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• libavformat/rtpdec : Fix RTP timestamp wraparound in Producer Reference Time

  7 juin, par Clément Péron

  libavformat/rtpdec : Fix RTP timestamp wraparound in Producer Reference Time
  The rtp_set_prft() function incorrectly calculates the timestamp delta
  
  when RTP timestamps wrap around the 32-bit boundary. The current code :
      delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp ;
  treats both timestamps as large positive values, causing wraparound to
  
  produce a large negative delta instead of the correct small positive delta.
  For example, with a 90kHz video clock :
  
   last_rtcp_timestamp = 0xFFFFFF00 (near wraparound)
  
   timestamp = 0x00000100 (after wraparound)
  
   Current result : delta ≈ -4.3 billion ticks ≈ -47,721 seconds
  
   Expected result : delta ≈ +512 ticks ≈ +0.006 seconds
  This causes prft->wallclock to jump backward by approximately :
  
   90kHz video : 47,721 seconds ( 13.25 hours)
  
   48kHz audio : 89,478 seconds ( 24.9 hours)
  
   8kHz audio : 536,871 seconds ( 6.2 days)
  Fix by casting the subtraction result to int32_t, which correctly
  
  handles wraparound through modular arithmetic :
      delta_timestamp = (int32_t)(timestamp - s->last_rtcp_timestamp) ;
  This ensures the delta is always in the range [-2^31, 2^31-1], making
  
  wraparound produce the correct small positive values.
  Fixes timing jumps in applications that rely on Producer Reference Time
  
  for media synchronization.
  Signed-off-by : Clément Péron <peron.clem@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/rtpdec.c

• 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!

• mediastreamvalidator. How can I validate local files

  20 juin 2012, par Agzam

  1) Is there any way to check segmented HLS files if they compliant and correct ?

  I guess mediastreamvalidator from Apple's HTTP Live Streaming Tools should do that, but when I run it against local index.m3u8, it says "Did not receive playlist".

  2) Is there any way to do stream validation on Windows ?

  3) Initially my problem was to do segmenting on Windows, after a lot of hours of searching I've found out that latest ffmpeg can do that. I'm not sure though if it creates correct HLS videos, it doesn't create .m3u8 playlists, which is not a big deal I guess, as long those segments are correct and use Access Unit Delimiter NALs and etc.