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• Capturing audio and video from different sources, how to sync ?

  16 février 2017, par aerodavo

  Here is my code :

  Lapaki:~ Lapaki$ /Users/Lapaki/Desktop/ffmpeg -f avfoundation -video_size 960x540 -pixel_format uyvy422 -framerate ntsc -thread_queue_size 8B -i "XI:none" -f avfoundation -thread_queue_size 8B -i "none:XI" -vf 'crop=iw-240:ih:120:0' -af 'asetpts=PTS+.58735/TB' -pix_fmt yuv420p -aspect 4:3 -s 720x480 -q:v 3 -maxrate 5000k -bufsize 2000k -acodec ac3 -ac 2 -ab 256k -ar 48000 -f dvd /Users/Lapaki/Desktop/FF\ Test/`date +%F`\ `date +%H_%M_%S`.mpg
  
  ffmpeg version 3.2.3-tessus Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  
    configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  Input #0, avfoundation, from 'XI:none':
  
    Duration: N/A, start: 648413.295900, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 960x540, 29.97 fps, 29.97 tbr, 1000k tbn, 1000k tbc
  
  Input #1, avfoundation, from 'none:XI':
  
    Duration: N/A, start: 648413.884042, bitrate: 3072 kb/s
  
      Stream #1:0: Audio: pcm_f32le, 48000 Hz, stereo, flt, 3072 kb/s
  
  Output #0, dvd, to '/Users/Lapaki/Desktop/FF Test/2017-02-16 04_16_33.mpg':
  
    Metadata:
  
      encoder         : Lavf57.56.101
  
      Stream #0:0: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 8:9 DAR 4:3], q=2-31, 200 kb/s, 29.97 fps, 90k tbn, 29.97 tbc
  
      Metadata:
  
        encoder         : Lavc57.64.101 mpeg2video
  
      Side data:
  
        cpb: bitrate max/min/avg: 5000000/0/200000 buffer size: 2000000 vbv_delay: -1
  
      Stream #0:1: Audio: ac3, 48000 Hz, stereo, fltp, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc57.64.101 ac3
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg2video (native))
  
    Stream #1:0 -> #0:1 (pcm_f32le (native) -> ac3 (native))
  
  Press [q] to stop, [?] for help
  
  [swscaler @ 0x7f8e0c8ab400] Warning: data is not aligned! This can lead to a speedloss
  
  frame=   33 fps=0.0 q=3.0 size=     266kB time=00:00:01.06 bitrate=2051.9kbits/sframe=   49 fps= 48 q=3.0 size=     444kB time=00:00:01.54 bitrate=2358.8kbits/sframe=   64 fps= 42 q=3.0 size=     652kB time=00:00:02.08 bitrate=2560.5kbits/sframe=   79 fps= 39 q=3.0 size=     838kB time=00:00:02.59 bitrate=2642.4kbits/sframe=   94 fps= 37 q=3.0 size=    1022kB time=00:00:03.07 bitrate=2720.0kbits/sframe=  109 fps= 36 q=3.0 size=    1208kB time=00:00:03.59 bitrate=2756.5kbits/sframe=  124 fps= 35 q=3.0 size=    1406kB time=00:00:04.07 bitrate=2830.0kbits/sframe=  127 fps= 35 q=3.0 Lsize=    1474kB time=00:00:04.19 bitrate=2876.4kbits/s dup=12 drop=0 speed=1.15x    
  
  video:1310kB audio:113kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3.604597%

  The start of Input #0 is 648413.295900, and the start of Input #1 is 648413.884042.

  I’ve been able to keep the audio and video in very close sync by subtracting the two values (which I assume are wallclock values), and using the asetpts audio filter to delay the audio stream of the recorded mpeg-2 file by that amount.

  I’d like to be able to do this exactly though, and that value changes slightly every time I start a new capture. Not to mention, I’d like to be able to do this reliably on different machines, where I assume the value will most likely be different, thus using a calculation as opposed to a fixed number is obviously the best option, if it’s possible.

  Is there a way to subtract the wallclock start time of input #0 from the wallclock start time of input #1 ? I’d like to do this inside the asetpts filter, instead of manually finding the difference from a previous run, which again is slightly different every time...

  I was thinking something like -af asetpts=PTS-([1:0]RTCSTART-[0:0]RTCSTART)/TB might work, but I have no idea how to format it.

  Thanks in advance !

• avformat/hlsenc:addition of #EXT-X-MEDIA tag and AUDIO attribute

  23 décembre 2017, par Vishwanath Dixit

  avformat/hlsenc:addition of #EXT-X-MEDIA tag and AUDIO attribute
  Signed-off-by : Steven Liu <lq@chinaffmpeg.org>
  

  • [DH] doc/muxers.texi
  • [DH] libavformat/dashenc.c
  • [DH] libavformat/hlsenc.c
  • [DH] libavformat/hlsplaylist.c
  • [DH] libavformat/hlsplaylist.h

• Media contaner which supports uncompressed audio and video

  21 juin 2017, par Denis Kokorin

  I want to produce uncompressed video and audio programmatically and feed it via stdin to ffmpeg.
  I need uncompressed, because I don’t want to encode video and audio in my code.

  I found here the way to create video only file from uncompressed rgb24 input stream. But I do need to create both video and audio.

  Which container supported by ffmpeg can contain uncompressed video and audio ?