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• Streaming a video with serial port [on hold]

  23 août 2019, par kanki66

  I need an application which is streaming an video recorded by an webcam over the serial port in c#

  I opened an port and can send some data over the port. I have access to the webcam and have an picture/video on the screen. For the webcam part i used this tutorial :
  https://en.code-bude.net/2013/01/02/how-to-easily-record-from-a-webcam-in-c/

  void videoSource_NewFrame(object sender, AForge.Video.NewFrameEventArgs eventArgs)
  
  {
  
    //Cast the frame as Bitmap object and don't forget to use ".Clone()" otherwise
  
    //you'll probably get access violation exceptions
  
    pictureBoxVideo.BackgroundImage = (Bitmap)eventArgs.Frame.Clone();
  
  }

  How can i convert now this to an video as small as possible and send it over the serial port, without saving the video like directly to the serial port ?

  Thank you very much for the help

• avformat/aviobuf : Add ffio_init_(read|write)_context()

  2 septembre 2023, par Andreas Rheinhardt

  avformat/aviobuf : Add ffio_init_(read|write)_context()
  Most users of ffio_init_context() simply want to wrap
  
  a buffer into an AVIOContext ; they do not provide
  
  function pointers at all.
  Therefore this commit adds shortcuts for these two common
  
  operations. This also allows to accept const data when reading
  
  (i.e. the const is now cast away at a central place in
  
  ffio_init_read_context() instead of at several callers).
  
  This also allows to constify the data in ff_text_init_buf().
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavformat/aacdec.c
  • [DH] libavformat/asfenc.c
  • [DH] libavformat/av1.c
  • [DH] libavformat/av1dec.c
  • [DH] libavformat/avio_internal.h
  • [DH] libavformat/aviobuf.c
  • [DH] libavformat/hls.c
  • [DH] libavformat/id3v2.c
  • [DH] libavformat/matroskadec.c
  • [DH] libavformat/mmst.c
  • [DH] libavformat/mov.c
  • [DH] libavformat/mpegts.c
  • [DH] libavformat/mpjpegdec.c
  • [DH] libavformat/oggenc.c
  • [DH] libavformat/rdt.c
  • [DH] libavformat/rtpdec_asf.c
  • [DH] libavformat/rtpdec_qt.c
  • [DH] libavformat/rtsp.c
  • [DH] libavformat/sapdec.c
  • [DH] libavformat/subtitles.c
  • [DH] libavformat/subtitles.h
  • [DH] libavformat/vividas.c

• How to adjust mpeg 2 ts start time with ffmpeg ?

  24 juillet 2019, par Maxim Kornienko

  I’m writing simple HLS (Http Live Streaming) java server to live cast (really live, not on demand) screenshow + voice. I constantly get chunks of image frames and audio samples as input to my service and produce mpeg 2 ts files + m3u8 playlist web page as output. The workflow is the following :

  1. Collect (buffer) source video frames and audio for certain period of time
  2. Convert series of video frames to h.264 encoded video file
  3. Convert audio samples to mp3 audio file

  4. Merge them to .ts file with ffmpeg command

     ffmpeg -i audio.mp3 -i video.mp4 -f mpegts -c:a copy -c:v copy -vprofile main -level:v 4.0 -vbsf h264_mp4toannexb -flags -global_header segment.ts

  5. Publish several .ts files on m3u8 playlist.

  The problem is resulting playlist interrupts after first segment is played. VLC logs following error :

  freetype error: Breaking unbreakable line
  
  ts error: libdvbpsi (PSI decoder): TS discontinuity (received 0, expected 4) for PID 17
  
  ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 0
  
  ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 4096
  
  core error: ES_OUT_SET_(GROUP_)PCR is called too late (pts_delay increased to 1000 ms)
  
  core error: ES_OUT_RESET_PCR called
  
  core error: Could not convert timestamp 185529572000
  
  ts error: libdvbpsi (PSI decoder): TS discontinuity (received 0, expected 4) for PID 17
  
  ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 0
  
  ts error: libdvbpsi (PSI decoder): TS duplicate (received 0, expected 1) for PID 4096
  
  core error: ES_OUT_SET_(GROUP_)PCR is called too late (jitter of 8653 ms ignored)
  
  core error: Could not get display date for timestamp 0
  
  core error: Could not convert timestamp 185538017000
  
  core error: Could not convert timestamp 185538267000
  
  core error: Could not convert timestamp 185539295977
  
  ...

  I guess the reason is that start time of segments do not belong to one stream, but it’s impossible to concat and resegment (with ffmepg -f segment) whole stream once new chunk is added. Tried adding #EXT-X-DISCONTINUITY tag to playlist as suggested here but it didn’t help. When I ffprobe them I get :

  Input #0, mpegts, from '26.ts':
  
  Duration: 00:00:10.02, start: 1.876978, bitrate: 105 kb/s
  
  Program 1
  
  Metadata:
  
    service_name    : Service01
  
    service_provider: FFmpeg
  
  Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 640x640, 4 fps, 4 tbr, 90k tbn, 8 tbc
  
  Stream #0:1[0x101]: Audio: mp3 ([3][0][0][0] / 0x0003), 48000 Hz, mono, s16p, 64 kb/s   

  Where start value in line Duration: 00:00:10.02, start: 1.876978, bitrate: 105 kb/s is more or less equal for all segments.
  When I check segments from available proven-to-work playlists (like http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8) they all have diffrenet start values for each segment, for example :

  Input #0, mpegts, from 'segm150518140104572-424570.ts':
  
  Duration: 00:00:06.17, start: 65884.808689, bitrate: 479 kb/s
  
  Program 257
  
  Stream #0:0[0x20]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 30 fps, 29.97 tbr, 90k tbn, 60 tbc
  
  Stream #0:1[0x21]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 115 kb/s
  
  Stream #0:2[0x22]: Data: timed_id3 (ID3  / 0x20334449)

  and the next after it

  Input #0, mpegts, from 'segm150518140104572-424571.ts':
  
  Duration: 00:00:06.22, start: 65890.814689, bitrate: 468 kb/s
  
  Program 257
  
  Stream #0:0[0x20]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 320x180 [SAR 1:1 DAR 16:9], 30 fps, 29.97 tbr, 90k tbn, 60 tbc
  
  Stream #0:1[0x21]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 124 kb/s
  
  Stream #0:2[0x22]: Data: timed_id3 (ID3  / 0x20334449)

  differ in the way that start time of segm150518140104572-424571.ts is equal to start time + duration of segm150518140104572-424570.ts.

  How could this start value be adjusted with ffmpeg ? Or maybe my whole aproach is wrong ? Unfortunately I couldn’t find on the internet working example of live (not on demand) video service implemented with ffmepg.