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• Node.js Stream Mp3 to http without having to save file

  21 août 2016, par user2758113

  I am trying to stream just audio from a youtube link straight to http with node.js.

  My code looks like this, I am using express 4.0.

  var express = require('express');
  
  var router = express.Router();
  
  var ytdl = require('ytdl');
  
  var ffmpeg = require('fluent-ffmpeg');
  
  var fs = require('fs');
  
  
  
  router.get('/', function(req, res) {
  
  
  
    var url = 'https://www.youtube.com/watch?v=GgcHlZsOgQo';
  
    var video = ytdl(url)
  
  
  
    res.set({
  
        "Content-Type": "audio/mpeg"
  
    })
  
  
  
    new ffmpeg({source: video})
  
        .toFormat('mp3')
  
        .writeToStream(res, function(data, err) {
  
          if (err) console.log(err)
  
        })
  
  
  
  });
  
  
  
  module.exports = router;

  Now, I’m able to stream the video’s audio to the response if I save the file then pipe it to the response, but I’d rather try to figure out some way to go from downloading to ffmpeg to response.

  Not sure if this is possible. The main goal is to keep it as light weight as possible, and not have to read from files.

  I’ve seen this code which is essentially what I’d like to do minus the saving to a file part.

  

• ffpmeg force overwrite (badly) detected input framerate

  20 juin 2020, par Longoon12000

  I am running IP Webcam on Android which provides an mpjpeg video stream. I have to limit the capture frame rate to 5fps to save on battery.

  


  However ffmpeg will still detect the input stream to be 25 fps, which causes it to be saved in the wrong speed causing timestamps and audio to be desynchronized.

  


  Input #0, mpjpeg, from 'https://***:***@smarthome:8080/video':
  Duration: N/A, bitrate: N/A
    Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1280x720 [SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc
Input #1, ogg, from 'https://***:***@smarthome:8080/audio.opus':
  Duration: N/A, start: 0.006500, bitrate: N/A
    Stream #1:0: Audio: opus, 48000 Hz, mono, fltp
    Metadata:
      ENCODER         : Lavf58.12.100
[stream_segment,ssegment @ 0x19b49a0] Opening '/mnt/nas/SecurityCamera/2020-06-20_14-26-04.mkv' for writing
Output #0, stream_segment,ssegment, to '/mnt/nas/SecurityCamera/%Y-%m-%d_%H-%M-%S.mkv':
  Metadata:
    encoder         : Lavf58.20.100
    Stream #0:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 25 tbr, 1k tbn, 25 tbc
    Stream #0:1: Audio: opus, 48000 Hz, mono, fltp
    Metadata:
      ENCODER         : Lavf58.12.100
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
[ogg @ 0x1753e00] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
frame=    8 fps=7.9 q=-1.0 size=N/A time=00:00:00.49 bitrate=N/A speed=0.489x


  


  As you can see it detects Input #0 to be 25 tbr, 25 tbn, 25 tbc which results in an expected 25 fps, however the fps (shown high right now but it slowly approaching 5fps) is way below 25 which causes the speed to be <1x.

  &#xA;

  I have tried to use -r 5 -i ... and -vsync 2 and different values for -enc_time_base none of which had any impact. From https://trac.ffmpeg.org/ticket/403 I've learned that -r only works on inputs with unknown fps. But my input doesn't have unknown fps, it has the wrong fps.

  &#xA;

  Is there any way to force overwrite the input fps so that I can get a proper speed of 1x and synchronized timestamps and audio ?

  &#xA;

• avcodec/av1dec : parse and export Metadata OBUs

  5 mars 2023, par James Almer

  avcodec/av1dec : parse and export Metadata OBUs
  This includes Mastering Display, Content light level, and some ITU-T T35
  
  metadata like closed captions and HDR10+.
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] libavcodec/av1dec.c
  • [DH] libavcodec/av1dec.h