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• Adding text to movie using ffmpeg

  23 août 2014, par microspace

  I use git to track *.ass subtitle files.
  Here is example of *.ass file :

  [Script Info]
  
  ; Script generated by Aegisub 3.1.2
  
  ; http://www.aegisub.org/
  
  Title: Default Aegisub file
  
  ScriptType: v4.00+
  
  
  
  [V4+ Styles]
  
  Format: Name, Fontname, Fontsize, PrimaryColour, SecondaryColour, OutlineColour,    BackColour, Bold, Italic, Underline, StrikeOut, ScaleX, ScaleY, Spacing, Angle, BorderStyle, Outline, Shadow, Alignment, MarginL, MarginR, MarginV, Encoding
  
  Style: Default,Arial,20,&H00FFFFFF,&H000000FF,&H00000000,&H00000000,0,0,0,0,100,100,0,0,1,2,2,2,10,10,10,1
  
  Style: titr,DejaVu     
  
  
  
  Sans,20,&H007DDBFA,&H000000FF,&H00000000,&HFF000000,0,0,0,0,100,100,0,0,1,2,2,1,10,10,10,1
  
  
  
  [Events]
  
  Format: Layer, Start, End, Style, Name, MarginL, MarginR, MarginV, Effect, Text
  
  Dialogue: 0,0:00:00.46,0:00:11.22,Default,,0,0,0,,Если это можно было бы
  
  Dialogue: 0,0:00:03.44,0:00:08.96,titr,,0,0,0,,{\pos(20,240)\fad(600,600)}бывший министр

  After commit I burn subtitles into video :

  ffmpeg -i video.avi -vf "ass=subtitle.ass" out.avi

  My goal is to show commit date for 10 second at the start of movie. This should be done automatically.

  1) It can be easily done with by modifying subtitle.ass itself, but I can’t do it after commit and there are other reasons.

  2) It can be done by ffmpeg from command line : How to use ffmpeg to add a text to avi video ?

  Problem is that in this case text will be shown for the whole lenght of movie.

  3) I can copy *.ass file to temporary directory, insert date, render and delete *.ass file.

  Is there a simpler way ?

• ffmpeg x264 codec increses the Video Bitrate

  13 août 2014, par Damen Salvatore

  I have Installed ffmpeg and x264 based on this post :

  Unknown encoder ’libx264’

  no errors receved. but when i want to convert an input stream from mumudvb to a lower bitrate with x264 codec the ffmpeg output will increase the bitrate to about 26000 ! im trying to get 512k Bitrate,
  the wierd thing is that my input’s bitrate is 3000k but when i get the ffmpeg’s output in vlc it shows inputs bitrate as 30000

  this is my config :

  Port 8888
  
  BindAddress 0.0.0.0
  
  MaxHTTPConnections 20000
  
  MaxClients 10000
  
  MaxBandwidth 1000000
  
  CustomLog -
  
  <feed>
  
  File /tmp/ch1.ffm
  
  FileMaxSize 100M
  
  ACL allow 127.0.0.1
  
  launch ffmpeg -i http://127.0.0.1:8081
  
  </feed>
  
  
  
  <stream ch1="ch1">
  
  Feed ch1.ffm
  
  Format mpegts
  
  AudioBitRate 64
  
  AudioChannels 2
  
  AudioSampleRate 44100
  
  #AVOptionAudio flags +global_header
  
  VideoBitRate 512
  
  VideoBufferSize 400
  
  VideoFrameRate 25
  
  VideoBitRateTolerance 100
  
  VideoSize 720x576
  
  VideoGopSize 12
  
  AudioCodec aac
  
  VideoCodec libx264
  
  AVOptionVideo bsf h264_mp4toannexb
  
  #AVOptionVideo threads 0
  
  AVOptionVideo threads_type frame
  
  AVOptionVideo coder 0
  
  AVOptionVideo bf 0
  
  AVOptionVideo flags +loop
  
  AVOptionVideo partitions +parti8x8+parti4x4+partp8x8+partb8x8
  
  AVOptionVideo me_method hex
  
  AVOptionVideo subq 7
  
  AVOptionVideo me_range 16
  
  AVOptionVideo g 250
  
  AVOptionVideo keyint_min 10
  
  AVOptionVideo sc_threshold 40
  
  AVOptionVideo i_qfactor 0.71
  
  AVOptionVideo b_strategy 1
  
  AVOptionVideo qcomp 0.6
  
  AVOptionVideo qmin 10
  
  AVOptionVideo qmax 51
  
  AVOptionVideo qdiff 4
  
  AVOptionVideo refs 3
  
  AVOptionVideo directpred 1
  
  AVOptionVideo trellis 1
  
  AVOptionVideo wpredp 0
  
  #AVOptionVideo flags +global_header
  
  
  
  StartSendOnKey
  
  </stream>

• wrong play audio samples

  8 août 2014, par Ivan Lisovich

  I have a problem with a ffmpeg and NAudio libs.
  I worked with the old ffmpeg library and there the audio plays correctly.
  read video in manage c++

  // Read frames and save to list audio frames
  
  while(av_read_frame(pFormatCtx, &packet) >= 0) 
  
  {
  
      if(packet.stream_index == videoStream) 
  
      {
  
          // reade image
  
      }
  
      else if(packet.stream_index == audioStream) 
  
      {
  
          int b = av_dup_packet(&packet);
  
          if(b >= 0) {
  
              int audio_pkt_size = packet.size;
  
              libffmpeg::uint8_t* audio_pkt_data = packet.data;
  
              while(audio_pkt_size > 0) 
  
              {
  
                  int got_frame = 0;
  
                  int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
  
                  if(len1 < 0)
  
                  {
  
                      /* if error, skip frame */
  
                      audio_pkt_size = 0;
  
                      break;
  
                  }
  
                  audio_pkt_data += len1;
  
                  audio_pkt_size -= len1;
  
                  if (got_frame)
  
                  {
  
                      int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
  
                      array<byte>^ managedBuf = gcnew array<byte>(data_size);
  
                      System::IntPtr iptr = System::IntPtr( frame.data[0] );
  
                      System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
  
                      audioData->Add(managedBuf);
  
                  }
  
              }
  
          }
  
      }
  
      // Free the packet that was allocated by av_read_frame
  
      libffmpeg::av_free_packet(&amp;packet);
  
  }
  
  </byte></byte>

  I return audioData to c# code and play in NAudio library
  play in c#

  var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
  
  var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
  
  var waveOut = new DirectSoundOut();
  
  waveOut.Init(waveProvider);
  
  waveOut.Play();
  
  foreach (byte[] data in audioData)
  
  {
  
      waveProvider.AddSamples(data, 0, data.Length);
  
  }

  but audio not playing.
  what am I doing wrong ?