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• FFmpeg make mpeg2 ts without discontinuity

  31 mai 2012, par user1427162

  I have many MOV files recorded with iPhone and I want to convert them to mpeg2 TS. I want to use them for live video streaming with HTTP Live Streaming protocol.

  I set my iPhone to continiously send MOV files to server. Every video clip is 5 seconds long. I want to make mpeg2 TS out of them and add their urls to m3u8 playlist.
  I managed to do all of that, but when I try to play the stream VLC player plays only first two files in playlist, and last file in playlist at that moment.
  I searched the internet and I think this has something to do with discontinuity.

  Is there any way to convert multiple MOV files into multiple mpeg2 TS segments without discontinuity ?

  Or maybe I'm doing something else wrong ?
  Here is my ffmpeg command :

  ffmpeg.exe -i input,MOV -f mpegts output.ts

  and here is my m3u8 list :

  #EXTM3U
  
  #EXT-X-PLAYLIST-TYPE:EVENT
  
  #EXT-X-TARGETDURATION:10
  
  #EXT-X-MEDIA-SEQUENCE:0
  
  #EXT-X-MEDIA-SEQUENCE:0
  
  #EXTINF:10,
  
  fileSequence0.ts
  
  #EXTINF:10,
  
  fileSequence1.ts
  
  #EXTINF:10,
  
  fileSequence2.ts

  Thanks in advance

• Maintaining exact aspect ratio when scaling videos using ffmpeg

  31 mai 2012, par Skkard

  I have a mkv video, which is a mix of multiple resolution recordings, e.g. I have the first few seconds of widescreen 16:9 (1024x576) resolution, and the rest of the video if 4:3 (768x576) resolution. I want to scale this video down 3 times, while copying all the other attributes (audio codec, subtitles etc.). I use ffmpeg -i  -vf scale=iw/2:-1 -acodec copy . Also, VLC detects it's resolution as 720x576.

  The problem is that after the scaling, the resolution constantly becomes 4:3 (360x288). How can I maintain the dynamic aspect ratio of the input video file i.e. the 16:9 parts to scale to 16:9, while the 4:3 parts scale to 4:3 ?

  update

  The player size actually changes, atleast in mplayer, when the resolution is switched. I figured out the main problem. It seems each frame is tagged with a Sample Aspect Ratio (SAR), so when the player plays it, it can find the display aspect ratio. This SAR value isn't getting copied over when encoding to MKV. When encoding to MPG, it does get copied over and I get an exact copy, with the player switching sizes, but not with MKV.

  Output of ffprobe -show_streams filename :

  ffprobe version 0.10.3 Copyright (c) 2007-2012 the FFmpeg developers
  
    built on May  9 2012 17:51:07 with gcc 4.7.0 20120505 (prerelease)
  
    configuration: --prefix=/usr --enable-libmp3lame --enable-libvorbis --enable-libxvid --enable-libx264 --enable-libvpx --enable-libtheora --enable-libgsm --enable-libspeex --enable-postproc --enable-shared --enable-x11grab --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libschroedinger --enable-libopenjpeg --enable-librtmp --enable-libpulse --enable-gpl --enable-version3 --enable-runtime-cpudetect --disable-debug --disable-static
  
    libavutil      51. 35.100 / 51. 35.100
  
    libavcodec     53. 61.100 / 53. 61.100
  
    libavformat    53. 32.100 / 53. 32.100
  
    libavdevice    53.  4.100 / 53.  4.100
  
    libavfilter     2. 61.100 /  2. 61.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0.  6.100 /  0.  6.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  Input #0, matroska,webm, from 'sample.mkv':
  
    Metadata:
  
      title           : Pan prstenu. Dve veze
  
    Duration: 00:00:29.80, start: 0.000000, bitrate: 3124 kb/s
  
      Stream #0:0(eng): Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 15000 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc (default)
  
      Stream #0:1(cze): Audio: mp2, 48000 Hz, stereo, s16, 128 kb/s (default)
  
  [STREAM]
  
  index=0
  
  codec_name=mpeg2video
  
  codec_long_name=MPEG-2 video
  
  codec_type=video
  
  codec_time_base=1/50
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  width=720
  
  height=576
  
  has_b_frames=1
  
  sample_aspect_ratio=64:45
  
  display_aspect_ratio=16:9
  
  pix_fmt=yuv420p
  
  level=8
  
  timecode=16:35:19:10
  
  id=N/A
  
  r_frame_rate=25/1
  
  avg_frame_rate=25/1
  
  time_base=1/1000
  
  start_time=0.000000
  
  duration=N/A
  
  nb_frames=N/A
  
  TAG:language=eng
  
  [/STREAM]
  
  [STREAM]
  
  index=1
  
  codec_name=mp2
  
  codec_long_name=MP2 (MPEG audio layer 2)
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=s16
  
  sample_rate=48000
  
  channels=2
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=125/3
  
  time_base=1/1000
  
  start_time=0.000000
  
  duration=N/A
  
  nb_frames=N/A
  
  TAG:language=cze
  
  [/STREAM]

• Get PTS from raw H264 mdat generated by iOS AVAssetWriter

  26 décembre 2012, par kolyuchiy

  I'm trying to simultaneously read and write H.264 mov file written by AVAssetWriter. I managed to extract individual NAL units, pack them into ffmpeg's AVPackets and write them into another video format using ffmpeg. It works and the resulting file plays well except the playback speed is not right. How do I calculate the correct PTS/DTS values from raw H.264 data ? Or maybe there exists some other way to get them ?

  Here's what I've tried :

  1. Limit capture min/max frame rate to 30 and assume that the output file will be 30 fps. In fact its fps is always less than values that I set. And also, I think the fps is not constant from packet to packet.

  2. Remember each written sample's presentation timestamp and assume that samples map one-to-one to NALUs and apply saved timestamp to output packet. This doesn't work.

  3. Setting PTS to 0 or AV_NOPTS_VALUE. Doesn't work.

  From googling about it I understand that raw H.264 data usually doesn't contain any timing info. It can sometimes have some timing info inside SEI, but the files that I use don't have it. On the other hand, there are some applications that do exactly what I'm trying to do, so I suppose it is possible somehow.