] = {}
peer_bin = Gst.parse_bin_from_description(PEER_BIN_DESC, False)
webrtcbin = peer_bin.get_by_name("webrtcbin")
webrtcbin.set_property("stun-server", "stun://stun.l.google.com:19302")
webrtcbin.set_property("bundle-policy", "max-bundle")
# webrtcbin.set_property("stun-server", "stun://stun.l.google.com:19302")
# webrtcbin.set_property("bundle-policy", "max-bundle")
#webrtcbin.set_property("latency", 40)
rtpbin = webrtcbin.get_by_name('rtpbin')
rtpbin.set_property("latency", 40)
##webrtcbin.set_property("latency", 40)
#rtpbin = webrtcbin.get_by_name('rtpbin')
#rtpbin.set_property("latency", 40)
audio_queue = peer_bin.get_by_name("audio-queue")
audio_sink_pad = Gst.GhostPad.new("audio_sink", audio_queue.get_static_pad("sink"))
@ -237,9 +259,16 @@ class WebRTCCamera(Thread):
video_src_pad.link(video_sink_pad)
peer_bin.sync_state_with_parent()
#test
#self.pipe.sync_children_states()
#webrtc.sync_children_states()
#debug
#video_caps = video_src_pad.get_current_caps()
#if not video_caps:
# video_caps = video_src_pad.query_caps()
#print(video_caps)
audio_src_pad.remove_probe(audio_block)
video_src_pad.remove_probe(video_block)
@ -247,6 +276,13 @@ class WebRTCCamera(Thread):
print("blocked")
return Gst.PadProbeReturn.OK
# def bus_message(self, bus, msg):
# t = msg.type
# print("BUS MESSGE")
# if t == gst.MessageType.ERROR:
# busError, detail = msg.parse_error()
# print("gstreamer bus message error: %s" % busError)
def close_client_pipeline(self, client_sid):
print("closing client pipeline")
webrtcbin = self._peers[client_sid]['webrtcbin']

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