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• New FATE Test Coverage System

  10 août 2010, par Multimedia Mike — FATE Server

  I’ve been feeling a bit scattered for the last week since I was fired from my volunteer position as the FFmpeg QA manager, wondering if there is anything I should attempt to do with the project. It can’t be denied that the new system is working well. But one area I’ve wondered about is test coverage.

  Under my old regime I tracked test coverage as a wiki page which was a highly flawed method— tedious and error-prone. There are those 2 adjectives again— tedious and error-prone ; whenever I see those, I search for automation methods. I think that might be more plausible thanks to the new FATE’s tighter integration with the FFmpeg build system.

  I don’t think anyone is working on this problem so I wanted to toss out a brainstorm :

  1. First, run ’ffmpeg -formats’, ’ffmpeg -codecs’, etc. and parse the output to collect a list of all the features (full list : -formats, -codecs, -bsfs, -protocols, -filters, -pix_fmts). Transform these lists into a standardized list of features, e.g., "DEVSD  ffvhuff         Huffyuv FFmpeg variant" represents features ’decode-video-ffvhuff’, ’encode-video-ffvhuff’, ’ffvhuff-horizband’, and ’ffvhuff-dr1’.
  2. Next, tag each individual test spec with the features that it exercises. E.g., test ’fate-vqa-cc’ exercises features ’demux-wsvqa’, ’decode-video-vqavideo’, and ’decode-audio-adpcm_ima_ws’.
  3. Finally, compare the data from parts 1 and 2. Print a list of all the features that are not exercised in FATE.

  I think a lot of this could be implemented at the GNU make level. Then again, I’m no expert on GNU make syntax so I may be overestimating its capabilities. Or there might be simpler ways to automatically track test coverage stats based on the improved testing infrastructure.

• Gracefully closing FFMPEG child processes from node

  10 juin 2019, par Gordon

  I am trying to record numerous audio feeds using ffmpeg. Node downloads a config file with the stream URLS (HLS M3U8 playlists), parses it and starts the appropriate number of ffmpeg instances. When I go to shut them down, nothing happens and I have to kill them with task manager, resulting in corrupt files. When I am debugging and hit control-c within a minute or two of starting the program, it works without issue. When I need to record more than 5-10 minutes that I have problems.

  I found this related question from 2013, and adapted it to fit my multiple stream situation.

  The recorder processes are started with the following code (inside the http request callback) :

  config.audio_config.forEach((channel,i)=>{
  
  self.liveChannels++;
  
  
  
  console.log(`   ${channel.number}`);
  
  self.Channels[i] = spawn('ffmpeg', ['-i', `${channel.base_url + channel.stream_ios}`,  '-c:v', 'none', '-c:a', 'copy', `Output\\${config.folder}\\${channel.number}.m4a`]);
  
  self.Channels[i].stdin.setEncoding('utf8');
  
  self.Channels[i].chNum = channel.number;
  
  
  
  self.Channels[i].on('exit',(code,sig)=>{
  
  console.log(`   Channel ${channel.number} recorder closed.`);
  
  self.liveChannels--;
  
  if(self.liveChannels === 0){
  
      process.exit();
  
  }
  
  });
  
  });
  
  console.log('Press Ctl-C to start Shutdown');

  My shutdown function (triggered by SIGINT to main process) is :

  function shutdown() {
  
      self.Channels.forEach((chan)=>{
  
          chan.stdin.write('q');
  
          chan.stdin.end(); //I have tried both with and without this line and chan.stdin.end('q')
  
      });
  
  
  
  }

  UPDATE :
  Switching to an AAC container file (simply changed the extension on the output) removed the need for a graceful FFMPEG exit. I still would like to know why sending ’q’ to stdin only kills the FFMPEG process for the first 10 minutes.

• How would I add an audio channel to an rtsp stream ?

  9 septembre 2022, par PlayerWet

  Good companions. It turns out that my Raspberry does not give more than itself when it comes to joining video with audio at good quality and sending it by rtsp. But I think I could send the video in rtsp format and then the audio in mp3, but I would need to join it again on another computer (nas Debian) on my home lan where I have the Shinobi program (security camera manager) installed.

  


  I would need something that can somehow grab the rtsp stream and another mp3 audio and merge them into a new rtsp stream. Is this possible ? or is it a crazy idea.

  


  On the one hand I send this, which is the transmission of the camera by rtsp through v4l2rtspserver :

  


  v4l2rtspserver -H 1080 -W 1920 -F 25 -P 8555 /dev/video0


  


  And separately I send an audio in mp3 with sound from a usb microphone through ffmpeg :

  


  ffmpeg -ac 1 -f alsa -i hw:1,0 -acodec libmp3lame -ab 32k -ac 1 -f rtp rtp://192.168.1.77:12348


  


  My idea is to put both things together on a nas server in a new rtsp stream (or another idea).

  


  But I don't know if with ffmpeg I can capture the video from an rtsp video stream and then be able to join it with the mp3 audio and form another new rtsp stream.

  


  Merge these two streams into one and reassemble an rtsp :

  


  rtsp://192.168.1.57:8555/unicast

rtp://192.168.1.77:12348


  


  I have tried this way but it gives me an error :

  


  ffmpeg \
    -i rtsp://192.168.1.57:8555/unicast \
    -i rtp://192.168.1.37:12348 \
    -acodec copy -vcodec libx264 \
    -f rtp_mpegts "rtp://192.168.1.77:5000?ttl=2"


  


  Error :

  


  [h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
    Last message repeated 1 times
[h264 @ 0x55ac6acaf4c0] decode_slice_header error
[h264 @ 0x55ac6acaf4c0] no frame!
[h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
    Last message repeated 1 times
[h264 @ 0x55ac6acaf4c0] decode_slice_header error
[h264 @ 0x55ac6acaf4c0] no frame!
[h264 @ 0x55ac6acaf4c0] non-existing PPS 0 referenced
    Last message repeated 1 times


  


  What am I doing wrong ?