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• Call to undefined method FFMpeg\Media\Video::addWatermark()

  13 avril 2023, par Amir Khan

  I m trying to add a watermark to a video that I have but it's giving me this error while applying a watermark

  


  The library is installed and working with the code commented but not when trying to add watermark

  


  use FFMpeg\FFMpeg;
use ProtoneMedia\LaravelFFMpeg\Filters\WatermarkFactory;
    
$ffmpeg = FFMpeg::create();
$video = $ffmpeg->open(public_path('video-making-test/test1.mp4'));
//    $video
//        ->filters()
//        ->resize(new \FFMpeg\Coordinate\Dimension(320, 240))
//        ->synchronize();
//    $video
//        ->frame(\FFMpeg\Coordinate\TimeCode::fromSeconds(10))
//        ->save(public_path('video-making-test/results/frame.jpg'));
    $video->addWatermark(function(WatermarkFactory $watermark) {
    $watermark->fromDisk('public')
        ->open('video-making-test/logo.png')
        ->right(25)
        ->bottom(25);
});


  


• Live streaming : node-media-server + Dash.js configured for real-time low latency

  7 juillet 2021, par Maoration

  We're working on an app that enables live monitoring of your back yard.
Each client has a camera connected to the internet, streaming to our public node.js server.

  



  I'm trying to use node-media-server to publish an MPEG-DASH (or HLS) stream to be available for our app clients, on different networks, bandwidths and resolutions around the world.

  



  Our goal is to get as close as possible to live "real-time" so you can monitor what happens in your backyard instantly.

  



  The technical flow already accomplished is :

  



  


  1. ffmpeg process on our server processes the incoming camera stream (separate child process for each camera) and publishes the stream via RTSP on the local machine for node-media-server to use as an 'input' (we are also saving segmented files, generating thumbnails, etc.). the ffmpeg command responsible for that is :

     



     -c:v libx264 -preset ultrafast -tune zerolatency -b:v 900k -f flv rtmp://127.0.0.1:1935/live/office

  


  2. node-media-server is running with what I found as the default configuration for 'live-streaming'

     



     private NMS_CONFIG = {
server: {
  secret: 'thisisnotmyrealsecret',
},
rtmp_server: {
  rtmp: {
    port: 1935,
    chunk_size: 60000,
    gop_cache: false,
    ping: 60,
    ping_timeout: 30,
  },
  http: {
    port: 8888,
    mediaroot: './server/media',
    allow_origin: '*',
  },
  trans: {
    ffmpeg: '/usr/bin/ffmpeg',
    tasks: [
      {
        app: 'live',
        hls: true,
        hlsFlags: '[hls_time=2:hls_list_size=3:hls_flags=delete_segments]',
        dash: true,
        dashFlags: '[f=dash:window_size=3:extra_window_size=5]',
      },
    ],
  },
},


     



     } ;

  


  3. As I understand it, out of the box NMS (node-media-server) publishes the input stream it gets in multiple output formats : flv, mpeg-dash, hls.
with all sorts of online players for these formats I'm able to access and the stream using the url on localhost. with mpeg-dash and hls I'm getting anything between 10-15 seconds of delay, and more.

  


  



  ---

  



  My goal now is to implement a local client-side mpeg-dash player, using dash.js and configure it to be as close as possible to live.

  



  my code for that is :

  



  


  


  &#xD;&#xA;&#xD;&#xA;    &#xD;&#xA;        &#xD;&#xA;        &#xD;&#xA;    &#xD;&#xA;    &#xD;&#xA;        <div>&#xD;&#xA;            <video autoplay="" controls=""></video>&#xD;&#xA;        </div>&#xD;&#xA;        <code class="echappe-js">&lt;script src=&quot;https://cdnjs.cloudflare.com/ajax/libs/dashjs/3.0.2/dash.all.min.js&quot;&gt;&lt;/script&gt;&#xD;&#xA;&#xD;&#xA;        &lt;script&gt;&amp;#xD;&amp;#xA;            (function(){&amp;#xD;&amp;#xA;                // var url = &quot;https://dash.akamaized.net/envivio/EnvivioDash3/manifest.mpd&quot;;&amp;#xD;&amp;#xA;                var url = &quot;http://localhost:8888/live/office/index.mpd&quot;;&amp;#xD;&amp;#xA;                var player = dashjs.MediaPlayer().create();&amp;#xD;&amp;#xA;                &amp;#xD;&amp;#xA;                &amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;                // config&amp;#xD;&amp;#xA;                targetLatency = 2.0;        // Lowering this value will lower latency but may decrease the player&amp;#x27;s ability to build a stable buffer.&amp;#xD;&amp;#xA;                minDrift = 0.05;            // Minimum latency deviation allowed before activating catch-up mechanism.&amp;#xD;&amp;#xA;                catchupPlaybackRate = 0.5;  // Maximum catch-up rate, as a percentage, for low latency live streams.&amp;#xD;&amp;#xA;                stableBuffer = 2;           // The time that the internal buffer target will be set to post startup/seeks (NOT top quality).&amp;#xD;&amp;#xA;                bufferAtTopQuality = 2;     // The time that the internal buffer target will be set to once playing the top quality.&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;                player.updateSettings({&amp;#xD;&amp;#xA;                    &amp;#x27;streaming&amp;#x27;: {&amp;#xD;&amp;#xA;                        &amp;#x27;liveDelay&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;liveCatchUpMinDrift&amp;#x27;: 0.05,&amp;#xD;&amp;#xA;                        &amp;#x27;liveCatchUpPlaybackRate&amp;#x27;: 0.5,&amp;#xD;&amp;#xA;                        &amp;#x27;stableBufferTime&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;bufferTimeAtTopQuality&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;bufferTimeAtTopQualityLongForm&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;bufferToKeep&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;bufferAheadToKeep&amp;#x27;: 2,&amp;#xD;&amp;#xA;                        &amp;#x27;lowLatencyEnabled&amp;#x27;: true,&amp;#xD;&amp;#xA;                        &amp;#x27;fastSwitchEnabled&amp;#x27;: true,&amp;#xD;&amp;#xA;                        &amp;#x27;abr&amp;#x27;: {&amp;#xD;&amp;#xA;                            &amp;#x27;limitBitrateByPortal&amp;#x27;: true&amp;#xD;&amp;#xA;                        },&amp;#xD;&amp;#xA;                    }&amp;#xD;&amp;#xA;                });&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;                console.log(player.getSettings());&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;                setInterval(() =&gt; {&amp;#xD;&amp;#xA;                  console.log(&amp;#x27;Live latency= &amp;#x27;, player.getCurrentLiveLatency());&amp;#xD;&amp;#xA;                  console.log(&amp;#x27;Buffer length= &amp;#x27;, player.getBufferLength(&amp;#x27;video&amp;#x27;));&amp;#xD;&amp;#xA;                }, 3000);&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;                player.initialize(document.querySelector(&quot;#videoPlayer&quot;), url, true);&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;            })();&amp;#xD;&amp;#xA;&amp;#xD;&amp;#xA;        &lt;/script&gt;&#xD;&#xA;    &#xD;&#xA;

  


  


  




  with the online test video (https://dash.akamaized.net/envivio/EnvivioDash3/manifest.mpd) I see that the live latency value is close to 2 secs (but I have no way to actually confirm it. it's a video file streamed. in my office I have a camera so I can actually compare latency between real-life and the stream I get).
however when working locally with my NMS, it seems this value does not want to go below 20-25 seconds.

  



  Am I doing something wrong ? any configuration on the player (client-side html) I'm forgetting ?
or is there a missing configuration I should add on the server side (NMS) ?

  


• How can a desktop node.js app play audio that may be controlled by the user like a media player ? [closed]

  19 février, par eedefeed

  I'm building a playlist manager that plays music. How can Node.JS play audio files quickly, reliably and with all the basic level features you would expect from a media player, namely :

  


  

  • play

  


  • pause

  


  • seek

  


  • stop

  


  • adjust volume

  


  


  I'm targetting windows/linux, but a Windows-only solution but be okay (for now.)

  


  I have tried a number of libraries and methods to play audio but it seems none of them are good enough :

  


  

  • Audic : it's reasonably good, but buggy. The play and pause functions sometimes get switched around. I also recall that there are some issues with uncaught exceptions somewhere in the dependencies that crash the entire app.

  


  • OBS : since the app is designed with broadcasting in mind, I've tried to use OBS's API to get it to play media. Unfortunately, it sometimes stops playback during some tracks, which is surprising since its underlying library, FFmpeg, plays them without issue.

  


  • node-groove : seems like its underlying library, libgroove, only supports linux. I can't find any builds to download, regardless.

  


  


  Attempts to use Speaker (which seems pretty good) have also failed because all the decoders have big issues :

  


  

  • Anything using lame - I want support for all audio, not specific formats.

  


  • fluent-FFmpeg - this is a wrapper around FFmpeg's CLI interface. It has no play/pause function, but bonus library fluent-FFmpeg-util adds this feature. Unfortunately, its pause takes about 4 seconds to work, which I'm guessing is to do with a buffer being exhausted. This is just too latent. Seek would also work by stopping the CLI process and reloading the file, which seems massively inefficient.

  


  • Node Vlc - promising but ancient library that gives me reams of node-gyp errors on install. Poorly documented and no explanation of what the library to do

  


  • VLC Client - this library has uncaught exceptions that crash the app. Wrapping in try/catch doesn't help.

  


  • sound play - doesn't support play/pause

  


  


  NPM's search function is filled with audio players designed to work in browsers, but I'm not building a web app. I guess it's an option but it seems inelegant to the point of rediculous.

  


  So it seems the best option centres around FFmpeg. FFmpeg has libraries, and I'm aware that node has ways to hook into those libraries via some sort of C or C++ compatibility layer. Unfortunately, official documentation is rather dense. Different unofficial guides seem to be recommending conflicting approaches (and might be dated), and it's difficult to work out whether myriad technologies are working in tandem or are alternatives, renames or replacements : node-gyp, node-api, addons, windows-build-tools, nan, C vs C++, Visual Studio. It's difficult to make any decisions or know where to start.

  


  Perhaps, also, another option is to use a Python library to interact with FFmpeg, since initial searches have indicated this might be possible. I wouldn't know whether this is a good option.

  


  So my question is : what's my best option to play audio ? Is it another NPM module that I'm not aware of ? Is it a compatibility layer with FFmpeg libraries ?