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• Real-Time Buffer Too Full (FFMPEG)

  25 janvier 2018, par Nimble

  So I’ve been having this issue with ffmpeg, it has been a journey getting the hardware and command to actually do what I want, but I still have one problem.

  Sometimes when I’m recording I just start dropping frames like crazy, this can be after an hour of recording or even ten hours in... Everything will be working fine and then suddenly I’ll start dropping frames due to "real-time buffer too full or near too full". This happens regardless of how low I put the bitrate, and the buffer size is high as it will allow, eventually I’ll just start dropping frames. Almost seems like it could be like a power saving feature kicking in but it’s too inconsistent it seems. Like I said sometimes I can go 10 hours without having this issue.

  Any ideas ?

  Here is my block of code :

  ffmpeg -guess_layout_max 0 -y -f dshow -video_size 3440x1440 -rtbufsize 2147.48M -pixel_format nv12 -framerate 200 ^
  
  -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" -map 0:0,0:1 -map 0:1 ^
  
  -preset: llhp -codec:v h264_nvenc -pix_fmt nv12 -b:v 250M -maxrate:v 250M -minrate:v 250M -bufsize:v 250M -b:a 320k ^
  
  -ac 2 -r 100 -async 1 -vsync 1 -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\PC%02d.mp4 ^
  
  -guess_layout_max 0 -f dshow -rtbufsize 2000M -i audio="Analog (3+4) (RME Fireface UC)" -map 1:0 -b:a 320k -ac 2 ^
  
  -af "adelay=200|200" -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\Voices\Theirs\TPC%02d.wav ^
  
  -guess_layout_max 0 -f dshow -rtbufsize 2000M -i audio="Analog (5+6) (RME Fireface UC)" -map 2:0 -b:a 320k -ac 2 ^
  
  -af "adelay=825|825" -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\Voices\Mine\MPC%02d.wav

  Here is the error, it repeated around 300 times before locking up ffmpeg forcing my to quit before starting the recording again :

  [dshow @ 0000019a596bdcc0] real-time buffer [Video (00 Pro Capture HDMI 4K+)] [video input] too full or near too full (62% of size: 2147480000 [rtbufsize parameter])! frame dropped!

• Annual Release of External-Videos plugin – we’ve hit v1.0

  13 janvier 2017, par silvia

  This is the annual release of my external-videos wordpress plugin and with the help of Andrew Nimmolo I’m proud to annouce we’ve reached version 1.0 !

  So yes, my external-videos wordpress plugin is now roughly 7 years old, who would have thought ! During the year, I don’t get the luxury of spending time on maintaining this open source love child of mine, but at Christmas, my bad conscience catches up with me – every year ! I then spend some time going through bug reports, upgrading the plugin to the latest wordpress version, upgrading to the latest video site APIs, testing functionality and of course making a new release.

  This year has been quite special. The power of open source has kicked in and a new developer took an interest in external-videos. Andrew Nimmolo submitted patches over all of 2016. He decided to bring the external-videos plugin into the new decade with a huge update to the layout of the settings pages, general improvements, and an all-round update of all the video site APIs which included removing their overly complex SDKs and going straight for the REST APIs.

  Therefore, I’m very proud to be able to release version 1.0 today. Thanks, Andrew !

  Enjoy – and I look forward to many more contributions – have a Happy 2017 !

  —

  NOTE : If you’re upgrading from an older version, you might need to remove and re-add your social video sites because the API details have changed a bit. Also, we noticed that there were layout issues on WordPress 4.3.7, so try and make sure your WordPress version is up to date.

  The post Annual Release of External-Videos plugin – we’ve hit v1.0 first appeared on ginger’s thoughts.

• avformat/pcm : factorize and improve determining the default packet size

  2 mars 2024, par Marton Balint

  avformat/pcm : factorize and improve determining the default packet size
   Remove the 1024 cap on the number of samples, for high sample rate audio it
  
    was suboptimal, calculate the low neighbour power of two for the number of
  
    samples (audio blocks) instead.
  
   Make the function work correctly also for non-pcm codecs by using the stream
  
    bitrate to estimate the target packet size. A previous version of this patch
  
    used av_get_audio_frame_duration2() the estimate the desired packet size, but
  
    for some codecs that returns the duration of a single audio frame regardless
  
    of frame_bytes.
  
   Fallback to 4096/block_align*block_align if bitrate is not available.
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] libavformat/pcm.c
  • [DH] libavformat/pcm.h
  • [DH] tests/ref/seek/lavf-al
  • [DH] tests/ref/seek/lavf-ul