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• Progress with rtc.io

  12 août 2014, par silvia

  At the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.

  

  Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.

  

  One of the most exciting opportunities is still under-exploited : the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.

  Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.

  

  We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone ?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections :

  

  For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.

  For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.

  

  So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course : enjoy WebRTC ! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team !

  On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating !!

  The post Progress with rtc.io first appeared on ginger’s thoughts.

• How to encode specific metadata version in FFMPEG ?

  11 février 2021, par Charlie Britton

  I am batch converting lots of songs into shorter "Advert" songs for SHOUTcast and to be recognised as adverts by the server. The song must have ":Advert" for both the title and the artist metadata tags. When I use the following command :

  



  ffmpeg -i "$i" -c copy -vn -map_metadata -1 -metadata title=":Advert" -metadata artist=":Advert" -t 120 "adverts/ADVERT_$i"


  



  I would expect it to output the song with only ":Advert" as title and artist metadata but when I import it into the radio playout software (using ID3 1.x tagging) the metadata has not copied across and is therefore lost. Output from ffmpeg :

  



  ffmpeg version 3.0.2 Copyright (c) 2000-2016 the FFmpeg developers
  built with Apple LLVM version 9.0.0 (clang-900.0.37)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/3.0.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --disable-lzma --enable-vda
  libavutil      55. 17.103 / 55. 17.103
  libavcodec     57. 24.102 / 57. 24.102
  libavformat    57. 25.100 / 57. 25.100
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 31.100 /  6. 31.100
  libavresample   3.  0.  0 /  3.  0.  0
  libswscale      4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[mp3 @ 0x7feba6800000] Skipping 0 bytes of junk at 230934.
[mjpeg @ 0x7feba7000600] Changing bps to 8
Input #0, mp3, from 'Joakim Karud - Vibe With Me.mp3':
  Metadata:
    major_brand     : dash
    minor_version   : 0
    compatible_brands: iso6mp41
    encoder         : Lavf56.40.101
    artist          : Joakim Karud
    title           : Vibe With Me
  Duration: 00:02:53.06, start: 0.025056, bitrate: 138 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
    Metadata:
      encoder         : Lavc56.60
    Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1280x720 [SAR 1:1 DAR 16:9], 90k tbr, 90k tbn, 90k tbc
    Metadata:
      comment         : Cover (front)
Output #0, mp3, to 'adverts/ADVERT_Joakim Karud - Vibe With Me.mp3':
  Metadata:
    TIT2            : :Advert
    TPE1            : :Advert
    TSSE            : Lavf57.25.100
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    1876kB time=00:02:00.00 bitrate= 128.1kbits/s speed=1.44e+03x
video:0kB audio:1876kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.024837%


  



  I believe this is happening because the tag names are different (e.g. title should be title, but is TIT2 when output. Please could someone specify how I could ensure that the metadata is encoded in the ID3 1.x format so that it is readable by the radio playout software. Many thanks.

  


• ffmpeg itsoffset doesnt apply offset to mp4

  26 juillet 2019, par brux

  The following command mixes an mp4 and an mp3 together, keeping the audio from the mp4.

  ffmpeg -i video.mp4 -i audio.mp3 -map 0:v -c:v copy -filter_complex '[0:a][1:a]amix[aout]' -map '[aout]' -shortest out.mp4

  The command works as expected.

  Now I want to offset the mp4 file (both the audio and video stream of the mp4) so that there is a delay of 500ms at the start of the mp4, here is my command :

  ffmpeg -itsoffset 00:00:00.500 -i video.mp4 -i audio.mp3 -map 0:v -c:v copy -filter_complex '[0:a][1:a]amix[aout]' -map '[aout]' -shortest out.mp4

  This doesnt work as expected, the output doesnt have the expected delay of 500ms at the start of the mp4 streams. It appears the output is just the same as the first command I ran.

  The version of ffmpeg I am using is :

  ffmpeg version n4.0-39-gda39990

  Here are the files I’m using :

  video.mp4

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/video.mp4':
  
        Metadata:
  
          major_brand     : mp42
  
          minor_version   : 0
  
          compatible_brands: isommp42
  
          creation_time   : 2019-07-26T03:28:49.000000Z
  
          com.android.version: 8.0.0
  
        Duration: 00:00:07.64, start: 0.000000, bitrate: 20534 kb/s
  
          Stream #0:0(eng): Video: h264 (avc1 / 0x31637661), yuvj420p(pc, smpte170m), 1920x1080, 20966 kb/s, SAR 1:1 DAR 16:9, 29.70 fps, 29.75 tbr, 90k tbn, 180k tbc (default)
  
          Metadata:
  
            rotate          : 270
  
            creation_time   : 2019-07-26T03:28:49.000000Z
  
            handler_name    : VideoHandle
  
          Side data:
  
            displaymatrix: rotation of 90.00 degrees
  
          Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)
  
          Metadata:
  
            creation_time   : 2019-07-26T03:28:49.000000Z
  
            handler_name    : SoundHandle

  audio.mp3

  Input #0, mp3, from '/storage/emulated/0/audio.mp3':
  
        Metadata:
  
          album           : Sunflower (Spider-Man: Into the Spider-Verse) - Single
  
          composer        : Louis Bell, Carter Lang, Austin Richard Post, Billy Walsh & Khalif Malik Ibin Shaman Brown
  
          genre           : Hip-Hop/Rap
  
          copyright       : This Compilation ℗ 2018 Republic Records, a division of UMG Recordings, Inc.
  
          title           : Sunflower (Spider-Man: Into the Spider-Verse)
  
          artist          : Post Malone & Swae Lee
  
          album_artist    : Post Malone & Swae Lee
  
          track           : 01/01
  
          TYER            : 2018-10-18T07:00:00Z
  
        Duration: 00:02:38.07, start: 0.000000, bitrate: 325 kb/s
  
          Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
  
          Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 1400x1400 [SAR 300:300 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
          Metadata:
  
            comment         : Cover (front)