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• Recording Video in splits using Web Media Recorder and ffmpeg

  5 août 2022, par NerosAngel

  I'm currently working on a project where I want to record and process video splits parallel ( in order to minimize time taken ) .

  


  I've been using MediaRecorder to capture the video and audio and send it to the server on where it is processed using ffmpeg. And it is finally concatenated using concat demuxer .

  


  Unfortunately because of how Web MediaRecorder works(see more https://stackoverflow.com/a/51355276) it creates some audio gaps in video.

  


  Audio gaps in video

  


  I've been trying to encode the video to remove this gap and processing the audio out separately in splits . Tried using raw PCM audio but eventually pinned down the issue is from the MediaRecorder's side.

  


  I think there might be some way to re encode the video to get rid of this gaps , Could someone tell me if this is possible or should I abandon this approach for it ?

  


• Media conversion on AWS

  27 juin 2022, par Gurmeet Singh

  I have an API written in nodeJS (/api/uploadS3) which is a PUT request and accepts a video file and a URL (AWS s3 URL in this case). Once called its task is to upload the file on the s3 URL.

  


  Now, users are uploading files to this node API in different formats (thanks to the different browsers recording videos in different formats) and I want to convert all these videos to mp4 and then store them in s3.

  


  I wanted to know what is the best approach to do this ?

  


  I have 2 solutions till now

  


  1. Convert on node server using ffmpeg -

  


  The issue with this is that ffmpeg can only execute a single operation at a time. And since I have only one server I will have to implement a queue for multiple requests which can lead to longer waiting times for users who are at the end of the queue. Apart from that, I am worried that during any ongoing video conversion if my node's traffic handling capability will be affected.

  


  Can someone help me understand what will be the effect of other requests coming to my server while video conversion is going on ? How will it impact the RAM, CPU usage and speed of processing other requests ?

  


  2. Using AWS lambda function -

  


  To avoid load on my node server I was thinking of using an AWS lambda server where my node API will upload the file to S3 in the format provided by the user. Once, done s3 will trigger a lambda function which can then take that s3 file and convert it into .mp4 using ffmpeg or AWS MediaConvert and once done it uploads the mp4 file to a new s3 path. Now I don't want the output path to be any s3 path but the path that was received by the node API in the first place.

  


  Moreover, I want the user to wait while all this happens as I have to enable other UI features based on the success or error of this upload.

  


  The query here is that, is it possible to do this using just a single API like /api/uploadS3 which —> uploads to s3 —> triggers lambda —> converts file —> uploads the mp4 version —> returns success or error.

  


  Currently, if I upload to s3 the request ends then and there. So is there a way to defer the API response until and unless all the operations have been completed ?

  


  Also, how will the lambda function access the path of the output s3 bucket which was passed to the node API ?

  


  Any other better approach will be welcomed.

  


  PS - the s3 path received by the node API is different for each user.

  


• Get composition time C++

  13 novembre 2019, par Andrea

  I made my own rtmp server using libav and ffmpeg. I receive as input either an flv file or an rtmp streaming "containing" an flv file.
  Since I manipulate the flv file and the relative composition time of each frame, I would like to know if there is a way to get this composition time.
  I thought that given my AVPacket, I could analyze the raw buffer in order to extract the right information since I know that the flv header is 11 bytes and then in the next 16 bytes I should find the composition time.
  But it doesn’t work.

  This is a rough example of code :

  AVPacket pkt;
  
  AVFormatContext *ifmt_ctx
  
  while(true)
  
  {
  
      AVStream *in_stream, *out_stream;
  
  
  
      ret = av_read_frame(ifmt_ctx, &pkt);
  
      //get the composite time
  
  }