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• ffmpeg cut video at specified start position

  15 septembre 2014, par user2648561

  I’m trying to use ffmpeg to cut out part of an avi file.
  I know similar questions have been asked before, but i’ve trawled through loads of other posts and I can’t see where I’m going wrong.

  The command line args i’m using to test on a 7 second video is :

  -ss 00:00:01 -t 00:00:01 -i "C :\recordings\2013-09-11\Camera 1\Camera 1_2013-09-11_22-23-43.avi" -vcodec copy -acodec copy "I :\clip2.avi"

  The output is :

  ffmpeg version N-56254-gb7bd688 Copyright (c) 2000-2013 the FFmpeg developers built on Sep 12 2013 21:00:14 with gcc 4.7.3 (GCC) configuration :
  —enable-gpl —enable-version3 —disable-w32threads —enable-avisynth —enable-bzlib —enable-fontconfig —enable-frei0r —enable-gnutls —enable-iconv —enable-libass —enable-libbluray —enable-libcaca —enable-libfreetype —enable-libgsm —enable-libilbc —enable-libmodplug —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-libopus —enable-librtmp —enable-libschroedinger —enable-libsoxr —enable-libspeex —enable-libtheora —enable-libtwolame —enable-libvidstab —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-libvorbis —enable-libvpx —enable-libx264 —enable-libxavs —enable-libxvid —enable-zlib
  libavutil 52. 43.100 / 52. 43.100
  libavcodec 55. 31.101 / 55. 31.101
  libavformat 55. 16.102 / 55. 16.102
  libavdevice 55. 3.100 / 55. 3.100
  libavfilter 3. 84.100 / 3. 84.100
  libswscale 2. 5.100 / 2. 5.100
  libswresample 0. 17.103 / 0. 17.103
  libpostproc 52. 3.100 / 52. 3.100

  Guessed Channel Layout for  Input Stream #0.1 : mono
  
  
  
  Input #0, avi, from 'C:\recordings\2013-09-11\Camera 1\Camera 1_2013-09-11_22-23-43.avi':
  
    Duration: 00:00:07.03, start: 0.000000, bitrate: 787 kb/s
  
      Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 800x448, 30 fps, 30 tbr, 30 tbn, 30 tbc
  
      Stream #0:1: Audio: adpcm_ms ([2][0][0][0] / 0x0002), 44100 Hz, mono, s16, 176 kb/s
  
  Output #0, avi, to 'I:\clip2.avi':
  
    Metadata:
  
      ISFT            : Lavf55.16.102
  
      Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 800x448, q=2-31, 30 fps, 30 tbn, 30 tbc
  
      Stream #0:1: Audio: adpcm_ms ([2][0][0][0] / 0x0002), 44100 Hz, mono, 176 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:1 -> #0:1 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=   88 fps=0.0 q=-1.0 Lsize=     226kB time=00:00:02.03 bitrate= 908.7kbits/s    
  
  
  
  video:146kB audio:66kB subtitle:0 global headers:0kB muxing overhead 6.345259%

  I end up with a clip which is 3 seconds long !!!??
  In the output I can see that the -ss flag is completely ignored.

  Can anyone see where I’m going wrong ?

  Thanks

• Is there a chance to check why a videostream on IOS does not work ?

  13 septembre 2013, par mazleu

  I have a very strange problem.

  To stream videos I use AMS (Adobe Media Server)
  This works fine for the most Videos.
  But 4 videos do not work on iOS devices. The first seconds work but then the Video break.
  All Videos (over 1000) are convertet with the same ffmpeg settings.

  I try other settings and other converters but the Result are the same.

  Is there any opportunity to show why IOS don't like this 4 Videos ?

  Mediainfo form a failed video :

  MediaInfoLib - v0.7.62
  
  
  
  General
  
  Complete name                       : ---
  
  Format                              : MPEG-4
  
  Format profile                      : Base Media
  
  Codec ID                            : isom
  
  File size                           : 91.3 MiB
  
  Duration                            : 24mn 27s
  
  Overall bit rate mode               : Variable
  
  Overall bit rate                    : 522 Kbps
  
  Encoded date                        : UTC 2013-09-12 14:24:13
  
  Tagged date                         : UTC 2013-09-12 14:24:13
  
  
  
  Video
  
  ID                                  : 1
  
  Format                              : AVC
  
  Format/Info                         : Advanced Video Codec
  
  Format profile                      : High@L4.1
  
  Format settings, CABAC              : Yes
  
  Format settings, ReFrames           : 4 frames
  
  Codec ID                            : avc1
  
  Codec ID/Info                       : Advanced Video Coding
  
  Duration                            : 24mn 27s
  
  Bit rate                            : 387 Kbps
  
  Maximum bit rate                    : 4 885 Kbps
  
  Width                               : 1 280 pixels
  
  Height                              : 720 pixels
  
  Display aspect ratio                : 16:9
  
  Frame rate mode                     : Constant
  
  Frame rate                          : 30.000 fps
  
  Color space                         : YUV
  
  Chroma subsampling                  : 4:2:0
  
  Bit depth                           : 8 bits
  
  Scan type                           : Progressive
  
  Bits/(Pixel*Frame)                  : 0.014
  
  Stream size                         : 67.7 MiB (74%)
  
  Writing library                     : x264 core 130 r2273 b3065e6
  
  Encoding settings                   : cabac=1 / ref=2 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=6 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=6 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=1 / keyint=250 / keyint_min=25 / scenecut=40 / intra_refresh=0 / rc_lookahead=30 / rc=crf / mbtree=1 / crf=28.0 / qcomp=0.70 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=50000 / vbv_bufsize=50000 / crf_max=0.0 / nal_hrd=none / ip_ratio=1.40 / aq=1:0.60
  
  Encoded date                        : UTC 2013-09-12 14:24:13
  
  Tagged date                         : UTC 2013-09-12 14:24:16
  
  
  
  Audio
  
  ID                                  : 2
  
  Format                              : AAC
  
  Format/Info                         : Advanced Audio Codec
  
  Format profile                      : LC
  
  Codec ID                            : 40
  
  Duration                            : 24mn 27s
  
  Bit rate mode                       : Variable
  
  Bit rate                            : 132 Kbps
  
  Maximum bit rate                    : 172 Kbps
  
  Channel(s)                          : 2 channels
  
  Channel positions                   : Front: L R
  
  Sampling rate                       : 48.0 KHz
  
  Compression mode                    : Lossy
  
  Delay relative to video             : 67ms
  
  Stream size                         : 22.7 MiB (25%)
  
  Encoded date                        : UTC 2013-09-12 14:24:15
  
  Tagged date                         : UTC 2013-09-12 14:24:16

• Desktop audio falls behind when recording microphone + desktop audio + screen using ffmpeg

  15 septembre 2013, par madr

  I have put together this script for recording the microphone, the desktop audio and the screen using ffmpeg :

  DATE=`which date`
  
  RESO=2560x1440
  
  FPS=30
  
  PRESET=ultrafast
  
  DIRECTORY=$HOME/Video/
  
  FILENAME=videocast`$DATE +%d%m%Y_%H.%M.%S`.mkv
  
  
  
  ffmpeg -y -vsync 1 \
  
  -f pulse -ac 2 -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor \
  
  -f pulse -ac 1 -ar 25000 -i alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono \
  
  -filter_complex aresample=async=1,amix=duration=shortest,apad \
  
  -f x11grab -r $FPS -s $RESO -i :0.0 \
  
  -acodec libvorbis \
  
  -vcodec libx264 -pix_fmt yuv420p -preset $PRESET -threads 0 \
  
  $DIRECTORY$FILENAME

  Everything is recorded and between the screen and the microphone sound there are no issues what so ever, however the desktop audio falls behind badly.

  It begins in sync but gets worse over time during playback, also in ffplay. It does not matter what application playing sound : both Youtube-videos in the browser, desktop sounds and Rhythmbox (playing a couple of seconds of song then stops, wait and repeat) gets out of sync.

  The terminal output complain about

  "ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred" 

  and similar but I do not know what that means.

  Full terminal output here :

  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Aug 11 2013 14:52:28 with gcc 4.8.1 (GCC) 20130725 (prerelease)
  
    configuration: --prefix=/usr --disable-debug --disable-static --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic --enable-postproc --enable-runtime-cpudetect --enable-shared --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libavresample   1.  1.  0 /  1.  1.  0
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  
  Input #0, pulse, from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor':
  
    Duration: N/A, start: 0.014093, bitrate: 1536 kb/s
  
      Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
  
  Guessed Channel Layout for  Input Stream #1.0 : mono
  
  Input #1, pulse, from 'alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono':
  
    Duration: N/A, start: 0.006172, bitrate: 400 kb/s
  
      Stream #1:0: Audio: pcm_s16le, 25000 Hz, mono, s16, 400 kb/s
  
  [x11grab @ 0x218a6e0] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 2560 height: 1440
  
  [x11grab @ 0x218a6e0] shared memory extension found
  
  Input #2, x11grab, from ':0.0':
  
    Duration: N/A, start: 1379021580.184321, bitrate: N/A
  
      Stream #2:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 2560x1440, -2147483 kb/s, 30 tbr, 1000k tbn, 30 tbc
  
  [libx264 @ 0x21ae560] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x21ae560] profile Constrained Baseline, level 5.0
  
  [libx264 @ 0x21ae560] 264 - core 133 r2339 585324f - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
  
  Output #0, matroska, to '/home/anders/Video/videocast12092013_23.33.00.mkv':
  
    Metadata:
  
      encoder         : Lavf55.12.100
  
      Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 25000 Hz, mono, fltp
  
      Stream #0:1: Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 2560x1440, q=-1--1, 1k tbn, 30 tbc
  
  Stream mapping:
  
    Stream #0:0 (pcm_s16le) -> aresample (graph 0)
  
    Stream #1:0 (pcm_s16le) -> amix:input1 (graph 0)
  
    amix (graph 0) -> Stream #0:0 (libvorbis)
  
    Stream #2:0 -> #0:1 (rawvideo -> libx264)
  
  Press [q] to stop, [?] for help
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred3.22 bitrate=10423.3kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred25.25 bitrate=11011.0kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred5.76 bitrate=11013.7kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred27.25 bitrate=11175.4kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred7.76 bitrate=11168.7kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred8.24 bitrate=11176.4kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred55.48 bitrate=11243.8kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  frame=12871 fps= 30 q=-1.0 Lsize=  542369kB time=00:07:09.31 bitrate=10349.3kbits/s    
  
  video:539762kB audio:2363kB subtitle:0 global headers:3kB muxing overhead 0.044476%
  
  [libx264 @ 0x21ae560] frame I:52    Avg QP:15.46  size:725888
  
  [libx264 @ 0x21ae560] frame P:12819 Avg QP:18.26  size: 40172
  
  [libx264 @ 0x21ae560] mb I  I16..4: 100.0%  0.0%  0.0%
  
  [libx264 @ 0x21ae560] mb P  I16..4:  2.6%  0.0%  0.0%  P16..4: 18.1%  0.0%  0.0%  0.0%  0.0%    skip:79.3%
  
  [libx264 @ 0x21ae560] coded y,uvDC,uvAC intra: 57.8% 49.8% 25.3% inter: 8.9% 8.7% 2.2%
  
  [libx264 @ 0x21ae560] i16 v,h,dc,p: 23% 29% 32% 16%
  
  [libx264 @ 0x21ae560] i8c dc,h,v,p: 45% 28% 18%  9%
  
  [libx264 @ 0x21ae560] kb/s:10306.26

  Please help me, I am really close to get this working !

  UPDATE : The desktop audio is out of sync when skipping filter_complex and microphone also, bit in a smaller amount. Using copy instead of libvorbis does not change anything either.