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• Live Stream using .m3u8 and .ts files with iPhone as server

  26 février 2015, par Bhumit

  I am trying to accomplish a task to live stream from iPhone camera. I have done some research and found that i can use .m3u8 files for streaming live video with should contain .ts(Mpeg-2) files .

  Now the file which i have on my iPhone is .mp4 file and it does not work with .m3u8, so i figured out i will have to convert .mp4 to .ts for that , but i have not succeeded in doing so.

  I found that it is possible to convert video ffmpeg lib as mentioned in this article here. I have successfully imported ffmpeg library but not able figure out how can i use it to convert a video as i am using this for first time.

  One another thing apple documentation says

  There are a number of hardware and software encoders that can create
  MPEG-2 transport streams carrying MPEG-4 video and AAC audio in real
  time.

  What is being said here ? is there any other way i can use .mp4 files for live streaming without converting them from iOS ?

  Let me know if i am not clear, i can provide more information .Any suggestion is appreciated. I would like to know am i on a right path here ?

  EDIT

  I am adding more info to my question, so basically what i am asking is , we can convert .mp4 video to .ts using following command

  ffmpeg -i file.mp4 -acodec libfaac -vcodec libx264 -an -map 0 -f segment -segment_time 10 -segment_list test.m3u8 -segment_format mpegts -vbsf h264_mp4toannexb -flags -global_header stream%05d.ts

  How can i use ffmpeg library to do what this command does in iOS.

• Node.js Live Streaming : Avoid buffering

  27 octobre 2012, par Shirish Kamath

  I've written a small nodeJS server that outputs system audio captured by ffmpeg on Windows (using DirectShow) to the browser as a streaming MP3 file. The audio needs to be as live as possible, with minimum/no buffering, and a "skipping" effect in the audio is perfectly acceptable.

  When I play the audio in Chrome using the HTML5 audio tag, there's a delay of about 8-10 secs over a low-latency LAN connection. I suspected this to be a client-side buffer, and used a Flash MP3 player on the client-side, which brought down the delay to 2-3 secs.

  Now, the buffering seems to taking place on the server-side. The documentation for NodeJS's response.write mentions that the data is written kernel buffers. How do I go about avoiding any buffering altogether or at least getting around it, so that the client always gets the latest audio data ? Strategies for handling 'drain' events to always push live data ?

  On the request object, I've used setNoDelay(true) to avoid the use of Nagle's algorithm. Following is a snippet of how data is written when the spawned ffmpeg process emits data.

  var clients = []; //List of client connections currently being served
  
  ffmpeg.stdout.on('data', function(data) {
  
      for(var i = 0; i < clients.length; i++){
  
          clients[i].res.write(data);
  
      }
  
  });

• ffmpeg - preserve time base and pts of all frames during transcode

  18 mars 2021, par jdizzle

  Context :

  


  I have an application that produces mp4s with HEVC encoding. I want to convert them to AVC for use in browser-based displaying. A very crucial part of how I want to use this is to preserve exact PTS times, as this is how we correlate the frames to other data streams not included in the video.

  


  Question :

  


  How do I make ffmpeg preserve this information across the transcode ? All the obvious flags seem to have no effect and ffmpeg just does whatever it wants.

  


  $ ffprobe -show_streams original.mp4 2>/dev/null | grep time_base
codec_time_base=16666667/500000000
time_base=1/1000


  


  Here is my convert command :

  


  $ ffmpeg -i original.mp4 -copyts -copytb 0 test.mp4


  


  And its result :

  


  $ ffprobe -show_streams test.mp4 2>/dev/null | grep time_base
codec_time_base=1/60
time_base=1/15360


  


  I would expect the time_bases to match. The PTS of the frames also don't match when doing ffprobe -show_frames

  


  EDIT :
@Gyan suggested using -video_track_timescale, but that didn't get the exact behavior I was looking for :

  


  $ sdiff <(ffprobe -show_frames test.mp4  | grep pkt_pts_time) <(ffprobe -show_frames original.mp4 | grep pkt_pts_time)
pkt_pts_time=0.000000                           pkt_pts_time=0.000000
pkt_pts_time=0.033000                           pkt_pts_time=0.033000
pkt_pts_time=0.067000                         | pkt_pts_time=0.066000
pkt_pts_time=0.100000                           pkt_pts_time=0.100000
pkt_pts_time=0.133000                           pkt_pts_time=0.133000
pkt_pts_time=0.167000                         | pkt_pts_time=0.166000
pkt_pts_time=0.200000                           pkt_pts_time=0.200000
pkt_pts_time=0.233000                           pkt_pts_time=0.233000
pkt_pts_time=0.267000                         | pkt_pts_time=0.266000
pkt_pts_time=0.300000                           pkt_pts_time=0.300000
pkt_pts_time=0.333000                           pkt_pts_time=0.333000
pkt_pts_time=0.367000                         | pkt_pts_time=0.366000
pkt_pts_time=0.400000                           pkt_pts_time=0.400000
pkt_pts_time=0.433000                           pkt_pts_time=0.433000
pkt_pts_time=0.467000                           pkt_pts_time=0.467000
pkt_pts_time=0.500000                           pkt_pts_time=0.500000
pkt_pts_time=0.533000                         | pkt_pts_time=0.532000
pkt_pts_time=0.567000                         | pkt_pts_time=0.565000
pkt_pts_time=0.600000                         | pkt_pts_time=0.598000
pkt_pts_time=0.633000                         | pkt_pts_time=0.631000
pkt_pts_time=0.667000                         | pkt_pts_time=0.665000
pkt_pts_time=0.700000                         | pkt_pts_time=0.698000
...