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• FFmpeg create output directory hierarchy

  23 avril 2017, par David

  Recently bought an ip-cam which outputs a RTSP stream. I’m using the segment option of FFmpeg to create 60 minute long recordings.

  I want FFmpeg to write the files to a directory based on Year/Month/Date, and write to a file Hour-Minute.mp4 For example :
  /raid1/homes/share/public/recordings/queue/bedroom/2017/04/23/13-05.mp4 for a recording started on 23 april 2017, 13:05.

  Unfortunately FFmpeg seems to not create the directory hierarchy. FFmpeg quits since the directory can not be found.

  Input #0, rtsp, from 'rtsp://192.168.1.240/unicast':
  
    Metadata:
  
      title           : LIVE555 Streaming Media v2014.07.04
  
      comment         : LIVE555 Streaming Media v2014.07.04
  
    Duration: N/A, start: 0.000750, bitrate: N/A
  
      Stream #0:0: Video: h264 (High), yuv420p, 1920x1080, 90k tbr, 90k tbn, 180k tbc
  
      Stream #0:1: Audio: pcm_alaw, 8000 Hz, 1 channels, s16, 64 kb/s
  
  [segment @ 0x2557300] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
  
  [segment @ 0x2557300] Failed to open segment '/raid1/homes/share/public/recordings/queue/bedroom/2017/04/23/14-19.mp4'
  
  Output #0, segment, to '/raid1/homes/share/public/recordings/queue/bedroom/%Y/%m/%d/%H-%M.mp4':
  
    Metadata:
  
      title           : LIVE555 Streaming Media v2014.07.04
  
      comment         : LIVE555 Streaming Media v2014.07.04
  
      encoder         : Lavf57.41.100
  
      Stream #0:0: Video: h264, yuv420p, 1920x1080, q=2-31, 90k tbr, 90k tbn, 90k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Could not write header for output file #0 (incorrect codec parameters ?): No such file or directory

  record.sh is as follows :

  #!/bin/sh
  
  ffmpeg -stimeout 600\
  
   -rtsp_transport udp \
  
   -i rtsp://192.168.1.240/unicast \
  
   -c copy \
  
   -map 0:0 \
  
   -f segment \
  
   -segment_time 3600 \
  
   -segment_wrap 100 \
  
   -segment_format mov \
  
   -strftime 1 \
  
   -reset_timestamps 1 \
  
   "/raid1/homes/share/public/recordings/queue/bedroom/%Y/%m/%d/%H-%M.mp4"

  I’ve tried not using a directory hierachy : "/raid1/homes/share/public/recordings/queue/bedroom/%Y-%m-%d_%H-%M.mp4". This works fine.

  $ ffmpeg -version
  
  ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
  
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
  
  libavutil      55. 28.100 / 55. 28.100
  
  libavcodec     57. 48.101 / 57. 48.101
  
  libavformat    57. 41.100 / 57. 41.100
  
  libavdevice    57.  0.102 / 57.  0.102
  
  libavfilter     6. 47.100 /  6. 47.100
  
  libavresample   3.  0.  0 /  3.  0.  0
  
  libswscale      4.  1.100 /  4.  1.100
  
  libswresample   2.  1.100 /  2.  1.100
  
  libpostproc    54.  0.100 / 54.  0.100

  Can FFmpeg create output directories on the go ?

• ffmpegmediaMetadataRetriever cannot be used for m3u8 format ?

  19 juillet 2017, par Nandz

  I want to get the frames from the video URL which is of m3u8 format. I couldn’t retrieve the frame at the specified time. When i tried with mp4 format, it worked. ( Am using android videoview)

  Here is my code.

   public class VideoViewAdapterClass extends RecyclerView.Adapter {
  
  
  
          private List<videomodelactivity> videoListModel;
  
          private Context context;
  
          public Bitmap bitmap;
  
         String videoPath = "https://xxxxxx.xxxxxx.xxxxx/HLS/xxxxxxxx/fruit_milkshake-master-playlist.m3u8";
  
  
  
          public VideoViewAdapterClass(List<videomodelactivity> videoList, Context context) {
  
              this.context = context;
  
  
  
              videoListModel = videoList;
  
  
  
          }
  
  
  
          @Override
  
          public VideoViewAdapterClass.ViewHolder onCreateViewHolder(ViewGroup parent, int viewType) {
  
  
  
              View itemLayoutView = LayoutInflater.from(parent.getContext()).inflate(
  
                      R.layout.customized_layout, null);
  
  
  
              ViewHolder viewHolder = new ViewHolder(itemLayoutView);
  
              return viewHolder;
  
  
  
          }
  
  
  
          @Override
  
          public void onBindViewHolder(final VideoViewAdapterClass.ViewHolder holder, int position) {
  
  
  
  
  
             final VideoModelActivity videoModel = videoListModel.get(position);
  
              holder.movieName.setText(videoModel.getMovieName());
  
  
  
              MediaMetadataRetriever mediaMetadataRetriever = null;
  
  
  
              try {
  
                  mediaMetadataRetriever = new MediaMetadataRetriever();
  
                  if (Build.VERSION.SDK_INT >= 14)
  
  
  
                      mediaMetadataRetriever.setDataSource(videoPath, new HashMap());
  
                  bitmap = mediaMetadataRetriever.getFrameAtTime(2000000, MediaMetadataRetriever.OPTION_CLOSEST_SYNC);
  
  
  
  
  
              } catch (Exception e) {
  
                  e.printStackTrace();
  
  
  
              } finally {
  
                  if (mediaMetadataRetriever != null)
  
                      mediaMetadataRetriever.release();
  
              }
  
  
  
              holder.vidImg.setImageBitmap(bitmap);
  
  
  
                      holder.vidImg.setOnClickListener(new View.OnClickListener() {
  
                  @Override
  
                  public void onClick(View view) {
  
  
  
                      Intent intent = new Intent (context, VideoViewActivity.class);
  
                      intent.setFlags(Intent.FLAG_ACTIVITY_NEW_TASK);
  
                      context.startActivity(intent);
  
  
  
                  }
  
              });
  
  
  
  
  
          }
  
  
  
          @Override
  
          public int getItemCount() {
  
              return videoListModel.size();
  
          }
  
  
  
          public class ViewHolder extends RecyclerView.ViewHolder {
  
  
  
              ImageView vidImg;
  
              TextView movieName;
  
              public ViewHolder(View itemView) {
  
                  super(itemView);
  
  
  
                  vidImg = (ImageView) itemView.findViewById(R.id.imageVid);
  
                  movieName = (TextView) itemView.findViewById(R.id.movieName);
  
              }
  
          }
  
      }
  
  </videomodelactivity></videomodelactivity>

• WebRTC predictions for 2016

  17 février 2016, par silvia

  I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

  WebRTC Browser support

  I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016 :

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

  Codec Observations

  With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

  However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

  Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

  I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability : I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

  The Enterprise Boundary

  Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary : firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

  The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

  SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

  We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

  Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

  We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

  What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

  I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.

  Summary

  So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress : interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

  —

  It’s worth mentioning Philipp Hancke’s tweet reply to my post :

  https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/ … — we saw some clever people come up with a solution already. Now it needs to be implemented

  The post WebRTC predictions for 2016 first appeared on ginger’s thoughts.