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• WebRTC Multi-Stream recording

  11 janvier 2021, par Tim Specht

  I'm currently trying to build a WebRTC streaming architecture that contains multiple users streaming content from their camera in the same "room" and a SFU / MCU on server-side "recording" the incoming video packets, merging them into one image and re-distributing them to the viewers as either RTMP or HLS for added scalability.

  


  Upon doing some initial research on this, Janus Gateway seems like a good fit for this given it's wide adoption across the space + their (seemingly) extensible plugin architecture. Thus, I'm currently trying to figure out what a recommended architecture for my use-case would look like.
I looked at the following plugins :

  


  

  • Janus Streaming

  


  • Janus Recordings

  


  


  While Janus and the Streaming plugin seem like a good start to get the broadcasting aspect within the group of casters in the room, I'm trying to piece together how I could combine the different video sources into a combined one (split horizontally for example if there are 2 casters active) and retransmit the final result as something optimized for broadcast-consumption like HLS. Some of the ways I could imagine doing that :

  


  

  • Implement a custom Janus plugin that transcodes the incoming buffers on the gateway itself

  


  • Forwarding the incoming packets via RTP to a Transcoding server

    

    • In this specific case I am not sure what would be best to implement that ? Are the video frames different tracks ? Could I stream all of them to the same port and have ffmpeg or something similar take care of the merging for me ?

    


    


  


  


• YouTube-DL Python details of extracted audio file are not displayed

  16 septembre 2020, par Sushil

  I have written a small piece of code in python to extract the audio from a YouTube video. Here is the code :

  


  from __future__ import unicode_literals
import youtube_dl

link = input("Enter the video link:")

ydl_opts = {
    'format': 'bestaudio/best',
    'postprocessors': [{
        'key': 'FFmpegExtractAudio',
        'preferredcodec': 'mp3',
        'preferredquality': '192',
    }],
}
with youtube_dl.YoutubeDL(ydl_opts) as ydl:
    info_dict = ydl.extract_info(link, download=False)
    video_title = info_dict.get('title', None)

path = f'D:\\{video_title}.mp3'

ydl_opts.update({'outtmpl':path})

with youtube_dl.YoutubeDL(ydl_opts) as ydl:
    ydl.download([link])


  


  This is the folder where the output audio file is saved :


  


  As you can see, all the details of the audio file are displayed, such as Date Modified, Type and Size.

  


  However, if I change path = f'D:\\{video_title}.mp3' to path = f'D:\\YT_Files\\{video_title}.mp3', then the file details are not getting displayed.


  


  Any idea about why this is so ? Is there any way to solve this problem ? Any help would be appreciated. Thanks.

  


• Audio offset get wrong after some time when streaming audios

  7 septembre 2024, par Antoine Grenard

  I use microsoft-cognitiveservices-speech-sdk (1.38.0) in order to do real time speech to text.
It seems like the offset is right when I send a full audio but it is wrong when I send it cut in a lot of audio chunks.

  


  The more there is audio chunks the more inaccurate the offset is :

  


  

  • No chunks : 1 726 300 000

  


  • 369 chunks of 0.5 seconds : 1 729 600 000

  


  • 923 chunks of 0.2 seconds : 1 744 600 000

  


  • 1443 chunks of 0.1 seconds : 1 757 900 000

  


  


  To reproduce here is some piece of code :

  


      const speechConfig = SpeechConfig.fromSubscription(<key>,  {console.log(event)}&#xA;    speechRecognizer.canceled = async (recognizer, event) => {console.log(event)}&#xA;    speechRecognizer.startContinuousRecognitionAsync();&#xA;&#xA;    for (let i = 1; i &lt;= 1443; i&#x2B;&#x2B;) {&#xA;      const formattedNumber = i.toString().padStart(4, &#x27;0&#x27;);&#xA;      const buffer = fs.readFileSync(`/var/tmp/chunks/output_${formattedNumber}.wav`);&#xA;      pushStream.write(buffer);&#xA;    }&#xA;</key>

  


  To create the audio chunks :

  


  ffmpeg -i  -f segment -segment_time 0.1 -c copy output_%04d.wav


  


  Here is the audio link : https://drive.google.com/file/d/1H_RJuqMiBaVkpo9XHrgp1bpuFdgQl64O/view?usp=sharing

  


  Thanks for your help