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• Why is my Youtube Video not downloading completely ? "Input buffer exhausted, packet corrupt"

  15 septembre 2021, par user16909319

  Introduction :

  


  I'm working on a Discord Music Bot for personal use only. Basically, when I play a 2min song with the bot, it plays the song for 1min 30secs, then skips it and plays the next one in the queue. The error is shown below :

  


  


  Error in Pull Function
  
IO error : Error number -10054 occurred
  
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000018f86a6f4c0] Packet corrupt (stream = 0, dts = 11154432).
  
Input buffer exhausted before END element found
  
Invalid Data found when processing Input
  
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000018f86a6f4c0] stream 0, offset 0x3e805c : partial file

  


  


  The code block which I think is causing the problem :

  


  Search song method

  


      async def search_song(self, amount, song, get_url=False):
        info = await self.bot.loop.run_in_executor(None, lambda: youtube_dl.YoutubeDL(ytdl_format_options).extract_info(
            f"ytsearch{amount}:{song}", download=False, ie_key="YoutubeSearch"))
        if len(info["entries"]) == 0: return None

        return [entry["webpage_url"] for entry in info["entries"]] if get_url else info


  


  Play_song method

  


      async def play_song(self, ctx, song):
        url = pafy.new(song).getbestaudio().url
        ctx.voice_client.play(discord.PCMVolumeTransformer(discord.FFmpegPCMAudio(url, executable="C:/ffmpeg/bin/ffmpeg.exe")),
                              after=lambda error: self.bot.loop.create_task(self.check_queue(ctx)))
        ctx.voice_client.source.volume = 0.5


  


  Formatting Options I provided :

  


  ytdl_format_options = {
    'format': 'bestaudio/best',
    'outtmpl': '%(extractor)s-%(id)s-%(title)s.%(ext)s',
    'restrictfilenames': True,
    'noplaylist': True,
    'nocheckcertificate': True,
    'ignoreerrors': True,
    'logtostderr': False,
    'quiet': True,
    'no_warnings': True,
    'default_search': 'auto',
    'source_address': '0.0.0.0'
}


  


  Solutions that I've tried :

  


  

  • Running it on both Replit and locally.

  


  • Redownloading FFmpeg

  


  • Ensuring FFmpeg, pafy, and youtube_dl are all up to date.

  


  


  Things to Note :

  


  

  • Playing a 2mins song, it stops after 1min 30 seconds and displays the error above. (75% of the song)

  


  • Playing a 1hr song, it still continues after 10 minutes.

  


  


  I do not have much experience in this yet so I'm not entirely sure where in my code is actually causing this issue or other ways which I can use to test and fix the issue.

  


• Using JavaCV/FFMPEG to push byte buffer image via RTMP

  6 mai 2022, par ljnoah

  I have a USB camera that returns frames as a byte buffer type to which I would like to push/send via rtmp using JavaCV library to be viewed using VLC. According to this link : http://bytedeco.org/javacpp-presets/ffmpeg/apidocs/org/bytedeco/ffmpeg/ffmpeg.html. It should be possible, but I don't really have any experience with ffmpeg, but from what I've read in their github it should be possible, only, thus far I was not able to find an example on how to do it. Basically, I would like to store a byte buffer and send an image that is in variable instead of grabbing frames from a camera as my USB camera is not detected by android and needs external libraries to work.

  


  private byte[] FrameData = new byte[384 * 288 * 4];
  //private final Bitmap bitmap = Bitmap.createBitmap(PREVIEW_WIDTH, PREVIEW_HEIGHT, // Bitmap.Config.ARGB_8888); // creates a bitmap with the proper format in-case its needed
  private final IFrameCallback mIFrameCallback = new IFrameCallback() {
      @Override
      public void onFrame(final ByteBuffer frameData) {
          frameData.clear();
          frameData.get(FrameData, 0, frameData.capacity());
          }
  };


  


  This callback gives me the frames from my camera on the fly and writes it to FrameData, which I can compress to a bitmap if needed.

  


  My camera preview where I call the callback above :

  


    public void handleStartPreview(Object surface) {
      if ((mUVCCamera == null)) return;
      try {
          mUVCCamera.setPreviewSize(mWidth, mHeight, 1, 26, UVCCamera.DEFAULT_PREVIEW_MODE, UVCCamera.DEFAULT_BANDWIDTH, 0);
      } catch (IllegalArgumentException e) {
          try {
              mUVCCamera.setPreviewSize(mWidth, mHeight, 1, 26, UVCCamera.DEFAULT_PREVIEW_MODE, UVCCamera.DEFAULT_BANDWIDTH, 0);
              Log.e(TAG, "handleStartPreview4");
          } catch (IllegalArgumentException e1) {
              callOnError(e1);
              return;
          }
      }
      int result = mUVCCamera.startPreview();
      mUVCCamera.setFrameCallback(mIFrameCallback, UVCCamera.PIXEL_FORMAT_RGBX);
      mUVCCamera.startCapture();
      startRecording();
  }


  


  My question is How can I use this example :

  


   String ffmpeg = Loader.load(org.bytedeco.ffmpeg.ffmpeg.class);
 ProcessBuilder pb = new ProcessBuilder(ffmpeg, "-i", "/path/to/input.mp4", "-vcodec", "h264", "/path/to/output.mp4");
 pb.inheritIO().start().waitFor();


  


  to push my frames from the camera that are stored FrameData byte buffer via RTMP/RTSP to my server IP, even If I need to compress it to a bitmap before...

  


• How to record/trim/combine audio seamlessly in a React/Node web app

  16 mai 2021, par Rayhan Memon

  I've developed a digital audio workstation for the browser that you can check out here. Essentially it helps authors narrate their own audiobooks themselves at home.

  


  I'm looking to dramatically improve the speed at which audio files are combined or trimmed.

  


  Right now, the user records some variable amount of audio (a line, paragraph, or entire passage). When the user stops recording, this clip is added to the main audio file for the section using ffmpeg.wasm like so :

  


          if (duration === 0) {
            //concatenate the two files (they should already be written to memory)
            await ffmpeg.run('-i', 'concat:fullAudio.mp3|clip.mp3', '-c', 'copy', 'fullAudio.mp3');

        } else {
            //Set the insert time to wherever the user's cursor is positioned
            let insertTime = duration;
            if (selectedObj) {
                insertTime = selectedObj.endTime;
            }

            //split the audio file into two parts at the point we want to insert the audio
            await ffmpeg.run('-i', 'fullAudio.mp3', '-t', `${insertTime}`, '-c', 'copy', 'part1.mp3', '-ss',  `${insertTime}`, '-codec', 'copy', 'part2.mp3');

            //concatenate the three files
            await ffmpeg.run('-i', 'concat:part1.mp3|clip.mp3', '-acodec', 'copy', 'intermediate.mp3');
            await ffmpeg.run('-i', 'concat:intermediate.mp3|part2.mp3', '-acodec', 'copy', 'fullAudio.mp3');
        }

        //Read the result from memory
        const data = ffmpeg.FS('readFile', 'fullAudio.mp3');

        //Create URL so it can be used in the browser
        const url = URL.createObjectURL(new Blob([data.buffer], { type: 'audio/mp3' }));
        globalDispatch({ type: "setFullAudioURL", payload: url });


  


  After every recorded clip, the user is forced to wait a few seconds for this concatenation process to finish up - and the longer the main file or recorded clip gets, the longer the user has to wait. Looking at other browser-based audio editors such as AudioMass, it clearly seems possible to make a seamless recording and editing experience with zero wait time, but I can't seem to figure out how to do the same within my react app.

  


  Is it possible to seamlessly combine or trim audio data within a React app ? Is FFMPEG the best way to go about it, or are there simpler ways using pure javascript ?