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• ffmpeg : capture rtsp stream continously without stoping

  21 décembre 2019, par Santhosh Yedidi

  I am trying to capture an rtsp stream

  ffmpeg -rtsp_transport tcp -i "rtsp://...." -acodec copy -vcodec copy file.ts

  But this will only record for few seconds and stop

  ffmpeg version n4.1.2 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181127
  
    configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  [rtsp @ 0x564a002b4e80] max delay reached. need to consume packet
  
  [rtsp @ 0x564a002b4e80] RTP: missed 3 packets
  
  [h264 @ 0x564a002b8a80] out of range intra chroma pred mode
  
  [h264 @ 0x564a002b8a80] error while decoding MB 8 43
  
  [h264 @ 0x564a002b8a80] concealing 1401 DC, 1401 AC, 1401 MV errors in I frame
  
  Input #0, rtsp, from 'rtsp://...':
  
    Metadata:
  
      title           : RTSP Session/2.0
  
    Duration: N/A, start: 0.463333, bitrate: N/A
  
      Stream #0:0: Video: h264 (Main), yuvj420p(pc, progressive), 1280x960, 15 fps, 100 tbr, 90k tbn, 30 tbc
  
  Output #0, mpegts, to 'file.ts':
  
    Metadata:
  
      title           : RTSP Session/2.0
  
      encoder         : Lavf58.20.100
  
      Stream #0:0: Video: h264 (Main), yuvj420p(pc, progressive), 1280x960, q=2-31, 15 fps, 100 tbr, 90k tbn, 90k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  frame=  380 fps= 16 q=-1.0 Lsize=    6962kB time=00:00:24.86 bitrate=2293.5kbits/s speed=1.06x    
  
  video:6416kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.512586%

  How to keep the recording continous

• "Invalid or unexpected token" error when trying to execute ffmpeg build on lambda

  4 janvier 2019, par almarc

  I have a node.js script that uses ffmpeg to convert mp4 downloaded from YT to mp3 and save to Amazon S3. Uploading using the serverless framework. The "ffmpeg" file is included in the main directory (with .yml), downloaded from here :
  https://johnvansickle.com/ffmpeg/

  The code :

  'use strict'
  
  process.env.PATH = process.env.PATH + ':/tmp/'
  
  process.env['FFMPEG_PATH'] = '/tmp/ffmpeg';
  
  const BIN_PATH = process.env['LAMBDA_TASK_ROOT']
  
  process.env['PATH'] = process.env['PATH'] + ':' + BIN_PATH;
  
  
  
  module.exports.download_mp3 = function (event, context, callback)
  
  {
  
    require('child_process').exec('cp /var/task/ffmpeg /tmp/.; chmod 755 
  
    /tmp/ffmpeg;', function (error, stdout, stderr) {
  
    if (error) 
  
    {
  
      console.log('An error occured', error);
  
      callback(null, null)
  
    } 
  
    else 
  
    {
  
      var ffmpeg = require('ffmpeg');
  
      const aws = require('aws-sdk')
  
      const s3 = new aws.S3()
  
      const ytdl = require('ytdl-core');
  
  
  
      function uploadFromStream(s3) {
  
        const stream = require('stream')
  
        var pass = new stream.PassThrough();
  
  
  
        var params = {Bucket: "some-bucket", Key: "some-key", Body: pass};
  
        s3.upload(params, function(err, data) {
  
          console.log(err, data);
  
        });
  
        console.log("Should be finished")
  
        callback(null)
  
      }
  
  
  
      let stream = ytdl("some-video-id", {
  
        quality: 'highestaudio',
  
        filter: 'audioonly'
  
      });
  
  
  
      ffmpeg(stream)
  
        .audioBitrate(128)
  
        .format('mp3')
  
        .on('error', (err) => console.error(err))
  
        .pipe(uploadFromStream(s3), {
  
          end: true
  
      });
  
    }})
  
  }

  When triggered, the function writes an error in logs :
  

  2019-01-04T14:50:54.525Z    21da4d49-1030-11e9-b901-0dc32b691a16     
  
  /var/task/ffmpeg:1
  
  (function (exports, require, module, __filename, __dirname) { ELF
  
  ^
  
  
  
  SyntaxError: Invalid or unexpected token
  
  at createScript (vm.js:80:10)
  
  at Object.runInThisContext (vm.js:139:10)
  
  at Module._compile (module.js:616:28)
  
  at Object.Module._extensions..js (module.js:663:10)
  
  at Module.load (module.js:565:32)
  
  at tryModuleLoad (module.js:505:12)
  
  at Function.Module._load (module.js:497:3)
  
  at Module.require (module.js:596:17)
  
  at require (internal/module.js:11:18)
  
  at /var/task/download.js:17:18

  It’s, most definetely, an error in the "ffmpeg" file I’ve mentioned above (link provided). But I don’t know what’s the exact issue, I followed the first answer here : https://stackoverflow.com/questions/47882810/lambda-not-connecting-to-ffmpeg to include the ffmpeg build.

• add black&silence to beginning of a video

  21 août 2020, par Richard

  Hi I am struggling to add black&silence to the begining of a video with ffmpeg. I did search a lot but they look too complex for me.
Below command is what I find to add black&silence to the end of of video, now how can I tune it to the beginning of a video ?

  


  ffmpeg -i input.mp4 -f lavfi -i color=s=1920x1080:d=10 -filter_complex [0:v][1]concat -af [0]apad -shortest output.mp4


  


  Looks I need to use adelay instead of apad, below is the command that makes sense to me, but the audio is not delayed.

  


  ffmpeg -i input.mp4 -f lavfi -i color=s=1920x1080:d=10 -filter_complex [1][0:v]concat -af [0]adelay=10  output.mp4


  


  Here is the input info and ffmpeg version :

  


  ffmpeg -i input.mp4 
ffmpeg version 4.2.1-static https://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
  configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --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. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.29.100
  Duration: 00:01:00.00, start: 0.000998, bitrate: 2526 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080, 2394 kb/s, 24 fps, 24 tbr, 16k tbn, 48 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 124 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
At least one output file must be specified


  


  Thanks !