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• How to convert an mp3 file to raw audio format using javascript

  4 octobre 2022, par Lord Bee

  I'm working on a project that involves song matching, so I am integrating with rapidApi's shazam endpoints. But the thing is, the song matching endpoint needs the audio snippet to be a base64 string of the audio in raw audio format. I know the API works. I downloaded a 3rd party application to do the conversion from mp3 to .raw, and converted it to base64 before making the request with it.

  


  Now, I need to integrate this flow programmatically. How do I convert an mp3 or any audio source really to a .raw file ? I've done a lot of searching but I can't find any solution.

  


• Python / Discord FFMPEG playing not a audio

  19 août 2022, par F L R U Z

  I wanted to play an mp3 file in a voice channel, however, it gives me an error... I have already installed FFMPEG correctly on my computer.

  


  My Code :

  


  @bot.command()
async def joinandplay(ctx):
   song = os.path.isfile("musik.mp3")
   voice_channel = bot.get_channel(940326390857883648)
   vc = await voice_channel.connect()
   voice = discord.FFmpegPCMAudio(song)
   vc.play(voice)


  


  The error :

  


  AttributeError: 'FFmpegPCMAudio' object has no attribute '_process'


  


• FFMPEG YouTube Live Too Fast

  6 avril 2017, par Liam Martens

  So I am streaming video and audio to YouTube as follows

  THE CONTEXT

  1. First I convert a graphic GIF to an MP4 file

  ffmpeg -f gif -i graphic.gif -c:v libx264 -pix_fmt yuv420p -vf scale=1280:-1 temp.mp4

  2. Then I overlay a PNG with text over the MP4

  ffmpeg -i temp.mp4 -i overlay.png -filter_complex "overlay=10:10" '.$graphicsPath.'/graphic.mp4

  3. Then I start the stream the video and combine it with audio using following code (sources.txt is just a concat list *)

  ffmpeg -f concat -i sources.txt -i music.mp3 \
  
      -c:v libx264 -c:a aac -shortest -deinterlace \
  
      -pix_fmt yuv420p -preset '.$encoding.' -r 30 -g 60 -b:v 2500k \
  
      -acodec libmp3lame -ar 44100 -threads 6 -qscale 3 -b:a 712000 \
  
      -maxrate 800k -bufsize 1400k \
  
      -f flv rtmp://a.rtmp.youtube.com/live2/KEY

  4. After the stream ends, the code starts over again with a new song to mimic a 247 stream.

  THE ISSUE

  So the issue I am having is that it appears to be streaming too fast. It’s like the opposite of buffering issues where the buffer is way too long (as in a full song buffered by the time the first one has finished if you open the stream)

  Does anyone know how I could throttle the output ? I have tried with maxrate and bufsize but no real result.

  * sources.txt example
  file ’graphic.mp4’
  file ’graphic.mp4’
  file ’graphic.mp4’
  file ’graphic.mp4’
  file ’graphic.mp4’
  file ’graphic.mp4’
  and so on