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• ffmpeg can't stream to remote client

  4 septembre 2014, par KFL

  I’m building a simple ffmpeg command line on my laptop to stream from its camera. The command line reads (verbose output at the botton) :

  host1> ffmpeg -v verbose \
  
                -f dshow \
  
                -i video="Camera":audio="Microphone" \
  
                -r 30 -g 0 -vcodec h264 -acodec libmp3lame \
  
                -tune zerolatency \
  
                -preset ultrafast \
  
                -f mpegts udp://12.34.56.78:12345

  Firstly, it works locally. I.e., I can view the output by using ffplay on the same host :

  host1> ffplay -hide_banner -v udp://12.34.56.78:12345

  Now what is NOT working is when I do this from another machine in the same network. It shows a nan progress :

  host2> ffplay -hide_banner -v udp://12.34.56.78:12345
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   

  I used ncat to dump the raw content. But there’s no output :

  host2>\ncat\ncat -v -u 12.34.56.78 12345
  
  Ncat: Version 5.59BETA1 ( http://nmap.org/ncat )
  
  Ncat: Connected to 12.34.56.78:12345.
  
  (...and nothing happen...)

  Note that I can exclude firewall issues as I used ncat to communicate with each other across the wire using the same port and protocol (UDP). This works and they can chat to each other :

  host1> ncat -l -u -p 12345
  
  host2> ncat -u 12.34.56.78 12345

  Any hint ?

  I’m using Windows x64 with FFMPEG 64bit installed from here. Below is the Output of my ffmpeg command :

  C:\ffmpeg\bin>ffmpeg -v verbose -f dshow -i video="Integrated Camera":audio="Microphone (Realtek High Definition Audio)" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://12.34.56.78:12345
  
  ffmpeg version N-66012-g97b8809 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Sep  1 2014 00:21:15 with gcc 4.8.3 (GCC)
  
    configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --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-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  3.100 / 56.  3.100
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  0.103 /  5.  0.103
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Guessed Channel Layout for  Input Stream #0.1 : stereo
  
  Input #0, dshow, from 'video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)':
  
    Duration: N/A, start: 171840.657000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo, bgr24, 640x480, 30 fps, 30 tbr, 10000k tbn, 30 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Matched encoder 'libx264' for codec 'h264'.
  
  [graph 0 input from stream 0:0 @ 0000000000470aa0] w:640 h:480 pixfmt:bgr24 tb:1/10000000 fr:10000000/333333 sar:0/1 sws_param:flags=2
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:iw h:ih flags:'0x4' interl:0
  
  [format @ 0000000004325a00] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed_null_0' and the filter 'format'
  
  [auto-inserted scaler 0 @ 0000000004326d00] w:640 h:480 fmt:bgr24 sar:0/1 -> w:640 h:480 fmt:yuv444p sar:0/1 flags:0x4
  
  No pixel format specified, yuv444p for H.264 encoding chosen.
  
  Use -pix_fmt yuv420p for compatibility with outdated media players.
  
  [graph 1 input from stream 0:1 @ 0000000000460c20] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [audio format for output stream 0:1 @ 00000000004601a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
  
  [auto-inserted resampler 0 @ 00000000004604a0] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
  
  [libx264 @ 000000000081bb20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 000000000081bb20] profile High 4:4:4 Intra, level 3.0, 4:4:4 8-bit
  
  [mpegts @ 000000000081abe0] muxrate VBR, pcr every 3 pkts, sdt every 200, pat/pmt every 40 pkts
  
  Output #0, mpegts, to 'udp://12.34.56.78:12345':
  
    Metadata:
  
      encoder         : Lavf56.3.100
  
      Stream #0:0: Video: h264 (libx264), yuv444p, 640x480, q=-1--1, 30 fps, 90k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc56.1.100 libx264
  
      Stream #0:1: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p
  
      Metadata:
  
        encoder         : Lavc56.1.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  
    Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  *** 1 dup!
  
  frame=  241 fps= 31 q=28.0 Lsize=    3439kB time=00:00:08.03 bitrate=3506.4kbits/s dup=1 drop=0
  
  video:3035kB audio:125kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.791966%
  
  Input file #0 (video=Integrated Camera:audio=Microphone (Realtek High Definition Audio)):
  
    Input stream #0:0 (video): 240 packets read (221184000 bytes); 240 frames decoded;
  
    Input stream #0:1 (audio): 16 packets read (1411200 bytes); 16 frames decoded (352800 samples);
  
    Total: 256 packets (222595200 bytes) demuxed
  
  Output file #0 (udp://12.34.56.78:12345):
  
    Output stream #0:0 (video): 241 frames encoded; 241 packets muxed (3108187 bytes);
  
    Output stream #0:1 (audio): 306 frames encoded (352512 samples); 307 packets muxed (128313 bytes);
  
    Total: 548 packets (3236500 bytes) muxed
  
  [libx264 @ 000000000081bb20] frame I:241   Avg QP:27.97  size: 12897
  
  [libx264 @ 000000000081bb20] mb I  I16..4: 100.0%  0.0%  0.0%
  
  [libx264 @ 000000000081bb20] coded y,u,v intra: 26.3% 0.5% 0.0%
  
  [libx264 @ 000000000081bb20] i16 v,h,dc,p: 19% 28% 21% 31%
  
  [libx264 @ 000000000081bb20] kb/s:3095.29
  
  [dshow @ 0000000000467720] real-time buffer[Integrated Camera] too full (90% of size: 3041280)! frame dropped!
  
  Received signal 2: terminating. (I pressed CTRL-C)

• ffmpeg silence detect and removal

  6 septembre 2014, par Zahlii

  I am trying to use the following command with the latest ffmpeg build to remove silence from my .mp3 files :

  ffmpeg -i SILENCE.mp3 -af silencedetect=n=-50dB:d=1 -y -ab 192k  SILENCE_OUT.mp3

  However, the following output is produced :

  ffmpeg version N-66154-g1654ca7 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Sep  5 2014 22:10:38 with gcc 4.8.3 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib
  
  modplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrw
  
  b --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinge
  
  r --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --en
  
  able-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
  
   --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-
  
  libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  4.100 / 56.  4.100
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  1.100 /  5.  1.100
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Input #0, mp3, from 'SILENCE.mp3':
  
    Metadata:
  
      title           : Snowblind (Featuring Tasha Baxter)
  
      artist          : Au5
  
      album           : Snowblind (Featuring Tasha Baxter)
  
      genre           : Electronica
  
      performer       : Au5
  
      track           : 1/1
  
      date            : 2014
  
      album_artist    : Au5,Tasha Baxter
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: isommp42
  
      encoder         : Lavf55.42.100
  
    Duration: 00:05:50.80, start: 0.025057, bitrate: 192 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 192 kb/s
  
  Output #0, mp3, to 'SILENCE_OUT.mp3':
  
    Metadata:
  
      TIT2            : Snowblind (Featuring Tasha Baxter)
  
      TPE1            : Au5
  
      TALB            : Snowblind (Featuring Tasha Baxter)
  
      TCON            : Electronica
  
      TPE3            : Au5
  
      TRCK            : 1/1
  
      TDRL            : 2014
  
      TPE2            : Au5,Tasha Baxter
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: isommp42
  
      TSSE            : Lavf56.4.100
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc56.1.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [silencedetect @ 0000000004398f40] silence_start: -0.00628118
  
  [silencedetect @ 0000000004398f40] silence_end: 3.21413 | silence_duration: 3.22
  
  041
  
  [silencedetect @ 0000000004398f40] silence_start: 343.844
  
  [libmp3lame @ 00000000043b2940] Trying to remove 1152 samples, but the queue is
  
  empty
  
  size=    8223kB time=00:05:50.79 bitrate= 192.0kbits/s
  
  video:0kB audio:8222kB subtitle:0kB other streams:0kB global headers:0kB muxing
  
  overhead: 0.011485%

  The generated audio file however still has the original length without any silence removed.
  See the following images :
  

  Any help is appreciated !

  EDIT :
  Alright, silence detect is only DETECTING the silence. Not removing it. I will try to post a solution for this.

• FFMPEG api convertion file error

  10 septembre 2014, par Thomas

  I’m trying to convert video thanks to the API (using the transcoding example) which I adapted a very few (like 5 lines) and force the codec to MJPEG. It encode 3 or 4 frames then it quit saying Invalid pts (3) <= last (3). Here’s the whole trace :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/thomas/test.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2014-09-05 15:42:24
  
    Duration: 00:00:29.70, start: 0.000000, bitrate: 943 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 935 kb/s, SAR 1:1 DAR 16:9, 60 fps, 60 tbr, 6k tbn, 50 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-09-05 15:42:24
  
        handler_name    : Core Media Data Handler
  
        encoder         : H.264
  
        timecode        : 00:00:00:00
  
      Stream #0:1(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-09-05 15:42:24
  
        handler_name    : Core Media Data Handler
  
        timecode        : 00:00:00:00
  
  Output #0, mov, to '/Users/thomas/test_MJPEG.mov':
  
      Stream #0:0: Video: mjpeg, yuvj420p, 1280x720, q=2-31, 128 kb/s, 50 tbc
  
      Stream #0:1: Data: none (tmcd / 0x64636D74), 0 kb/s
  
  [mov @ 0x7ffd6202e600] Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead.
  
  [mov @ 0x7ffd6202e600] Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead.
  
  [swscaler @ 0x7ffd6204d000] deprecated pixel format used, make sure you did set range correctly
  
  Pushing decoded frame to filters
  
  Pulling filtered frame from filters
  
  Encoding frame
  
  Pulling filtered frame from filters
  
  Pushing decoded frame to filters
  
  Pulling filtered frame from filters
  
  Encoding frame
  
  Pulling filtered frame from filters
  
  Pushing decoded frame to filters
  
  Pulling filtered frame from filters
  
  Encoding frame
  
  Pulling filtered frame from filters
  
  Pushing decoded frame to filters
  
  Pulling filtered frame from filters
  
  Encoding frame
  
  Pulling filtered frame from filters
  
  Pushing decoded frame to filters
  
  Pulling filtered frame from filters
  
  Encoding frame
  
  [mjpeg @ 0x7ffd6202ec00] Invalid pts (3) <= last (3)
  
  Error occurred: Operation not permitted

  I don’t know how to fix this, am I using invalid settings ?

  Here’s the movie I’m working with

  FYI, I’d like to achieve the same thing than the command ffmpeg -i test.mov -vcodec mjpeg testMJPEG.mov