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• Is it possible to merge 3 videos with ffmpeg by -f concat and crossfade without video content

  10 août 2019, par user3792705

  I want to merge 3 .mov files quickly without losing any resolution. I want to be able to distinguish the 3 pieces of videos after merge.

  "ffmpeg -f concat" does not lose resolution and quick without crossfade.

  But, I can’t distinguish 3 videos.

  As far as I know ffmpeg filter can be used add crossfade, but it have to use video start/end content to do the merger, which might involve transcoding. It won’t be fast compared with ’concat’, which won’t do transcoding, but simply copying.

  Here is the content (ffmpeg -i video.mov) of one of 3 videos :

  ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with Apple LLVM version 10.0.1 (clang-1001.0.46.3)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.3 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/openjdk-12.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/openjdk-12.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  
    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
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '..../(edited)/VMEK8375.MOV':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2019-06-30T01:28:04.000000Z
  
      com.apple.quicktime.model: iPhone
  
      com.apple.quicktime.software: ZHIYUN
  
      com.apple.quicktime.creationdate: 2019-06-30T09:28:04Z
  
    Duration: 00:00:07.61, start: 0.000000, bitrate: 4386 kb/s
  
      Stream #0:0(und): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv, smpte170m/bt709/bt709), 1280x720, 4329 kb/s, 30.01 fps, 30 tbr, 600 tbn, 600 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-06-30T01:28:04.000000Z
  
        handler_name    : Core Media Video
  
        encoder         : HEVC
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 94 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-06-30T01:28:04.000000Z
  
        handler_name    : Core Media Audio

  If I don’t care about crossfade with video content, just some ’nice’ black screen in between (It would be nice I add some text, like date + time on the black screen) is good enough for me. Is it possible to do ’concat’ and simple crossfade without video ’content’ ?

• H.264 conversion with FFmpeg (from a RTP stream)

  12 juillet 2014, par Toby

  Environment :

  I have an IP Camera, which is capable of streaming it’s data over RTP in a H.264 encoded format. This raw stream is recorded from the ethernet. With that data I have to work.

  Goal :

  In the end I want to have a *.mp4 file, which I can play with common Media Players (like VLC or Windows MP).

  What have I done so far :

  I take that raw stream data I have and parse it. Since the data has been transmitted via RTP I need to take care of the NAL Bytes, SPS and PPS.

  1. Write a raw file

  First I determine the type of each frame received over Ethernet. To do so, I parse the first two bytes of every RTP Payload, so I can get the 8 NAL Unit Bit, the Fragment Type Bits and the Start, Reserved and End Bit. In the payload, they’re arranged like this :

  Byte 1: [          3 NAL Unit Bits          | 5 Fragment Type Bits]
  
  Byte 2: [Start Bit | Reserved Bit | End Bit | 5 NAL Unit Bits]

  From this I can determine :

  • Start and End of
    a Video Frame -> Start Bit and End Bit
  • Type of the Payload -> 5 Fragment Type Bits
  • NAL Unit Byte

  The Fragment types which are necessary in my case are :

  Fragment Type  7 = SPS
  
  Fragment Type  8 = PPS
  
  Fragment Type 28 = Video Fragment

  The NAL Byte is created by putting the NAL Unit Bits from Byte 1 and 2 together.

  Now depending on the fragmentation type I do the following :

  SPS/PPS :

  1. Write the NAL Prefix (0x00 0x00 0x01) and then the SPS or PPS data

  Fragmentation with Start Bit

  1. Write NAL Prefix
  2. Write NAL Unit Byte
  3. Write remaining raw data

  Fragmentation without Start Bit

  1. Write raw data

  This means my raw file looks something like this :

  [NAL Prefix][SPS][NAL Prefix][PPS][NAL Prefix][NAL Unit Byte][Raw Video Data][Raw Video Data]....[NAL Prefix][NAL Unit Byte][Raw Video Data]...

  For every PPS and SPS I find in my stream data, I just write a NAL Prefix ( 0x00 0x00 0x01 ) and then the SPS/PPS itself.

  Now I can’t play this data with some media player, which leads me to :

  2. Convert the file

  Since I wanted to avoid working much with codecs I just went to use an existing application -> FFmpeg. This I am calling with those parameters :

  ffmpeg.exe -f h264 -i <rawinputfile> -vcodec copy -r 25 <outputfilename>.mp4</outputfilename></rawinputfile>

  -f h264 : This should tell ffmpeg I have a h264 coded stream

  -vcodec copy : Quote from the manpage :

  Force video codec to codec. Use the "copy" special value to tell that the raw codec data must be copied as is.

  -r 25 : Sets the framerate to 25 FPS.

  When I call ffmpeg with those parameters I get an .mp4 File, which I can play with VLC and Windows MP, so it actually works. But the file now looks a bit different from my raw file.

  This leads me to my question :

  What did I actually do ?

  My problem is not that it is not working. I just want/need to know what I have actually done with calling ffmpeg. I had a raw H264 file which I could not play. After using FFmpeg I can play it.

  There are the following differences between the original raw file (which I have written) and the one written by FFmpeg :

  1. Header : The FFmpeg File has like about 0x30 Bytes of Header
  2. Footer : The FFmpeg File also has a footer
  3. Changed Prefix and 2 new Bytes :

  While a new Video Frame from the Raw File started like
  [NAL Prefix][NAL Unit Byte][Raw Video Data] in the new file it looks like this :

  [0x00 0x00][2 "Random" Bytes][NAL Unit Byte][Raw Video Data].....[0x00 0x00[2 other "Random" Bytes][NAL Unit Byte][Raw Video Data]...

  I understand that the Video Stream needs a container format (correct me if I am wrong but I assume that the new header and footer are responsible for that). But why does it change actually some Bytes in the raw data ? It can’t be some decoding since the stream itself should get decoded by the player and not ffmpeg.

  As you can see I don’t need a new solution for my problem as far more an explanation (so I can explain it by myself). What does ffmpeg actually do ? And why does it change some bytes within the video data ?

• avcodec/flac_parser : Fix off-by-one error

  6 octobre 2019, par Andreas Rheinhardt

  avcodec/flac_parser : Fix off-by-one error
  The flac parser uses a fifo to buffer its data. Consequently, when
  
  searching for sync codes of flac packets, one needs to take care of
  
  the possibility of wraparound. This is done by using an optimized start
  
  code search that works on each of the continuous buffers separately and
  
  by explicitly checking whether the last pre-wrap byte and the first
  
  post-wrap byte constitute a valid sync code.
  Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched
  
  for (the start of) a sync code because a header that might be found in this
  
  region might not be completely available. These bytes ought to be searched
  
  lateron when more data is available or when flushing.
  Unfortunately there was an off-by-one error in the calculation of the
  
  length to search of the post-wrap buffer : It was too large, because the
  
  calculation was based on the amount of bytes available in the fifo from
  
  the last pre-wrap byte onwards. This meant that a header might be
  
  parsed twice (once prematurely and once regularly when more data is
  
  available) ; it could also mean that an invalid header will be treated as
  
  valid (namely if the length of said invalid header is
  
  MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the
  
  last byte of this potential header happens to be the right CRC-8).
  Should a header be parsed twice, the second instance will be the best child
  
  of the first instance ; the first instance's score will be
  
  FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than
  
  the second instance's score. So the frame belonging to the first
  
  instance will be output and it will be done as a zero length frame (the
  
  difference of the header's offset and the child's offset). This has
  
  serious consequences when flushing, as returning a zero length buffer
  
  signals to the caller that no more data will be output ; consequently the
  
  last frames not yet output will be dropped.
  Furthermore, a "sample/frame number mismatch in adjacent frames" warning
  
  got output when returning the zero-length frame belonging to the first
  
  header, because the child's sample/frame number of course didn't match
  
  the expected sample frame/number given its parent.
  filter/hdcd-mix.flac from the FATE-suite was affected by this (the last
  
  frame was omitted) which is the reason why several FATE-tests needed to
  
  be updated.
  Fixes ticket #5937.
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/flac_parser.c
  • [DH] tests/fate/filter-audio.mak