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enhancement #48: rename example folder to examples. provide simple re…
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Original file line number | Diff line number | Diff line change |
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import argparse | ||
from typing import Optional | ||
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import gi | ||
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gi.require_version("Gst", "1.0") | ||
from gi.repository import Gst | ||
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from gst_signalling.utils import find_producer_peer_id_by_name | ||
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class GstRecorder: | ||
def __init__( | ||
self, signalling_host: str, signalling_port: int, peer_id: Optional[str] = None, peer_name: Optional[str] = None | ||
) -> None: | ||
Gst.init(None) | ||
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self.pipeline = Gst.Pipeline.new("webRTC-recorder") | ||
self.source = Gst.ElementFactory.make("webrtcsrc") | ||
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if not self.pipeline: | ||
print("Pipeline could be created.") | ||
exit(-1) | ||
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if not self.source: | ||
print( | ||
"webrtcsrc component could not be created. Please make sure that the plugin is installed \ | ||
(see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/main/net/webrtc)" | ||
) | ||
exit(-1) | ||
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self.pipeline.add(self.source) | ||
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if peer_id is None: | ||
peer_id = find_producer_peer_id_by_name(signalling_host, signalling_port, peer_name) | ||
print(f"found peer id: {peer_id}") | ||
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self.source.connect("pad-added", self.webrtcsrc_pad_added_cb) | ||
signaller = self.source.get_property("signaller") | ||
signaller.set_property("producer-peer-id", peer_id) | ||
signaller.set_property("uri", f"ws://{signalling_host}:{signalling_port}") | ||
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def webrtcsrc_pad_added_cb(self, webrtcsrc: Gst.Element, pad: Gst.Pad) -> None: | ||
if pad.get_name().startswith("video"): # type: ignore[union-attr] | ||
videodepay = Gst.ElementFactory.make("rtph264depay") | ||
assert videodepay is not None | ||
gdppay = Gst.ElementFactory.make("gdppay") | ||
assert gdppay is not None | ||
filesink = Gst.ElementFactory.make("filesink") | ||
assert filesink is not None | ||
filesink.set_property("location", f"{pad.get_name()}.gdp") | ||
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self.pipeline.add(videodepay) | ||
self.pipeline.add(gdppay) | ||
self.pipeline.add(filesink) | ||
videodepay.link(gdppay) | ||
gdppay.link(filesink) | ||
pad.link(videodepay.get_static_pad("sink")) # type: ignore[arg-type] | ||
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videodepay.sync_state_with_parent() | ||
gdppay.sync_state_with_parent() | ||
filesink.sync_state_with_parent() | ||
elif pad.get_name().startswith("audio"): # type: ignore[union-attr] | ||
audiodepay = Gst.ElementFactory.make("rtpopusdepay") | ||
assert audiodepay is not None | ||
gdppay = Gst.ElementFactory.make("gdppay") | ||
assert gdppay is not None | ||
filesink = Gst.ElementFactory.make("filesink") | ||
assert filesink is not None | ||
filesink.set_property("location", f"{pad.get_name()}.gdp") | ||
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self.pipeline.add(audiodepay) | ||
self.pipeline.add(gdppay) | ||
self.pipeline.add(filesink) | ||
audiodepay.link(gdppay) | ||
gdppay.link(filesink) | ||
pad.link(audiodepay.get_static_pad("sink")) # type: ignore[arg-type] | ||
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audiodepay.sync_state_with_parent() | ||
gdppay.sync_state_with_parent() | ||
filesink.sync_state_with_parent() | ||
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def __del__(self) -> None: | ||
Gst.deinit() | ||
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def get_bus(self) -> Gst.Bus: | ||
return self.pipeline.get_bus() | ||
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def record(self) -> None: | ||
# Start playing | ||
ret = self.pipeline.set_state(Gst.State.PLAYING) | ||
if ret == Gst.StateChangeReturn.FAILURE: | ||
print("Error starting playback.") | ||
exit(-1) | ||
print("recording ... (ctrl+c to quit)") | ||
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def stop(self) -> None: | ||
print("stopping") | ||
self.pipeline.send_event(Gst.Event.new_eos()) | ||
self.pipeline.set_state(Gst.State.NULL) | ||
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def process_msg(bus: Gst.Bus) -> bool: | ||
msg = bus.timed_pop_filtered(10 * Gst.MSECOND, Gst.MessageType.ANY) | ||
if msg: | ||
if msg.type == Gst.MessageType.ERROR: | ||
err, debug = msg.parse_error() | ||
print(f"Error: {err}, {debug}") | ||
return False | ||
elif msg.type == Gst.MessageType.EOS: | ||
print("End-Of-Stream reached.") | ||
return False | ||
# else: | ||
# print(f"Message: {msg.type}") | ||
return True | ||
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def main() -> None: | ||
parser = argparse.ArgumentParser(description="webrtc gstreamer simple recorder") | ||
parser.add_argument("--signaling-host", default="127.0.0.1", help="Gstreamer signaling host") | ||
parser.add_argument("--signaling-port", default=8443, help="Gstreamer signaling port") | ||
parser.add_argument( | ||
"--remote-producer-peer-id", | ||
type=str, | ||
help="producer peer_id", | ||
) | ||
parser.add_argument( | ||
"--remote-producer-peer-name", | ||
type=str, | ||
help="producer name", | ||
) | ||
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args = parser.parse_args() | ||
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if args.remote_producer_peer_id is None and args.remote_producer_peer_name is None: | ||
exit("You must set either remote_producer_peer_id or remote_producer_peer_name") | ||
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recorder = GstRecorder( | ||
args.signaling_host, args.signaling_port, args.remote_producer_peer_id, args.remote_producer_peer_name | ||
) | ||
recorder.record() | ||
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# Wait until error or EOS | ||
bus = recorder.get_bus() | ||
try: | ||
while True: | ||
if not process_msg(bus): | ||
break | ||
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except KeyboardInterrupt: | ||
print("User exit") | ||
finally: | ||
# Free resources | ||
recorder.stop() | ||
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if __name__ == "__main__": | ||
main() | ||
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""" | ||
This will create video_x.gdp streams. You can mux them using: | ||
gst-launch-1.0 \ | ||
mp4mux name=mux ! filesink location=recording.mp4 \ | ||
filesrc location=video_0.gdp ! gdpdepay ! h264parse ! queue ! mux. \ | ||
filesrc location=video_1.gdp ! gdpdepay ! h264parse ! queue ! mux. \ | ||
filesrc location=audio_0.gdp ! gdpdepay ! opusparse ! queue ! mux. | ||
""" |