Sur d’autres sites (10310)

• FFMPEG fade in/out expressed in milliseconds

  22 juillet 2020, par orestino

  I working on a programmatic sketch to cut, crop and fade in/out audio from a single input video file.
Everything works pretty well except for the audio fade in fade out.

  


  Everything work fine if I set a fade in and fade out time of 1 (or more) seconds. Here's my command :

  


  ffmpeg -ss 0:1:11.10 -i INPUT_FILE.mp4 -filter:a afade=t=in:st=0:d=1.0,afade=t=out:st=55.867:d=1.0 -filter:v crop=720:720:138:0 -to 0:0:56.86 OUTPUT_FILE.mp4


  


  Fade filter stop working if I set time less than 1 secs (say 100ms like the following example).

  


  ffmpeg -ss 0:1:11.10 -i INPUT_FILE.mp4 -filter:a afade=t=in:st=0:d=0.1,afade=t=out:st=56.767:d=0.1 -filter:v crop=720:720:138:0 -to 0:0:56.86 OUTPUT_FILE.mov


  


  I mean, ffmpeg is not complaining about any error but the output video file simply has no fade in or fade out.

  


  Why ?
What I'm doing wrong ?

  


  Thank you so much for your support

  


  Edit 22/07/2020 :
here's the output from the second command :

  


  ffmpeg -ss 0:1:11.10 -i 2020-07-20_00-13-35.mkv -filter:a afade=t=in:st=0:d=0.1,afade=t=out:st=56.767:d=0.1 -filter:v crop=720:720:138:0 -to 0:0:56.86 ./_selections/2020-07-20_00-13-35_R1_crop2_w_fadein_fadeout_0:1:11.10.mov
ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  WARNING: library configuration mismatch
  avcodec     configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared --enable-version3 --disable-doc --disable-programs --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract --enable-libvo_amrwbenc
  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
Input #0, matroska,webm, from '2020-07-20_00-13-35.mkv':
  Metadata:
    ENCODER         : Lavf57.83.100
  Duration: 00:03:19.23, start: 0.000000, bitrate: 2661 kb/s
    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1440x900, 30 fps, 30 tbr, 1k tbn, 60 tbc (default)
    Metadata:
      DURATION        : 00:03:19.233000000
    Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp (default)
    Metadata:
      title           : audio desktop
      DURATION        : 00:03:19.065000000
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
[libx264 @ 0x564dad2f9680] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x564dad2f9680] profile High, level 3.1
[libx264 @ 0x564dad2f9680] 264 - core 152 r2854 e9a5903 - H.264/MPEG-4 AVC codec - Copyleft 2003-2017 - 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=6 lookahead_threads=1 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, mov, to './_selections/2020-07-20_00-13-35_R1_crop2_w_fadein_fadeout_0:1:11.10.mov':
  Metadata:
    encoder         : Lavf57.83.100
    Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 720x720, q=-1--1, 30 fps, 15360 tbn, 30 tbc (default)
    Metadata:
      DURATION        : 00:03:19.233000000
      encoder         : Lavc57.107.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
    Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      title           : audio desktop
      DURATION        : 00:03:19.065000000
      encoder         : Lavc57.107.100 aac
frame= 1706 fps=140 q=-1.0 Lsize=    2685kB time=00:00:56.86 bitrate= 386.7kbits/s speed=4.67x    
video:1681kB audio:942kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.351145%
[libx264 @ 0x564dad2f9680] frame I:7     Avg QP:15.76  size: 60653
[libx264 @ 0x564dad2f9680] frame P:448   Avg QP:23.13  size:  1372
[libx264 @ 0x564dad2f9680] frame B:1251  Avg QP:31.17  size:   545
[libx264 @ 0x564dad2f9680] consecutive B-frames:  1.3%  2.3%  1.4% 95.0%
[libx264 @ 0x564dad2f9680] mb I  I16..4: 29.2% 37.7% 33.1%
[libx264 @ 0x564dad2f9680] mb P  I16..4:  0.1%  0.4%  0.5%  P16..4:  2.1%  0.2%  0.2%  0.0%  0.0%    skip:96.4%
[libx264 @ 0x564dad2f9680] mb B  I16..4:  0.0%  0.3%  0.2%  B16..8:  1.5%  0.1%  0.1%  direct: 0.0%  skip:97.8%  L0:51.7% L1:46.3% BI: 2.0%
[libx264 @ 0x564dad2f9680] 8x8 transform intra:43.2% inter:27.7%
[libx264 @ 0x564dad2f9680] coded y,uvDC,uvAC intra: 35.4% 42.2% 40.1% inter: 0.3% 0.5% 0.3%
[libx264 @ 0x564dad2f9680] i16 v,h,dc,p: 61% 37%  2%  0%
[libx264 @ 0x564dad2f9680] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30%  7% 59%  2%  0%  0%  0%  1%  1%
[libx264 @ 0x564dad2f9680] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 17% 25% 10%  4%  5%  5%  7%  7%
[libx264 @ 0x564dad2f9680] i8c dc,h,v,p: 66% 19%  7%  7%
[libx264 @ 0x564dad2f9680] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x564dad2f9680] ref P L0: 65.1% 15.3% 13.6%  5.9%
[libx264 @ 0x564dad2f9680] ref B L0: 68.1% 27.1%  4.8%
[libx264 @ 0x564dad2f9680] ref B L1: 95.3%  4.7%
[libx264 @ 0x564dad2f9680] kb/s:242.09
[aac @ 0x564dad2f25a0] Qavg: 606.451


  


• The SSIM values calculated using FFMPEG are not what I expected

  18 mai 2020, par fd-alexer

  I'm trying to encode an m2ts (MPEG-2 Transport Stream) file to mp4 (H.264) and get the ssim value.
  
I did get some SSIM values, but the results were contrary to my expectations.

  



  Are there wrong command options for ffmpeg ?

  



  Encoding and SSIM calculation commands

  



  # encode&#xA;$ ffmpeg -hide_banner -fflags &#x2B;discardcorrupt -i input.m2ts \&#xA;  -c:v libx264 -crf <crf> -preset:v medium \&#xA;  -c:a copy -bsf:a aac_adtstoasc \&#xA;  output_ff_crf-<crf>.mp4&#xA;&#xA;# calculate ssim&#xA;$ ffmpeg -hide_banner -i <a> -i <b> \&#xA;  -lavfi  "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]ssim" \&#xA;  -f null -&#xA;</b></a></crf></crf>

  



  The results of the SSIM

  



  (a) A=input.m2ts,          B=input.m2ts,           ssim=0.973266
(b) A=input.m2ts,          B=output_ff_crf-0.mp4,  ssim=0.813347
(c) A=input.m2ts,          B=output_ff_crf-30.mp4, ssim=0.819897
(d) A=output_ff_crf-0.mp4, B=output_ff_crf-0.mp4,  ssim=1.000000
(e) A=output_ff_crf-0.mp4, B=output_ff_crf-30.mp4, ssim=0.972911


  



  

  • (d)(e) : These are what I expected.

  


  • (a) : The files are the same, but ssim≠1.

  


  • (b)(c) : SSIMs with CRF=0 and CRF=30 have almost the same value, although the image quality is different.

  


  



  In the case of HandBrakeCLI

  



  To determine if there was a problem with input.m2ts, I ran HandBrakeCLI with almost the same parameters as ffmpeg.

  



  # encode&#xA;HandBrakeCLI --verbose --format av_mp4 --encoder x264 --quality <crf> --x264-preset medium \&#xA;  --aencoder copy \&#xA;  --input input.m2ts --output output_hb_crf-<crf>.mp4&#xA;&#xA;# calculate ssim (same as ffmpeg)&#xA;$ ffmpeg -hide_banner -i <a> -i <b> \&#xA;  -lavfi  "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]ssim" \&#xA;  -f null -&#xA;</b></a></crf></crf>

  



  (b') A=input.m2ts,          B=output_hb_crf-0.mp4,  ssim=0.999999
(c') A=input.m2ts,          B=output_hb_crf-30.mp4, ssim=0.972886
(d') A=output_hb_crf-0.mp4, B=output_hb_crf-0.mp4,  ssim=1.000000
(e') A=output_hb_crf-0.mp4, B=output_hb_crf-30.mp4, ssim=0.972886


  



  It's all as I expected. (although A is not ssim=1.0) 
Therefore, I don't see a problem with input.m2ts.

  



  Informations about video files and tools

  



  Results of the ffprobe

  



  input.m2ts

  



  [mpeg2video @ 0x5655577c1680] Invalid frame dimensions 0x0.
    Last message repeated 1 times
[mpegts @ 0x5655577bd080] start time for stream 2 is not set in estimate_timings_from_pts
[mpegts @ 0x5655577bd080] PES packet size mismatch
Input #0, mpegts, from 'input.m2ts':
  Duration: 00:30:02.68, start: 39593.392600, bitrate: 19019 kb/s
  Program 211
    Stream #0:0[0x140]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x141]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 255 kb/s
    Stream #0:2[0x138]: Data: bin_data ([6][0][0][0] / 0x0006)
Unsupported codec with id 100359 for input stream 2


  



  output_ff_crf-0.mp4

  



  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output_ff_crf-0.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf57.83.100
  Duration: 00:30:02.67, start: 0.000000, bitrate: 109301 kb/s
    Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 109040 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 252 kb/s (default)
    Metadata:
      handler_name    : SoundHandler


  



  output_hb_crf-0.mp4

  



  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output_hb_crf-0.mp4':
  Metadata:
    major_brand     : mp42
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 2020-05-17T06:22:06.000000Z
    encoder         : HandBrake 1.1.0 2018042400
  Duration: 00:30:02.22, start: 0.000000, bitrate: 109661 kb/s
    Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 109405 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default)
    Metadata:
      creation_time   : 2020-05-17T06:22:06.000000Z
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 252 kb/s (default)
    Metadata:
      creation_time   : 2020-05-17T06:22:06.000000Z
      handler_name    : Stereo


  



  Tools

  



  version

  



  

  • ffmpeg : 3.4.6-0ubuntu0.18.04.1

  


  • HandBrakeCLI : 1.1.0

  


  



  ldd

  



  $ ldd /usr/bin/ffmpeg
...
libx264.so.152 => /usr/lib/x86_64-linux-gnu/libx264.so.152 (0x00007efbf1f33000)
...

$ ldd /usr/bin/HandBrakeCLI
...
libx264.so.152 => /usr/lib/x86_64-linux-gnu/libx264.so.152 (0x00007efbfb38f000)
...


  



  ffmpeg and HandBrakeCLI are using the same libx264.

  


• Using ffmpeg to convert the audio of an RTSP stream to PCM alaw and hosting an RTSP stream as the output

  24 octobre 2020, par Sirisian

  I have an RTSP stream on my network that is H264 for video and aac for audio. I need the audio converted to PCM alaw with the video just passed through.

  


  I tried :

  


  ffmpeg -rtsp_transport udp -i rtsp://10.0.0.10/main -vcodec copy -acodec pcm_alaw -rtsp_flags listen -f rtsp rtsp://localhost:8001/main


  


  The output I get is :

  


  ffmpeg version 4.3.1-2020-10-01-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.2.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
[rtsp @ 0000025ebdf1db00] Interleaved RTP mode is not supported yet.
Input #0, rtsp, from 'rtsp://10.0.0.10/main':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (aac (native) -> pcm_alaw (native))
Press [q] to stop, [?] for help


  


  It's running, but it's not listening on port 8001. I've tried both UDP and TCP, so I think I am misunderstanding what "-rtsp_flags listen" does for ffmpeg ? My goal is to start an "RTSP server" locally and allow say VLC or any other player to connect and play it. Is this something ffmpeg can do or do I need to send the stream to something else to allow VLC or other media players to connect to it locally ? (The hardware I'm running isn't terribly fast, so I'd like to do this with the least amount of operations possible).

  


  For reference if I run :

  


  ffplay -rtsp_flags listen rtsp://localhost:8001/main


  


  Then run :

  


  ffmpeg -rtsp_transport udp -i rtsp://10.0.0.10/main -vcodec copy -acodec pcm_alaw -f rtsp rtsp://localhost:8001/main


  


  It plays fine in the ffplay application. So it's like the "-rtsp_flags listen" only works with the input portion of ffplay and ffmpeg.