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• dca_parser : ensure frame duration matches the actual sample rate

  3 février 2016, par Hendrik Leppkes

  dca_parser : ensure frame duration matches the actual sample rate
  The parser only parses the core, and thus has a duration relative
  
  to the core sample rate only, not the actual stream sample rate.
  FATE references changed due to now correct timestamps.
  

  • [DH] libavcodec/dca_parser.c
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_0
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_0-dmix_2
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_0-dmix_6
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_1
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_1-dmix_2
  • [DH] tests/ref/fate/dca-xll_51_16_192_768_1-dmix_6
  • [DH] tests/ref/fate/dca-xll_71_24_96_768
  • [DH] tests/ref/fate/dca-xll_71_24_96_768-dmix_2
  • [DH] tests/ref/fate/dca-xll_71_24_96_768-dmix_6
  • [DH] tests/ref/fate/dca-xll_x96_51_24_96_1509
  • [DH] tests/ref/fate/dca-xll_x96_51_24_96_1509-dmix_2
  • [DH] tests/ref/fate/dca-xll_x96_51_24_96_1509-dmix_6

• ffmpeg how to ensure audio+video synchronization ?

  22 juin 2020, par roko

  im using this conmmand :

  


  /usr/bin/ffmpeg \
-f pulse -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor \
-f pulse -i alsa_input.pci-0000_00_1b.0.analog-stereo \
-f x11grab -video_size 2560x1600 -framerate 8 -i :0.0 \
-filter_complex "amix=inputs=2[a]" \
-map 2:v -map '[a]' \
-c:a aac -b:a 128k \
-c:v h264_nvenc -b:v 1500k -maxrate 1500k -minrate 1500k \
-override_ffserver -g 16 http://10.100.102.18:8090/feed1.ffm


  


  my goal is to record screens continuously for long time (all the year)

  


  this is my ffmpeg version (running on RedHat 7.4) :

  


  ffmpeg version 3.4.7 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-39)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --disable-encoder=libopus --enable-libpulse --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100


  


  when i run these command i get warning : non monotonous dts in output stream 0:1. this is the output :

  


  [pulse @ 0x13832c0] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
Output #0, ffm, to 'http://193.100.200.206:8090/feed1.ffm':
  Metadata:
    creation_time   : now
    encoder         : Lavf57.83.100
    Stream #0:0: Video: h264 (h264_nvenc) (Main), bgr0(progressive), 1920x1080, q=-1--1, 1500 kb/s, 8 fps, 1000k tbn, 8 tbc
    Metadata:
      encoder         : Lavc57.107.100 h264_nvenc
    Side data:
      cpb: bitrate max/min/avg: 1500000/0/1500000 buffer size: 3000000 vbv_delay: -1
    Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc57.107.100 aac
[aac @ 0x13dda80] Queue input is backward in time0:00:00.27 bitrate=5907.7kbits/s speed=0.552x    
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320000, current: -195354; changing to 320001. This may result in incorrect timestamps in the output file.                                                              
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320001, current: -174021; changing to 320002. This may result in incorrect timestamps in the output file.                                                              
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320002, current: -152688; changing to 320003. This may result in incorrect timestamps in the output file.                                                              
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320003, current: -131354; changing to 320004. This may result in incorrect timestamps in the output file.                                                              
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320004, current: -110021; changing to 320005. This may result in incorrect timestamps in the output file.                                                              
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320005, current: -88688; changing to 320006. This may result in incorrect timestamps in the output file.                                                               
[ffm @ 0x13bd5c0] Non-monotonous DTS in output stream 0:1; previous: 320006, current: -67354; changing to 320007. This may result in incorrect timestamps in the output file


  


  when i use -async 1 the warning disapear but, when i play audio file, the audio is Stretches/squeezes for few seconds at the begining and in the end.

  


  how can i ensure that audio and video always stay synchronize ?

  


  does another ffmpeg version fix this problem ? (can i install ffmpeg 4.2.3 on redhat 7.4/7.7) (i know that there is asyncts flag but i dont have it in my version)

  


  Please help me to find solution in order to keep synchronization for long recording.

  


• avcodec/alsdec : Set channels from data after data is set

  18 mars 2022, par Michael Niedermayer

  avcodec/alsdec : Set channels from data after data is set
  Fixes : out of array write
  
  Fixes : 45624/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-6473487382872064
  
  Fixes : 45626/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-4874997192065024
  Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
  
  Reviewed-by : James Almer <jamrial@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavcodec/alsdec.c