Here, Data transfer is for controlling video pause and video record. We are using iMX8Mini Eval board - for streaming video to Android Mobile via USB-OTG. We would like to know, whether video stream does not affected with any command sent over same USB-OTG.
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I use getUserMedia to the the audio stream, and pass the stream into web audio using createMediaStreamSource. I then want to stream the raw audio data into a webrtc data channel.
There isn’t a data channel destination node in web audio. I’ve only been able to access the raw audio data from inside an audio worklet, but I don’t know how to get that data into a data channel. How should I go about streaming raw audio from getUserMedia data into a data channel?
I'm working on an application that transmits video in low quality using webrtc. Periodically I want to send from same camera single frame in high resolution.
When I try to acquire another stream using getUserMedia I get same low quality one and when I try to pass some constraints to force higher resolution then then operation fails with overconstrained error (even though normally when there is no other stream it works fine).
Is it even possible to have at the same time many streams with different parameters from same device? Or is it possible acquire high resolution image without requesting for a new stream?
as the title shows, is there any methods I can use to play multiple videos continuously using simple rtmp client(my rtmp server is wowza)? Here is the way I think:
when the first video is about to be finished，open a new thread to send a new createStream command and a new play command and get the video rtmp packet and put them into a buffer list, when the first video is finished, then play the video rtmp in the buffer list..
Can this way be available or are there any other recommended methods to achieve it? Any suggestion will be appreciated, thanks!
While the functionality is not built-in, Wowza does have a module called StreamPublisher to allow you to implement a server-side type of playlist. The source code for the module is available on GitHub. A scheduled playlist of multiple VOD files is streamed as a live stream, similar to a TV broadcast.
Suppose we get an audio stream by using getUserMedia(gUM) API. I am broadcasting this stream to other users using WebRTC.
Is it possible to reduce the volume of the audio getting streamed.
Note : I am not looking to reduce device mic volume, because I understand we cannot control through browser and I dont want to
Try with GainNode from WebAudio. But the best thing will be to do this on the receiving end.
how to interrupt rtmp flv broadcast and publish live rtmp broadcast on red5?
I am using osmf strobe player. I have my flv playlist working but when I broadcast live from my webcam what is the formula to stop the flv streams, then play flv countdown video then connect live broadcast from web cam?
Here is how I would do it, from a high-level since I don't have the code for what you're asking and "easy" transitions between streams isn't built-in to the server.
First, create a signaling or event system within your app to accept actions triggered by your broadcaster. Using the signaling system, transition your subscribers / viewers by sending triggered events telling their players to stop playing a current video and start a new one. I suggest using Shared Objects for this when passing signals around. Use server-side methods called by your broadcaster to send the signals on the Share Object. The "play" functionality is the easy part since you simply provide the stream name in your signal / event.