Closed thierrygayet closed 3 years ago
@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.
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='
Hope it will help!
BR Thierry
@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?
@michael-riha using audiomixer is an appropriate workaround
Hi Mathieu,
I am testing a local record using the cefsrc and cefdemux modules. What I observe is a problem out of the sound and I would like to have your opinion on what can happen.
The pipeline I use for a local recording is:
As you can see, I get an empty file:
What I can see is that I do not get a dot file for the PLAYING state; i just get READY then PAUSE.
However, when I use another source for the sound, namely the gstreamer module audiotestsrc, what I get seems ok with sound this time:
Do you have an idea what could happen? Any idea?
Thank you in advance for your return.
Thierry