Comments (4)
Hi,
The problem with a webrtc source is that the audio may come late compared to the video. One of the solutions to get around this problem was to add a fake audio with zero volume and to mix it with the audio from the webrtc :
gst-launch-1.0 rtspclientsink stream-name='stream_@channelid@' name=r user-agent='cef-player/1.0' location=@rtsp_url@ cefsrc url='<url_webrtc_source>' ! queue ! cefdemux name=t t. ! video/x-raw ! videoconvert ! x264enc threads=4 tune=zerolatency speed-preset=ultrafast ! video/x-h264 ! queue ! r. audiotestsrc volume=0.0 is-live=true ! queue ! audiomixer name=m ! audioconvert ! voaacenc bitrate=128000 ! aacparse ! queue ! r. t. ! audio/x-raw ! queue ! m."
Hope it will help!
BR
Thierry
from gstcefsrc.
@thierrygayet just saw your issue and I wanted to ask if you already have an idea or answer?
I used this very nice plugin for video only in python and docker and it works like a charm.
Today I wanted to add audio, so I used the cefdemux
and nothing works anymore, even if I use
the demux for just video.
As audio_%u
is a sometimes
-pad I even added a SIGNAL callback in python
cefdemux.connect("pad-added", on_incoming_audio_pad)
but this is function on_incoming_audio_pad
is never called.
@MathieuDuponchelle any idea?
CLI like @thierrygayet shown does not work either.
Thank you in advance, for this nice work and would appreciate if you have an answer as well.
from gstcefsrc.
@thierryg okay, I see.
Very hacky but I will try it, thank you so far!
@MathieuDuponchelle what would be the common way to handle this challenge?
The cefmuxer
has this sometimes
-pad and mp4mux
needs to request_pad
before getting into PLAYING
- OR PAUSED
-state?
from gstcefsrc.
@michael-riha using audiomixer is an appropriate workaround
from gstcefsrc.
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