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• Using FFmpeg with URL input causes SIGSEGV in AWS Lambda (Python runtime)

  26 mars, par Dave94

  I'm trying to implement a video converting solution on AWS Lambda following their article named Processing user-generated content using AWS Lambda and FFmpeg. However when I run my command with subprocess.Popen() it returns -11 which translates to SIGSEGV (segmentation fault). I've tried to process the video with the newest (4.3.1) static build from John Van Sickle's site as with the "official" ffmpeg-lambda-layer but it seems like it doesn't matter which one I use, the result is the same.

  If I download the video to the Lambda's /tmp directory and add this downloaded file as an input to FFmpeg it works correctly (with the same parameters). However I'm trying to prevent this as the /tmp directory's max. size is only 512 MB which is not quite enough for me.

  The relevant code which returns SIGSEGV:

  ffmpeg_cmd = '/opt/bin/ffmpeg -stream_loop -1 -i "' + s3_source_signed_url + '" -i /opt/bin/audio.mp3 -i /opt/bin/watermark.png -shortest -y -deinterlace -vcodec libx264 -pix_fmt yuv420p -preset veryfast -r 30 -g 60 -b:v 4500k -c:a copy -map 0:v:0 -map 1:a:0 -filter_complex scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,overlay=(W-w)/2:(H-h)/2,format=yuv420p -loglevel verbose -f flv -'
  command1 = shlex.split(ffmpeg_cmd)
  p1 = subprocess.Popen(command1, stdout=subprocess.PIPE, stderr=subprocess.PIPE)
  stdout, stderr = p1.communicate()
  print(p1.returncode) #prints -11
  

  stderr of FFmpeg:

  ffmpeg version 4.1.3-static https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2019 the FFmpeg developers
    built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
    configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzvbi --enable-libzimg
    libavutil      56. 22.100 / 56. 22.100
    libavcodec     58. 35.100 / 58. 35.100
    libavformat    58. 20.100 / 58. 20.100
    libavdevice    58.  5.100 / 58.  5.100
    libavfilter     7. 40.101 /  7. 40.101
    libswscale      5.  3.100 /  5.  3.100
    libswresample   3.  3.100 /  3.  3.100
    libpostproc    55.  3.100 / 55.  3.100
  [tcp @ 0x728cc00] Starting connection attempt to 52.219.74.177 port 443
  [tcp @ 0x728cc00] Successfully connected to 52.219.74.177 port 443
  [h264 @ 0x729b780] Reinit context to 1280x720, pix_fmt: yuv420p
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'https://bucket.s3.amazonaws.com --> presigned url with 15 min expiration time':
    Metadata:
      major_brand     : mp42
      minor_version   : 0
      compatible_brands: mp42mp41isomavc1
      creation_time   : 2015-09-02T07:42:42.000000Z
    Duration: 00:00:15.64, start: 0.000000, bitrate: 2640 kb/s
      Stream #0:0(und): Video: h264 (High), 1 reference frame (avc1 / 0x31637661), yuv420p(tv, bt709, left), 1280x720 [SAR 1:1 DAR 16:9], 2475 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc (default)
      Metadata:
        creation_time   : 2015-09-02T07:42:42.000000Z
        handler_name    : L-SMASH Video Handler
        encoder         : AVC Coding
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 160 kb/s (default)
      Metadata:
        creation_time   : 2015-09-02T07:42:42.000000Z
        handler_name    : L-SMASH Audio Handler
  [mp3 @ 0x733f340] Skipping 0 bytes of junk at 1344.
  Input #1, mp3, from '/opt/bin/audio.mp3':
    Metadata:
      encoded_by      : Logic Pro X
      date            : 2021-01-03
      coding_history  : 
      time_reference  : 158760000
      umid            : 0x0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000004500F9E4
      encoder         : Lavf58.49.100
    Duration: 00:04:01.21, start: 0.025057, bitrate: 320 kb/s
      Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
      Metadata:
        encoder         : Lavc58.97
  Input #2, png_pipe, from '/opt/bin/watermark.png':
    Duration: N/A, bitrate: N/A
      Stream #2:0: Video: png, 1 reference frame, rgba(pc), 701x190 [SAR 1521:1521 DAR 701:190], 25 tbr, 25 tbn, 25 tbc
  [Parsed_scale_0 @ 0x7341140] w:1920 h:1080 flags:'bilinear' interl:0
  Stream mapping:
    Stream #0:0 (h264) -> scale
    Stream #2:0 (png) -> overlay:overlay
    format -> Stream #0:0 (libx264)
    Stream #1:0 -> #0:1 (copy)
  Press [q] to stop, [?] for help
  [h264 @ 0x72d8600] Reinit context to 1280x720, pix_fmt: yuv420p
  [Parsed_scale_0 @ 0x733c1c0] w:1920 h:1080 flags:'bilinear' interl:0
  [graph 0 input from stream 0:0 @ 0x7669200] w:1280 h:720 pixfmt:yuv420p tb:1/25 fr:25/1 sar:1/1 sws_param:flags=2
  [graph 0 input from stream 2:0 @ 0x766a980] w:701 h:190 pixfmt:rgba tb:1/25 fr:25/1 sar:1521/1521 sws_param:flags=2
  [auto_scaler_0 @ 0x7670240] w:iw h:ih flags:'bilinear' interl:0
  [deinterlace_in_2_0 @ 0x766b680] auto-inserting filter 'auto_scaler_0' between the filter 'graph 0 input from stream 2:0' and the filter 'deinterlace_in_2_0'
  [Parsed_scale_0 @ 0x733c1c0] w:1280 h:720 fmt:yuv420p sar:1/1 -> w:1920 h:1080 fmt:yuv420p sar:1/1 flags:0x2
  [Parsed_pad_1 @ 0x733ce00] w:1920 h:1080 -> w:1920 h:1080 x:0 y:0 color:0x000000FF
  [Parsed_setsar_2 @ 0x733da00] w:1920 h:1080 sar:1/1 dar:16/9 -> sar:1/1 dar:16/9
  [auto_scaler_0 @ 0x7670240] w:701 h:190 fmt:rgba sar:1521/1521 -> w:701 h:190 fmt:yuva420p sar:1/1 flags:0x2
  [Parsed_overlay_3 @ 0x733e440] main w:1920 h:1080 fmt:yuv420p overlay w:701 h:190 fmt:yuva420p
  [Parsed_overlay_3 @ 0x733e440] [framesync @ 0x733e5a8] Selected 1/50 time base
  [Parsed_overlay_3 @ 0x733e440] [framesync @ 0x733e5a8] Sync level 2
  [libx264 @ 0x72c1c00] using SAR=1/1
  [libx264 @ 0x72c1c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
  [libx264 @ 0x72c1c00] profile Progressive High, level 4.0, 4:2:0, 8-bit
  [libx264 @ 0x72c1c00] 264 - core 157 r2969 d4099dd - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=9 lookahead_threads=3 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=1 keyint=60 keyint_min=6 scenecut=40 intra_refresh=0 rc_lookahead=10 rc=abr mbtree=1 bitrate=4500 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  Output #0, flv, to 'pipe:':
    Metadata:
      major_brand     : mp42
      minor_version   : 0
      compatible_brands: mp42mp41isomavc1
      encoder         : Lavf58.20.100
      Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 4500 kb/s, 30 fps, 1k tbn, 30 tbc (default)
      Metadata:
        encoder         : Lavc58.35.100 libx264
      Side data:
        cpb: bitrate max/min/avg: 0/0/4500000 buffer size: 0 vbv_delay: -1
      Stream #0:1: Audio: mp3 ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp, 320 kb/s
      Metadata:
        encoder         : Lavc58.97
  frame=   27 fps=0.0 q=32.0 size=     247kB time=00:00:00.03 bitrate=59500.0kbits/s speed=0.0672x
  frame=   77 fps= 77 q=27.0 size=    1115kB time=00:00:02.03 bitrate=4478.0kbits/s speed=2.03x
  frame=  126 fps= 83 q=25.0 size=    2302kB time=00:00:04.00 bitrate=4712.4kbits/s speed=2.64x
  frame=  177 fps= 87 q=26.0 size=    3576kB time=00:00:06.03 bitrate=4854.4kbits/s speed=2.97x
  frame=  225 fps= 88 q=25.0 size=    4910kB time=00:00:07.96 bitrate=5047.8kbits/s speed=3.13x
  frame=  272 fps= 89 q=27.0 size=    6189kB time=00:00:09.84 bitrate=5147.9kbits/s speed=3.22x
  frame=  320 fps= 90 q=27.0 size=    7058kB time=00:00:11.78 bitrate=4907.5kbits/s speed=3.31x
  frame=  372 fps= 91 q=26.0 size=    8098kB time=00:00:13.84 bitrate=4791.0kbits/s speed=3.4x
  

  And that's the end of it. It should continue to do the processing until 00:04:02 as that's my audio's length but it stops here every time (approximately this is my video length).

  The relevant code which works correctly:

  ffmpeg_cmd = '/opt/bin/ffmpeg -stream_loop -1 -i "' + '/tmp/' + s3_source_key + '" -i /opt/bin/audio.mp3 -i /opt/bin/watermark.png -shortest -y -deinterlace -vcodec libx264 -pix_fmt yuv420p -preset veryfast -r 30 -g 60 -b:v 4500k -c:a copy -map 0:v:0 -map 1:a:0 -filter_complex scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,overlay=(W-w)/2:(H-h)/2,format=yuv420p -loglevel verbose -f flv -'
  command1 = shlex.split(ffmpeg_cmd)
  p1 = subprocess.Popen(command1, stdout=subprocess.PIPE, stderr=subprocess.PIPE)
  stdout, stderr = p1.communicate()
  print(p1.returncode) #prints 0
  

  With this code it repeats the video as many times as it has to do to be as long as the audio.

  Both versions work correctly on my computer.

  This question is almost the same but in my case FFmpeg is able to access the signed URL.

• How is better to use ffmpeg with the ASP.NET Core, on Linux ?

  26 mars, par user8245660

  I've got the project (ASP.NET Core, on Linux) where are the several tasks, which relate to the video converting and extracting frames from the video file.

  I thought about the two possible options:

  1. using ffmpeg console utility
  2. using P/Invoke with the libavcodec library and others, which ffmpeg uses

  The second option is miles harder and may be very impractical, because it reminds me developing the new wrapper/library, instead of using the ready products.

  I've done googling, but there aren't well-done projects for the C#/ASP.NET Core on Linux platform. There are some good for C++ and Python, but NOT for the C# on Linux/.NET Core.

  I decided to look at the first option, which I suppose would be more easier and practical. But, there are some weak places, which can produce many problems. We understand (I suppose), that using the 1st option, the end-developer shall use the process forking. So, there could be possible problems with the process idle and other possible issues...

  I'm asking about your practice, because it's my first experience on Linux platform with the video converting/sampling using C#. I've used the Expression Encoder .NET library on Windows platform, but it's other story and it makes no sense, right now.

  May be, there are other options, which I can't see right at the moment. I dislike the 1st option because of possible unhandled exceptions, because ffmpeg with such role becomes the black box for the ASP.NET Core backend.

• Remove audio tracks with FFmpeg [closed]

  25 mars, par David Hildreth

  I have some ProRes MOV files that have good audio on the first two tracks and then tone on the last 14. I'd love to remove the last 14 tracks, but I can't seem to do it.

  I've been trying variations on

  ffmpeg -i input_video.mov -map 0 -map -0:a:2 -map -0:a:3 -c copy output_video.mov
  

  but when I drag the resulting file into Premiere, the last 14 tracks are still included. How do I ditch those audio tracks?

• Broken Pipe Error On Moviepy When Receiving a SIGINT OR SIGTERM

  25 mars, par Pablo Estrada

  I'm trying to write a video with moviepy on a temp dir.

  Here's a sample code:

  video_file_name = '{}/{}'.format(tempfile.mkdtemp(), 'somefile.mp4')
  clip.write_videofile(video_file_name, audio = False, threads =4,logger = None)
  

  This works perfectly fine. However In my app context I want to be able to catch SIGTERM or SIGINT signals and wait until this write finishes before actually killing the app.

  I do it this way:

  signal.signal(signal.SIGINT, self.exit_gracefully)
  signal.signal(signal.SIGTERM, self.exit_gracefully)
  
  def exit_gracefully():
      # Some logic to wait for video processing to finish, for now we'll wait 1 hour to give enough time.
      time.sleep(3600)
  

  The problem is that when I do this, moviepy raises an exception like this:

  /lib/python3.7/site-packages/moviepy/video/io/ffmpeg_writer.py", line 136, in write_frame
      self.proc.stdin.write(img_array.tobytes())
  BrokenPipeError: [Errno 32] Broken pipe
  
  During handling of the above exception, another exception occurred:
  
        File "", line 2, in write_videofile
        File "lib/python3.7/site-packages/moviepy/decorators.py", line 54, in requires_duration
          return f(clip, *a, **k)
        File "", line 2, in write_videofile
        File "lib/python3.7/site-packages/moviepy/decorators.py", line 135, in use_clip_fps_by_default
          return f(clip, *new_a, **new_kw)
        File "", line 2, in write_videofile
        File "lib/python3.7/site-packages/moviepy/decorators.py", line 22, in convert_masks_to_RGB
          return f(clip, *a, **k)
        File "lib/python3.7/site-packages/moviepy/video/VideoClip.py", line 307, in write_videofile
          logger=logger)
        File "lib/python3.7/site-packages/moviepy/video/io/ffmpeg_writer.py", line 228, in ffmpeg_write_video
          writer.write_frame(frame)
        File "lib/python3.7/site-packages/moviepy/video/io/ffmpeg_writer.py", line 180, in write_frame
          raise IOError(error)
      OSError: [Errno 32] Broken pipe
      
      MoviePy error: FFMPEG encountered the following error while writing file /tmp/tmpt0o8ae6e/1633997015.176414_re_saved.mp4:
      
       b''
  

  Is there a way I can go around this error and allow my write file to finish? Another context is that this write_videofile() function is executed on a separate thread, I'm not sure if that's useful but it might help to give more insights.

  Any ideas on how I can solve this?

  Thanks!

• How to grab a single image from RTSP stream using FFMPEG

  25 mars, par sealfab

  I have seen several other related questions but they all seem to be related to grabbing a still shot every X number of seconds. How can I grab 1 image when the command is run.

  I was trying

  ffmpeg -y -i rtsp://admin:admin@192.168.10.113:554/live -f image2 -updatefirst 1 do.jpg