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• Ffmpeg send duration of video to client (using node-fluent-ffmpeg)

  26 mai 2013, par Vprnl

  I'm really new to the world of ffmpeg so please excuses me if this is a stupid queston.

  I'm using the module Node-fluent-ffmpeg to stream a movie and convert it from avi to webm with FFMPEG.

  So far so good (it plays the video), but I'm having trouble parsing the duration to the player. It also gives me an error even though I plays the video.

  my code is as followed :

  var stat = fs.statSync(movie);
  
  
  
  var start = 0;
  
  var end = 0;
  
  var range = req.header('Range');
  
  if (range != null) {
  
  start = parseInt(range.slice(range.indexOf('bytes=')+6,
  
    range.indexOf('-')));
  
  end = parseInt(range.slice(range.indexOf('-')+1,
  
    range.length));
  
  }
  
  if (isNaN(end) || end == 0) end = stat.size-1;
  
  if (start > end) return;
  
  
  
  var duration = (end / 1024) * 8 / 1024;
  
  
  
  res.writeHead(206, { // NOTE: a partial http response
  
      'Connection':'close',
  
      'Content-Type':'video/webm',
  
      'Content-Length':end - start,
  
      'Content-Range':'bytes '+start+'-'+end+'/'+stat.size,
  
      'Transfer-Encoding':'chunked'
  
  });
  
  
  
  var proc = new ffmpeg({ source: movie, nolog: true, priority: 1, timeout:15000})
  
      .toFormat('webm')
  
      .addOptions(['-probesize 900000', '-analyzeduration 0', '-minrate 1024k', '-maxrate 1024k', '-bufsize 1835k', '-t '+duration+' -ss'])
  
      .writeToStream(res, function(retcode, error){
  
      if (!error){
  
          console.log('file has been converted succesfully',retcode);
  
      }else{
  
          console.log('file conversion error',error);
  
      }
  
  });

  I set the header with a start and a end based on this article : http://delog.wordpress.com/2011/04/25/stream-webm-file-to-chrome-using-node-js/

  I calculate the length in seconds in the variable duration.

  The error FFmpeg is giving me is :

      file conversion error ffmpeg version N-52458-gaa96439 Copyright (c) 2000-2013 the FFmpeg developers
  
        built on Apr 24 2013 22:19:32 with gcc 4.8.0 (GCC)
  
        configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --e
  
      nable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable
  
      -libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --ena
  
      ble-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwola
  
      me --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enabl
  
      e-libxvid --enable-zlib
  
        libavutil      52. 27.101 / 52. 27.101
  
        libavcodec     55.  6.100 / 55.  6.100
  
        libavformat    55.  3.100 / 55.  3.100
  
        libavdevice    55.  0.100 / 55.  0.100
  
        libavfilter     3. 60.101 /  3. 60.101
  
        libswscale      2.  2.100 /  2.  2.100
  
        libswresample   0. 17.102 /  0. 17.102
  
        libpostproc    52.  3.100 / 52.  3.100
  
      Input #0, avi, from 'C:/temp/test.avi':
  
        Metadata:
  
          encoder         : Nandub v1.0rc2
  
        Duration: 00:01:09.78, start: 0.000000, bitrate: 1517 kb/s
  
          Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 640x352, 23.98 tbr, 23.98 tbn, 23.98 tbc
  
          Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 222 kb/s
  
      [libvpx @ 0036db20] v1.2.0
  
      Output #0, webm, to 'pipe:1':
  
        Metadata:
  
          encoder         : Lavf55.3.100
  
          Stream #0:0: Video: vp8, yuv420p, 640x352, q=-1--1, 200 kb/s, 1k tbn, 23.98 tbc
  
          Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp
  
      Stream mapping:
  
        Stream #0:0 -> #0:0 (msmpeg4 -> libvpx)
  
        Stream #0:1 -> #0:1 (mp3 -> libvorbis)

  The client side player (which is VideoJs) says the file is infinite/NaN in length.

  I feel like I'm pretty close to a solution but my inexperience with the subject matter prohibits me from getting it to work. If I'm unclear in any way please let me know. (I have a tendency of explaining things fuzzy.)

  Thanks in advance !

  [EDIT]

  I removed the duration bit because it has nothing to do with the issue. I checked the response header of the client and saw :

   Accept-Ranges:bytes
  
   Connection:keep-alive
  
   Content-Length:13232127
  
   Content-Range:bytes 0-13232127/13232128
  
   Content-Type:video/webm

  Why can't the client figure out the duration even though it receives it in the header ?

• Discord.py music bot doesn't play next song in queue

  10 mai 2019, par Lewis H

  This is my code for the bot I’m trying to create, it plays the music fine.

  ytdl_options = {
  
      'format': 'bestaudio/best',
  
      'restrictfilenames': True,
  
      'noplaylist': True,
  
      'nocheckcertificate': True,
  
      'quiet':True,
  
      'ignoreerrors': False,
  
      'logtostderr': False,
  
      'no_warnings': True,
  
      'default_search': 'auto',
  
      'source_address': '0.0.0.0' # using ipv4 since ipv6 addresses causes issues sometimes
  
      }
  
  
  
  
  
          # ffmpeg options
  
  ffmpeg_options= {
  
      'options': '-vn'
  
      }
  
  
  
  
  
  
  
  @bot.command()
  
  async def play(ctx, url:str = None): 
  
      queue = {} 
  
  
  
      channel = ctx.author.voice.channel    
  
      if ctx.voice_client is not None:  
  
          await ctx.voice_client.move_to(channel)
  
      elif ctx.author.voice and ctx.author.voice.channel:       
  
          await channel.connect()
  
  
  
      if not url:
  
          await ctx.send("Try adding a URL. e.g. !play https://youtube.com/watch?v=XXXXXXXXX")
  
  
  
      if ctx.voice_client is not None:
  
          vc = ctx.voice_client #vc = voice client, retrieving it
  
          ytdl = youtube_dl.YoutubeDL(ytdl_options)
  
  
  
  
  
          loop = asyncio.get_event_loop()
  
          data = await loop.run_in_executor(None, lambda: ytdl.extract_info(url))
  
  
  
          if 'entries' in data:
  
              data = data['entries'][0] 
  
          svr_id = ctx.guild.id
  
          if svr_id in queue:
  
              queue[svr_id].append(data)
  
  
  
          else:
  
              queue[svr_id] = [data]
  
          await ctx.send(data.get('title') + " added to queue")
  
  
  
          source = ytdl.prepare_filename(queue[svr_id][0])
  
          def pop_queue():
  
              if queue[svr_id] != []:
  
                  queue[svr_id].pop(0)
  
                  data = queue[svr_id][0]
  
              else:
  
                  vc.stop()
  
  
  
          if not vc.is_playing():
  
              vc.play(discord.FFmpegPCMAudio(source, **ffmpeg_options), after=lambda: pop_queue())

  The next song downloads and queues it fine, but once the first song finishes, it doesn’t play the next one. I can’t figure out how to make it play after the first song has commenced. I have the after= set to remove the top item of the queue, but how do I get it to play again ? Thanks

• Why doesn't the lame mp3 codec (libmp3lame) create XingHeaders ?

  5 novembre 2012, par user784637

  I downloaded a video from youtube whose audio was encoded in aac and transcoded it to an mp3 using the libmp3lame codec.

  $ ffmpeg -i video.mp4 -vn -acodec libmp3lame -- test.mp3

  Now when I try to write id3 tags to it using the taglib example tagwriter I get the following warning :

  $ ./testwriter -t 'stuff' test.mp3
  
  TagLib: MPEG::XingHeader::parse() -- Xing header doesn't contain the total stream size.

  The id3 tags are written and display correctly, but the lack of the XingHeaders causes older players to loop the song several times.

  Why doesn't the lame mp3 codec create XingHeaders ? How do I create XingHeaders ?