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• FFmpeg for marking time video based on a reference date

  3 août 2016, par Denio Mariz

  I am trying to mark a timestamp in a video using Drawtext filter.
  FFmpeg easily marks timestamps based on localtime, gmtime or even PTS. However, I want to assign a reference time (start time) for the timestamp in order to represent the time the video was recorded (not encoded).

  Reading the documentation, I found that option "basetime" can be used for this purpose. However it seems that is not working or I am missing something.

  The command line I am using is :

  ffmpeg -y -i input.mp4 -filter_complex drawtext="fontfile=/tmp/UbuntuMono-B.ttf: fontsize=36: fontcolor=yellow: box=1: boxcolor=black@0.4: text='Wall Clock Time\: %{gmtime\:%Y-%m-%d %T}': basetime=1456007118" output.mp4

  By using basetime=1456007118", it was expected the start time was set to ’02/20/2016 20:25:18’ since 1456007118 is the UTC time for that time and date :

  date -d '02/20/2016 20:25:18' +"%s" # format MM/DD/AAAA hh:mm:ss
  
  1456007118

  However, no error is issued by FFMpeg and the video is marked with current GMT, ignoring "basetime" option.

  Any hint ?
  Thanks.

  Complete information about FFmpeg version and output is :

     ffmpeg -y -i /home/denio/Videos/Interstellar_2014_Trailer_4_5.1-1080p-HDTN.mp4 -filter_complex drawtext="fontfile=/tmp/UbuntuMono-B.ttf: fontsize=36: fontcolor=yellow: box=1: boxcolor=black@0.4: text='Wall Clock Time\: %{gmtime\:%Y-%m-%d %T}': basetime=1470226363" /tmp/x.mp4
   ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 5.3.1 (Ubuntu 5.3.1-14ubuntu2.1) 20160413
     configuration: --enable-libxavs --enable-bzlib --enable-libfaac --enable-libfreetype --enable-libfontconfig --enable-libmp3lame --enable-libschroedinger --enable-libspeex --enable-libvorbis --enable-libx264 --enable-libx265 --enable-libxvid --enable-zlib --enable-x11grab --enable-static --enable-pthreads --enable-gpl --enable-nonfree --enable-version3 --disable-ffserver --enable-libgsm --enable-librtmp --enable-libvpx --enable-libschroedinger --enable-libopencore-amrnb --enable-libopenjpeg
     libavutil      55. 28.100 / 55. 28.100
     libavcodec     57. 48.101 / 57. 48.101
     libavformat    57. 41.100 / 57. 41.100
     libavdevice    57.  0.101 / 57.  0.101
     libavfilter     6. 47.100 /  6. 47.100
     libswscale      4.  1.100 /  4.  1.100
     libswresample   2.  1.100 /  2.  1.100
     libpostproc    54.  0.100 / 54.  0.100
   ...
   ...

• Manipulate video with ffmpeg without losing quality

  17 août 2016, par TheChymera

  I am rotating a video with ffmpeg with the following code :

  ffmpeg -i nd750_a0040.MOV -vf "transpose=dir=clock, transpose=dir=clock" out.mkv

  The resulting file is almost 10x smaller than the input.
  I have found this question which addresses a related question and suggsts passing to ffmpeg the codec and bitrate obtained from the following :

  bitratev="$(ffmpeg -i "$1" -f null - |& grep video: | awk -F'[:|kB]' '{print $2}')"
  
  codecv="$(ffprobe -loglevel error -select_streams v:0 -show_entries stream=codec_name -of default=nk=1:nw=1 "$1")"

  however, both of these commands give me the same output for both files : 2643 and h264 respectively.
  Am I correct in assuming that ffmpeg keeps these values the same for the output - by default ?

  However, if I inspect the files with ffmpeg -i I get different bitrate values :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'nd750_a0040.MOV':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537331968
  
      compatible_brands: qt  niko
  
      creation_time   : 2016-06-18 04:28:03
  
    Duration: 00:15:40.74, start: 0.000000, bitrate: 11569 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, smpte170m/bt709/bt470m), 1920x1080, 10029 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
  
      Metadata:
  
        creation_time   : 2016-06-18 04:28:03
  
      Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 2 channels, s16, 1536 kb/s (default)
  
      Metadata:
  
        creation_time   : 2016-06-18 04:28:03

  and

  Input #0, matroska,webm, from 'out.mkv':
  
    Metadata:
  
      COMPATIBLE_BRANDS: qt  niko
  
      MAJOR_BRAND     : qt  
  
      MINOR_VERSION   : 537331968
  
      ENCODER         : Lavf56.40.101
  
    Duration: 00:15:40.74, start: 0.000000, bitrate: 1445 kb/s
  
      Stream #0:0(eng): Video: h264 (High), yuvj420p(pc), 1920x1080, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc (default)
  
      Metadata:
  
        CREATION_TIME   : 2016-06-18 04:28:03
  
        LANGUAGE        : eng
  
        ENCODER         : Lavc56.60.100 libx264
  
        DURATION        : 00:15:40.742000000
  
      Stream #0:1(eng): Audio: vorbis, 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        CREATION_TIME   : 2016-06-18 04:28:03
  
        LANGUAGE        : eng
  
        ENCODER         : Lavc56.60.100 libvorbis
  
        DURATION        : 00:15:40.743000000

  So I have a few questions :

  • Which bitrate is the correct one for each video ?
  • Is all of the information loss between these two files covered by the bitrate (or does ffmpeg by default change other things that lead to a lower file size as well - if so, what ?) ?
  • How do I make sure nothing else changes but the container format and the rotation ?

• How improves Video Player processing using Qt and FFmpeg ?

  13 septembre 2016, par Eric Menezes

  A time ago, I started to develop a video player/analyser. For beeing an analyser as well, the application should have inside its buffer the next frames and the previous as well. That’s where the complication begins.

  For that, we started to use an VideoProducer that decodes the frames and audio from video (using ffmpeg), added it into a buffer from where the video and audio consumers retrieve that objects (VideoFrame and AudioChunk). For this job, we have some QThreads which is one producer, 2 consumers and (the biggest trouble maker) 2 workers that is used to retrieve objects from producer’s buffer and insert them into a circular buffer (that because of previous frames). These workers are important because of the backwards buffering job (this player should play backwards too).

  So, now the player is running well, but not so good. It’s notable that is losing performance. Like removing producer buffer and using just the circular. But still, some questions remains :

  • Should I continue using QThread with reimplemented run() ? I read that works better with Signals & Slots ;

  • If Signals & Slots worth it, the producer still needs to reimplement QThread::run(), right ?

  • Cosidering that buffer must have some previous frames and bad quality videos will be reproduced, is that (VideoProducer insert objects into a Buffer, AudioConsumer and FrameConsumer retrieve these objects from Buffer and display/reproducer them) the better way ?

  • What is the best way to sync audio and video ? The sync using audio pts is doing well, but some troubles appear sometimes ; and

  • For buffering backwards, ffmpeg does not provide frames this way, so I need to seek back, decode older frames, reorder them and prepend to the buffer. This job has been done by that Workers, another QThread the keep consuming from Producer buffer and, if buffering backwards, asks for seek and do the reorder job. I can just guess that it is bad. And I assume that do this reorder job should be done at Producer level. Is that any way to do this better ?

  I know it’s a lot of questions, and I’m sorry for that, but I don’t know where to find these answers.

  Thanks for helping.

  For better understanding, heres how its been done :

  • VideoProducer -> Decoder QThread. Runs in a loop decoding and enqueuing frames into a Buffer.

  • FrameConsumer -> Video consumer. Retrieves frames from frame CircularBuffer in a loop using another QThread. Display the frame and sleep few mseconds based on video fps and AudioConsumer clock time.

  • AudioConsumer -> Audio consumer and video’s clock. Works with signals using QAudioOutput::notify() to retrieve chunks of audio from audio CircularBuffer and insert them into QAudioOutput buffer. When decodes the first frame, its pts is used to start the clock. (if a seek has been called, the next audio frame will mark the clock start time)

  • Worker -> Each stream (audio and video) has one. It’s a QThread running in a loop (run() reimplemented) retrieving objects from Buffer and inserting (backwards or forward) to CircularBuffer.

  And another ones that manage UI, filters, some operations with frames/chunks...