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• tests/checkasm : add checkasm tests for swscale ops

  25 mars, par Niklas Haas

  tests/checkasm : add checkasm tests for swscale ops
  Because of the lack of an external ABI on low-level kernels, we cannot
  
  directly test internal functions. Instead, we construct a minimal op chain
  
  consisting of a read, the op to be tested, and a write.
  The bigger complication arises from the fact that the backend may generate
  
  arbitrary internal state that needs to be passed back to the implementation,
  
  which means we cannot directly call `func_ref` on the generated chain. To get
  
  around this, always compile the op chain twice - once using the backend to be
  
  tested, and once using the reference C backend.
  The actual entry point may also just be a shared wrapper, so we need to
  
  be very careful to run checkasm_check_func() on a pseudo-pointer that will
  
  actually be unique for each combination of backend and active CPU flags.
  

  • [DH] tests/checkasm/Makefile
  • [DH] tests/checkasm/checkasm.c
  • [DH] tests/checkasm/checkasm.h
  • [DH] tests/checkasm/sw_ops.c
  • [DH] tests/fate/checkasm.mak

• Use ffmpeg libraries to convert stream formats

  17 septembre 2021, par Syrinx

  I'm attempting to write a small program and link it to a minimal set of ffmpeg libraries, like libavformat and whatever other libraries I need.

  


  I am looking for documentation to get me started, or maybe a quick fix to the example program I am using.

  


  I know ffmpeg (the project) provides example programs to help developers get started. I'm using the transcoding example, as it's close to my end goal, but it exits during init with an error about an audio issue.

  


  Here I am using the transcoding example program that come with ffmpeg v4.4, on Ubuntu 18.04. My input source has one video channel (h264) and one audio channel (pcm_mulaw).

  


  $ LD_LIBRARY_PATH=../../dist/lib ./transcoding rtsp://ip-camera/stream out.flv
...
  Stream #0:0: Video: h264, yuv420p, 1280x720, q=2-31, 20 tbn
  Stream #0:1: Audio: pcm_mulaw, 8000 Hz, 0 channels, s16
auto_resampler_0 @ 0x55da787fa140] [SWR @ 0x55da787fa5c0] Rematrix is needed between mono and 0 channels but there is not enough information to do it
[auto_resampler_0 @ 0x55da787fa140] Failed to configure output pad on auto_resampler_0


  


  In libswresample/swresample.c :

  


  320      if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
321          av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
322                 "but there is not enough information to do it\n", l1, l2);
323          ret = AVERROR(EINVAL);
324          goto fail;
325      }


  


  I'd really like it if I could make the transcoding example program work (fix it, or maybe use it appropriately if I am misunderstanding something). But short of that, where should I look for documentation about using the ffmpeg libraries ?

  


  I don't even care about the audio. If I can just disable the audio, I would be happy with that solution. I tried tracking the "-an" option to ffmpeg (the program) to see how it does that in source code, but the options handling is a mess and I can't distinguish the parts of the code that I need from all the noise.

  


  ffmpeg has web pages like this that aren't useful at all. There is documentation in the source code that looks like it should be viewed as HTML, but I don't see it exported anywhere. "make doc" generates a very small set of man pages that are insufficient to get me started.

  


• Playing RTSP in WPF application with low latency using FFMPEG / FFMediaElement (FFME)

  22 mars 2019, par Paboka

  I’m trying to use FFMediaElement (FFME, WPF MediaElement replacement based on FFmpeg) component to play RSTP live video in my WPF application.

  I have a good connection to my camera and I want to play it with minimum available latency.

  I’ve reduced the latency by changing ProbeSize to its minimal value :

  private void Media_OnMediaInitializing(object Sender, MediaInitializingRoutedEventArgs e)
  
  {
  
    e.Configuration.GlobalOptions.ProbeSize = 32;
  
  }

  But I still have about 1 second of latency since the very beginning of the stream. I mean, when I start playing, I have to wait for 1 second till the video appears and then I have 1s of latency.

  I’ve also tried to change following parameters :

  e.Configuration.GlobalOptions.EnableReducedBuffering = true;
  
  e.Configuration.GlobalOptions.FlagNoBuffer = true;
  
  e.Configuration.GlobalOptions.MaxAnalyzeDuration = TimeSpan.Zero;

  but it gave no result.

  I measured time-interval between FFmpeg output lines (the number in the first column is the time, elapsed from the previous line, ms)

   ----     OpenCommand: Entered
  
     39     FFInterop.Initialize: FFmpeg v4.0
  
      0     EVENT START: MediaInitializing
  
      0     EVENT DONE : MediaInitializing
  
    379     EVENT START: MediaOpening
  
      0     EVENT DONE : MediaOpening
  
      0     COMP VIDEO: Start Offset:      0,000; Duration:        N/A
  
     41     SYNC-BUFFER: Started.
  
    609     SYNC-BUFFER: Finished. Clock set to 1534932751,634
  
      0     EVENT START: MediaOpened
  
      0     EVENT DONE : MediaOpened
  
      0     EVENT START: BufferingStarted
  
      0     EVENT DONE : BufferingStarted
  
      0     OpenCommand: Completed
  
      0     V BLK: 1534932751,634 | CLK: 1534932751,634 | DFT:    0 | IX:   0 | PQ:     0,0k | TQ:     0,0k
  
      0     Command Queue (1 commands): Before ProcessNext
  
      0        Play - ID: 404 Canceled: False; Completed: False; Status: WaitingForActivation; State: 
  
     94     V BLK: 1534932751,675 | CLK: 1534932751,699 | DFT:   24 | IX:   1 | PQ:     0,0k | TQ:     0,0k

  So, the most the process of "sync-buffering" takes the most of the time.

  Is there any parameter of FFmpeg which allows reducing a size of the buffer ?