Recherche avancée

Sur d’autres sites (8425)

• C# .net mvc Process return encoding progress (FFMpeg) to client initiated with ajax

  19 décembre 2014, par user1585895

  I am using FFMpeg to encode wav to mp3s, and on completion download the final .zip to the client.

  All works great, and I can Debug.WriteLine the progress of the encoding, but I would like to return _percentage back to the client to do some UI updates based on the value of _percentage and if all encoding is done, Im not sure how to approach this.

  My thinking is as follows :

  • ajax post to CreateAndDownloadAlbum(List trackIds, int productId)
  • loop through List and create a new Process, run
    — parse StdError to get encoding percentage in myProcess_ErrorData(object source, DataReceivedEventArgs e)
    — send _percentage value at timed intervals back to CreateAndDownloadAlbum
    — update UI based on value of _percentage, when all is complete break and call public ActionResult SendFileDownload()

  Any input would be great.

  Thanks !

  front end AJAX call to post list of file Ids to download

  var data = $('#downloadAlbum').serialize();
  
  $.ajax({
  
      url: "/Admin/CreateAndDownloadAlbum",
  
      method: "POST",
  
      data: data,
  
      dataType: 'json',
  
      success: function(result) {
  
           // wait here for _percentage
  
           // if not 100, call again to get new _percentage value
  
           // if 100, update UI, move to next file being encoded
  
           // if all encoding is complete, call SendFileDownload() in controller
  
      }
  
  });

  then the controller code

  [HttpPost]
  
      public void CreateAndDownloadAlbum(List<trackstodownload> trackIds, int productId)
  
      {
  
          _numTracksToDownload = trackIds.Count;
  
          var product = _productRepository.GetProductById(productId);
  
          var artist = _artistRepository.GetArtistById(product.ArtistId);
  
  
  
          var folderGuid = Guid.NewGuid();
  
          _zipFolder = string.Concat(artist.ArtistName.ToUpper().Replace(" ", "_"), "[", product.ProductName.ToUpper().Replace(" ", "_"), "].zip");
  
          _mp3FolderPath = Server.MapPath("/Files/" + productId + "/" + folderGuid);
  
          _zipDownloadPath = Server.MapPath("/Delivery/" + _zipFolder);
  
  
  
          if (!Directory.Exists(Server.MapPath("/Files/" + productId + "/" + folderGuid)))
  
          {
  
              Directory.CreateDirectory(Server.MapPath("/Files/" + productId + "/" + folderGuid));
  
          }
  
  
  
          foreach (var z in trackIds)
  
          {
  
              var track = _trackRepository.GetTrackById(z.TrackId);
  
              var process = new Process();
  
              var startInfo = new ProcessStartInfo
  
              {
  
                  WindowStyle = System.Diagnostics.ProcessWindowStyle.Hidden,
  
                  FileName = "F:\\Development\\ffmpeg\\ffmpeg.exe",
  
                  RedirectStandardOutput = true,
  
                  RedirectStandardError = true,
  
                  UseShellExecute = false
  
              };
  
              var startFile = Path.Combine("F:\\Development","Files", track.ProductId.ToString(), track.TrackFileName);
  
              var endFile = Path.Combine("F:\\Development", "Files", track.ProductId.ToString(), folderGuid.ToString(), track.TrackFileName.Replace(".wav", ".mp3"));
  
  
  
              startInfo.Arguments = "-i " + startFile + " -b:a 320k " + endFile + " -stats";
  
              process.StartInfo = startInfo;
  
              process.EnableRaisingEvents = true;
  
              process.Exited += myProcess_exited;
  
              process.OutputDataReceived += myProcess_OutputData;
  
              process.ErrorDataReceived += myProcess_ErrorData;
  
  
  
              try
  
              {
  
                  process.Start();
  
                  process.BeginOutputReadLine();
  
                  process.BeginErrorReadLine();
  
  
  
                  if (process.HasExited == false)
  
                  {
  
                      process.WaitForExit();
  
                  }
  
              }
  
              catch (Exception)
  
              {
  
  
  
                  throw;
  
              }
  
          }
  
  
  
          SendFileDownload();
  
  
  
      }
  
  
  
      public ActionResult SendFileDownload()
  
      {
  
          // zip and move to delivery folder
  
          using (var zip = new ZipFile())
  
          {
  
              zip.AddDirectory(_mp3FolderPath, _zipFolder);
  
              zip.Save(_zipDownloadPath);
  
          }
  
  
  
          // delete items in temp guid folder
  
          var downloadedMessageInfo = new DirectoryInfo(_mp3FolderPath);
  
          foreach (var f in downloadedMessageInfo.GetFiles())
  
          {
  
              f.Delete();
  
          }
  
  
  
          // delete temp folder
  
          Directory.Delete(_mp3FolderPath);
  
  
  
          // download file
  
          string file = _zipDownloadPath;
  
          return File(file, "application/force-download", Path.GetFileName(file));
  
  
  
      }
  
  
  
      public void myProcess_exited(Object source, EventArgs e)
  
      {
  
          Debug.WriteLine("myProcess_exited ===================================");
  
          _duration = new TimeSpan(0, 0, 0);
  
          _frameTime = new TimeSpan(0, 0, 0);
  
          _percentage = 0;
  
          _numEncodedTracks++; // using this to tell me if all tracks have been encoded
  
      }
  
  
  
      public void myProcess_OutputData(object source, DataReceivedEventArgs e)
  
      {
  
      }
  
  
  
      public void myProcess_ErrorData(object source, DataReceivedEventArgs e)
  
      {
  
          string strMessage = e.Data;
  
  
  
          if (!string.IsNullOrEmpty(strMessage) &amp;&amp; (strMessage.Contains("Duration: ") || strMessage.Contains("size=")))
  
          {
  
              if (_duration != null)
  
              {
  
                  if (strMessage.Contains("Duration: "))
  
                  {
  
                      _strDuration = strMessage.Substring(strMessage.IndexOf("Duration: ") + ("Duration: ").Length,
  
                          ("00:00:00.00").Length);
  
  
  
                      char[] d_delHrMn = new char[] { ':' };
  
                      string[] d_tempHrMn = _strDuration.Split(d_delHrMn, StringSplitOptions.RemoveEmptyEntries);
  
  
  
                      char[] d_delSec = new char[] { '.' };
  
                      string[] d_tempSec = d_tempHrMn[2].Split(d_delSec, StringSplitOptions.RemoveEmptyEntries);
  
  
  
                      var d_hour = d_tempHrMn[0] == "0" ? 0 : Convert.ToInt32(d_tempHrMn[0]);
  
                      var d_min = d_tempHrMn[1] == "0" ? 0 : Convert.ToInt32(d_tempHrMn[1]);
  
                      var d_sec = d_tempSec[0] == "0" ? 0 : Convert.ToInt32(d_tempSec[0]);
  
  
  
                      _duration = new TimeSpan(d_hour, d_min, d_sec);
  
                  }
  
              }
  
  
  
              if (strMessage.Contains("size="))
  
              {
  
                  _strFrameTime = strMessage.Substring(strMessage.IndexOf("time=") + ("time=").Length,
  
                      ("00:00:00.00").Length);
  
  
  
                  char[] f_delHrMn = new char[] { ':' };
  
                  string[] f_tempHrMn = _strFrameTime.Split(f_delHrMn, StringSplitOptions.RemoveEmptyEntries);
  
  
  
                  char[] f_delSec = new char[] { '.' };
  
                  string[] f_tempSec = f_tempHrMn[2].Split(f_delSec, StringSplitOptions.RemoveEmptyEntries);
  
  
  
                  var f_hour = f_tempHrMn[0] == "0" ? 0 : Convert.ToInt32(f_tempHrMn[0]);
  
                  var f_min = f_tempHrMn[1] == "0" ? 0 : Convert.ToInt32(f_tempHrMn[1]);
  
                  var f_sec = f_tempSec[0] == "0" ? 0 : Convert.ToInt32(f_tempSec[0]);
  
  
  
                  _frameTime = new TimeSpan(f_hour, f_min, f_sec);
  
  
  
              }
  
  
  
              if (_strDuration != "00:00:00.00" &amp;&amp; _strFrameTime != "00:00:00.00" &amp;&amp; _percentage &lt; 100)
  
              {
  
                  _percentage = _frameTime.TotalMilliseconds / _duration.TotalMilliseconds * 100;
  
                  Debug.WriteLine(_percentage + " || " + _frameTime + " " + _duration);
  
              }
  
          }
  
      }
  
  </trackstodownload>

• Using Gstreamer or ffmpeg to create rtsp client on Android

  9 décembre 2014, par Pankaj Bansal

  I want to stream a rtsp stream on android and I finally have come to
  conclusion that I can’t use android API’s MediaPlayer,Videoview etc because
  latency is big issue for me. I need an latency of <500 ms. Now I am
  planning to use Gstreamer or ffmpeg to create an android rtsp client. I just have few
  doubts

  1. Will the Gstreamer or ffmpeg client be able to provide latency <500ms. I read there are
     some parameters which I can tweak to get very low latency. Just want to
     confirm. I have very good network bandwidth. The frame size is generally
     1920X1080.

  2. I read Gstreamer is one made one level above ffmpeg and uses ffmpeg
     codecs to work. I want to know which one is easier to work with for creating an android client. Working on Gstreamer or workig directly on ffmpeg.

  3. If I use Gstreamer android client, Will I have to use the Gstreamer server as well to stream the data ? Currently I am using Live555 RTSP server to stream data

• ffmpeg can't stream to remote client

  4 septembre 2014, par KFL

  I’m building a simple ffmpeg command line on my laptop to stream from its camera. The command line reads (verbose output at the botton) :

  host1> ffmpeg -v verbose \
  
                -f dshow \
  
                -i video="Camera":audio="Microphone" \
  
                -r 30 -g 0 -vcodec h264 -acodec libmp3lame \
  
                -tune zerolatency \
  
                -preset ultrafast \
  
                -f mpegts udp://12.34.56.78:12345

  Firstly, it works locally. I.e., I can view the output by using ffplay on the same host :

  host1> ffplay -hide_banner -v udp://12.34.56.78:12345

  Now what is NOT working is when I do this from another machine in the same network. It shows a nan progress :

  host2> ffplay -hide_banner -v udp://12.34.56.78:12345
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   

  I used ncat to dump the raw content. But there’s no output :

  host2>\ncat\ncat -v -u 12.34.56.78 12345
  
  Ncat: Version 5.59BETA1 ( http://nmap.org/ncat )
  
  Ncat: Connected to 12.34.56.78:12345.
  
  (...and nothing happen...)

  Note that I can exclude firewall issues as I used ncat to communicate with each other across the wire using the same port and protocol (UDP). This works and they can chat to each other :

  host1> ncat -l -u -p 12345
  
  host2> ncat -u 12.34.56.78 12345

  Any hint ?

  I’m using Windows x64 with FFMPEG 64bit installed from here. Below is the Output of my ffmpeg command :

  C:\ffmpeg\bin>ffmpeg -v verbose -f dshow -i video="Integrated Camera":audio="Microphone (Realtek High Definition Audio)" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://12.34.56.78:12345
  
  ffmpeg version N-66012-g97b8809 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Sep  1 2014 00:21:15 with gcc 4.8.3 (GCC)
  
    configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --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-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  3.100 / 56.  3.100
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  0.103 /  5.  0.103
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Guessed Channel Layout for  Input Stream #0.1 : stereo
  
  Input #0, dshow, from 'video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)':
  
    Duration: N/A, start: 171840.657000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo, bgr24, 640x480, 30 fps, 30 tbr, 10000k tbn, 30 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Matched encoder 'libx264' for codec 'h264'.
  
  [graph 0 input from stream 0:0 @ 0000000000470aa0] w:640 h:480 pixfmt:bgr24 tb:1/10000000 fr:10000000/333333 sar:0/1 sws_param:flags=2
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:iw h:ih flags:'0x4' interl:0
  
  [format @ 0000000004325a00] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:640 h:480 fmt:bgr24 sar:0/1 -> w:640 h:480 fmt:yuv444p sar:0/1 flags:0x4
  
  No pixel format specified, yuv444p for H.264 encoding chosen.
  
  Use -pix_fmt yuv420p for compatibility with outdated media players.
  
  [graph 1 input from stream 0:1 @ 0000000000460c20] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:1 @ 00000000004601a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
  
  [auto-inserted resampler 0 @ 00000000004604a0] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
  
  [libx264 @ 000000000081bb20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 000000000081bb20] profile High 4:4:4 Intra, level 3.0, 4:4:4 8-bit
  
  [mpegts @ 000000000081abe0] muxrate VBR, pcr every 3 pkts, sdt every 200, pat/pmt every 40 pkts
  
  Output #0, mpegts, to 'udp://12.34.56.78:12345':
  
    Metadata:
  
      encoder         : Lavf56.3.100
  
      Stream #0:0: Video: h264 (libx264), yuv444p, 640x480, q=-1--1, 30 fps, 90k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc56.1.100 libx264
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
  
      Metadata:
  
        encoder         : Lavc56.1.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  
    Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  *** 1 dup!
  
  frame=  241 fps= 31 q=28.0 Lsize=    3439kB time=00:00:08.03 bitrate=3506.4kbits/s dup=1 drop=0
  
  video:3035kB audio:125kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.791966%
  
  Input file #0 (video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)):
  
    Input stream #0:0 (video): 240 packets read (221184000 bytes); 240 frames decoded;
  
    Input stream #0:1 (audio): 16 packets read (1411200 bytes); 16 frames decoded (352800 samples);
  
    Total: 256 packets (222595200 bytes) demuxed
  
  Output file #0 (udp://12.34.56.78:12345):
  
    Output stream #0:0 (video): 241 frames encoded; 241 packets muxed (3108187 bytes);
  
    Output stream #0:1 (audio): 306 frames encoded (352512 samples); 307 packets muxed (128313 bytes);
  
    Total: 548 packets (3236500 bytes) muxed
  
  [libx264 @ 000000000081bb20] frame I:241   Avg QP:27.97  size: 12897
  
  [libx264 @ 000000000081bb20] mb I  I16..4: 100.0%  0.0%  0.0%
  
  [libx264 @ 000000000081bb20] coded y,u,v intra: 26.3% 0.5% 0.0%
  
  [libx264 @ 000000000081bb20] i16 v,h,dc,p: 19% 28% 21% 31%
  
  [libx264 @ 000000000081bb20] kb/s:3095.29
  
  [dshow @ 0000000000467720] real-time buffer[Integrated Camera] too full (90% of size: 3041280)! frame dropped!
  
  Received signal 2: terminating. (I pressed CTRL-C)