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Sur d’autres sites (8115)

  • Interpreting ffmpeg output in Python

    11 juin 2020, par Luka Milivojevic

    I started working in FFmpeg and I want to create a list that will contain start and end timestamps of silence intervals. I did print out these intervals using the FFmpeg but I need to format that output so it looks a bit more readable, so that is why I want to create a list out of it and then print it using a custom function. I know that I should go with regex here but I am not sure how should I write it nor how should I read the FFmpeg console output. My function for silence detection looks like :

    



    def detect_silence_ffmpeg():
    command = r"ffmpeg -i audio.wav -af silencedetect=n=-40dB:d=0.5 -f null - "
    subprocess.call(command, shell=True)


    



    And the output of this function on a 7 second long sample video is :

    



    ffmpeg version git-2020-06-03-b6d7c4c Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 49.100 / 56. 49.100
  libavcodec     58. 90.100 / 58. 90.100
  libavformat    58. 44.100 / 58. 44.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 84.100 /  7. 84.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'audio.wav':
  Metadata:
    encoder         : Lavf58.44.100
  Duration: 00:00:07.34, bitrate: 1411 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf58.44.100
    Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Metadata:
      encoder         : Lavc58.90.100 pcm_s16le
[silencedetect @ 00000202fc71e680] silence_start: 0
[silencedetect @ 00000202fc71e680] silence_end: 1.16374 | silence_duration: 1.16374
[silencedetect @ 00000202fc71e680] silence_start: 1.94558
[silencedetect @ 00000202fc71e680] silence_end: 3.41345 | silence_duration: 1.46787
[silencedetect @ 00000202fc71e680] silence_start: 3.8578
[silencedetect @ 00000202fc71e680] silence_end: 5.84844 | silence_duration: 1.99063
[silencedetect @ 00000202fc71e680] silence_start: 6.43653
size=N/A time=00:00:07.33 bitrate=N/A speed= 308x    
video:0kB audio:1264kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[silencedetect @ 00000202fc71e680] silence_end: 7.33868 | silence_duration: 0.902154


    



    And this code should be implemented on an hour or so long videos so I really need to find a way to format this output a bit better than this. That would be it, any help would be much appreciated :)

    



    P.S : the idea is that this should work on Windows mainly, but if the cross-platform is possible too it would be great.

    


  • ffmpeg capturing image from rtmp stream

    14 juin 2020, par Lewis Day

    I am passing this command via ssh ;

    



        $rtmp_address = 'rtmp://198.251.69.110/live/';
    $stream_link = "" . $rtmp_address . "" . $stream_key . "";
echo $ssh->exec('ffmpeg -i "' . $stream_link . ' live=1" -f image2 -vcodec png -vframes 1 -s 180x145 -compression_level 100 /var/www/vhosts/flamingocams.net/httpdocs/images/' . $username . '.png -y');


    



    however getting this response ;

    



    


    ffmpeg version N-53084-gd29aaf12f4-static
 https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2020 the FFmpeg
 developers built with gcc 8 (Debian 8.3.0-6) configuration :
 —enable-gpl —enable-version3 —enable-static —disable-debug —disable-ffplay —disable-indev=sndio —disable-outdev=sndio —cc=gcc —enable-fontconfig —enable-frei0r —enable-gnutls —enable-gmp —enable-libgme —enable-gray —enable-libaom —enable-libfribidi —enable-libass —enable-libvmaf —enable-libfreetype —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-librubberband —enable-libsoxr —enable-libspeex —enable-libsrt —enable-libvorbis —enable-libopus —enable-libtheora —enable-libvidstab —enable-libvo-amrwbenc —enable-libvpx —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxml2 —enable-libdav1d —enable-libxvid —enable-libzvbi —enable-libzimg libavutil 56. 50.100 / 56. 50.100 libavcodec 58. 90.100 / 58. 90.100 libavformat 58. 44.100 / 58. 44.100 libavdevice 58. 9.103 / 58. 9.103 libavfilter 7. 84.100 / 7. 84.100 libswscale 5. 6.101 / 5. 6.101 libswresample 3. 6.100 / 3. 6.100 libpostproc 55. 6.100 / 55. 6.100 [rtmp @ 0x730fe40] Detected librtmp style URL parameters, these aren't supported by the
 libavformat internal RTMP handler currently enabled. See the
 documentation for the correct way to pass parameters.

    


    



    could anyone help with what is going wrong.

    


  • FFMPEG HTTP to RTP then RTP to HTTP with OPUS

    20 juin 2020, par Brad Hambleton

    I'm taking a HTTP output to FFMPEG and copying the audio (no video) to an RTP :
ffmpeg -i http://192.168.0.40:20110 -c:a copy -f rtp rtp ://192.168.87.40:20210 ?pkt_size=1328 -sdp_file opus.sdp

    


    At the other end receiving the RTP and pushing it back to HTTP :
ffmpeg -re -protocol_whitelist rtp,file,udp -i opus.sdp -c:a copy -listen 1 -method GET -f opus http://192.168.87.40:20220

    


    2 Problems :

    


      

    1. Currently the encoding process doesn't optimize packets.
92 1.004672 192.168.0.40 192.168.0.40 UDP 392 52954 → 20210 Len=332
93 1.004727 192.168.0.40 192.168.0.40 UDP 392 52954 → 20210 Len=332
94 1.004789 192.168.0.40 192.168.0.40 UDP 392 52954 → 20210 Len=332
95 1.004855 192.168.0.40 192.168.0.40 UDP 392 52954 → 20210 Len=332
96 1.004908 192.168.0.40 192.168.0.40 UDP 392 52954 → 20210 Len=332

      2. 


    


    Each packet length is 332, which leaves a lot of wasted space. I'd like to get close to 1500 (Stack 4 together I get 1328 which is close enough)
Is there a command in the FFMPEG/RTP that will optimize packets ?
I added ?pkt_size=1328 to the RTP however that only sets max, not preferred.

    


      

    1. I get the following error when I try to HTTP to RTP via copy :
C :\Decode>ffmpeg -re -protocol_whitelist rtp,file,udp -i opus.sdp -c:a copy -listen 1 -method GET -f opus http://192.168.0.40:20220
ffmpeg version git-2020-05-23-26b4509 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 9.3.1 (GCC) 20200523
configuration : —enable-gpl —enable-version3 —enable-sdl2 —enable-fontconfig —enable-gnutls —enable-iconv —enable-libass —enable-libdav1d —enable-libbluray —enable-libfreetype —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-libopus —enable-libshine —enable-libsnappy —enable-libsoxr —enable-libsrt —enable-libtheora —enable-libtwolame —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxml2 —enable-libzimg —enable-lzma —enable-zlib —enable-gmp —enable-libvidstab —enable-libvmaf —enable-libvorbis —enable-libvo-amrwbenc —enable-libmysofa —enable-libspeex —enable-libxvid —enable-libaom —disable-w32threads —enable-libmfx —enable-ffnvcodec —enable-cuda-llvm —enable-cuvid —enable-d3d11va —enable-nvenc —enable-nvdec —enable-dxva2 —enable-avisynth —enable-libopenmpt —enable-amf
libavutil 56. 48.100 / 56. 48.100
libavcodec 58. 87.101 / 58. 87.101
libavformat 58. 43.100 / 58. 43.100
libavdevice 58. 9.103 / 58. 9.103
libavfilter 7. 83.100 / 7. 83.100
libswscale 5. 6.101 / 5. 6.101
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
Input #0, sdp, from 'opus.sdp' :
Metadata :
title : No Name
Duration : N/A, start : 0.000000, bitrate : N/A
Stream #0:0 : Audio : opus, 48000 Hz, stereo, fltp
[opus @ 00000221a9a4d280] No extradata present
Could not write header for output file #0 (incorrect codec parameters ?) : Invalid data found when processing input
Stream mapping :
Stream #0:0 -> #0:0 (copy)
Last message repeated 1 times

      2. 


    


    Tried a variety of additions to the RTP to HTTP CLI to get it to work, but still nothing.

    


    -flags -global_header -reconnect_streamed 1 -headers "X-Forwarded-For : 13.14.15.66"

    


    Is there a specific OPUS or HTTP header that can be added to get it to work. Decoding and Encoding does work for RTP to HTTP, the idea isn't to decode/encode at either point, just to copy the audio, change the container..

    


    Cheers

    


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