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• Reading file stream from Google Cloud Storage to ffmpeg (using fluent-ffmpeg)

  27 juillet 2018, par ekuusi

  I’m trying to run ffmpeg on a NodeJS backend with fluent-ffmpeg, reading input files from Google Cloud Storage. Everything works fine if I download the file first :

  const file = storage
  
      .bucket('example_bucket')
  
      .file('examplefile.mp4');
  
  
  
  file.download({destination: 'test.mp4'}, (err) => {
  
  
  
      let command = ffmpeg()
  
      .input('test.mp4')
  
      .duration(10)
  
      .format('mp4');
  
  
  
      command.save('test_out.mp4');
  
  
  
  });
  
  
  
  res.json([{
  
      message: 'Command sent!'
  
  }]);

  But if I try to use a readable stream as the input, it fails :

  const file = storage
  
      .bucket('example_bucket')
  
      .file('examplefile.mp4');
  
  
  
  
  
  var filestream = file.createReadStream()
  
  
  
  let command = ffmpeg()
  
      .input(filestream)
  
      .duration(10)
  
      .format('mp4');
  
  
  
  
  
  command.save('test_out.mp4');
  
  });
  
  
  
  res.json([{
  
      message: 'Command sent!'
  
  }]);

  Here is the full output of ffmpeg when trying to do the conversion. It seems to read the details of the file fine but for some reason it fails, saying "Cannot process video : ffmpeg exited with code 1 : pipe:0 : Invalid data found when processing input"

  Spawned Ffmpeg with command: ffmpeg -i pipe:0 -y -t 10 -f mp4 test_out.mp4
  
  Stderr output: ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
  
  Stderr output:   built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
  
  Stderr output:   configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable-libass --enable-libbluray --enable-lzma --enable-gnutls --enable-fontconfig --enable-libfreetype --enable-libfribidi --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --disable-indev=jack --enable-opencl --disable-outdev=xv --enable-audiotoolbox --enable-videotoolbox --enable-sdl2 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-x86asm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
  
  Stderr output:   libavutil      56. 14.100 / 56. 14.100
  
  Stderr output:   libavcodec     58. 18.100 / 58. 18.100
  
  Stderr output:   libavformat    58. 12.100 / 58. 12.100
  
  Stderr output:   libavdevice    58.  3.100 / 58.  3.100
  
  Stderr output:   libavfilter     7. 16.100 /  7. 16.100
  
  Stderr output:   libavresample   4.  0.  0 /  4.  0.  0
  
  Stderr output:   libswscale      5.  1.100 /  5.  1.100
  
  Stderr output:   libswresample   3.  1.100 /  3.  1.100
  
  Stderr output:   libpostproc    55.  1.100 / 55.  1.100
  
  Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] stream 2, offset 0x30: partial file
  
  Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1920x1080, 4647 kb/s): unspecified pixel format
  
  Stderr output: Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  Stderr output: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:0':
  
  Stderr output:   Metadata:
  
  Stderr output:     major_brand     : isom
  
  Stderr output:     minor_version   : 512
  
  Stderr output:     compatible_brands: isomiso2avc1mp41
  
  Stderr output:     encoder         : Lavf58.12.100
  
  Stderr output:     location-eng    : +60.2121+024.8754/
  
  Stderr output:     location        : +60.2121+024.8754/
  
  Stderr output:   Duration: 00:02:23.13, bitrate: N/A
  
  Stderr output:     Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), none, 1920x1080, 4647 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 120k tbc (default)
  
  Stderr output:     Metadata:
  
  Stderr output:       handler_name    : VideoHandler
  
  Stderr output:     Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
  
  Stderr output:     Metadata:
  
  Stderr output:       handler_name    : SoundHandler
  
  Stderr output:     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
  
  Stderr output:     Metadata:
  
  Stderr output:       handler_name    : TimeCodeHandler
  
  Stderr output: Stream mapping:
  
  Input is aac (mp4a / 0x6134706D) audio with h264 (avc1 / 0x31637661) video
  
  Stderr output:   Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  
  Stderr output:   Stream #0:1 -> #0:1 (aac (native) -> aac (native))
  
  Stderr output: [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb6db80a200] stream 0, offset 0x34: partial file
  
  Stderr output: pipe:0: Invalid data found when processing input
  
  Stderr output: Cannot determine format of input stream 0:0 after EOF
  
  Stderr output: Error marking filters as finished
  
  Stderr output: Conversion failed!
  
  Stderr output: 
  
  Cannot process video: ffmpeg exited with code 1: pipe:0: Invalid data found when processing input
  
  Cannot determine format of input stream 0:0 after EOF
  
  Error marking filters as finished
  
  Conversion failed!

• Does PTS have to start at 0 ?

  5 juillet 2018, par stevendesu

  I’ve seen a number of questions regarding video PTS values not starting at zero, or asking how to make them start at zero. I’m aware that using ffmpeg I can do something like ffmpeg -i <video> -vf="setpts=PTS-STARTPTS" <output></output></video> to fix this kind of thing

  However it’s my understanding that PTS values don’t have to start at zero. For instance, if you join a live stream then odds are it has been going on for an hour and the PTS is already somewhere around 3600000+ but your video player faithfully displays everything just fine. Therefore I would expect there to be no problem if I intentionally created a video with a PTS value starting at, say, the current wall clock time.

  I want to send a live stream using ffmpeg, but embed the current time into the stream. This can be used both for latency calculation while the stream is live, and later to determine when the stream was originally aired. From my understanding of PTS, something as simple as this should probably work :

  ffmpeg -i video.flv -vf="setpts=RTCTIME" rtmp://<output>
  
  </output>

  When I try this, however, ffmpeg outputs the following :

  frame=   93 fps= 20 q=-1.0 Lsize=    9434kB time=535020:39:58.70 bitrate=   0.0kbits/s speed=1.35e+11x

  Note the extremely large value for "time", the bitrate (0.0kbits), and the speed (135000000000x !!!)

  At first I thought the issue might be my timebase, so I tried the following :

  ffmpeg -i video.flv -vf="settb=1/1K,setpts=RTCTIME/1K" rtmp://<output>
  
  </output>

  This puts everything in terms of milliseconds (1 PTS = 1 ms) but I had the same issue (massive time, zero bitrate, and massive speed)

  Am I misunderstanding something about PTS ? Is it not allowed to start at non-zero values ? Or am I just doing something wrong ?

  Update

  After reviewing @Gyan’s answer, I formatted my command like so :

  ffmpeg -re -i video.flv -vf="settb=1/1K, setpts=(RTCTIME-RTCSTART)/1K" -output_ts_offset $(date +%s.%N) rtmp://<output>
  
  </output>

  This way the PTS values would match up to "milliseconds since the stream started" and would be offset by the start time of the stream (theoretically making PTS = timestamp on the server)

  This looked like it was encoding better :

  frame=  590 fps=7.2 q=22.0 size=   25330kB time=00:01:21.71 bitrate=2539.5kbits/s dup=0 drop=1350 speed=   1x 

  Bitrate was now correct, time was accurate, and speed was not outrageous. The frames per second was still a bit off, though (the source video is 24 fps but it’s reporting 7.2 frames per second)

  When I tried watching the stream from the other end, the video was out of sync with the audio and played at about double normal speed for a while, then the video froze and the audio continued without it

  Furthermore, when I dumped the stream to a file (ffmpeg -i rtmp://<output> dump.mp4</output>) and look at the PTS timestamps with ffprobe (ffprobe -show_entries packet=codec_type,pts dump.mp4 | grep "video" -B 1 -A 2) the timestamps didn’t seem to show server time at all :

  ...
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=131072
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=video
  
  pts=130048
  
  [/PACKET]
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=129536
  
  [/PACKET]
  
  [PACKET]
  
  codec_type=video
  
  pts=130560
  
  [/PACKET]
  
  --
  
  [PACKET]
  
  codec_type=video
  
  pts=131584
  
  [/PACKET]

  Is the problem just an incompatibility with RTMP ?

  Update 2

  I’ve removed the video filter and I’m now encoding like so :

  ffmpeg -re -i video.flv -output_ts_offset $(date +%s.%N) rtmp://<output>
  
  </output>

  This is encoding correctly :

  frame=  910 fps= 23 q=25.0 size=   12027kB time=00:00:38.97 bitrate=2528.2kbits/s speed=0.981x 

  In order to verify that the PTS values are correct, I’m dumping the output to a file like so :

  ffmpeg -i rtmp://<output> -copyts -write_tmcd 0 dump.mp4
  
  </output>

  I tried saving it as dump.flv (since it’s RTMP) however this threw the error :

  [flv @ 0x5600f24b4620] Audio codec mp3 not compatible with flv

  This is a bit weird since the video isn’t mp3-encoded (it’s speex) - but whatever.

  While dumping this file the following error pops up repeatedly :

  frame=    1 fps=0.0 q=0.0 size=       0kB time=00:00:09.21 bitrate=   0.0kbits/s dup=0 dr
  
  43090023 frame duplication too large, skipping
  
  43090027 frame duplication too large, skipping
  
      Last message repeated 3 times
  
  43090031 frame duplication too large, skipping
  
      Last message repeated 3 times
  
  43090035 frame duplication too large, skipping

  Playing the resulting video in VLC plays an audio stream but displays no video. I then attempt to probe this video with ffprobe to look at the video PTS values :

  ffprobe -show_entries packet=codec_type,pts dump.mp4 | grep "video" -B 1 -A 2

  This returns only a single video frame whose PTS is not large like I would expect :

  [PACKET]
  
  codec_type=video
  
  pts=1020
  
  [/PACKET]

  This has been a surprisingly difficult task

• HLS : How to detect out of order segments in media playlist ?

  27 juin 2018, par anirudh612

  What would be an efficient way to detect if an http live streaming VOD playlist has segments out of order (and count how many segments are out of order) ? They are ordered correctly based on the #EXT-X-PROGRAM-DATETIME tag but the segment decoding timestamps in some cases are out of order. Currently, the workflow I’m following is :

  1. Convert the HLS stream into an mp4 using ffmpeg :

     ffmpeg -i http://localhost:8080/test/unsorted.m3u8 -c copy -bsf:a aac_adtstoasc test/unsorted.mp4 &> test/unsorted_ffmpeg.log

  2. Inspect the logs and count number of occurrences of "Non-monotonous DTS in output stream" log messages :

     [mp4 @ 0x7fe74f01b000] Non-monotonous DTS in output stream 0:1 ; previous : 12063760, current : 11866128 ; changing to 12063761. This may result in incorrect timestamps in the output file.

     However, this requires downloading and reading all of the ts segments and is an expensive operation. Is there a more efficient way to determine out of order DTS or PTS in chunks using ffmpeg or ffprobe ?