rtmp | Golang rtmp server - Run a simple server | Stream Processing library
kandi X-RAY | rtmp Summary
kandi X-RAY | rtmp Summary
Run a simple server. Use avconv to publish stream. Use avplay to play stream.
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Top functions reviewed by kandi - BETA
- handlePlay is a play event handler
- listenEvent listens for events
- serve reads messages from a MsgStream
- Read a message from the underlying reader
- ReadAMF reads an AMF object from an io . Reader
- WriteAMF writes an AMF object to the given io . Writer .
- Handles CONNECT command
- parseChal parses b and returns dig digest .
- SimpleServer starts accepting incoming connections
- handShake is the handshaking handshake
rtmp Key Features
rtmp Examples and Code Snippets
Community Discussions
Trending Discussions on rtmp
QUESTION
my setup for codecov has worked well so far
- you can regular updates with each pr commits here
- I haven't change my repo settings
as I've inadvertently pushed a folder that I wasn't supposed to,
then I merged a pr to remove said folderhere is my codecov.yml
- on the aforementioned last pr linked above the github action ci complained with the log below
ANSWER
Answered 2021-Jun-06 at 17:47Codecov has some heisenberg issues. If you don't have a token, please add one otherwise try to:
- Force-push to retrigger Codecov
- Rotate your token.
QUESTION
I cannot find any information how to handle the situation like this:
Stream starts: about 3 o'clock.
1.Before the person who is streaming (let's call him a streamer) start to stream I would like to have static image saying something like: 'The event will start soon'.
2.Streamer start pushing his stream to RTMP endpoint but he's late and starts at 3.02. Up until 3.02 the same picture should be visible (as in point 1).
3.Streamer should finish at 4 o'clock but he finishes 5 minutes before 4 (pushing stop
at his device).
4.Now, ending screen should be visible from 5 minutes to four and later.
I know that inputs should be switched in order to change a view and this can be scheduled for fixed time, but I would like this to be switched dynamically, ie. when streamer starts pushing to RTMP URL and stops pushing to RTMP URL (from eg. Larix software). How to handle that in AWS Media Live?
...ANSWER
Answered 2021-Jun-08 at 18:40Thank you for asking this question on stackoverflow, the easiest way to achieve what you are looking to do is by using an Input Prepare Scheduled Action. The channel will then monitor the input and raise an alarm if the RTMP source is not there. When the RTMP source begins then the alarm will remit, you can send the alarms to a lambda that will look for these alarms and can do the switch from slate MP4 to the RTMP source when it sees the RTMP input missing alarm was cleared. This can also be done for when RTMP input goes away.
Information on Prepare Inputs: https://docs.aws.amazon.com/medialive/latest/ug/feature-prepare-input.html
Global configuration - Input loss behavior: https://docs.aws.amazon.com/medialive/latest/ug/creating-a-channel-step3.html
Zach
QUESTION
2 FFMPEG process
(1) generating a ffmpeg x11grab to a .mp4 (2) take the .mp4 and restream it simultaneously to multiple rtmp endpoints
ISSUE the generated file in (1) have this error "moov atom not found"
This is the command that generate (1) :
...ANSWER
Answered 2021-Jun-02 at 03:01With those changes, I'm able to acheive 3 to 4 stable delay ;)
LINE 79 of
I REPLACED
QUESTION
I am trying to build a web app for users to easily add text (as open caption) and other assets in my app as overlays in real-time to their YouTube live stream video.
They will use their camera to record their video, and select from my app which text should be added to the video.
Then, the video will be sent to Youtube live through their API.
Here are my questions:
First of all, I was wondering if mixing video + subtitle and sending it to Youtube's rtmp url can be done from the client side, so it's simple and lightweight.
Second, should I encode the output being sent to Youtube? Can this be done from the browser too? I'm only seeing a few node.js frameworks, and even they're not very mature (or is Webcodecs for this purpose?). Is a web app a poor choice for this task?
Lastly, if I do need a server to process the video, where should the encoding happen (from the user's machine, or in the server, or both?)? Is my server most likely going to be the bottleneck given YouTube's infrastructure, since video files are huge and my server is limited?
I am new to video streaming, so please excuse my lack of understanding of the subject. Also, if there's any good resource for my problem, please share them with me.
...ANSWER
Answered 2021-May-29 at 18:13First of all, I was wondering if mixing video + subtitle and sending it to Youtube's rtmp url can be done from the client side, so it's simple and lightweight.
You can do the video compositing and audio mixing and what not, but browsers don't support RTMP. To get the data to an RTMP server, you need to send it to a server where it is proxied off to the final URL.
They will use their camera to record their video, and select from my app which text should be added to the video.
Yeah, that's no problem at all. Draw everything to a canvas every frame.
Second, should I encode the output being sent to Youtube?
Yes, you must. Check out the Media Recorder API.
Lastly, if I do need a server to process the video, where should the encoding happen (from the user's machine, or in the server, or both?)?
The video has to be encoded client-side to get to the server in the first place. The server can then hopefully just repackage with flv and send it along. If the browser doesn't support H.264 in its Media Recorder API, then you'll have an intermediary codec like VP8, and you'll have to transcode server-side.
A few years ago, I wrote a tutorial on how to do all of these steps here: https://github.com/fbsamples/Canvas-Streaming-Example Note that the tutorial is in the context of Facebook, but this should teach you the concepts.
QUESTION
Get that error when try to get stream from any HLS source.
I tried to add videojs-contrib-hls lib , but its dont help. Maybe should i try some other player, and what player will properly work with hls sources?
...ANSWER
Answered 2021-May-26 at 12:14I used another player vue-vjs-hls. On this player hls work good, dont now why hls source not works at vue-video-player and video.js. What strange becouse vue-vjs-hls use video.js as core.
QUESTION
I've been learning GStreamer to manage and forward streams from one RTMP server to another. I've hit a roadblock when trying to replicate the tutorial found here, albeit without the data buffer. Whenever I run the program, it stops when I set the rtmpsink
URL. It doesn't throw an error on that line or report that the pipeline couldn't start to ret
, it just stops. The first part of the code is below
ANSWER
Answered 2021-May-20 at 09:45g_object_set()
needs to end with a NULL
parameter to indicate an end as you can set multiple parameters with one call.
Try:
QUESTION
I am able to stream with ffmpeg an mp4 file with h264 encoded video and aac encoded audio to flash flv output. The command I used is:
ffmpeg -re -i bigbuckbunny_HD_60fps.mp4 -c copy -f flv rtmp://192.168.3.64:1935/main/stream0
When I am trying to use the tee option to duplicate the output I am failing with an error. This is the command I use:
ffmpeg -re -i bigbuckbunny_HD_60fps.mp4 -map 0 -c copy -f tee "[f=flv]rtmp://192.168.3.64:1935/main/stream0|[f=flv]rtmp://192.168.3.64:1935/main/stream1"
However, it fails. What I am missing? The output I have from ffmpeg is:
...ANSWER
Answered 2021-May-19 at 19:28According to ffmpeg -f tee
refuses to do anything useful add -tag:v 7
(or -vtag 7
, same thing):
QUESTION
I am able to stream with ffmpeg an mp4 file with h264 encoded video and aac encoded audio to flash flv output. The command I used is:
...ANSWER
Answered 2021-May-19 at 18:09You have to add the -map
option when using the tee muxer. From the tee muxer documentation:
Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select output streams. So all streams intended for output must be specified using
-map
.
Example to include all streams from the input (-map 0
):
QUESTION
I tried to add an already active stream to a new broadcast, and can't get the broadcast started. The steps I took were.
- Created a new Broadcast.
ANSWER
Answered 2021-May-18 at 20:36I figured it out.
Apparently you cannot have a brodcast created with enableAutoStart=true and then add an active stream. It seems that enableAutoStart=true fails the broadcast transition API calls to change the status to testing or live or complete.
To get this to work, I stopped then started sending to the stream, which caused the stream to transition to inactive then back to active. The transition caused the broadcast to start.
Alternatively, to get this to work without the restart of the stream, I did the following:
- create the broadcast with enableAutoStart=false
- bind the active stream to the broadcast (as in the question).
- transition the broadcast to testing, then to live.
This seems to work fine.
Would have been nice to have the error message for transitioning indicate it was the enableAutoStart which was the problem.
QUESTION
I would like to find the streams associated with a broadcast with 'ready' status. I've been looking at the broadcasts using this call, and don't see either the streams or a key I can use to correlate them:
...ANSWER
Answered 2021-May-17 at 21:14My mistake, adding contentDetails to part fixed this.
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