Recherche avancée

Sur d’autres sites (6367)

• FFMPEG with SSM is not using the specified network interface

  23 janvier 2014, par caspar

  I'm trying to use ffmpeg/ffprobe to join a SSM stream on a server with multiple network interfaces. I'm using the following command to initialize the input :

  c:\ffmpeg\bin>ffprobe "udp://232.2.4.206:24206?localaddr=10.15.248.217&sources=10.15.248.210,10.15.248.211,10.15.248.212,10.15.248.213&connect=1&fifo_size=1000000" -v 9 -loglevel 99

  The routing on the server can't be changed, though I can confirm other applications running on the same server are able to join and receive the multicast signal. The issue is, I believe, with the localaddr parameter, which is ignored it seems. Using Wireshark I can see that the wrong interface is being used (i.e. not the 10.15.248.217 interface).

  The output of the above command is :

  ffprobe version N-60087-g94a5241 Copyright (c) 2007-2014 the FFmpeg developers
  
  built on Jan 21 2014 22:06:13 with gcc 4.8.2 (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-libwavpack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
  libavutil      52. 63.100 / 52. 63.100
  
  libavcodec     55. 48.102 / 55. 48.102
  
  libavformat    55. 25.101 / 55. 25.101
  
  libavdevice    55.  5.102 / 55.  5.102
  
  libavfilter     4.  1.100 /  4.  1.100
  
  libswscale      2.  5.101 /  2.  5.101
  
  libswresample   0. 17.104 /  0. 17.104
  
  libpostproc    52.  3.100 / 52.  3.100
  
  [udp @ 000000000272dcc0] end receive buffer size reported is 65536

  Anyone have experience with this use case ? Perhaps this is a bug that needs raising ?

  EDIT : I found out that if I remove the &sources= parameter, the localaddr is used and the request is passed through the correct interface, however as I need to join a SSM stream, this still blocks me from continuing.

• Why is ffmpeg start time non-zero in .ts format ?

  22 janvier 2014, par geekydel

  I'm using ffmpeg to transcode video to .ts format, and I am getting unexpected start times in the output file.

  To simplify things, I've started with a nice simple AVI file (no audio) :

  ffmpeg -i in.avi 
  
  ...
  
  Input #0, avi, from 'in.avi':
  
    Metadata:
  
      encoder         : Lavf55.25.100
  
    Duration: 00:00:05.00, start: 0.000000, bitrate: 448 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 480x270 [SAR 1:1 DAR 16:9], 24 tbr, 24 tbn, 24 tbc

  Duration=5s, startTime=0s, as expected.

  However, if I transcode to .ts file, with no customisation :

  ffmpeg -i in.avi -y out.ts
  
  ...
  
  Input #0, avi, from 'in.avi':
  
    Metadata:
  
      encoder         : Lavf55.25.100
  
    Duration: 00:00:05.00, start: 0.000000, bitrate: 448 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 480x270 [SAR 1:1 DAR 16:9], 24 tbr, 24 tbn, 24 tbc
  
  Output #0, mpegts, to 'out.ts':
  
    Metadata:
  
      encoder         : Lavf55.25.100
  
      Stream #0:0: Video: mpeg2video, yuv420p, 480x270 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 24 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mpeg4 -> mpeg2video)
  
  Press [q] to stop, [?] for help
  
  frame=  120 fps=0.0 q=31.0 Lsize=     315kB time=00:00:04.95 bitrate= 519.9kbits/s dup=1 drop=0    
  
  video:277kB audio:0kB subtitle:0 global headers:0kB muxing overhead 13.491544%

  Then I get a very odd parameters (Duration=4:96s, startTime=1.441667) :

  ffmpeg -i out.ts
  
  ...
  
  Input #0, mpegts, from 'out.ts':
  
    Duration: 00:00:04.96, start: 1.441667, bitrate: 519 kb/s
  
    Program 1 
  
      Metadata:
  
        service_name    : Service01
  
        service_provider: FFmpeg
  
      Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv), 480x270 [SAR 1:1 DAR 16:9], max. 104857 kb/s, 24 fps, 24 tbr, 90k tbn, 48 tbc

  Now, I can just about understand why the transcode might lose a couple of frames, which explains the duration, but I can't see why the start time should be any different to the AVI file's start time.

  I have tried transcoding to .mp4, .webm and .mov, and in each case we get the 'correct' start time of 0.0s. Can anybody help explain why .ts files behave differently ?

  Thanks in advance !

  (ffmpeg version information :)

  ffmpeg version N-60031-ga459891 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Jan 21 2014 05:31:54 with gcc 4.6 (Debian 4.6.3-1)
  
    configuration: --prefix=/root/ffmpeg-static/64bit --extra-cflags='-I/root/ffmpeg-static/64bit/include -static' --extra-ldflags='-L/root/ffmpeg-static/64bit/lib -static' --extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared --disable-ffserver --disable-doc --enable-bzlib --enable-zlib --enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl --enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray --enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3 --enable-libvpx
  
    libavutil      52. 63.100 / 52. 63.100
  
    libavcodec     55. 48.102 / 55. 48.102
  
    libavformat    55. 25.100 / 55. 25.100
  
    libavdevice    55.  5.102 / 55.  5.102
  
    libavfilter     4.  1.100 /  4.  1.100
  
    libswscale      2.  5.101 /  2.  5.101
  
    libswresample   0. 17.104 /  0. 17.104
  
    libpostproc    52.  3.100 / 52.  3.100

• Removing audio noise (hissing) in C++ from 16bit PCM

  4 juillet 2016, par hockeyislife

  I have been going through the posts of stack overflow but I am not understanding how to remove hissing sound from my audio being grabbed from the microphone.

  I implemented a simple low pass filter but I must be doing something wrong.

  unsigned short *buf = "audio data in PCM format";
  
  double out_sample = 0;
  
  int sample_size = "number of samples of audio";
  
  for (int n = 0; n < sample_size/2; n++)
  
  {
  
      out_sample = (out_sample * 90 + buf[n] * 10) / 100;
  
      buf[n] = (unsigned short) out_sample;
  
  }

  The above produces really corrupt audio.

  I know I need to make a low pass filter on the PCM data. Can anyone shed some light into what I am doing wrong.

  Thanks in advance.