Recherche avancée

Sur d’autres sites (13934)

• FFmpeg : h264 output broken

  22 juin 2023, par puaaaal

  When I try to encode anything (no -c copy) using the h264 codec, the output is broken in the sense that it can't be played by any standard player (like Windows Media Player). VLC works mostly fine, but also here I had problems, that the video did not align properly. When I use ffplay though, it works fine. When I try to play my source material, which is also encoded with h264, all those methods to play it work fine. (I use mp4 as a container for all these tests)

  


  Command to reproduce :

  


  ffmpeg -f lavfi -i "testsrc=d=10" -vcodec h264 test_h264.mp4


  


  When I use mpeg4 instead, it works fine :

  


  ffmpeg -f lavfi -i "testsrc=d=10" -vcodec mpeg4 test_mpeg4.mp4


  


  Is there something I am missing here, or might this be a bug ?

  


  ffmpeg -version :

  


  ffmpeg version n6.0-22-g549430e14d-20230607 Copyright (c) 2000-2023 the FFmpeg developers
built with gcc 13.1.0 (crosstool-NG 1.25.0.196_227d99d)
configuration: --prefix=/ffbuild/prefix --pkg-config-flags=--static --pkg-config=pkg-config --cross-prefix=x86_64-w64-mingw32- --arch=x86_64 --target-os=mingw32 --enable-gpl --enable-version3 --disable-debug --enable-shared --disable-static --disable-w32threads --enable-pthreads --enable-iconv --enable-libxml2 --enable-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-lzma --enable-fontconfig --enable-libvorbis --enable-opencl --disable-libpulse --enable-libvmaf --disable-libxcb --disable-xlib --enable-amf --enable-libaom --enable-libaribb24 --enable-avisynth --enable-chromaprint --enable-libdav1d --enable-libdavs2 --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-frei0r --enable-libgme --enable-libkvazaar --enable-libass --enable-libbluray --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librist --enable-libssh --enable-libtheora --enable-libvpx --enable-libwebp --enable-lv2 --enable-libvpl --enable-openal --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopenmpt --enable-librav1e --enable-librubberband --enable-schannel --enable-sdl2 --enable-libsoxr --enable-libsrt --enable-libsvtav1 --enable-libtwolame --enable-libuavs3d --disable-libdrm --disable-vaapi --enable-libvidstab --enable-vulkan --enable-libshaderc --disable-libplacebo --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libzimg --enable-libzvbi --extra-cflags=-DLIBTWOLAME_STATIC --extra-cxxflags= --extra-ldflags=-pthread --extra-ldexeflags= --extra-libs=-lgomp --extra-version=20230607
libavutil      58.  2.100 / 58.  2.100
libavcodec     60.  3.100 / 60.  3.100
libavformat    60.  3.100 / 60.  3.100
libavdevice    60.  1.100 / 60.  1.100
libavfilter     9.  3.100 /  9.  3.100
libswscale      7.  1.100 /  7.  1.100
libswresample   4. 10.100 /  4. 10.100
libpostproc    57.  1.100 / 57.  1.100


  


• Encode video of powerpoint presentation for HTML5 playback

  17 avril 2013, par user2291446

  We have a number of powerpoint presentations that have been converted to 16:9
  aspect ratio and then converted into mp4 "master videos" with an "apple TV" 720p
  profile. These powerpoint presentations are voice annotated. So in essence, we
  show a slide and then let the annotation sound play for a while, then go to the
  next slide, and so on. The resulting mp4 master video is somewhere around 900MB
  on average.

  Here is an example of the master video

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4' :
  Metadata :
      major_brand : isom
      minor_version : 512
      compatible_brands : isomiso2avc1mp41
      creation_time : 1970-01-01 00:00:00
      encoder : Lavf52.104.0
    Duration : 02:00:57.65, start : 0.000000, bitrate : 970 kb/s
      Stream #0:0(und) : Video : h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 836 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc
      Metadata :
        creation_time : 1970-01-01 00:00:00
        handler_name : VideoHandler
      Stream #0:1(und) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 127 kb/s
      Metadata :
        creation_time : 1970-01-01 00:00:00
        handler_name : SoundHandler
  

  We are trying to get these presentations to play on the web on as many
  devices/browsers as possible including some that don't do HTML5 (IE7/IE8). We
  have narrowed down our player of choice which is mediaElement and have extracted
  some "cue points" from the powerpoint presentation that mark where the slides
  are changing. We have also captured thumbnails for those cuepoints such that we
  now have a nice list of thumbnails for each slide and an associated cuepoint in
  the video where the particular slide begins.

  Here comes the problem...due to the large size of the master video it is not
  practical for us to use the master video with our mediaElement player. We do
  need to transcode the master video to mp4 and ogv in order to get decent
  device/browser coverage.

  We do not seem to be able to find a suitable transcoding strategy to reduce the
  size of the video. We have played with numerous ffmpeg settings and were able to
  reduce the size but when we do so we compromise the ability to jump to specific
  cue points.

  It works well for browsers that do HTML5 video natively (Chrome and Firefox) but
  not for the flash fallback of mediaElement (IE7/IE8) which uses the mp4 file and
  seemingly is very tied to the number and frequency of key frames in the video in
  order to allow for clean seeking and skipping using the cue points.

  Seeing that we are talking about a video that has only slides (practically 90
  static images per presentation) and some sound we imagine it must be possible to
  transcode as such that the keyframes fall at the cue points or near the
  cuepoints, and that the size of the video could be drastically reduced while
  still allowing for smooth seeking and skipping.

• How to solve Jiiter Buffer problem in receiving audio RTP stream (bad sound quality) in PJSIP ?

  1er août 2019, par Mayur Patel

  I’m a newbie to pjsip and want to build an RTP stream receiver using pjsip.

  Setup :

  I want to use specific L16/16000/1 codec and have also enabled it in "config_site.h" during compiling the pjsip project and checked that its available

  Receiver :

  • BeagleBone
  • CrossCompiled Pjsip and Installed all req. libs and sample apps

  Sender :

  • Another Windows PC in the same Network using FFmpeg to transmit Audio Stream via Multicast

  I got to know about streamutil.c(pjsip sample-apps) which does similar things to send and receive both. Now for the sake of easyness, I’m using the same Cross-Compiled binary streamutil.

  SENDER :

  ..\ffmpeg -re -stream_loop -1 -i test.mp3 -ar 16000 -acodec pcm_s16be -b:a 128k -ac 1 -payload_type 123 -f rtp udp://239.255.255.211:5500?pkt_size=652
  
  
  
  ......
  
  Output #0, rtp, to 'udp://239.255.255.211:5500?pkt_size=652':
  
    Metadata:
  
      title           : -----
  
      artist          : --------
  
      album           : -------
  
      date            : 2019
  
      track           : 1
  
      encoder         : Lavf58.20.100
  
      Stream #0:0: Audio: pcm_s16be, 16000 Hz, mono, s16, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 pcm_s16be
  
  SDP:
  
  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=GREATEST HITS (2) [1 HOUR 20 MINUTES LONG]
  
  c=IN IP4 239.255.255.211/5
  
  t=0 0
  
  a=tool:libavformat 58.26.101
  
  m=audio 5500 RTP/AVP 123
  
  b=AS:256
  
  a=rtpmap:97 L16/16000/1
  
  a=rtpmap:123 L16/16000/1
  
  a=control:streamid=
  
  
  
  size=     833kB time=00:00:25.91 bitrate= 263.4kbits/s speed=   1x

  RECIEVER LOG :

  ./streamutil --mcast-addr=239.255.255.211 --recv-only --codec=L16/16000/1
  
  ...
  
  ...
  
  17:05:05.178     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
  
  17:05:05.246     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked
  
  17:05:05.266     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
  
  17:05:05.325     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked
  
  17:05:05.344     strm0x55dee1537f48  Jitter buffer starts returning normal frames (after 1 empty/lost)
  
  17:05:05.422     strm0x55dee1537f48  Jitter buffer empty (prefetch=0), plc invoked

  Tried So far :

  • set different payload_type
  • set specific codec in streamutil as parameter
  • all other parameters in FFmpeg ex. bitrate, clockrate, channels

  Check working stream

  I am facing no issue, if I use a *.sdp file to receive RTP stream in VLC.

  SDP file :

  v=0
  
  o=- 0 0 IN IP4 127.0.0.1
  
  s=GREATEST HITS (2) [1 HOUR 20 MINUTES LONG]
  
  c=IN IP4 239.255.255.211/5
  
  t=0 0
  
  a=tool:libavformat 58.26.101
  
  m=audio 5500 RTP/AVP 123
  
  b=AS:256
  
  a=rtpmap:97 PCMU/8000/1
  
  a=rtpmap:123 PCMU/8000/1
  
  a=control:streamid=

  I have googled a lot but stuck now at this problem.
  So finally my question is that,
  How can I get the same Output via Pjsip without this Jitter Buffer logging and dropped sound ?

  Any help would be greatly appreciated.!