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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 hangs when -vcodec copy specified (called from Java via ProcessBuilder)

  24 juin 2015, par Inglonias

  I’m trying to use ffmpeg to export an array of bytes to a video file, but the people I work with insist that I use -vcodec copy in the arguments for it. This, however, causes the code to hang, whereas if I don’t use -vcodec copy, the code will not hang. I don’t know what the problem is, and I’ve been trying to debug this code for the past two hours.

  Here is the relevant section of code. I’ve added comments above and below the line where the code hangs. Can anybody help me ?

          // This is the tricky part. We need to build an ffmpeg process that
  
          // takes input from stdin, and then plug Java into that.
  
          ProcessBuilder ffmpegBuilder = new ProcessBuilder();
  
          String[] cmd = {"ffmpeg", "-i", "-","-vcodec", "copy", directory
  
                  + "/" + fileName};
  
          StringBuilder combinedCmd = new StringBuilder();
  
          for (String s : cmd) {
  
              combinedCmd.append(s);
  
              combinedCmd.append(" ");
  
          }
  
          mLogger.log(Level.INFO,"Final command is " + combinedCmd.toString());
  
          ffmpegBuilder.command(cmd);
  
          ffmpegBuilder.redirectErrorStream(true); // So that stdout and stderr go
  
                                                      // to the same stream.
  
          byte[] dataToWrite = new byte[data.size()];
  
          for (int i = 0; i < dataToWrite.length; i++) {
  
              dataToWrite[i] = data.get(i); // Is there really STILL no better way
  
                                              // to convert an ArrayList to an
  
                                              // array?!
  
          }
  
          try {
  
              Process ffmpeg = ffmpegBuilder.start();
  
              OutputStream stdin = ffmpeg.getOutputStream();
  
              BufferedReader stdout = new BufferedReader(new InputStreamReader(
  
                      ffmpeg.getInputStream()));
  
  //HANGS AT THIS LINE vvvvvvvvvvvvvvvv
  
              stdin.write(dataToWrite);
  
  //HANGS AT THIS LINE ^^^^^^^^^^^^^^^^
  
  
  
              String line = "I know a song that gets on everybody's nerves...";
  
              while ((line != null) && stdout.ready()) {
  
                  line = stdout.readLine();
  
                  mLogger.log(Level.INFO, line);
  
              }
  
              try {
  
                  ffmpeg.waitFor(2, TimeUnit.SECONDS);
  
                  ffmpeg.destroyForcibly();
  
              } catch (InterruptedException e) {
  
                  // TODO Auto-generated catch block
  
                  e.printStackTrace();
  
              }
  
          } catch (IOException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }

• Audio playing back at the wrong speed using FFmpeg on Android

  11 avril 2019, par Kyborg2011

  General problem in the following :
  I decode the audio as follows :

  ReSampleContext* rsc = av_audio_resample_init(
  
          1, aCodecCtx->channels,
  
          aCodecCtx->sample_rate, aCodecCtx->sample_rate,
  
          av_get_sample_fmt("u8"), aCodecCtx->sample_fmt,
  
          16, 10, 0, 1);
  
  
  
  while (av_read_frame(pFormatCtx, &packet)>= 0) {
  
      if (aCodecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {
  
          int data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE * 2;
  
          int size=packet.size;
  
          int decoded = 0;
  
  
  
           while(size > 0) {
  
               int len = avcodec_decode_audio3(aCodecCtx, pAudioBuffer, 
  
                   &data_size, &packet);
  
  
  
               //Сonvert audio to sample 8bit
  
               out_size = audio_resample(rsc, outBuffer, pAudioBuffer, len);
  
  
  
               jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
  
  
               memcpy(bytes, outBuffer, out_size);
  
               (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
               (*env)->CallStaticVoidMethod(env, cls, mid, array, out_size, number);
  
  
  
               size -= len;
  
               number++;
  
           }
  
       }
  
   }

  Next release it AudioTrack. After that, I hear that song that was necessary, but with noise and speed of 2 times larger. In what may be the problem ?

  UPDATE :
  This is Java code :

  public static AudioTrack track;
  
  public static byte[] bytes;
  
  protected void onCreate(Bundle savedInstanceState) {
  
      super.onCreate(savedInstanceState);
  
      setContentView(R.layout.main);
  
      int bufSize = 2048;
  
      track = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_OUT_MONO, 
  
                  AudioFormat.ENCODING_PCM_8BIT, bufSize, AudioTrack.MODE_STREAM); 
  
  
  
      bytes = new byte[bufSize];
  
      Thread mAudioThread = new Thread(new Runnable() {
  
          public void run() {
  
              int res = main(2, "/sdcard/muzika_iz_reklami_bmw_5_series_-_bmw_5_series.mp3", bytes);
  
              System.out.println(res);
  
          } 
  
      });
  
      mAudioThread.setPriority(Thread.MAX_PRIORITY);
  
      mAudioThread.start(); 
  
  }
  
  
  
  private static void play(byte[] play, int length, int p) {
  
      if (p==0){
  
          track.play();
  
      }
  
      track.write(play, 0, length); 
  
  }