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• HTTP Live Streaming Mac app

  4 mars 2013, par Diederik Hoogenboom

  I am developing a Mac app which needs to provide a HTTP Live Stream (just the last 2 seconds or so) for the main screen (Desktop).

  I was thinking of the following process :

  1. Create a AVCaptureSession with a AVCaptureScreenInput as input (sessionPreset = AVCaptureSessionPresetPhoto)
  2. Add a AVCaptureVideoDataOutput output to the session
  3. Capture the frames (in kCVPixelFormatType_32BGRA format) in captureOutput:didDropSampleBuffer:fromConnection : and write these to a ffmpeg process for segmenting (using a pipe or something) that creates the MPEG-TS and playlist files.
  4. Use an embedded HTTP server to server up the segmented files and playlist file.

  Is this the best approach and is there is no way to circumvent the ffmpeg part for encoding and segmenting the video stream ?

  What is the best way to pipe the raw frames to ffmpeg ?

• using ffmpeg to create frames from live streaming rtmp url getting failed to read rtmp packet header

  30 décembre 2013, par Nikita

  I want to create video frames per second from rtmp live streaming url using ffmpeg.
  Running command in command prompt "ffmpeg -i -r 1 thumb%03d.jpg" but getting "RTMP_ReadPacket, failed to read RTMP packet header
  rtmp ://ipaddress/oflaDemo/livestream/sample : Unknown error occurred"
  using red5 & adobe live encoder for live streaming.

• Node.JS Live Streaming Audio with FFMPEG

  20 avril 2021, par nicnacnic

  I'm trying to create an Express server to live stream audio captured from another application (Discord in this case). I'm able to get a server up and running, but there are a couple issues that need to be solved. Here's my server code so far.

  


  const app = express();
app.get("/", function(req, res) {
    res.sendFile(__dirname + "/index.html");
});
app.get("/audio", function(req, res) {
    const stream = ffmpeg(audio).inputOptions(["-f", "s16le", "-ar", "48k", "-ac", "2"]).format('wav');
    res.writeHead(200, { "Content-Type": "audio/wav" });
    stream.pipe(res);
});
app.listen(8080)


  


  

  1. Silent sections of audio need to be added. When there's no activity on the input, there's no data written to the audio variable. This causes weird behavior, for example I can speak and the audio comes through a second later. Then, if I wait 10 seconds then speak again, the audio comes through 4-5 seconds later. I believe this is a problem with the way I'm using ffmpeg to transcode, but I have no idea how to fix it.

  


  2. Refreshing the client crashes the program. Every time I refresh the client I get an ffmpeg error. Error: Output stream closed. This error doesn't happen if I close it, only on reload.

  


  3. The audio is not synced between clients. Every time I open a new connection, the audio starts playing from the beginning instead of being synced with each other and playing the audio live.

  


  


  This is how it's supposed to work : it captures audio from my app in PCM, converts the audio to WAV with ffmpeg, and then streams the audio live to the clients. The audio needs to be synced with all the clients as best as possible to reduce delay. And I'm using fluent-ffmpeg instead of just regular ffmpeg for the transcoding.
Thanks !