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• avformat/libsrt : fix race condition with libsrt_network_wait_fd and epoll

  31 janvier 2021, par Marton Balint

  avformat/libsrt : fix race condition with libsrt_network_wait_fd and epoll
  The way SRT's async / epoll-based IO works is that the event status is stored
  
  in the epoll containers. That is, if an event occurs on an SRT socket, and that
  
  SRT socket isn't part of any epoll container, then that event is lost. If we
  
  later add that socket to an epoll container, we still won't receive the event
  
  even if it wasn't serviced.
  Therefore we create the epoll and put the fd into it right after the connection
  
  is established.
  See http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2021-January/275334.html
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] libavformat/libsrt.c

• FFMPEG decode from RTP dump file into mp3 file

  5 février 2021, par pingvincible

  I'm trying to save RTP stream into mp3 file. I use this command :

  


  ffmpeg -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3


  


  I get this result :

  


  user@pc:~/$ ffmpeg-amrnb -loglevel debug -protocol_whitelist file -f rtp -i microphone.rtpdump -f mp3 microphone.mp3
ffmpeg version N-100958-g4f3d8cb554 Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
  configuration: --enable-gpl --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-nonfree --enable-version3
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.120.100 / 58.120.100
  libavformat    58. 65.101 / 58. 65.101
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.102.100 /  7.102.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'rtp'.
Reading option '-i' ... matched as input url with argument 'microphone.rtpdump'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option 'microphone.mp3' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url microphone.rtpdump.
Applying option f (force format) with argument rtp.
Successfully parsed a group of options.
Opening an input file: microphone.rtpdump.
[rtp @ 0x556947200580] Unable to receive RTP payload type 97 without an SDP file describing it
microphone.rtpdump: Invalid data found when processing input


  


  It looks like that microphone.rtpdump file format is correct as ffmpeg can find RTP payload type 97. The problem is that I don't understand how to use SDP file in this situation.

  


  I have an SDP file for this payload type which I use, when I send data over network. It looks like this :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 58.65.101
m=audio 1234 RTP/AVP 97
b=AS:12
a=rtpmap:97 AMR/8000/1
a=fmtp:97 octet-align=1


  


  And now I want to decode RTP stream from file, not by receiving it from network.

  


  How to adapt my SDP file to read RTP stream from file ?

  


  UPDATE : My rtpdump file is not a real rtpdump file format. It is just payloads from UDP packets written into file without any additional headers.

  


• vstack results in out of sync audio

  12 février 2021, par Arlo

  When I stack two videos using vstack, the result has audio sync issues for the bottom video.

  


  My starting point : four separate RTP tracks captured from a 2 person video chat :

  


   Actor1Video.webm,
 Actor1Audio.webm,
 Actor2Video.webm,
 Actor2Audio.webm


  


  I use vstack to put Actor1 on top and Actor2 on bottom :

  


  ffmpeg -i Actor1Video.webm -i Actor2Video.webm -i Actor1Audio.webm -i Actor2Audio.webm  -filter_complex "[1][0]scale2ref=oh*mdar:ih[2nd][ref];[ref][2nd]vstack=inputs=2[v];[2:a][3:a]join=inputs=2:channel_layout=stereo:map=0.0-FL|1.0-FR[a]" -c:a libfdk_aac -map "[v]" -map "[a]"  -vsync 2 ActorsCombined.mp4


  


  Here's the log :

  


  ffmpeg version git-2021-02-08-89f78dd Copyright (c) 2000-2021 the FFmpeg developers
  built with Apple clang version 11.0.3 (clang-1103.0.32.62)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/HEAD-89f78dd_6 --enable-shared --cc=clang --host-cflags= --host-ldflags= --enable-gpl --enable-libaom --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-libsnappy --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-demuxer=dash --disable-libjack --disable-indev=jack --enable-opencl --enable-videotoolbox --disable-htmlpages --enable-libfdk-aac --enable-nonfree
  libavutil      56. 64.100 / 56. 64.100
  libavcodec     58.121.100 / 58.121.100
  libavformat    58. 67.100 / 58. 67.100
  libavdevice    58. 11.103 / 58. 11.103
  libavfilter     7.103.100 /  7.103.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, matroska,webm, from 'Actor1Video.webm':
  Metadata:
    title           : FFmpeg
    ENCODER         : Lavf58.29.100
  Duration: 447576:28:17.41, start: 1611273978.135000, bitrate: N/A
  Stream #0:0: Video: vp8, yuv420p(tv, bt470bg/unknown/unknown, progressive), 1280x720, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 1k tbn, 1k tbc (default)
    Metadata:
      DURATION        : 447576:28:17.408999
Input #1, matroska,webm, from 'Actor2Video.webm':
  Metadata:
    title           : FFmpeg
    ENCODER         : Lavf58.29.100
  Duration: 447576:28:17.45, start: 1611273978.257000, bitrate: N/A
  Stream #1:0: Video: vp8, yuv420p(tv, bt470bg/unknown/unknown, progressive), 320x180, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 1k tbn, 1k tbc (default)
    Metadata:
      DURATION        : 447576:28:17.453999
Input #2, matroska,webm, from 'Actor1Audio.webm':
  Metadata:
    title           : FFmpeg
    ENCODER         : Lavf58.29.100
  Duration: 447576:28:17.49, start: 1611273978.112000, bitrate: N/A
  Stream #2:0: Audio: opus, 48000 Hz, stereo, fltp (default)
    Metadata:
      DURATION        : 447576:28:17.492000
Input #3, matroska,webm, from 'Actor2Audio.webm':
  Metadata:
    title           : FFmpeg
    ENCODER         : Lavf58.29.100
  Duration: 447576:28:17.45, start: 1611273978.208000, bitrate: N/A
  Stream #3:0: Audio: opus, 48000 Hz, stereo, fltp (default)
    Metadata:
      DURATION        : 447576:28:17.447999
File 'ActorsCombined.mp4' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:0 (vp8) -> scale2ref:ref
  Stream #1:0 (vp8) -> scale2ref:default
  Stream #2:0 (opus) -> join:input0
  Stream #3:0 (opus) -> join:input1
  vstack -> Stream #0:0 (libx264)
  join -> Stream #0:1 (libfdk_aac)
Press [q] to stop, [?] for help
[libx264 @ 0x7ff0c1831a00] using SAR=1/1
[libx264 @ 0x7ff0c1831a00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x7ff0c1831a00] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0x7ff0c1831a00] 264 - core 161 r3043 59c0609 - H.264/MPEG-4 AVC codec - Copyleft 2003-2021 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'ActorsCombined.mp4':
  Metadata:
    title           : FFmpeg
    encoder         : Lavf58.67.100
  Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p(progressive), 1280x1440 [SAR 1:1 DAR 8:9], q=2-31, 29.97 fps, 30k tbn (default)
    Metadata:
      encoder         : Lavc58.121.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
  Stream #0:1: Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 139 kb/s (default)
    Metadata:
      encoder         : Lavc58.121.100 libfdk_aac
frame=36626 fps= 15 q=-1.0 Lsize=  389420kB time=00:21:59.38 bitrate=2417.9kbits/s dup=0 drop=34791 speed=0.535x    
video:365641kB audio:22446kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.343645%
[libx264 @ 0x7ff0c1831a00] frame I:158   Avg QP:15.51  size:107833
[libx264 @ 0x7ff0c1831a00] frame P:9670  Avg QP:18.71  size: 25824
[libx264 @ 0x7ff0c1831a00] frame B:26798 Avg QP:24.90  size:  4018
[libx264 @ 0x7ff0c1831a00] consecutive B-frames:  0.6%  5.2%  0.6% 93.5%
[libx264 @ 0x7ff0c1831a00] mb I  I16..4: 13.2% 75.5% 11.3%
[libx264 @ 0x7ff0c1831a00] mb P  I16..4:  1.2%  3.6%  0.2%  P16..4: 43.1% 10.4%  5.9%  0.0%  0.0%    skip:35.6%
[libx264 @ 0x7ff0c1831a00] mb B  I16..4:  0.1%  0.1%  0.0%  B16..8: 28.3%  0.7%  0.1%  direct: 2.3%  skip:68.5%  L0:45.1% L1:53.6% BI: 1.3%
[libx264 @ 0x7ff0c1831a00] 8x8 transform intra:71.6% inter:85.4%
[libx264 @ 0x7ff0c1831a00] coded y,uvDC,uvAC intra: 50.4% 77.2% 47.8% inter: 6.9% 17.0% 3.8%
[libx264 @ 0x7ff0c1831a00] i16 v,h,dc,p: 37% 28% 14% 22%
[libx264 @ 0x7ff0c1831a00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 17% 25%  4%  6%  7%  5%  6%  5%
[libx264 @ 0x7ff0c1831a00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 35% 24% 16%  4%  6%  5%  4%  4%  2%
[libx264 @ 0x7ff0c1831a00] i8c dc,h,v,p: 60% 16% 17%  6%
[libx264 @ 0x7ff0c1831a00] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x7ff0c1831a00] ref P L0: 63.1%  9.9% 20.5%  6.6%
[libx264 @ 0x7ff0c1831a00] ref B L0: 90.0%  8.9%  1.1%
[libx264 @ 0x7ff0c1831a00] ref B L1: 94.7%  5.3%
[libx264 @ 0x7ff0c1831a00] kb/s:2270.36


  


  The resulting file begins in sync, but after a few minutes the bottom video is suddenly out of sync with its audio.

  


  The strange thing is, if I merge these videos with their audio separately, without using vstack, there's no sync issue :

  


  ffmpeg -i Actor1Video.webm -i Actor1Audio.webm -vsync 2 Actor1.mp4 &&
ffmpeg -i Actor2Video.webm -i Actor2Audio.webm -vsync 2 Actor2.mp4


  


  When I do the above, the two videos are perfectly in sync. But if I take these two mp4s and stack them, I have the same issue where the bottom video goes out of sync.

  


  Any suggestions ?