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• ffmpeg - when cutting video, last frame is green [on hold]

  4 juillet 2016, par Mister Fresh

  In a php web app, the user can upload a video and cut it in samples. Php launches a shell command with shell_exec() to start ffmpeg and cut the video.

  It works except that the last frame shows a green splash in the cut sample.

  Server is on linux. In dev environment (mac) with latest ffmpeg, the problem cannot be reproduced.

  Command is the following :

  ffmpeg -i /var/www/user_dir/web/projects/127472/cuts/1146 -ss 00:00:45 -t 15 -vcodec libx264 -s 640x360 -strict experimental -v quiet -y /var/www/user_dir/web/projects/127472/cuts/sample1954_small.mp4

  Console output :

  ffmpeg version 0.8.17-6:0.8.17-1, 
  
  
  
  Copyright (c) 2000-2014 the Libav developers
  
  
  
  built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  
  
  configuration: --arch=amd64 --enable-pthreads --enable-runtime-cpudetect --extra-version='6:0.8.17-1' --libdir=/usr/lib/x86_64-linux-gnu --prefix=/usr --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-librtmp --enable-libopencv --enable-libopenjpeg --enable-libpulse --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-vaapi --enable-vdpau --enable-libvorbis --enable-libvpx --enable-zlib --enable-gpl --enable-postproc --enable-swscale --enable-libcdio --enable-x11grab --enable-libx264 --enable-libxvid --shlibdir=/usr/lib/x86_64-linux-gnu --enable-shared --disable-static
  
  
  
  libavutil    51. 22. 3 / 51. 22. 3
  
  
  
  libavcodec   53. 35. 0 / 53. 35. 0
  
  
  
  libavformat  53. 21. 1 / 53. 21. 1
  
  
  
  libavdevice  53.  2. 0 / 53.  2. 0
  
  
  
  libavfilter   2. 15. 0 /  2. 15. 0
  
  
  
  libswscale    2.  1. 0 /  2.  1. 0
  
  
  
  libpostproc  52.  0. 0 / 52.  0. 0
  
  
  
  The ffmpeg program is only provided for script compatibility and will be removed
  
  in a future release. It has been deprecated in the Libav project to allow for
  
  incompatible command line syntax improvements in its replacement called avconv
  
  (see Changelog for details). Please use avconv instead.
  
  
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/var/www/user_dir/web/projects/127472/cuts/1146':
  
  
  
  Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf56.15.103
  
      genre           : Blues
  
  
  
  Duration: 00:01:29.47, start: 0.036281, bitrate: 808 kb/s
  
  
  
  Stream #0.0(und): Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 675 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc
  
  
  
  Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s
  
  [buffer @ 0x30a2ca0] w:1920 h:1080 pixfmt:yuv420p
  
  [scale @ 0x309bf00] w:1920 h:1080 fmt:yuv420p -> w:640 h:360 fmt:yuv420p flags:0x4
  
  [libx264 @ 0x30a3200] using SAR=1/1
  
  [libx264 @ 0x30a3200] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2
  
  [libx264 @ 0x30a3200] profile Main, level 3.0
  
  [libx264 @ 0x30a3200] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - 
  
  
  
      Copyleft 2003-2012 - http://www.videolan.org/x264.html 
  
  - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.25 aq=1:1.00
  
  
  
  Output #0, mp4, to '/var/www/user_dir/web/projects/127472/cuts/sample1954_small.mp4':
  
  
  
  Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      genre           : Blues
  
      encoder         : Lavf53.21.1
  
  
  
  Stream #0.0(und): Video: libx264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], q=-1--1, 25 tbn, 25 tbc
  
  
  
  Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 200 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
    Stream #0.1 -> #0.1
  
  
  
  Press ctrl-c to stop encoding
  
  
  
  [buffer @ 0x30a2ca0] Buffering several frames is not supported. Please consume all available frames before adding a new one.
  
  
  
  Last message repeated 1124 times 565kB time=12.80 bitrate= 361.7kbits/s    ts/s    
  
  frame=  375 fps= 17 q=28.0 Lsize=     742kB time=14.96 bitrate= 406.4kbits/s    
  
  video:360kB audio:371kB global headers:0kB muxing overhead 1.515374%
  
  frame I:3     Avg QP:16.97  size:  6922
  
  
  
  [libx264 @ 0x30a3200] frame P:158   Avg QP:24.40  size:  1858
  
  
  
  [libx264 @ 0x30a3200] frame B:214   Avg QP:28.69  size:   253
  
  
  
  [libx264 @ 0x30a3200] consecutive B-frames: 13.9% 28.3%  5.6% 52.3%
  
  
  
  [libx264 @ 0x30a3200] mb I  I16..4: 70.8%  0.0% 29.2%
  
  
  
  [libx264 @ 0x30a3200] mb P  I16..4:  4.8%  0.0%  1.6%  P16..4: 16.0%  3.9%  2.0%  0.0%  0.0%    skip:71.9%
  
  
  
  [libx264 @ 0x30a3200] mb B  I16..4:  0.5%  0.0%  0.1%  B16..8: 11.1%  0.7%  0.1%  direct: 0.1%  skip:87.4%  L0:44.2% L1:44.7% BI:11.1%
  
  
  
  [libx264 @ 0x30a3200] coded y,uvDC,uvAC intra: 19.3% 30.9% 23.5% inter: 2.3% 2.3% 1.3%
  
  
  
  [libx264 @ 0x30a3200] i16 v,h,dc,p: 39% 47%  2% 12%
  
  
  
  [libx264 @ 0x30a3200] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 20% 25% 32%  5%  5%  3%  4%  3%  3%
  
  
  
  [libx264 @ 0x30a3200] i8c dc,h,v,p: 51% 40%  7%  2%
  
  
  
  [libx264 @ 0x30a3200] Weighted P-Frames: Y:0.0% UV:0.0%
  
  
  
  [libx264 @ 0x30a3200] ref P L0: 63.7%  4.9% 19.3% 12.1%
  
  
  
  [libx264 @ 0x30a3200] ref B L0: 67.6% 32.4%
  
  
  
  [libx264 @ 0x30a3200] kb/s:196.48

• Latency and DAF in RTP transmissions

  24 février 2023, par jfernandz

  I'm trying to perform some tests for audio RTP transmissions to know their technical limitations. The idea is to prevent DAF effect in this kind of transmissions, I'm assuming a latency lower than 50ms will prevent it. But there is another handicap in my analysis, the RTP transmission must be over WiFi.

  


  For this tests I'm trying to transmit raw audio (not sure if skipping the encoding stage will improve latency) through ffmpeg between two different laptops, so I'm running ffmpeg in the first laptop (172.20.1.2) as :

  


  $ ffmpeg -f pulse -i 56 -c copy -f rtp rtp://172.20.1.5:10000

  


  which produces the following output :

  


  ffmpeg version n5.1.2 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from '56':
  Duration: N/A, start: 1677234050.938677, bitrate: 1536 kb/s
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Output #0, rtp, to 'rtp://172.20.1.5:10000':
  Metadata:
    encoder         : Lavf59.27.100
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536

Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=     322kB time=00:00:01.67 bitrate=1573.6kbits/s speed=1.06x


  


  I'm assuming the shown SDP is a valid one :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536


  


  So I saved it in a file called ccopy.sdp on the second laptop (172.20.1.5). However, when I run ffplay in this other laptop as :

  


  $ ffplay -protocol_whitelist file,rtp,udp -i ccopy.sdp

  


  I can see there is problems with this SDP :

  


  ffplay version n5.1.2 Copyright (c) 2003-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[sdp @ 0x7f8eec000c80] Could not find codec parameters for stream 0 (Audio: none, 0 channels): unknown codec
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, sdp, from 'ccopy.sdp':
  Metadata:
    title           : No Name
  Duration: N/A, bitrate: N/A
  Stream #0:0: Audio: none, 0 channels
Failed to open file 'ccopy.sdp' or configure filtergraph
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   


  


  Not sure if I'm doing something wrong or this is because of I cannot actually use pcm_s16le for an RTP transmission. Moreover ... Is there some argument for ffmpeg that I can use to improve this RTP transmission and reduce latency under 50ms.

  


  Thank you all :-)

  


  PS : When I don't use -c copy argument for ffmpeg and therefore I have this SDP

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:768
a=rtpmap:97 PCMU/48000/2


  


  The RTP transmission works as I expect, but with a significant DAF.

  


• Record sound with ffmpeg on ubuntu 12.04 [closed]

  27 juin 2012, par vzybilly

  I have been working for a few days on trying to get ffmpeg to record sound, a short list of what I've tried :

  #Crappy screen grab
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -f mp4 ./out
  
  #awesome screen grab, grabbing sound but non out.
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -f alsa -ac 2 -i pulse -vcodec libx264 -s "1366x768" -acodec libmp3lame -ab 128k -threads 0 -f mp4 ~/Desktop/vid
  
  #audio test, no audio in file.
  
  #ffmpeg -f alsa -ac 2 -i pulse -acodec libmp3lame -ab 128k -threads 0 -f mp3 ./test.mp3
  
  #awesome screen grab.
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -threads 0 -sameq -an -f mp4 ~/Desktop/vid[/CODE]I'm running ubuntu 12.04 from beta(ish)

  it would be awesome if someone could help me get this to work all in one line or (the way i'm going) multiple instances of ffmpeg (screen grab, microphone, program)

  I have also tried the pavucontrol with doing the monitoring of when recording audio, but that does not help either.

  Thanks for all of your help, vzybilly 

  EDIT :
  This one crashed.

  $ ffmpeg -f alsa -ac 2 -i plughw:0,0 -f x11grab -r 100 -s 1366x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 3 testVid.mkv
  
  ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers
  
    built on Jun 12 2012 16:37:58 with gcc 4.6.3
  
  *** THIS PROGRAM IS DEPRECATED ***
  
  This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
  
  [alsa @ 0x8fce240] capture with some ALSA plugins, especially dsnoop, may hang.
  
  [alsa @ 0x8fce240] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, alsa, from 'plughw:0,0':
  
    Duration: N/A, start: 433.999945, bitrate: N/A
  
      Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  [x11grab @ 0x8fde820] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1366 height: 768
  
  [x11grab @ 0x8fde820] shared memory extension  found
  
  [x11grab @ 0x8fde820] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, x11grab, from ':0.0':
  
    Duration: N/A, start: 1340805516.368518, bitrate: N/A
  
      Stream #1.0: Video: rawvideo, bgra, 1366x768, -2147483 kb/s, 100 tbr, 1000k tbn, 100 tbc
  
  File 'testVid.mkv' already exists. Overwrite ? [y/N] y
  
  Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p'
  
  [buffer @ 0x8fde700] w:1366 h:768 pixfmt:bgra
  
  [avsink @ 0x8fcdf20] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
  
  [scale @ 0x8ff3ce0] w:1366 h:768 fmt:bgra -> w:1366 h:768 fmt:yuv420p flags:0x4
  
  [libx264 @ 0x8fdd920] lookaheadless mb-tree requires intra refresh or infinite keyint
  
  [libx264 @ 0x8fdd920] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2
  
  [libx264 @ 0x8fdd920] profile Constrained Baseline, level 4.2
  
  [libx264 @ 0x8fdd920] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - 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=3 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=1.25 aq=0
  
  Output #0, matroska, to 'archinstall4.mkv':
  
    Metadata:
  
      encoder         : Lavf53.21.0
  
      Stream #0.0: Video: libx264, yuv420p, 1366x768, q=-1--1, 1k tbn, 100 tbc
  
      Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Stream mapping:
  
    Stream #1.0 -> #0.0
  
    Stream #0.0 -> #0.1
  
  Press ctrl-c to stop encoding
  
  [alsa @ 0x8fce240] ALSA buffer xrun.
  
  [matroska @ 0x8fcd980] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 213 >= 213
  
  av_interleaved_write_frame(): Invalid argument

  Any thoughts ?

  EDIT & ANSWER :
  Got it all working with a script :

  #!/bin/bash
  
  #vzybilly
  
  #these are temp files
  
  aud="aud.mp3"
  
  vid="vid.mp4"
  
  #grab audio & pid
  
  ffmpeg -f alsa -ac 2 -i plughw:0,0 $aud &
  
  audPID=$!
  
  #grab screen & pid
  
  ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -threads 0 -sameq -an -f mp4 $vid &
  
  vidPID=$!
  
  #wait, till name given (that means stop)
  
  read -p "Stop by giving an Output video name?" out
  
  #stop audio and video with pids
  
  kill -n 2 $audPID
  
  kill -n 2 $vidPID
  
  echo "$out"
  
  #combine to the target output file
  
  ffmpeg -i $aud -i $vid -acodec copy -vcodec copy "$out"
  
  #purge the temp files
  
  rm $aud
  
  rm $vid