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• C# How do I set the volume of sound bytes[]

  23 juillet 2016, par McLucario

  Im trying to change the volume of sound bytes[] in C#. Im reading a sound file with FFMPEG and and wanna change the volume on the fly. I found some examples and but I didnt understand them.

  public void SendAudio(string pathOrUrl)
  
  {
  
      cancelVid = false;
  
      isPlaying = true;
  
  
  
      mProcess = Process.Start(new ProcessStartInfo
  
      { // FFmpeg requireqs us to spawn a process and hook into its stdout, so we will create a Process
  
          FileName = "ffmpeg",
  
          Arguments = "-i " + (char)34 + pathOrUrl + (char)34 + // Here we provide a list of arguments to feed into FFmpeg. -i means the location of the file/URL it will read from
  
          " -f s16le -ar 48000 -ac 2 pipe:1", // Next, we tell it to output 16-bit 48000Hz PCM, over 2 channels, to stdout.
  
          UseShellExecute = false,
  
          RedirectStandardOutput = true, // Capture the stdout of the process
  
          Verb = "runas"
  
      });
  
  
  
      while (!isRunning(mProcess)) { Task.Delay(1000); }
  
  
  
      int blockSize = 3840; // The size of bytes to read per frame; 1920 for mono
  
      byte[] buffer = new byte[blockSize];
  
      byte[] gainBuffer = new byte[blockSize];
  
      int byteCount; 
  
  
  
      while (true && !cancelVid) // Loop forever, so data will always be read
  
      {
  
          byteCount = mProcess.StandardOutput.BaseStream // Access the underlying MemoryStream from the stdout of FFmpeg
  
          .Read(buffer, 0, blockSize); // Read stdout into the buffer
  
  
  
          if (byteCount == 0) // FFmpeg did not output anything
  
              break; // Break out of the while(true) loop, since there was nothing to read.
  
  
  
          if (cancelVid)
  
              break;
  
  
  
          disAudioClient.Send(buffer, 0, byteCount); // Send our data to Discord
  
      }
  
      disAudioClient.Wait(); // Wait for the Voice Client to finish sending data, as ffMPEG may have already finished buffering out a song, and it is unsafe to return now.
  
      isPlaying = false;
  
      Console.Clear();
  
      Console.WriteLine("Done Playing!");

• ffmpeg concat 2 avi files loosing the sound

  14 juin 2016, par Nir Diamant

  Hey i have 2 AVI files that i want to concatenate them to one output.

  the files path is written in text file (mylist.txt) :

  file1.avi - contain video only. file2.avi - contain video and audio.

  i execute the next command :

  ffmpeg -f concat -i C:\ffmpeg\mylist.txt -c copy C:\ffmpeg\output.avi

  mylist.txt=

  file ’C :\motionbee\ffmpeg\bin\file1.avi’

  file ’C :\motionbee\ffmpeg\bin\file2.avi’

  the output file is concatenate of the 2 files. but the sound from file2.avi is not there.

  so how can i contact them without delete the sound ???

  console log :

  C:\motionbee\ffmpeg\bin>ffmpeg -f concat -i C:\motionbee\ffmpeg\bin\mylist.txt -
  
  c copy C:\motionbee\ffmpeg\bin\output.avi
  
  ffmpeg version N-78949-g6f5048f Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
  
  ibilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopenc
  
  ore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --ena
  
  ble-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable
  
  -libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --ena
  
  ble-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx
  
  264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable
  
  -lzma --enable-decklink --enable-zlib
  
    libavutil      55. 19.100 / 55. 19.100
  
    libavcodec     57. 27.101 / 57. 27.101
  
    libavformat    57. 28.100 / 57. 28.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 39.100 /  6. 39.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, concat, from 'C:\motionbee\ffmpeg\bin\mylist.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 523 kb/s
  
      Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 720
  
  x480 [SAR 32:27 DAR 16:9], 523 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 30k tbc
  
  File 'C:\motionbee\ffmpeg\bin\output.avi' already exists. Overwrite ? [y/N] y
  
  Output #0, avi, to 'C:\motionbee\ffmpeg\bin\output.avi':
  
    Metadata:
  
      ISFT            : Lavf57.28.100
  
      Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 720x480 [SAR 32:27 D
  
  AR 16:9], q=2-31, 523 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [concat @ 0000000000452620] New audio stream 0:1 at pos:10038 and DTS:100.104s
  
  frame= 6000 fps=0.0 q=-1.0 Lsize=   13020kB time=00:03:20.23 bitrate= 532.7kbits
  
  /s speed=1.39e+003x
  
  video:12870kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing
  
   overhead: 1.159987%

• Envelope pattern in SoX (Sound eXchange) or ffmpeg

  26 mai 2016, par pJay

  I’ve been using SoX to generate white noise. I’m after a way of modulating the volume across the entire track in a way that will create a pattern similar to this :

  White noise envelope effect

  I’ve experimented with fade, but that fades in to 100% volume and fades out to 0% volume, which is just a pain in this instance.

  The tremolo effect isn’t quite what I’m after either, as the frequency of the pattern will be changing over time.

  The only other alternative is to split the white noise file into separate files, apply fade and then apply trim to either end so it doesn’t fade all the way, but this seems like a lot of unnecessary processing.

  I’ve been checking out this example Using SoX to change the volume level of a range of time in an audio file, but I don’t think it’s quite what I’m after.

  I’m using the command-line in Ubuntu with SoX, but I’m open to suggestions with ffmpeg, or any other Linux based command-line solution.