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• FireFox Video throws warning but video actually works ?

  21 février 2021, par SdBill
  

  • OS : Ubuntu 18.04

  


  • FF : 85.0.1

  


  • Error/warning : Cannot play media. No decoders for requested formats : video/mp4, video/mp4

  


  


  Same error for video/ogg

  


  Here are my questions : 1) The video still plays fine once loaded, and there is no error in Chrome or Chromium. Why does the video work fine after loading but throws the error on load ? 2) Is there anything that can be done without re-encoding over 2 gigs of video ?

  


  Context : this is an old no-profit site that used Flash for video and we really don't want to throw a lot of time at, but there are gigs of videos. I converted all .flv files and .mpg files to .mp4 using the most simple of ffmpeg commands, examples :

  


  ffmpeg -i video-source.flv video-source.mp4
ffmpeg -i video-source.mpg video-source.mp4
ffmpeg -i video-source.mpg video-source.ogg


  


  As I watched the ffmpeg output, it looked to me like the codec was H264 (at least, I think that is what I am seeing, not a video expert.)

  


  Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> theora (libtheora))
  Stream #0:1 -> #0:1 (aac (native) -> vorbis (libvorbis))


  


  Server response in a direct request to the mp4 files is

  


  Content-Type
    video/mp4


  


  I have seen the documentation and posts on fragmented mp4 and if re-encoding is the only option, we're probably going to abandon as it does play once loaded.

  


  Code is simplistic, using an html5 doctype :

  


    <video width="320" height="240" controls="controls">&#xA;   <source src="/images/video/mp4/video-source.mp4" type="video/mp4">&#xA;   <source src="/images/video/ogg/video-source.ogg" type="video/ogg">&#xA;   Your browser does not support the video tag.&#xA;  </source></source></video>&#xA;

  


  Is the only option here to pander to FireFox and re-encode everything since as mentioned, it plays fine once loaded and throws no error in Chrome ?

  


• MediaPlayer miscalculates audio duration. getCurrentPosition() goes back when audio reaches the end

  26 mars 2017, par Kacy

  I’m streaming .ogg files from my server. The MediaPlayer plays the song to completion, but when it reaches the end of the song, getCurrentPosition() returns a time less than the actual duration of the song. For 1 song it’s only off by a second, but for another song it’s off by 13 seconds.

  I have a SeekBar that’s updated every second by calling getCurrentPosition(), and once the bar reaches the end, it actually jumps back a few seconds. And I can no longer use seekTo() into the segment of the song that’s past the miscalculated duration because I receive the error : Attempt to seek to past end of file.

  Chrome’s html video element simply displays the songs with the incorrect durations.

  Other tools like ffmpeg or this app on Windows 10 called PowerMediaPlayer both calculate the songs’ durations perfectly.

  I believe this problem is related to this answer, but the solution assumes ffmpeg also miscalculates the duration, which is not the case here. Using the -vn flag when converting audio files to ogg didn’t change anything.

  What are my options ? Is there a way to get the MediaPlayer to calculate the correct duration ?

  Update :

  Converting to mp3 has no issues (although I used audioBitrate('96k')instead of audioQuality(2), but I require a solution to get ogg files working. Below is how I’m using ffmpeg to perform the conversion. Another thing worth noting is that when I used 'vorbis' instead of 'libvorbis' for the codec the durations were off by less than a second, if at all. However, I must use 'libvorbis' because 'vorbis' is experimental and results in significanlty inconsistent bitrates across different files given the same audio quality parameter. (And it ignores audioBitrate() entirely.)

  ffmpeg( filepath )
  
      .toFormat( 'ogg' )
  
      .audioCodec( 'libvorbis' )
  
      .audioQuality( 2 )
  
      .output( destination )
  
      .run();

• Can one "fix" a video with non monotonic DTS ?

  30 novembre 2020, par amn

  I've got a number of WebM containers (input.webm), each with a VP9 video track and an Opus audio track. These containers (files) were generated with an older version of Google Chrome using the MediaRecorder Web API, by a live media recording Web application.

  


  As part of an attempt to make the material available to a certain overly stubborn Windows DirectShow application that is otherwise refusing to import WebM files — despite the fact that we've got DirectShow filters for demuxing WebM and decoding VP9 and Opous installed and verified to be working on the application host — I have tried to introduce a re-muxing batch job where the media is re-muxed into an MPEG-4 container, stuffing the same VP9 and Opus tracks in the MP4 file, for intended consumption by said Windows application.

  


  However, on at least one of these WebM files, FFmpeg refuses to proceed with generating the output, using ffmpeg -i input.webm -c copy output.mp4 :

  


  [mp4 @ 00000208bee17840] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 53744 >= 53744
av_interleaved_write_frame(): Invalid argument


  


  Given how some of the material in the WebM collection are quite lengthy (on the order of hours), I don't want to transcode video (or audio, for that matter) tracks, instead opting to try and stuff them into an MPEG-4 container. As far as I understand, VP9 in MPEG-4 is rather well supported, too — certainly FFmpeg has been able to re-mux most of the WebM files we've got into MPEG-4 containers which the aforementioned application has been able to import, too.

  


  Is there anything I can do to "fix" [WebM] files where DTS is out of order ? Is this some error during original encoding of the media file in question ? Google Chrome does use FFmpeg for encoding media, by the way, at least I know earlier versions did.