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• Updating ffmpeg breaks opencv library

  7 octobre 2019, par Amit

  I had manually compiled opencv ver3.2 library on mac system, which had been working fine till now. Recently it seems that i had updated the version for ffmpeg, which is managed by brew package manager, leading to breakage of opencv library looking for libavcodec.57.dylib.

  Upon looking into library path, i found that libavcodec.dylib is symlinked to libavcodec.58.dylib, which is the most probable cause to above problem.

  Now i have two options,
  
   Recompile opencv to link against the latest version of libavcodec
  
   Revert the version of libavcodec to 57

  I am looking for a solution in which opencv works independently of the version of libavcodec and should dynamically link with current installed version of library. I should only be recompiling opencv when it is essentially required, like change in API of libavcodec.

• Video Ending prematurely with websocket and ffmpeg

  4 septembre 2024, par Zaid

  I am working on a screen recording application and it allows users to record their screen and I use websocket to send byte data in real time to my fastapi python server. The bytes are send every 2 second and I use ffmpeg to keep saving the bytes in the output mp4 video file. Everything was working fine when I had the server running on my local machine, however, I just deployed the server to EC2 instance us-ease-1 and when I try to record the videos, the videos are really short, IE, if I record a 30 second video, it only has the video for 3 second. Sometimes it saves 90% of the video and sometimes less. I am not sure what is the problem here.
I have been trying to debug the code for the past few days, but no success

  


  Here is my code :-

  


  FRONTEND

  


  const recorder = new MediaRecorder(stream, {
            mimeType: 'video/webm;codecs=H264',
            videoBitsPerSecond: 8000000
        });

recorder.ondataavailable = (e: BlobEvent) => {
    socketRef.socket.send(e.data)
}


  


  And here is my python code :-

  


  @router.websocket("/stream")
async def websocket_endpoint(websocket: WebSocket, token: str = Query(...), videoId: str = Query(...), authorize: AuthJWT = Depends()):
    await manager.connect(websocket)
    dataNumber = 1

    recordingFile = os.path.join(temp_dir, f"recording_{videoId}.mp4")

    command = [
        'ffmpeg', 
        '-y',
        '-i', 
        '-', 
        '-codec:v', 
        'copy', 
        '-f', 'mp4',
        recordingFile,
    ]

    process = subprocess.Popen(command, stdin=subprocess.PIPE)
    try:
        while True:
            try:
                data = await websocket.receive_bytes()
                if not data:
                    break
                process.stdin.write(data)
                await websocket.send_json({"chunkNumber": dataNumber, "status": 200})
                dataNumber = dataNumber + 1
            except RuntimeError:
                break      
    except WebSocketDisconnect:
        print(f"Client disconnected: {websocket.client.host}")
    finally:
        manager.disconnect(websocket)

        # Close stdin to signal EOF
        process.stdin.close()

        # Wait for FFmpeg to finish processing
        process.wait()

        # Ensure that the process is terminated
        process.terminate()  


  


  I also get this error in the console :-


  


• Setting up RTP on Nginx

  2 février 2021, par Swap

  I'm trying to use Janus Media Server to relay WebRTC streams to a particular RTP host/port, from where ffmpeg can pick it up as an input and convert it further to an rtmp stream, which can then be used to broadcast to various social media platforms (such as, YouTube, Twitch, Facebook, etc.)

  


  My inspiration for this has been the following blog - https://www.meetecho.com/blog/firefox-webrtc-youtube-kinda/

  


  Specifically, I'm trying to replicate the following architecture -

  


  

  


  And Janus, as per their documentation, has a very neat API for doing it -

  


  {&#xA;    "request" : "rtp_forward",&#xA;    "room" : <unique numeric="numeric" of="of" the="the" room="room" publisher="publisher" is="is" in="in">,&#xA;    "publisher_id" : <unique numeric="numeric" of="of" the="the" publisher="publisher" to="to" relay="relay" externally="externally">,&#xA;    "host" : "<host address="address" to="to" forward="forward" the="the" rtp="rtp" and="and" packets="packets">",&#xA;    "host_family" : "",&#xA;    "audio_port" : <port to="to" forward="forward" the="the" audio="audio" rtp="rtp" packets="packets">,&#xA;    "audio_ssrc" : <audio ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_pt" : <audio payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_rtcp_port" : <port to="to" contact="contact" receive="receive" audio="audio" rtcp="rtcp" feedback="feedback" from="from" the="the" and="and" currently="currently" unused="unused" for="for">,&#xA;    "video_port" : <port to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets">,&#xA;    "video_ssrc" : <video ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "video_pt" : <video payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "video_rtcp_port" : <port to="to" contact="contact" receive="receive" video="video" rtcp="rtcp" feedback="feedback" from="from" the="the" optional="optional">,&#xA;    "simulcast" : ,&#xA;    "video_port_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" second="second" substream="substream"></if>layer to>,&#xA;    "video_ssrc_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_pt_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_port_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" third="third" substream="substream"></if>layer to>,&#xA;    "video_ssrc_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "video_pt_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "data_port" : <port to="to" forward="forward" the="the" messages="messages">,&#xA;    "srtp_suite" : <length of="of" authentication="authentication" tag="tag" or="or" optional="optional">,&#xA;    "srtp_crypto" : "<key to="to" use="use" as="as" crypto="crypto" encoded="encoded" key="key" in="in" optional="optional">"&#xA;}&#xA;</key></length></port></port></video></video></port></port></audio></audio></port></host></unique></unique>

  


  For this, I've setup a Nginx server, where I've also installed Janus and everything's been running smoothly so far. But I'm quite clueless as to how to setup my Nginx server so that it accepts RTP connections (which will be forwarded as RTMP using ffmpeg).

  


  Please guide me to any relevant resources that would help me achieve this. Thanks in advance !