for a prototype of a project i would like to use a "GardePro E8 Wildlife Camera" to Capture wildlife photos. Then the photos are going to be directly sent to a Jetson Nano device using wifi, where the photos are going to be processed using artificial intelligence. But i have a general problem with connecting the camera with Jetson Nano because of the exclusive connection through the mobile application. For this reason, i would like to ask how is that possible or what is the best way to accomplish the connection.
Its my first post in this community, if i didnt explain everything or forgot something, please dont hesitate and just ask.
Thanx alot for the help
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I'm trying to stream a microphone/audio to multiple clients.
the broadcaster is a screenless raspberry, so I can't open a Webbrowser and click on "share mircophone"
The clients will be using their smartphone to listen.
the latency must be super low.
I did not find any WebRTC Demo that worked. All of them are either p2p or the scalable Broadcasting from muaz khan is only working for the initiator; not clients.
I came across Janus (which I didn't really understand what exactly this is doing) but I don't get how to install this and how to configure it.
Is there any way to easily share the microphone's output via WebRTC? Something like Apache hosting a simple website where the microphone audio is hosted on?
Thanks for all the ideas on how to solve it!
Is there any way to easily share the microphone's output via WebRTC?
No. There's nothing easy or simple about WebRTC.
the broadcaster is a screenless raspberry, so I can't open a Webbrowser and click on "share mircophone"
This is the simplest option... running a browser. Are you sure you need to actually allow it to access the audio device?
In the past, I've used a flag on Chromium to get around this problem. I don't remember exactly what that flag was, but looking at the list, it might have been...
--use-fake-ui-for-media-stream
You might also be able to use --enable-kiosk-mode.
At a minimum, if you were to open the browser interactively and enable access, that page would get automatic access in the future.
I did not find any WebRTC Demo that worked. All of them are either p2p
WebRTC is peer-to-peer, but remember that the "server" can be one of those "peers".
Finally, you can look into using GStreamer, but don't expect anything quick and easy. https://github.com/centricular/gstwebrtc-demos
I have developed a video chat app with webrtc api. I have fallowed the steps given by webrtc. Video working fine. But there is a lot of noice from my laptop. sound is not clear.
But in google developed demo site https://apprtc.appspot.com/ works with out any noise(better compare with us).
I fallowed the same procedure what they did. But no luck.
But in headset this echo is not hearing. This happens when we haering the sound from laptop without headset.
Please give me some suggestion on this.
Thanks in advance. Looking foward for the response.
Give a look at this demo webrtc conferencing supports four callers. This link describes the implentation details architecture
I have a thinkpad laptop and I would like to send AT commands to my modem once it is connected to the network. The interface that is used is '/dev/ttyACM0'. What is the best way to send my commands there? I tried gnokii but wasn't able to connect to my modem :/ I also tried echo'ing commands to the device, this worked, but it is not easy to get the output of the command to proceed with the computation. Maybe one of you guys has a better way to do this. TIA
Have you tried minicom? (assuming from your interface that you are using Linux)
Introduction to minicom.
Entering "minicom" in www.alternativeto.net gives a link to another terminal emulator for Linux called Cutecom, but I have no personal experience of it.
CRTMP Server is great tool... Nat Traversal when client is behind router working great.
...
Tested Android 2.2, 2.3, 4.1, RTSP streaming ok (rtmp flash also ok).
But RTSP on RealPlayer (Helix DNA 10.0 onS60) always shows 'can not play media; or 'can not connect' (connection is surely established - checked with wireshark).
...
(it is programming related problem, because i am willing to explore CRTMP code to accomplish solution)
BBC RTSP channel (wowza is behind) is showed well in symbian realPlayer, but streaming source:
ffmpeg -i rtsp://[bbc_channel_address] -c copy -f rtsp rtsp://[crtmp_server_addr]:8554/ch
... is ok for android, but not working for realPlayerS60.
Does anybody have a clue about reason?
Having RTSP working is not enough. Different phones have different requirements in terms of A/V codecs quality. The content you want to deliver may be too high quality for that device. This assumption fits well with what you said (working on android, but not on realPlayer)
You can ask this questions over the crtmpserver's mailing list. Consult http://rtmpd.com/resources/ for details about the mailing list
Edit:
Or the codecs you try to push from crtmpserver towards the phone are not even supported, let alone hitting the quality max limit
Does anyone know if it is possible to access the iPad's camera connection kit? I would like to read the files off the connected mass storage device. Would this be possible or is this something that only Apple can do in their apps.
Thanks
I know this is an old question, but google brought me here so I thought I'd add this link for the next person to come along.
The good news is this. USB drives do mount properly and show up in the system as /dev/disk2s1. Yay. You can even add more drives via a hub. The iPad supports both FAT and HFS+ drives.
The bad news is this. As iPhone developer Dustin Howett discovered,
that mount point is sandboxed away from normal developer use. You
cannot read from or write to that disk using standard iPhone SDK
applications.