Currently, I'm using a combination of the Windows version of butt and VLC Media Player in order to rebroadcast an icecast stream from a remote icecast server to my local icecast server, whilst disconnecting whenever a certain duration of silence is detected.
I would really prefer if there would be an easier way to do this on linux from the command line, so that it can run 24/7 without me having to constantly fix the VLC stream player every so often.
I need a way for if silence is detected on the stream, it disconnects the relay and will reconnect whenever audio is detected again.
As I've said before, I've been successfully using butt and VLC Media Player on Windows to do this, however would prefer a linux alternative.
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In Ant Media Server after recording stream on Windows using API, the VOD plays fine on Windows. But when playing the same VOD on macOS using Quick Time Player v10.5, the video freezes after some seconds and audio continues.
VODs playback with Quick Time Player is fine for recordings made on macOS.
How can I overcome this and is it an expected behaviour!
TL;DR;
Transcode the video with ffmpeg after recording or add at least one adaptive bitrate on the Ant Media Server side.
This is a known issue in quick time player. This problem also exists for MacOS/iOS and Safari. Let me tell the cause of the problem and offer a solution.
Problem:
The resolution may be changed in WebRTC sessions according to the network conditions so that the resolution of the recording is being changed to lower or higher resolution.
Most of the players and browsers can handle that. On the other hand, Safari and Quick Time Player cannot handle resolution changes and the problem you mention appear.
Solution:
Transcoding the stream into a specific resolution with ffmpeg or using adaptive bitrate on the server side resolves this issue. Typical ffmpeg command is sufficient
ffmpeg -i INPUT.mp4 OUTPUT.mp4
A. Oguz antmedia.io
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I am searching for a fast way to stream my Dektop to an UDP adress. I already thought of using the OBS Studio as capture device for ffmpeg or something like that,but could not find a way to do this. The Vlc player for example is way to slow.
FFmpeg can already capture the desktop: https://trac.ffmpeg.org/wiki/Capture/Desktop
I need to live broadcast multiple RTSP streams out of the audio mixing software StudioOne. For this I am using Jack Audio Connection Kit as the connector. I've already tried using IceCast with Darkice but the latency went up to 6+ seconds which won't work for the project that I'm working on. That's why I'm using the Wowza media server which does RTSP streaming instead of HTTP.
That's where I'm stuck as I need some way of getting the streams from Jack Audio to Wowza on a MAC OS machine. I've tried using FFMpeg but FFMpeg doesn't have the feature to get input from Jack Audio on it's OSX version. I can try to port my whole setup onto an Ubuntu but the mixing software StudioOne isn't available on Ubuntu. I can try using Wine to port StudioOne to Linux but I'm not sure it'll be a good idea for real time mixer to be used as a port, especially when latency is involved.
Is there some other way I can get input from Jack Audio to Wowza Media Server on my MAC?
JACK on OS X is now in FFmpeg as of this commit (67f8a0be545).
Once you have JACK installed, you can compile FFmpeg from source and support should be automatically compiled into FFmpeg.
I've successfully gotten ffmpeg to stream live video from the built-in webcam on my macbook pro to my rtmp server but I cannot figure out how to get it to also send audio from the built-in microphone.
I've tried both the qtkit device as well as the avfoundation. It appears that neither support an audio stream.
Does ffmpeg support audio capture on a mac?
All of the examples I can find only show audio capture working with the DirectShow device.
Turns out it isn't supported at this time. With the help of some of the ffmpeg-devel folks I was able to get it working with a patch. I've applied the patch to a fork available here:
https://github.com/realrunner/FFmpeg
I'm going to set up HTTP Live Streaming server.
In development environment, only Windows PC is available for the server's client due to network restriction and other regulations.
How can I confirm that
server is publishing HLS movie and
real-time movie's delay by viewing it
only with Windows software?
This page shows that browser available on Windows does not support HLS.
I'm prefer solution with free software, so JW Player may not be a best solution.
I'm not sure that I can check delay on-the-fly with hls-client or ffmpeg.
VLC, ffmpeg, and XBMC support playing HLS on Windows. A longer list can be found here: http://en.wikipedia.org/wiki/HTTP_Live_Streaming#Clients