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• Error finding watermark path using ffmpeg in asp.net application

  27 août 2013, par irfanmcsd

  I am using .net ffmpeg wrapper to post watermark on videos. Posting watermark works fine if i execute ffmpeg command directly but failed to find suitable watermark png file location if command executed via asp.net application.

  here is sample ffmpeg command

  string RootPath = HttpContext.Current.Server.MapPath(HttpContext.Current.Request.ApplicationPath);
  
  _mhandler.FFMPEGPath = RootPath + "/ffmpeg_aug_2013/bin/ffmpeg.exe";
  
  _mhandler.InputPath = RootPath + "/contents/original";
  
  _mhandler.OutputPath = RootPath + "/contents/mp4";
  
  _mhandler.BackgroundProcessing = false;
  
  _mhandler.FileName = "wildlife.wmv";
  
  _mhandler.OutputFileName = "wildlife_ddd";
  
  string presetpath = RootPath + "/ffmpeg_aug_2013/presets/libx264-ipod640.ffpreset";
  
  _mhandler.OutputExtension = ".mp4";
  
  _mhandler.Parameters = "-s 640x380 -b:v 500k -bufsize 500k -b:a 128k -ar 44100 -c:v libx264 -vf \"movie = watermark.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:main_h-overlay_h-10 [out]\"";
  
  _mhandler.Parameters = _mhandler.Parameters + " -fpre \"" + presetpath + "\"";
  
  VideoInfo info =  _mhandler.Process();

  i tried direct code too

  string _out = "";
  
  Process _process = new Process();
  
  _process.StartInfo.UseShellExecute = false;
  
  _process.StartInfo.RedirectStandardInput = true;
  
  //_process.StartInfo.RedirectStandardOutput = true;
  
  _process.StartInfo.RedirectStandardError = true;
  
  _process.StartInfo.CreateNoWindow = true;
  
  _process.StartInfo.WindowStyle = ProcessWindowStyle.Hidden;
  
  _process.StartInfo.FileName = _ffmpegpath;
  
  _process.StartInfo.Arguments = cmd;
  
  if (_process.Start())
  
  {             
  
      _process.WaitForExit(ExitProcess);
  
      _out = _process.StandardError.ReadToEnd();
  
      if (!_process.HasExited)
  
        _process.Kill();
  
  
  
      return _out;
  
   }

  ffmpeg error output as

  FFMPEG Output:ffmpeg version N-55753-g88909be Copyright (c) 2000-2013
  the FFmpeg developers built on Aug 24 2013 21:40:51 with gcc 4.7.3
  (GCC) configuration : —enable-gpl —enable-version3
  —disable-w32threads —enable-avisynth —enable-bzlib —enable-fontconfig —enable-frei0r —enable-gnutls —enable-iconv —enable-libass —enable-libbluray —enable-libcaca —enable-libfreetype —enable-libgsm —enable-libilbc —enable-libmodplug —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-libopus —enable-librtmp —enable-libschroedinger —enable-libsoxr —enable-libspeex —enable-libtheora —enable-libtwolame —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-libvorbis —enable-libvpx —enable-libx264 —enable-libxavs —enable-libxvid —enable-zlib libavutil 52. 42.100 / 52. 42.100 libavcodec 55. 29.100 / 55. 29.100 libavformat 55. 14.102 / 55. 14.102 libavdevice 55. 3.100
  / 55. 3.100 libavfilter 3. 82.102 / 3. 82.102 libswscale 2. 5.100 / 2.
  5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 [asf @ 024c9960] Stream #0 : not enough frames to estimate rate ; consider increasing probesize Guessed Channel Layout for Input Stream

  0.0 : stereo Input #0, asf, from 'F :\own\mhp_new/contents/original\wildlife.wmv' : Metadata :

  SfOriginalFPS : 299700 WMFSDKVersion : 11.0.6001.7000 WMFSDKNeeded :
  0.0.0.0000 comment : Footage : Small World Productions, Inc ; Tourism New Zealand | Producer : Gary F. Spradling | Music : Steve Ball title :
  Wildlife in HD copyright : © 2008 Microsoft Corporation IsVBR : 0
  DeviceConformanceTemplate : AP@L3 Duration : 00:00:30.09, start :
  0.000000, bitrate : 6977 kb/s Stream #0:0(eng) : Audio : wmav2 (a1[0][0] / 0x0161), 44100 Hz, stereo, fltp, 192 kb/s Stream

  0:1(eng) : Video : vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 1280x720, 5942 kb/s, 29.97 tbr, 1k tbn, 1k tbc [image2 @ 024c76e0]

  Could find no file with path 'watermark.png' and index in the range
  0-4 [Parsed_movie_0 @ 024c0540] Failed to avformat_open_input
  'watermark.png' [AVFilterGraph @ 024ca100] Error initializing filter
  'movie' with args 'watermark.png' Error opening filters ! Error Code= 0

  Error on point ( Could find no file with path 'watermark.png' ) shows watermark.png file not found.
  I place watermark.png file in the following locations but still can't found

  i : application root

  ii : root where actual aspx page located

  iii : ffmpeg root

  iv : ffmpeg/bin/

  I also used complete path but still can't detected.

  Note : if i use same ffmpeg command in php and place watermark.png on location where actual php page exist watermark properly detected and command executed properly, but same approach not working in asp.net

  Can any one help me where should i place watermark.png file so that script can access it.

• FFSERVER - streaming an ASF video as Webm output

  30 mai 2014, par Emmanuel Brunet

  I’m trying to stream an IP webcam ASF live stream to a ffserver to output a webm video format. The server starts successfully but the ffserver commands used to feed the ffserver fails and generates a core dump.

  Input stream

  $ ffprobe http://account:password@webcam/videostream.asf
  
  
  
  Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc
  
      Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, 1 channels, s16p, 32 kb/s

  ffserver configuration

  my ffserver configuration is :

  Port 8091
  
  RTSPPort 554
  
  BindAddress 192.168.1.62
  
  MaxHTTPConnections 1000
  
  MaxClients 100
  
  MaxBandwidth 1000
  
  CustomLog -
  
  
  
  <feed>
  
          File /tmp/webcam.ffm
  
          FileMaxSize 500M
  
          ACL allow localhost
  
          ACL allow 192.168.0.0 192.168.255.255
  
  
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed webcam.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 640x480           # Video resolution
  
     VideoFrameRate 25           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
     AVOptionVideo cpu-used 0
  
     AVOptionVideo qmin 10
  
     AVOptionVideo qmax 42
  
     AVOptionVideo quality good
  
     AVOptionAudio flags +global_header
  
     PreRoll 15
  
     StartSendOnKey
  
     # VideoBitRate 32            # Video bitrate
  
  </stream>
  
  
  
  <stream>
  
          Format status
  
          # Only allow local people to get the status
  
          ACL allow localhost
  
          ACL allow 192.168.0.0 192.168.255.255
  
  </stream>

  ffmpeg feed

  I run the following command that fails

  $ ffmpeg  -i http://account:password@webcam/videostream.asf http://192.168.1.62:8091/webcam.ffm
  
  http://192.168.1.62:8091/webcam.ffm
  
  Input #0, asf, from 'http://account:password@webcam/videostream.asf':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 25 tbr, 1k tbn, 1k tbc
  
      Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s
  
  [swscaler @ 0x36a80c0] deprecated pixel format used, make sure you did set range correctly
  
  Segmentation fault

  I tryed

  $ ffmpeg  -i http://account:password@webcam/videostream.asf -pix_fmt yuv420p  http://192.168.1.62:8091/webcam.ffm

  But it raises the same error.

  Thanks for your help

  Edit

  For an easy testing (I thought), I tried to publish the whole ASF stream as is, meaning connecting the ASF webcam output stream to the ffserver that outputs ASF format too.
  And thus with mirrored encoding so I changed the ffserver configuration to

  ...
  
  <stream>
  
      Feed webcam.ffm
  
      Format asf
  
      VideoFrameRate 25
  
      VideoSize 640X480
  
      VideoBitRate 256
  
      VideoBufferSize 1000
  
      VideoGopSize 30
  
      AudioBitRate 32
  
      StartSendOnKey
  
  </stream>
  
  ...

  And the output is now :

  Input #0, asf, from 'http://admin:alpha1237@webcam/videostream.asf':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc), 640x480, 1k tbr, 1k tbn, 1k tbc
  
      Stream #0:1: Audio: adpcm_ima_wav ([17][0][0][0] / 0x0011), 8000 Hz, mono, s16p, 32 kb/s
  
  [swscaler @ 0x3d620c0] deprecated pixel format used, make sure you did set range correctly
  
  Output #0, ffm, to 'http://192.168.1.62:8091/webcam.ffm':
  
    Metadata:
  
      creation_time   : now
  
      encoder         : Lavf55.40.100
  
      Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 32 kb/s
  
      Metadata:
  
        encoder         : Lavc55.64.100 wmav2
  
      Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 640x480, q=2-31, 256 kb/s, 1k fps, 1000k tbn, 1k tbc
  
      Metadata:
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (adpcm_ima_wav -> wmav2)
  
    Stream #0:0 -> #0:1 (mjpeg -> msmpeg4)
  
  Press [q] to stop, [?] for help
  
  Segmentation fault

  I can’t even forward the stream.

• ffmpeg stream chrome kiosk mode ubuntu 16.04 server

  15 février 2021, par Raul

  I have a weird out-of-sync issue while using ffmpeg to stream to youtube live a chrome browser from an ub untu 16.04 server.

  



  Issue : output video streamed to youtube has audio/video out of sync, sometimes with as much as 3s

  



  Current flow :

  



  1) start pulseaudio - we using something like this to start it :

  



  pulseaudio --start -vvv --disallow-exit --log-target=syslog --high-priority --exit-idle-time=-1 --daemonize


  



  2) start Xvfb

  



  Xvfb :0 -ac -screen 0 1920x1080x24


  



  3) start chrome linux in kiosk mode

  



  google-chrome --kiosk --disable-gpu --incognito --no-first-run --disable-java --disable-plugins --disable-translate --disk-cache-size=$((1024 * 1024)) --disk-cache-dir=/tmp/chrome/ --user-data-dir=/tmp/chrome/ --force-device-scale-factor=1 --window-size=1920,1080 --window-position=0,0 LOCATION_URL


  



  4) start ffmpeg

  



  ffmpeg -y \
  -thread_queue_size 8192 -rtbufsize 250M -f x11grab -video_size 1920x1080 -framerate 24 -i :0 \
  -thread_queue_size 8192 -channel_layout stereo -f alsa -i pulse \
  -c:v libx264 -pix_fmt yuv420p -c:v libx264 -g 48 -crf 24 -filter:v fps=24 -preset ultrafast -tune zerolatency \
  -c:a aac -strict -2 -channel_layout stereo -ab 96k -ac 2 -flags +global_header \
  -f flv YOUTUBE_LIVE_STREAMING_RTMP


  



  Note : this is running on an amazon ec2 instance, meaning there is no soundcard, so alsa and pulseaudio are creating a dummy audio card. However, the latency does not come from there. Logs :

  



  Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Final latency 69.66 ms = 23.22 ms + 2*11.61 ms + 23.22 ms


  



  At this point, here's what we observed :

  



  


  1. if we start ffmpeg exactly after issuing the command to start chrome, we see the DTS errors from ffmpeg. Audio is out of sync with the video and has delay of 3-5seconds AHEAD. We also noticed the out of sync remains the same for the full duration of the stream

  


  2. if we start ffmpeg after around 10seconds, audio and video are almost in sync. We then manually added a -itsoffset -0.125 to the ffmpeg command and everything is perfect.

  


  



  Questions :

  



  

  1. Why would ffmpeg have so much lag if it's started right after chrome ?

  


  2. Is starting the ffmpeg after 10s or X seconds the expected behavior ? That is, is this because the system needs to wait for audio/video signals to be "ready" or something ?

  


  3. Is there a way to 100% calculate or know when Chrome is fully ready and start ffmpeg ? We found sometimes it takes 5s, sometimes 10. Depends on the URL we load.

  


  4. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything. And a restart is required to "re-balance" the audio/video inputs and get them back in sync.

  


  5. Can pulseaudio be the problem in this scenario ?

  


  



  Thank you

  



  UPDATE Dec 20

  



  We were able to do some tricks to force chrome to start the audio on page load, and that will force connect to pulseaudio. Doing that, plus adding a 3s delay for ffmpeg to start, there is no more delay when ffmpeg starts.
However, our app is a webRTC app, and we have a STRANGER thing happening : if we start the page with no webcam/audio, once the webcam/audio is enabled, ffmpeg (while showing no errors) has a delay of 2s or so. While keep talking, in about max 30s, that delay is GONE.

  



  So the new questions are :

  



  

  1. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything.

  


  2. What could cause the initial audio/video out of sync issue and then catching up ?