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• videostreaming in java with bytedeco maven dependencies

  22 octobre 2020, par faenschi

  we try to setup a videostreamig service in java. our applications runs in a docker container on openshift. therefore we install ffmpeg in the dockerfile.

  


  in java we use the bytedeco maven dependencies. but i'm really not sure which dependency version runs with which ffmpeg version installed in the container. is there somewhere a overview ? does anybody have a similar setup ?

  


  at the moment we use the 1.5.4 maven dependencies and ffmpgen v4.3.1 in the container :

  


   <dependency>&#xA;        <groupid>org.bytedeco</groupid>&#xA;        <artifactid>javacv-platform</artifactid>&#xA;        <version>1.5.4</version>&#xA;    </dependency>&#xA;&#xA;    <dependency>&#xA;        <groupid>org.bytedeco</groupid>&#xA;        <artifactid>javacpp-platform</artifactid>&#xA;        <version>1.5.4</version>&#xA;    </dependency>&#xA;&#xA;   &#xA;    <dependency>&#xA;        <groupid>org.bytedeco</groupid>&#xA;        <artifactid>ffmpeg-platform</artifactid>&#xA;        <version>4.3.1-1.5.4</version>&#xA;    </dependency>&#xA;

  


  this does not work. the pod crashes as soon as i start the streaming. if i use ffmpeg v 3.x with a older centos base image and the dependency above the pod does not crash..but the streaming does not work either.

  


  any help appreciated
angela

  


• FFMPEG audio not lining up

  31 août 2020, par Chris Kooken

  I am using OpenTok to build a live video platform. It generates webm files from each users stream.

  


  I am using FFMPEG to convert webm (WEBRTC) videos to MP4s to edit in my NLE. The problem I am having is my audio is drifting. I THINK it is because the user pauses the audio during the stream. This is the command i'm running

  


  ffmpeg -acodec libopus -i 65520df3-1033-480e-adde-1856d18e2352.webm  -max_muxing_queue_size 99999999 65520df3-1033-480e-adde-1856d18e2352.webm.new.mp4


  


  The problem is I think, whenever the user muted themselves, there are no frames. But the PTS is in tact.

  


  This is from the OpenTok documentation (my WebRTC platform)

  


  


  Audio and video frames may not arrive with monotonic timestamps ; frame rates are not always consistent. This is especially relevant if either the video or audio track is disabled for a time, using one of publishVideo or publishAudio publisher properties.

  


  


  


  Frame presentation timestamps (PTS) are written based on NTP
timestamps taken at the time of capture, offset by the timestamp of
the first received frame. Even if a track is muted and later unmuted,
the timestamp offset should remain consistent throughout the duration
of the entire stream. When decoding in post-processing, a gap in PTS
between consecutive frames will exist for the duration of the track
mute : there are no "silent" frames in the container.

  


  


  How can I convert these files and have them play in sync ? Note, when I play in QuickTime or VLC, the files are synced correctly.

  


  EDIT
I've gotten pretty close with this command :

  


   ffmpeg -acodec libopus -i $f -max_muxing_queue_size 99999999 -vsync 1 -af aresample=async=1 -r 30 $f.mp4


  


  But every once in a while, I get a video where the audio starts right away, and they wont actually be talking in the video until half-way thought the video. My guess is they muted themselves during the video conference... so in some cases audio is 5-10 mins ahead. Again, plays fine in quicktime, but pulled into my NLE, its way off.

  


• FFMPEG OpenTok archive audio drift

  25 août 2020, par Chris Kooken

  I am using OpenTok to build a live video platform. It generates webm files from each users stream.

  


  I am using FFMPEG to convert webm (WEBRTC) videos to MP4s to edit in my NLE. The problem I am having is my audio is drifting. I THINK it is because the user pauses the audio during the stream. This is the command i'm running

  


  ffmpeg -acodec libopus -i 65520df3-1033-480e-adde-1856d18e2352.webm  -max_muxing_queue_size 99999999 65520df3-1033-480e-adde-1856d18e2352.webm.new.mp4


  


  The problem is I think, whenever the user muted themselves, there are no frames. But the PTS is in tact.

  


  This is from the OpenTok documentation (my WebRTC platform)

  


  


  Audio and video frames may not arrive with monotonic timestamps ; frame rates are not always consistent. This is especially relevant if either the video or audio track is disabled for a time, using one of publishVideo or publishAudio publisher properties.

  


  


  


  Frame presentation timestamps (PTS) are written based on NTP
timestamps taken at the time of capture, offset by the timestamp of
the first received frame. Even if a track is muted and later unmuted,
the timestamp offset should remain consistent throughout the duration
of the entire stream. When decoding in post-processing, a gap in PTS
between consecutive frames will exist for the duration of the track
mute : there are no "silent" frames in the container.

  


  


  How can I convert these files and have them play in sync ? Note, when I play in QuickTime or VLC, the files are synced correctly.

  


  EDIT
I've gotten pretty close with this command :

  


   ffmpeg -acodec libopus -i $f -max_muxing_queue_size 99999999 -vsync 1 -af aresample=async=1 -r 30 $f.mp4


  


  But every once in a while, I get a video where the audio starts right away, and they wont actually be talking in the video until half-way thought the video. My guess is they muted themselves during the video conference... so in some cases audio is 5-10 mins ahead. Again, plays fine in quicktime, but pulled into my NLE, its way off.