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• Why am I getting blips when encoding a sound file using Java JNA ?

  21 mars 2014, par yonran

  I have implemented a hello world libavcodec using JNA to generate a wav file containing a pure 440Hz sine wave. But when I actually run the program the wav file contains annoying clicks and blips (compare to pure sin wav created from the C program). How am I calling avcodec_encode_audio2 wrong ?

  Here is my Java code. All the sources are also at github in case you want to try to compile it.

  import java.io.IOException;
  
  import java.nio.ByteBuffer;
  
  import java.nio.ByteOrder;
  
  import java.nio.IntBuffer;
  
  import java.util.Objects;
  
  
  
  import javax.sound.sampled.AudioFormat;
  
  import javax.sound.sampled.AudioSystem;
  
  import javax.sound.sampled.DataLine;
  
  import javax.sound.sampled.LineUnavailableException;
  
  import javax.sound.sampled.TargetDataLine;
  
  
  
  
  
  public class Sin {
  
      /**
  
       * Abstract class that allows you to put the initialization and cleanup
  
       * code at the same place instead of separated by the big try block.
  
       */
  
      public static abstract class SharedPtr<t> implements AutoCloseable {
  
          public T ptr;
  
          public SharedPtr(T ptr) {
  
              this.ptr = ptr;
  
          }
  
          /**
  
           * Abstract override forces method to throw no checked exceptions.
  
           * Subclasses will call a C function that throws no exceptions.
  
           */
  
          @Override public abstract void close();
  
      }
  
  
  
      /**
  
       * @param args
  
       * @throws IOException 
  
       * @throws LineUnavailableException 
  
       */
  
      public static void main(String[] args) throws IOException, LineUnavailableException {
  
          final AvcodecLibrary avcodec = AvcodecLibrary.INSTANCE;
  
          final AvformatLibrary avformat = AvformatLibrary.INSTANCE;
  
          final AvutilLibrary avutil = AvutilLibrary.INSTANCE;
  
          avcodec.avcodec_register_all();
  
          avformat.av_register_all();
  
          AVOutputFormat.ByReference format = null;
  
          String format_name = "wav", file_url = "file:sinjava.wav";
  
          for (AVOutputFormat.ByReference formatIter = avformat.av_oformat_next(null); formatIter != null; formatIter = avformat.av_oformat_next(formatIter)) {
  
              formatIter.setAutoWrite(false);
  
              String iterName = formatIter.name;
  
              if (format_name.equals(iterName)) {
  
                  format = formatIter;
  
                  break;
  
              }
  
          }
  
          Objects.requireNonNull(format);
  
          System.out.format("Found format %s%n", format_name);
  
          AVCodec codec = avcodec.avcodec_find_encoder(format.audio_codec);  // one of AvcodecLibrary.CodecID
  
          Objects.requireNonNull(codec);
  
          codec.setAutoWrite(false);
  
          try (
  
              SharedPtr<avformatcontext> fmtCtxPtr = new SharedPtr<avformatcontext>(avformat.avformat_alloc_context()) {@Override public void close(){if (null!=ptr) avformat.avformat_free_context(ptr);}};
  
              ) {
  
              AVFormatContext fmtCtx = Objects.requireNonNull(fmtCtxPtr.ptr);
  
              fmtCtx.setAutoWrite(false);
  
              fmtCtx.setAutoRead(false);
  
              fmtCtx.oformat = format; fmtCtx.writeField("oformat");
  
  
  
              AVStream st = avformat.avformat_new_stream(fmtCtx, codec);
  
              if (null == st)
  
                  throw new IllegalStateException();
  
              AVCodecContext c = st.codec;
  
              if (null == c)
  
                  throw new IllegalStateException();
  
              st.setAutoWrite(false);
  
              fmtCtx.readField("nb_streams");
  
              st.id = fmtCtx.nb_streams - 1; st.writeField("id");
  
              assert st.id >= 0;
  
              System.out.format("New stream: id=%d%n", st.id);
  
  
  
              if (0 != (format.flags &amp; AvformatLibrary.AVFMT_GLOBALHEADER)) {
  
                  c.flags |= AvcodecLibrary.CODEC_FLAG_GLOBAL_HEADER;
  
              }
  
              c.writeField("flags");
  
  
  
              c.bit_rate = 64000; c.writeField("bit_rate");
  
              int bestSampleRate;
  
              if (null == codec.supported_samplerates) {
  
                  bestSampleRate = 44100;
  
              } else {
  
                  bestSampleRate = 0;
  
                  for (int offset = 0, sample_rate = codec.supported_samplerates.getInt(offset); sample_rate != 0; codec.supported_samplerates.getInt(++offset)) {
  
                      bestSampleRate = Math.max(bestSampleRate, sample_rate);
  
                  }
  
                  assert bestSampleRate > 0;
  
              }
  
              c.sample_rate = bestSampleRate; c.writeField("sample_rate");
  
              c.channel_layout = AvutilLibrary.AV_CH_LAYOUT_STEREO; c.writeField("channel_layout");
  
              c.channels = avutil.av_get_channel_layout_nb_channels(c.channel_layout); c.writeField("channels");
  
              assert 2 == c.channels;
  
              c.sample_fmt = AvutilLibrary.AVSampleFormat.AV_SAMPLE_FMT_S16; c.writeField("sample_fmt");
  
              c.time_base.num = 1;
  
              c.time_base.den = bestSampleRate;
  
              c.writeField("time_base");
  
              c.setAutoWrite(false);
  
  
  
              AudioFormat javaSoundFormat = new AudioFormat(bestSampleRate, Short.SIZE, c.channels, true, ByteOrder.nativeOrder() == ByteOrder.BIG_ENDIAN);
  
              DataLine.Info javaDataLineInfo = new DataLine.Info(TargetDataLine.class, javaSoundFormat);
  
              if (! AudioSystem.isLineSupported(javaDataLineInfo))
  
                  throw new IllegalStateException();
  
              int err;
  
              if ((err = avcodec.avcodec_open(c, codec)) &lt; 0) {
  
                  throw new IllegalStateException();
  
              }
  
              assert c.channels != 0;
  
  
  
              AVIOContext.ByReference[] ioCtxReference = new AVIOContext.ByReference[1];
  
              if (0 != (err = avformat.avio_open(ioCtxReference, file_url, AvformatLibrary.AVIO_FLAG_WRITE))) {
  
                  throw new IllegalStateException("averror " + err);
  
              }
  
              try (
  
                  SharedPtr ioCtxPtr = new SharedPtr(ioCtxReference[0]) {@Override public void close(){if (null!=ptr) avutil.av_free(ptr.getPointer());}}
  
                  ) {
  
                  AVIOContext.ByReference ioCtx = Objects.requireNonNull(ioCtxPtr.ptr);
  
                  fmtCtx.pb = ioCtx; fmtCtx.writeField("pb");
  
                  int averr = avformat.avformat_write_header(fmtCtx, null);
  
                  if (averr &lt; 0) {
  
                      throw new IllegalStateException("" + averr);
  
                  }
  
                  st.read();  // it is modified by avformat_write_header
  
                  System.out.format("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d%n", fmtCtx.nb_streams, st.time_base.num, st.time_base.den, st.avg_frame_rate.num, st.avg_frame_rate.den); 
  
                  avformat.avio_flush(ioCtx);
  
                  int frame_size = c.frame_size != 0 ? c.frame_size : 4096;
  
                  int expectedBufferSize = frame_size * c.channels * (Short.SIZE/8);
  
                  boolean supports_small_last_frame = c.frame_size == 0 ? true : 0 != (codec.capabilities &amp; AvcodecLibrary.CODEC_CAP_SMALL_LAST_FRAME);
  
                  int bufferSize = avutil.av_samples_get_buffer_size((IntBuffer)null, c.channels, frame_size, c.sample_fmt, 1);
  
                  assert bufferSize == expectedBufferSize: String.format("expected %d; got %d", expectedBufferSize, bufferSize);
  
                  ByteBuffer samples = ByteBuffer.allocate(expectedBufferSize);
  
                  samples.order(ByteOrder.nativeOrder());
  
                  int audio_time = 0;  // unit: (c.time_base) s = (1/c.sample_rate) s
  
                  int audio_sample_count = supports_small_last_frame ?
  
                      3 * c.sample_rate :
  
                      3 * c.sample_rate / frame_size * frame_size;
  
                  while (audio_time &lt; audio_sample_count) {
  
                      int frame_audio_time = audio_time;
  
                      samples.clear();
  
                      int nb_samples_in_frame = 0;
  
                      // encode a single tone sound
  
                      for (; samples.hasRemaining() &amp;&amp; audio_time &lt; audio_sample_count; nb_samples_in_frame++, audio_time++) {
  
                          double x = 2*Math.PI*440/c.sample_rate * audio_time;
  
                          double y = 10000 * Math.sin(x);
  
                          samples.putShort((short) y);
  
                          samples.putShort((short) y);
  
                      }
  
                      samples.flip();
  
                      try (
  
                              SharedPtr<avframe> framePtr = new SharedPtr<avframe>(avcodec.avcodec_alloc_frame()) {@Override public void close() {if (null!=ptr) avutil.av_free(ptr.getPointer());}};
  
                              ) {
  
                          AVFrame frame = Objects.requireNonNull(framePtr.ptr);
  
                          frame.setAutoRead(false);  // will be an in param
  
                          frame.setAutoWrite(false);
  
                          frame.nb_samples = nb_samples_in_frame; frame.writeField("nb_samples"); // actually unused during encoding
  
                          // Presentation time, in AVStream.time_base units.
  
                          frame.pts = avutil.av_rescale_q(frame_audio_time, c.time_base, st.time_base);  // i * codec_time_base / st_time_base
  
                          frame.writeField("pts");
  
  
  
                          assert c.channels > 0;
  
                          int bytesPerSample = avutil.av_get_bytes_per_sample(c.sample_fmt);
  
                          assert bytesPerSample > 0;
  
                          if (0 != (err = avcodec.avcodec_fill_audio_frame(frame, c.channels, c.sample_fmt, samples, samples.capacity(), 1))) {
  
                              throw new IllegalStateException(""+err);
  
                          }
  
                          AVPacket packet = new AVPacket();  // one of the few structs from ffmpeg with guaranteed size
  
                          avcodec.av_init_packet(packet);
  
                          packet.size = 0;
  
                          packet.data = null;
  
                          packet.stream_index = st.index; packet.writeField("stream_index");
  
                          // encode the samples
  
                          IntBuffer gotPacket = IntBuffer.allocate(1);
  
                          if (0 != (err = avcodec.avcodec_encode_audio2(c, packet, frame, gotPacket))) {
  
                              throw new IllegalStateException("" + err);
  
                          } else if (0 != gotPacket.get()) {
  
                              packet.read();
  
                              averr = avformat.av_write_frame(fmtCtx, packet);
  
                              if (averr &lt; 0)
  
                                  throw new IllegalStateException("" + averr);
  
                          }
  
                          System.out.format("encoded frame: codec time = %d; pts=%d = av_rescale_q(%d,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d%n",
  
                                  frame_audio_time,
  
                                  frame.pts,
  
                                  frame_audio_time, st.codec.time_base.num,st.codec.time_base.den,st.time_base.num,st.time_base.den,
  
                                  1.*frame_audio_time/c.sample_rate, frame.nb_samples, 1.*frame.nb_samples/c.sample_rate, gotPacket.array()[0], packet.size);
  
                      }
  
                  }
  
                  if (0 != (err = avformat.av_write_trailer(fmtCtx))) {
  
                      throw new IllegalStateException();
  
                  }
  
                  avformat.avio_flush(ioCtx);
  
              }
  
          }
  
          System.out.println("Done writing");
  
      }
  
  }
  
  </avframe></avframe></avformatcontext></avformatcontext></t>

  I also rewrote it in C, and the C version works fine without any blips. But I can’t figure out how I am using the library differently ; all the library function calls should be identical !

  //! gcc --std=c99 sin.c $(pkg-config --cflags --libs libavutil libavformat libavcodec) -o sin
  
  // sudo apt-get install libswscale-dev
  
  #include 
  
  #include 
  
  #include 
  
  #include 
  
  
  
  #include <libavutil></libavutil>opt.h>
  
  #include <libavutil></libavutil>mathematics.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  int main(int argc, char *argv[]) {
  
    const char *format_name = "wav", *file_url = "file:sin.wav";
  
    avcodec_register_all();
  
    av_register_all();
  
    AVOutputFormat *format = NULL;
  
    for (AVOutputFormat *formatIter = av_oformat_next(NULL); formatIter != NULL; formatIter = av_oformat_next(formatIter)) {
  
      int hasEncoder = NULL != avcodec_find_encoder(formatIter->audio_codec);
  
      if (0 == strcmp(format_name, formatIter->name)) {
  
        format = formatIter;
  
        break;
  
      }
  
    }
  
    printf("Found format %s\n", format->name);
  
    AVCodec *codec = avcodec_find_encoder(format->audio_codec);
  
    if (! codec) {
  
      fprintf(stderr, "Could not find codec %d\n", format->audio_codec);
  
      exit(1);
  
    }
  
    AVFormatContext *fmtCtx = avformat_alloc_context();
  
    if (! fmtCtx) {
  
      fprintf(stderr, "error allocating AVFormatContext\n");
  
      exit(1);
  
    }
  
    fmtCtx->oformat = format;
  
    AVStream *st = avformat_new_stream(fmtCtx, codec);
  
    if (! st) {
  
      fprintf(stderr, "error allocating AVStream\n");
  
      exit(1);
  
    }
  
    if (fmtCtx->nb_streams != 1) {
  
      fprintf(stderr, "avformat_new_stream should have incremented nb_streams, but it&#39;s still %d\n", fmtCtx->nb_streams);
  
      exit(1);
  
    }
  
    AVCodecContext *c = st->codec;
  
    if (! c) {
  
      fprintf(stderr, "avformat_new_stream should have allocated a AVCodecContext for my stream\n");
  
      exit(1);
  
    }
  
    st->id = fmtCtx->nb_streams - 1;
  
    printf("Created stream %d\n", st->id);
  
    if (0 != (format->flags &amp; AVFMT_GLOBALHEADER)) {
  
      c->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
    }
  
    c->bit_rate = 64000;
  
    int bestSampleRate;
  
    if (NULL == codec->supported_samplerates) {
  
      bestSampleRate = 44100;
  
      printf("Setting sample rate: %d\n", bestSampleRate);
  
    } else {
  
      bestSampleRate = 0;
  
      for (const int *sample_rate_iter = codec->supported_samplerates; *sample_rate_iter != 0; sample_rate_iter++) {
  
        if (*sample_rate_iter >= bestSampleRate)
  
          bestSampleRate = *sample_rate_iter;
  
      }
  
      printf("Using best supported sample rate: %d\n", bestSampleRate);
  
    }
  
    c->sample_rate = bestSampleRate;
  
    c->channel_layout = AV_CH_LAYOUT_STEREO;
  
    c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
  
    c->time_base.num = 1;
  
    c->time_base.den = c->sample_rate;
  
    if (c->channels != 2) {
  
      fprintf(stderr, "av_get_channel_layout_nb_channels returned %d instead of 2\n", c->channels);
  
      exit(1);
  
    }
  
    c->sample_fmt = AV_SAMPLE_FMT_S16;
  
    int averr;
  
    if ((averr = avcodec_open2(c, codec, NULL)) &lt; 0) {
  
      fprintf(stderr, "avcodec_open2 returned error %d\n", averr);
  
      exit(1);
  
    }
  
    AVIOContext *ioCtx = NULL;
  
    if (0 != (averr = avio_open(&amp;ioCtx, file_url, AVIO_FLAG_WRITE))) {
  
      fprintf(stderr, "avio_open returned error %d\n", averr);
  
      exit(1);
  
    }
  
    if (ioCtx == NULL) {
  
      fprintf(stderr, "AVIOContext should have been set by avio_open\n");
  
      exit(1);
  
    }
  
    fmtCtx->pb = ioCtx;
  
    if (0 != (averr = avformat_write_header(fmtCtx, NULL))) {
  
      fprintf(stderr, "avformat_write_header returned error %d\n", averr);
  
      exit(1);
  
    }
  
    printf("Wrote header. fmtCtx->nb_streams=%d, st->time_base=%d/%d; st->avg_frame_rate=%d/%d\n", fmtCtx->nb_streams, st->time_base.num, st->time_base.den, st->avg_frame_rate.num, st->avg_frame_rate.den);
  
    int align = 1;
  
    int sample_size = av_get_bytes_per_sample(c->sample_fmt);
  
    if (sample_size != sizeof(int16_t)) {
  
      fprintf(stderr, "expected sample size=%zu but got %d\n", sizeof(int16_t), sample_size);
  
      exit(1);
  
    }
  
    int frame_size = c->frame_size != 0 ? c->frame_size : 4096;
  
    int bufferSize = av_samples_get_buffer_size(NULL, c->channels, frame_size, c->sample_fmt, align);
  
    int expectedBufferSize = frame_size * c->channels * sample_size;
  
    int supports_small_last_frame = c->frame_size == 0 ? 1 : 0 != (codec->capabilities &amp; CODEC_CAP_SMALL_LAST_FRAME);
  
    if (bufferSize != expectedBufferSize) {
  
      fprintf(stderr, "expected buffer size=%d but got %d\n", expectedBufferSize, bufferSize);
  
      exit(1);
  
    }
  
    int16_t *samples = (int16_t*)malloc(bufferSize);
  
  
  
    uint32_t audio_time = 0;  // unit: (1/c->sample_rate) s
  
    uint32_t audio_sample_count = supports_small_last_frame ?
  
      3 * c->sample_rate :
  
      3 * c->sample_rate / frame_size * frame_size;
  
    while (audio_time &lt; audio_sample_count) {
  
      uint32_t frame_audio_time = audio_time; // unit: (1/c->sample_rate) s
  
      AVFrame *frame = avcodec_alloc_frame();
  
      if (frame == NULL) {
  
        fprintf(stderr, "avcodec_alloc_frame failed\n");
  
        exit(1);
  
      }
  
      for (uint32_t i = 0; i != frame_size &amp;&amp; audio_time &lt; audio_sample_count; i++, audio_time++) {
  
        samples[2*i] = samples[2*i + 1] = 10000 * sin(2*M_PI*440/c->sample_rate * audio_time);
  
        frame->nb_samples = i+1;  // actually unused during encoding
  
      }
  
      // frame->format = c->sample_fmt;  // unused during encoding
  
      frame->pts = av_rescale_q(frame_audio_time, c->time_base, st->time_base);
  
      if (0 != (averr = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*)samples, bufferSize, align))) {
  
        fprintf(stderr, "avcodec_fill_audio_frame returned error %d\n", averr);
  
        exit(1);
  
      }
  
      AVPacket packet;
  
      av_init_packet(&amp;packet);
  
      packet.data = NULL;
  
      packet.size = 0;
  
      int got_packet;
  
      if (0 != (averr = avcodec_encode_audio2(c, &amp;packet, frame, &amp;got_packet))) {
  
        fprintf(stderr, "avcodec_encode_audio2 returned error %d\n", averr);
  
        exit(1);
  
      }
  
      if (got_packet) {
  
          packet.stream_index = st->index;
  
        if (0 &lt; (averr = av_write_frame(fmtCtx, &amp;packet))) {
  
          fprintf(stderr, "av_write_frame returned error %d\n", averr);
  
          exit(1);
  
        } else if (averr == 1) {
  
          // end of stream wanted.
  
        }
  
      }
  
      printf("encoded frame: codec time = %u; format pts=%ld = av_rescale_q(%u,%d/%d,%d/%d) (%.02fs) contains %d samples (%.02fs); got_packet=%d; packet.size=%d\n",
  
          frame_audio_time,
  
          frame->pts,
  
          frame_audio_time, c->time_base.num, c->time_base.den, st->time_base.num, st->time_base.den,
  
          1.*frame_audio_time/c->sample_rate, frame->nb_samples, 1.*frame->nb_samples/c->sample_rate, got_packet, packet.size);
  
      av_free(frame);
  
    }
  
    free(samples);
  
    cleanupFile:
  
    if (0 != (averr = av_write_trailer(fmtCtx))) {
  
      fprintf(stderr, "av_write_trailer returned error %d\n", averr);
  
      exit(1);
  
    }
  
  
  
    avio_flush(ioCtx);
  
    avio_close(ioCtx);
  
    avformat_free_context(fmtCtx);
  
  }

• The New Samples Regime

  1er décembre 2011, par Multimedia Mike — General

  A little while ago, I got a big head over the fact that I owned and controlled the feared and revered MPlayer samples archive. This is the repository that retains more than a decade of multimedia samples.

  Conflict
  Where once there was one multimedia project (FFmpeg), there are now 2 (also Libav). There were various political and technical snafus regarding the previous infrastructure. I volunteered to take over hosting the vast samples archive (53 GB at the time) at samples.mplayerhq.hu (s.mphq for this post).

  However, a brand new server is online at samples.libav.org (s.libav for this post).

  Policies
  The server at s.libav will be the authoritative samples repository going forward. Why does s.libav receive the honor ? Mostly by virtue of having more advanced features. My simple (yet bandwidth-rich) web hosting plan does not provide for rsync or anonymous FTP services, both of which have traditionally been essential for the samples server. In the course of hosting s.mphq for the past few months, a few more discrepancies have come to light– apparently, the symlinks weren’t properly replicated. And perhaps most unusual is that if a directory contains a README file, it won’t be displayed in the directory listing (which frustrated me greatly when I couldn’t find this README file that I carefully and lovingly crafted years ago).

  The s.mphq archive will continue to exist — nay, must exist — going forward since there are years’ worth of web links pointing into it. I’ll likely set up a mirroring script that periodically (daily) rsyncs from s.libav to my local machine and then uses lftp (the best facility I have available) to mirror the files up to s.mphq.

  Also, since we’re starting fresh with a new upload directory, I think we need to be far more ruthless about policing its content. This means making sure that anything that is uploaded has an accompanying file which explains why it’s there and ideally links the sample to a bug report somewhere. No explanation = sample terminated.

  RSS
  I think it would be nifty to have an RSS feed that shows the latest samples to appear in the repository. I figure that I can use the Unix ‘find’ command on my local repository in concert with something like PyRSS2Gen to accomplish this goal.

  Monetization
  In the few months that I have been managing the repository, I have had numerous requests for permission to leech the entire collection in one recursive web-suck. These requests often from commercial organizations who wish to test their multimedia product on a large corpus of diverse samples. Personally, I believe the archive makes a rather poor corpus for such an endeavor, but so be it. Go ahead ; hosting this archive barely makes a dent in my fairly low-end web hosting plan. However, at least one person indicated that it might be easier to mail a hard drive to me, have me copy it, and send it back.

  This got me thinking about monetization opportunities. Perhaps, I should provide a service to send HDs filled with samples for the cost of the HD, shipping, and a small donation to the multimedia projects. I immediately realized that that is precisely the point at which the vast multimedia samples archive — with all of its media of questionable fair use status — would officially run afoul of copyright laws.

  Which brings me to…

  Clean Up
  I think we need to clean up some samples, starting with the ones that were marked not-readable in the old repository. Apparently, some ‘samples’ were, e.g., full anime videos and were responsible for a large bandwidth burden when linked from various sources.

  We multimedia nerds are a hoarding lot, never willing to throw anything away. This will probably the most challenging proposal to implement.

• Using FFMpeg with Runtime.exec() to do a simple transcoding

  1er novembre 2011, par Adam Ingmansson

  I know there are many questions touching this subject, but none have helped me solve my issue.

  Purpose :

  Transcoding a video taken,from a queue, from .mov to h.264 (for now only that)

  Solution :

  Building a java application that gets the next in the queue, transcodes it then repeat

  Problem :

  Running ffmpeg using Runtime.exec() is not working.
  Im using the StreamGobbler from this tutorial to capturing the output from the process.

  This code shows how i start my process :

  String[] command = new String[]{"./ffmpeg/ffmpeg","-i",videoFile.getPath(),"-vcodec","libx264","-fpre",preset,folder + recID + ".flv"};
  
  System.out.println("Running command..");
  
  Process p = r.exec(command);
  
  
  
   // any error message?
  
  StreamGobbler errorGobbler = new 
  
  StreamGobbler(p.getErrorStream(), "ERROR");            
  
  
  
  // any output?
  
  StreamGobbler outputGobbler = new 
  
  StreamGobbler(p.getInputStream(), "OUT");
  
  
  
  // kick them off
  
  errorGobbler.start();
  
  outputGobbler.start();
  
  
  
  //logProcessOutputAndErrors(p);
  
  
  
  int res = p.waitFor();
  
  if (res != 0) {
  
      throw new Exception("Encoding error: "+String.valueOf(res));
  
  }

  and this is the current modified version of StreamGobbler (the important part)

  InputStreamReader isr = new InputStreamReader(is);
  
  BufferedReader br = new BufferedReader(isr);
  
  String line=null;
  
  int c = 0;
  
  StringBuilder str = new StringBuilder();
  
  
  
  while (true) {
  
      c = br.read();
  
  }

  Sometimes ffmpeg just stalls, maybe waiting for my input (although there is no indication on screen).

  Sometimes it just ends.

  Sometimes (when I added the line "System.out.print((char) c) ;" in the while-loop above) i got loads of "¿¿ï" repeated over and over again, wich might be the actual encoding of the video wich I managed to capture instead of to a file.

  For those who wonders why i dont just go with a commandline or maybe even php :

  The purpose is an application that will run 24/7 transcoding anything and everything from a queue. The files are pretty large to begin with and takes about 15 min to transcode.