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• FFmpeg get buffer from stream

  4 janvier 2020, par nonoom

  I have an express route, the user gives me a youtube url, with ytdl-core, i get the stream,
  and finally pass it to fluent-ffmpeg, and I need to get a buffer from the ffmpeg stream to finally send the picture to another api (i don’t save it)
  My code looks like :

  router.get("/youtubegifupload", (req,res)=>{ //get video youtube + informations --> create gif --> upload aws s3
  
  
  
      try{
  
      stream = ytdl("https://www.youtube.com/watch?v=xYQfk5275NA"); //i took random url for test purposes
  
      }catch(error){
  
        res.status(500).json({success : false, message: "Error while getting youtube video ! "})
  
      }
  
      buffer= []
  
      modifiedstream = ffmpeg(stream)
  
      .fps(30)
  
      .setStartTime(10)
  
      .duration(20)
  
      .noAudio()
  
      .videoCodec('gif')
  
      .format('gif')
  
      .save('./jpp3.gif')
  
      .on('progress', (prog)=>{
  
        console.log(prog);
  
      })
  
      .on('data',(buffer)=>{
  
      })
  
      .on('error', ((error)=>{
  
        res.status(500).json({success : false, message: "Error while modifying youtube video ! "})
  
      }))
  
      .on('end', ()=>{
  
          console.log("send to imgur")
  
          //request imgur - pass buffer as image
  
      }) 
  
  });

  Everything worked well with save(path) at the end of the ffmpeg operations, but it was for test purposes (actually i don’t want to save),
  Now, my goal is to get a buffer, so i tried to add .on(’data’, .. add the buffer to the array, and concat.
  But here is my problem :
  Without save(), it do not even launch progress(), ffmpeg is printing nothing, no end, ...

  How to make it work without save() ? How to get this stream and concat into a buffer ?

  *I tried : *

  -Verified imports, added const ffmpegPath = require(’@ffmpeg-installer/ffmpeg’).path ;

  -https://github.com/fluent-ffmpeg/node-fluent-ffmpeg/issues/747 - added videoCodec

  Thanks for your help

• Transcription via OpenAi's whisper : AssertionError : incorrect audio shape

  1er avril 2024, par muratowski

  I'm trying to use OpenAI's open source Whisper library to transcribe audio files.

  


  Here is my script's source code :

  


  import whisper

model = whisper.load_model("large-v2")

# load the entire audio file
audio = whisper.load_audio("/content/file.mp3")
#When i write that code snippet here ==> audio = whisper.pad_or_trim(audio) the first 30 secs are converted and without any problem they are converted.

# make log-Mel spectrogram and move to the same device as the model
mel = whisper.log_mel_spectrogram(audio).to(model.device)

# detect the spoken language
_, probs = model.detect_language(mel)
print(f"Detected language: {max(probs, key=probs.get)}")

# decode the audio
options = whisper.DecodingOptions(fp16=False)
result = whisper.decode(model, mel, options)

# print the recognized text if available
try:
    if hasattr(result, "text"):
        print(result.text)
except Exception as e:
    print(f"Error while printing transcription: {e}")

# write the recognized text to a file
try:
    with open("output_of_file.txt", "w") as f:
        f.write(result.text)
        print("Transcription saved to file.")
except Exception as e:
    print(f"Error while saving transcription: {e}")


  


  In here :

  


  # load the entire audio file
audio = whisper.load_audio("/content/file.mp3")


  


  when I write below : " audio = whisper.pad_or_trim(audio) ", the first 30 secs of the sound file is transcribed without any problem and language detection works as well,

  


  but when I delete it and want the whole file to be transcribed, I get the following error :

  


  


  AssertionError : incorrect audio shape

  


  


  What should I do ? Should I change the structure of the sound file ? If yes, which library should I use and what type of script should I write ?

  


• FFMPEG Multiple outputs from BlackMagic input

  24 septembre 2018, par leo_dragons

  I’m currently needing help to achieve multiple outputs from one input.
  Right now, the outputs are set like this :

  ffmpeg -re -f decklink -i "DeckLink Mini Recorder" -y -pix_fmt yuv420p -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 128k -ar 44100 -async 1 -b:v 2300k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 1280x720 -bufsize 1000k -maxrate 3072k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_720p"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 900k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 854x480 -bufsize 400k -maxrate 1000k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_480p_hq"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 550k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 854x480 -bufsize 400k -maxrate 500k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_480p_lq"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 450k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 640x360 -bufsize 400k -maxrate 500k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_360p" 

  This uses quite a lot of processing power, and also generates unnecessary latency (since I have to stream to WOWZA first, then back to FFMPEG and then back to WOWZA).

  And I want to optimize this.

  I’ve been trying several methods, but I only managed to overflow the decklink buffer. How could I solve this ?