コード例 #1
0
ファイル: RecorderRtpImpl.java プロジェクト: natib/libjitsi
/**
 * A <tt>Recorder</tt> implementation which attaches to an <tt>RTPTranslator</tt>.
 *
 * @author Vladimir Marinov
 * @author Boris Grozev
 */
public class RecorderRtpImpl
    implements Recorder, ReceiveStreamListener, ActiveSpeakerChangedListener, ControllerListener {
  /**
   * The <tt>Logger</tt> used by the <tt>RecorderRtpImpl</tt> class and its instances for logging
   * output.
   */
  private static final Logger logger = Logger.getLogger(RecorderRtpImpl.class);

  // values hard-coded to match chrome
  // TODO: allow to set them dynamically
  private static final byte redPayloadType = 116;
  private static final byte ulpfecPayloadType = 117;
  private static final byte vp8PayloadType = 100;
  private static final byte opusPayloadType = 111;
  private static final Format redFormat = new VideoFormat(Constants.RED);
  private static final Format ulpfecFormat = new VideoFormat(Constants.ULPFEC);
  private static final Format vp8RtpFormat = new VideoFormat(Constants.VP8_RTP);
  private static final Format vp8Format = new VideoFormat(Constants.VP8);
  private static final Format opusFormat =
      new AudioFormat(Constants.OPUS_RTP, 48000, Format.NOT_SPECIFIED, Format.NOT_SPECIFIED);

  private static final int FMJ_VIDEO_JITTER_BUFFER_MIN_SIZE = 300;

  /** The <tt>ContentDescriptor</tt> to use when saving audio. */
  private static final ContentDescriptor AUDIO_CONTENT_DESCRIPTOR =
      new ContentDescriptor(FileTypeDescriptor.MPEG_AUDIO);

  /** The suffix for audio file names. */
  private static final String AUDIO_FILENAME_SUFFIX = ".mp3";

  /** The suffix for video file names. */
  private static final String VIDEO_FILENAME_SUFFIX = ".webm";

  static {
    Registry.set("video_jitter_buffer_MIN_SIZE", FMJ_VIDEO_JITTER_BUFFER_MIN_SIZE);
  }

  /** The <tt>RTPTranslator</tt> that this recorder is/will be attached to. */
  private RTPTranslatorImpl translator;

  /**
   * The custom <tt>RTPConnector</tt> that this instance uses to read from {@link #translator} and
   * write to {@link #rtpManager}.
   */
  private RTPConnectorImpl rtpConnector;

  /** Path to the directory where the output files will be stored. */
  private String path;

  /** The <tt>RTCPFeedbackMessageSender</tt> that we use to send RTCP FIR messages. */
  private RTCPFeedbackMessageSender rtcpFeedbackSender;

  /**
   * The {@link RTPManager} instance we use to handle the packets coming from
   * <tt>RTPTranslator</tt>.
   */
  private RTPManager rtpManager;

  /**
   * The instance which should be notified when events related to recordings (such as the start or
   * end of a recording) occur.
   */
  private RecorderEventHandlerImpl eventHandler;

  /**
   * Holds the <tt>ReceiveStreams</tt> added to this instance by {@link #rtpManager} and additional
   * information associated with each one (e.g. the <tt>Processor</tt>, if any, used for it).
   */
  private final HashSet<ReceiveStreamDesc> receiveStreams = new HashSet<ReceiveStreamDesc>();

  private final Set<Long> activeVideoSsrcs = new HashSet<Long>();

  /**
   * The <tt>ActiveSpeakerDetector</tt> which will listen to the audio receive streams of this
   * <tt>RecorderRtpImpl</tt> and notify it about changes to the active speaker via calls to {@link
   * #activeSpeakerChanged(long)}
   */
  private ActiveSpeakerDetector activeSpeakerDetector = null;

  StreamRTPManager streamRTPManager;

  private SynchronizerImpl synchronizer;
  private boolean started = false;

  /**
   * Constructor.
   *
   * @param translator the <tt>RTPTranslator</tt> to which this instance will attach in order to
   *     record media.
   */
  public RecorderRtpImpl(RTPTranslator translator) {
    this.translator = (RTPTranslatorImpl) translator;
    activeSpeakerDetector = new ActiveSpeakerDetectorImpl();
    activeSpeakerDetector.addActiveSpeakerChangedListener(this);
  }

  /** Implements {@link Recorder#addListener(Recorder.Listener)}. */
  @Override
  public void addListener(Listener listener) {}

  /** Implements {@link Recorder#removeListener(Recorder.Listener)}. */
  @Override
  public void removeListener(Listener listener) {}

  /** Implements {@link Recorder#getSupportedFormats()}. */
  @Override
  public List<String> getSupportedFormats() {
    return null;
  }

  /** Implements {@link Recorder#setMute(boolean)}. */
  @Override
  public void setMute(boolean mute) {}

  /**
   * Implements {@link Recorder#getFilename()}. Returns null, since we don't have a (single)
   * associated filename.
   */
  @Override
  public String getFilename() {
    return null;
  }

  /**
   * Sets the instance which should be notified when events related to recordings (such as the start
   * or end of a recording) occur.
   */
  public void setEventHandler(RecorderEventHandler eventHandler) {
    if (this.eventHandler == null
        || (this.eventHandler != eventHandler && this.eventHandler.handler != eventHandler)) {
      if (this.eventHandler == null) this.eventHandler = new RecorderEventHandlerImpl(eventHandler);
      else this.eventHandler.handler = eventHandler;
    }
  }

  /**
   * {@inheritDoc}
   *
   * @param format unused, since this implementation records multiple streams using potentially
   *     different formats.
   * @param dirname the path to the directory into which this <tt>Recorder</tt> will store the
   *     recorded media files.
   */
  @Override
  public void start(String format, String dirname) throws IOException, MediaException {
    if (logger.isInfoEnabled()) logger.info("Starting, format=" + format + " " + hashCode());
    path = dirname;

    MediaService mediaService = LibJitsi.getMediaService();

    /*
     * Note that we use only one RTPConnector for both the RTPTranslator
     * and the RTPManager instances. The this.translator will write to its
     * output streams, and this.rtpManager will read from its input streams.
     */
    rtpConnector = new RTPConnectorImpl(redPayloadType, ulpfecPayloadType);

    rtpManager = RTPManager.newInstance();

    /*
     * Add the formats that we know about.
     */
    rtpManager.addFormat(vp8RtpFormat, vp8PayloadType);
    rtpManager.addFormat(opusFormat, opusPayloadType);
    rtpManager.addReceiveStreamListener(this);

    /*
     * Note: When this.rtpManager sends RTCP sender/receiver reports, they
     * will end up being written to its own input stream. This is not
     * expected to cause problems, but might be something to keep an eye on.
     */
    rtpManager.initialize(rtpConnector);

    /*
     * Register a fake call participant.
     * TODO: can we use a more generic MediaStream here?
     */
    streamRTPManager =
        new StreamRTPManager(
            mediaService.createMediaStream(
                new MediaDeviceImpl(new CaptureDeviceInfo(), MediaType.VIDEO)),
            translator);

    streamRTPManager.initialize(rtpConnector);

    rtcpFeedbackSender = translator.getRtcpFeedbackMessageSender();

    translator.addFormat(streamRTPManager, opusFormat, opusPayloadType);

    // ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, redFormat,
    // redPayloadType);
    // ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager, ulpfecFormat,
    // ulpfecPayloadType);
    // ((RTPTranslatorImpl)videoRTPTranslator).addFormat(streamRTPManager,
    // mediaFormatImpl.getFormat(), vp8PayloadType);

    started = true;
  }

  @Override
  public void stop() {
    if (started) {
      if (logger.isInfoEnabled()) logger.info("Stopping " + hashCode());

      // remove the recorder from the translator (e.g. stop new packets from
      // being written to rtpConnector
      if (streamRTPManager != null) streamRTPManager.dispose();

      HashSet<ReceiveStreamDesc> streamsToRemove = new HashSet<ReceiveStreamDesc>();
      synchronized (receiveStreams) {
        streamsToRemove.addAll(receiveStreams);
      }

      for (ReceiveStreamDesc r : streamsToRemove) removeReceiveStream(r, false);

      rtpConnector.rtcpPacketTransformer.close();
      rtpConnector.rtpPacketTransformer.close();
      rtpManager.dispose();

      started = false;
    }
  }

  /**
   * Implements {@link ReceiveStreamListener#update(ReceiveStreamEvent)}.
   *
   * <p>{@link #rtpManager} will use this to notify us of <tt>ReceiveStreamEvent</tt>s.
   */
  @Override
  public void update(ReceiveStreamEvent event) {
    if (event == null) return;
    ReceiveStream receiveStream = event.getReceiveStream();

    if (event instanceof NewReceiveStreamEvent) {
      if (receiveStream == null) {
        logger.warn("NewReceiveStreamEvent: null");
        return;
      }

      final long ssrc = getReceiveStreamSSRC(receiveStream);

      ReceiveStreamDesc receiveStreamDesc = findReceiveStream(ssrc);

      if (receiveStreamDesc != null) {
        String s = "NewReceiveStreamEvent for an existing SSRC. ";
        if (receiveStream != receiveStreamDesc.receiveStream)
          s += "(but different ReceiveStream object)";
        logger.warn(s);
        return;
      } else receiveStreamDesc = new ReceiveStreamDesc(receiveStream);

      if (logger.isInfoEnabled()) logger.info("New ReceiveStream, ssrc=" + ssrc);

      // Find the format of the ReceiveStream
      DataSource dataSource = receiveStream.getDataSource();
      if (dataSource instanceof PushBufferDataSource) {
        Format format = null;
        PushBufferDataSource pbds = (PushBufferDataSource) dataSource;
        for (PushBufferStream pbs : pbds.getStreams()) {
          if ((format = pbs.getFormat()) != null) break;
        }

        if (format == null) {
          logger.error("Failed to handle new ReceiveStream: " + "Failed to determine format");
          return;
        }

        receiveStreamDesc.format = format;
      } else {
        logger.error("Failed to handle new ReceiveStream: " + "Unsupported DataSource");
        return;
      }

      int rtpClockRate = -1;
      if (receiveStreamDesc.format instanceof AudioFormat)
        rtpClockRate = (int) ((AudioFormat) receiveStreamDesc.format).getSampleRate();
      else if (receiveStreamDesc.format instanceof VideoFormat) rtpClockRate = 90000;
      getSynchronizer().setRtpClockRate(ssrc, rtpClockRate);

      // create a Processor and configure it
      Processor processor = null;
      try {
        processor = Manager.createProcessor(receiveStream.getDataSource());
      } catch (NoProcessorException npe) {
        logger.error("Failed to create Processor: ", npe);
        return;
      } catch (IOException ioe) {
        logger.error("Failed to create Processor: ", ioe);
        return;
      }

      if (logger.isInfoEnabled()) logger.info("Created processor for SSRC=" + ssrc);

      processor.addControllerListener(this);
      receiveStreamDesc.processor = processor;

      final int streamCount;
      synchronized (receiveStreams) {
        receiveStreams.add(receiveStreamDesc);
        streamCount = receiveStreams.size();
      }

      /*
       * XXX TODO IRBABOON
       * This is a terrible hack which works around a failure to realize()
       * some of the Processor-s for audio streams, when multiple streams
       * start nearly simultaneously. The cause of the problem is currently
       * unknown (and synchronizing all FMJ calls in RecorderRtpImpl
       * does not help).
       * XXX TODO NOOBABRI
       */
      if (receiveStreamDesc.format instanceof AudioFormat) {
        final Processor p = processor;
        new Thread() {
          @Override
          public void run() {
            // delay configuring the processors for the different
            // audio streams to decrease the probability that they
            // run together.
            try {
              int ms = 450 * (streamCount - 1);
              logger.warn(
                  "Sleeping for "
                      + ms
                      + "ms before"
                      + " configuring processor for SSRC="
                      + ssrc
                      + " "
                      + System.currentTimeMillis());
              Thread.sleep(ms);
            } catch (Exception e) {
            }

            p.configure();
          }
        }.run();
      } else {
        processor.configure();
      }
    } else if (event instanceof TimeoutEvent) {
      if (receiveStream == null) {
        // TODO: we might want to get the list of ReceiveStream-s from
        // rtpManager and compare it to our list, to see if we should
        // remove a stream.
        logger.warn("TimeoutEvent: null.");
        return;
      }

      // FMJ silently creates new ReceiveStream instances, so we have to
      // recognize them by the SSRC.
      ReceiveStreamDesc receiveStreamDesc = findReceiveStream(getReceiveStreamSSRC(receiveStream));
      if (receiveStreamDesc != null) {
        if (logger.isInfoEnabled()) {
          logger.info("ReceiveStream timeout, ssrc=" + receiveStreamDesc.ssrc);
        }

        removeReceiveStream(receiveStreamDesc, true);
      }
    } else if (event != null && logger.isInfoEnabled()) {
      logger.info("Unhandled ReceiveStreamEvent (" + event.getClass().getName() + "): " + event);
    }
  }

  private void removeReceiveStream(ReceiveStreamDesc receiveStream, boolean emptyJB) {
    if (receiveStream.format instanceof VideoFormat) {
      rtpConnector.packetBuffer.disable(receiveStream.ssrc);
      emptyPacketBuffer(receiveStream.ssrc);
    }

    if (receiveStream.dataSink != null) {
      try {
        receiveStream.dataSink.stop();
      } catch (IOException e) {
        logger.error("Failed to stop DataSink " + e);
      }

      receiveStream.dataSink.close();
    }

    if (receiveStream.processor != null) {
      receiveStream.processor.stop();
      receiveStream.processor.close();
    }

    DataSource dataSource = receiveStream.receiveStream.getDataSource();
    if (dataSource != null) {
      try {
        dataSource.stop();
      } catch (IOException ioe) {
        logger.warn("Failed to stop DataSource");
      }
      dataSource.disconnect();
    }

    synchronized (receiveStreams) {
      receiveStreams.remove(receiveStream);
    }
  }

  /**
   * Implements {@link ControllerListener#controllerUpdate(ControllerEvent)}. Handles events from
   * the <tt>Processor</tt>s that this instance uses to transcode media.
   *
   * @param ev the event to handle.
   */
  public void controllerUpdate(ControllerEvent ev) {
    if (ev == null || ev.getSourceController() == null) {
      return;
    }

    Processor processor = (Processor) ev.getSourceController();
    ReceiveStreamDesc desc = findReceiveStream(processor);

    if (desc == null) {
      logger.warn("Event from an orphaned processor, ignoring: " + ev);
      return;
    }

    if (ev instanceof ConfigureCompleteEvent) {
      if (logger.isInfoEnabled()) {
        logger.info(
            "Configured processor for ReceiveStream ssrc="
                + desc.ssrc
                + " ("
                + desc.format
                + ")"
                + " "
                + System.currentTimeMillis());
      }

      boolean audio = desc.format instanceof AudioFormat;

      if (audio) {
        ContentDescriptor cd = processor.setContentDescriptor(AUDIO_CONTENT_DESCRIPTOR);
        if (!AUDIO_CONTENT_DESCRIPTOR.equals(cd)) {
          logger.error(
              "Failed to set the Processor content "
                  + "descriptor to "
                  + AUDIO_CONTENT_DESCRIPTOR
                  + ". Actual result: "
                  + cd);
          removeReceiveStream(desc, false);
          return;
        }
      }

      for (TrackControl track : processor.getTrackControls()) {
        Format trackFormat = track.getFormat();

        if (audio) {
          final long ssrc = desc.ssrc;
          SilenceEffect silenceEffect;
          if (Constants.OPUS_RTP.equals(desc.format.getEncoding())) {
            silenceEffect = new SilenceEffect(48000);
          } else {
            // We haven't tested that the RTP timestamps survive
            // the journey through the chain when codecs other than
            // opus are in use, so for the moment we rely on FMJ's
            // timestamps for non-opus formats.
            silenceEffect = new SilenceEffect();
          }

          silenceEffect.setListener(
              new SilenceEffect.Listener() {
                boolean first = true;

                @Override
                public void onSilenceNotInserted(long timestamp) {
                  if (first) {
                    first = false;
                    // send event only
                    audioRecordingStarted(ssrc, timestamp);
                  } else {
                    // change file and send event
                    resetRecording(ssrc, timestamp);
                  }
                }
              });
          desc.silenceEffect = silenceEffect;
          AudioLevelEffect audioLevelEffect = new AudioLevelEffect();
          audioLevelEffect.setAudioLevelListener(
              new SimpleAudioLevelListener() {
                @Override
                public void audioLevelChanged(int level) {
                  activeSpeakerDetector.levelChanged(ssrc, level);
                }
              });

          try {
            // We add an effect, which will insert "silence" in
            // place of lost packets.
            track.setCodecChain(new Codec[] {silenceEffect, audioLevelEffect});
          } catch (UnsupportedPlugInException upie) {
            logger.warn("Failed to insert silence effect: " + upie);
            // But do go on, a recording without extra silence is
            // better than nothing ;)
          }
        } else {
          // transcode vp8/rtp to vp8 (i.e. depacketize vp8)
          if (trackFormat.matches(vp8RtpFormat)) track.setFormat(vp8Format);
          else {
            logger.error("Unsupported track format: " + trackFormat + " for ssrc=" + desc.ssrc);
            // we currently only support vp8
            removeReceiveStream(desc, false);
            return;
          }
        }
      }

      processor.realize();
    } else if (ev instanceof RealizeCompleteEvent) {
      desc.dataSource = processor.getDataOutput();

      long ssrc = desc.ssrc;
      boolean audio = desc.format instanceof AudioFormat;
      String suffix = audio ? AUDIO_FILENAME_SUFFIX : VIDEO_FILENAME_SUFFIX;

      // XXX '\' on windows?
      String filename = getNextFilename(path + "/" + ssrc, suffix);
      desc.filename = filename;

      DataSink dataSink;
      if (audio) {
        try {
          dataSink = Manager.createDataSink(desc.dataSource, new MediaLocator("file:" + filename));
        } catch (NoDataSinkException ndse) {
          logger.error("Could not create DataSink: " + ndse);
          removeReceiveStream(desc, false);
          return;
        }

      } else {
        dataSink = new WebmDataSink(filename, desc.dataSource);
      }

      if (logger.isInfoEnabled())
        logger.info(
            "Created DataSink ("
                + dataSink
                + ") for SSRC="
                + ssrc
                + ". Output filename: "
                + filename);
      try {
        dataSink.open();
      } catch (IOException e) {
        logger.error("Failed to open DataSink (" + dataSink + ") for" + " SSRC=" + ssrc + ": " + e);
        removeReceiveStream(desc, false);
        return;
      }

      if (!audio) {
        final WebmDataSink webmDataSink = (WebmDataSink) dataSink;
        webmDataSink.setSsrc(ssrc);
        webmDataSink.setEventHandler(eventHandler);
        webmDataSink.setKeyFrameControl(
            new KeyFrameControlAdapter() {
              @Override
              public boolean requestKeyFrame(boolean urgent) {
                return requestFIR(webmDataSink);
              }
            });
      }

      try {
        dataSink.start();
      } catch (IOException e) {
        logger.error(
            "Failed to start DataSink (" + dataSink + ") for" + " SSRC=" + ssrc + ". " + e);
        removeReceiveStream(desc, false);
        return;
      }

      if (logger.isInfoEnabled()) logger.info("Started DataSink for SSRC=" + ssrc);

      desc.dataSink = dataSink;

      processor.start();
    } else if (logger.isDebugEnabled()) {
      logger.debug(
          "Unhandled ControllerEvent from the Processor for ssrc=" + desc.ssrc + ": " + ev);
    }
  }

  /**
   * Restarts the recording for a specific SSRC.
   *
   * @param ssrc the SSRC for which to restart recording. RTP packet of the new recording).
   */
  private void resetRecording(long ssrc, long timestamp) {
    ReceiveStreamDesc receiveStream = findReceiveStream(ssrc);

    // we only restart audio recordings
    if (receiveStream != null && receiveStream.format instanceof AudioFormat) {
      String newFilename = getNextFilename(path + "/" + ssrc, AUDIO_FILENAME_SUFFIX);

      // flush the buffer contained in the MP3 encoder
      String s = "trying to flush ssrc=" + ssrc;
      Processor p = receiveStream.processor;
      if (p != null) {
        s += " p!=null";
        for (TrackControl tc : p.getTrackControls()) {
          Object o = tc.getControl(FlushableControl.class.getName());
          if (o != null) ((FlushableControl) o).flush();
        }
      }

      if (logger.isInfoEnabled()) {
        logger.info("Restarting recording for SSRC=" + ssrc + ". New filename: " + newFilename);
      }

      receiveStream.dataSink.close();
      receiveStream.dataSink = null;

      // flush the FMJ jitter buffer
      // DataSource ds = receiveStream.receiveStream.getDataSource();
      // if (ds instanceof net.sf.fmj.media.protocol.rtp.DataSource)
      //    ((net.sf.fmj.media.protocol.rtp.DataSource)ds).flush();

      receiveStream.filename = newFilename;
      try {
        receiveStream.dataSink =
            Manager.createDataSink(
                receiveStream.dataSource, new MediaLocator("file:" + newFilename));
      } catch (NoDataSinkException ndse) {
        logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ndse);
        removeReceiveStream(receiveStream, false);
      }

      try {
        receiveStream.dataSink.open();
        receiveStream.dataSink.start();
      } catch (IOException ioe) {
        logger.warn("Could not reset recording for SSRC=" + ssrc + ": " + ioe);
        removeReceiveStream(receiveStream, false);
      }

      audioRecordingStarted(ssrc, timestamp);
    }
  }

  private void audioRecordingStarted(long ssrc, long timestamp) {
    ReceiveStreamDesc desc = findReceiveStream(ssrc);
    if (desc == null) return;

    RecorderEvent event = new RecorderEvent();
    event.setType(RecorderEvent.Type.RECORDING_STARTED);
    event.setMediaType(MediaType.AUDIO);
    event.setSsrc(ssrc);
    event.setRtpTimestamp(timestamp);
    event.setFilename(desc.filename);

    if (eventHandler != null) eventHandler.handleEvent(event);
  }

  /**
   * Handles a request from a specific <tt>DataSink</tt> to request a keyframe by sending an RTCP
   * feedback FIR message to the media source.
   *
   * @param dataSink the <tt>DataSink</tt> which requests that a keyframe be requested with a FIR
   *     message.
   * @return <tt>true</tt> if a keyframe was successfully requested, <tt>false</tt> otherwise
   */
  private boolean requestFIR(WebmDataSink dataSink) {
    ReceiveStreamDesc desc = findReceiveStream(dataSink);
    if (desc != null && rtcpFeedbackSender != null) {
      return rtcpFeedbackSender.sendFIR((int) desc.ssrc);
    }

    return false;
  }

  /**
   * Returns "prefix"+"suffix" if the file with this name does not exist. Otherwise, returns the
   * first inexistant filename of the form "prefix-"+i+"suffix", for an integer i. i is bounded by
   * 100 to prevent hanging, and on failure to find an inexistant filename the method will return
   * null.
   *
   * @param prefix
   * @param suffix
   * @return
   */
  private String getNextFilename(String prefix, String suffix) {
    if (!new File(prefix + suffix).exists()) return prefix + suffix;

    int i = 1;
    String s;
    do {
      s = prefix + "-" + i + suffix;
      if (!new File(s).exists()) return s;
      i++;
    } while (i < 1000); // don't hang indefinitely...

    return null;
  }

  /**
   * Finds the <tt>ReceiveStreamDesc</tt> with a particular <tt>Processor</tt>
   *
   * @param processor The <tt>Processor</tt> to match.
   * @return the <tt>ReceiveStreamDesc</tt> with a particular <tt>Processor</tt>, or <tt>null</tt>.
   */
  private ReceiveStreamDesc findReceiveStream(Processor processor) {
    if (processor == null) return null;

    synchronized (receiveStreams) {
      for (ReceiveStreamDesc r : receiveStreams) if (processor.equals(r.processor)) return r;
    }

    return null;
  }

  /**
   * Finds the <tt>ReceiveStreamDesc</tt> with a particular <tt>DataSink</tt>
   *
   * @param dataSink The <tt>DataSink</tt> to match.
   * @return the <tt>ReceiveStreamDesc</tt> with a particular <tt>DataSink</tt>, or <tt>null</tt>.
   */
  private ReceiveStreamDesc findReceiveStream(DataSink dataSink) {
    if (dataSink == null) return null;

    synchronized (receiveStreams) {
      for (ReceiveStreamDesc r : receiveStreams) if (dataSink.equals(r.dataSink)) return r;
    }

    return null;
  }

  /**
   * Finds the <tt>ReceiveStreamDesc</tt> with a particular SSRC.
   *
   * @param ssrc The SSRC to match.
   * @return the <tt>ReceiveStreamDesc</tt> with a particular SSRC, or <tt>null</tt>.
   */
  private ReceiveStreamDesc findReceiveStream(long ssrc) {
    synchronized (receiveStreams) {
      for (ReceiveStreamDesc r : receiveStreams) if (ssrc == r.ssrc) return r;
    }

    return null;
  }

  /**
   * Gets the SSRC of a <tt>ReceiveStream</tt> as a (non-negative) <tt>long</tt>.
   *
   * <p>FMJ stores the 32-bit SSRC values in <tt>int</tt>s, and the <tt>ReceiveStream.getSSRC()</tt>
   * implementation(s) don't take care of converting the negative <tt>int</tt> values sometimes
   * resulting from reading of a 32-bit field into the correct unsigned <tt>long</tt> value. So do
   * the conversion here.
   *
   * @param receiveStream the <tt>ReceiveStream</tt> for which to get the SSRC.
   * @return the SSRC of <tt>receiveStream</tt> an a (non-negative) <tt>long</tt>.
   */
  private long getReceiveStreamSSRC(ReceiveStream receiveStream) {
    return 0xffffffffL & receiveStream.getSSRC();
  }

  /**
   * Implements {@link ActiveSpeakerChangedListener#activeSpeakerChanged(long)}. Notifies this
   * <tt>RecorderRtpImpl</tt> that the audio <tt>ReceiveStream</tt> considered active has changed,
   * and that the new active stream has SSRC <tt>ssrc</tt>.
   *
   * @param ssrc the SSRC of the new active stream.
   */
  @Override
  public void activeSpeakerChanged(long ssrc) {
    if (eventHandler != null) {
      RecorderEvent e = new RecorderEvent();
      e.setAudioSsrc(ssrc);
      // TODO: how do we time this?
      e.setInstant(System.currentTimeMillis());
      e.setType(RecorderEvent.Type.SPEAKER_CHANGED);
      e.setMediaType(MediaType.VIDEO);
      eventHandler.handleEvent(e);
    }
  }

  private void handleRtpPacket(RawPacket pkt) {
    if (pkt != null && pkt.getPayloadType() == vp8PayloadType) {
      int ssrc = pkt.getSSRC();
      if (!activeVideoSsrcs.contains(ssrc & 0xffffffffL)) {
        synchronized (activeVideoSsrcs) {
          if (!activeVideoSsrcs.contains(ssrc & 0xffffffffL)) {
            activeVideoSsrcs.add(ssrc & 0xffffffffL);
            rtcpFeedbackSender.sendFIR(ssrc);
          }
        }
      }
    }
  }

  private void handleRtcpPacket(RawPacket pkt) {
    getSynchronizer().addRTCPPacket(pkt);
    eventHandler.nudge();
  }

  public SynchronizerImpl getSynchronizer() {
    if (synchronizer == null) synchronizer = new SynchronizerImpl();
    return synchronizer;
  }

  public void setSynchronizer(Synchronizer synchronizer) {
    if (synchronizer instanceof SynchronizerImpl) {
      this.synchronizer = (SynchronizerImpl) synchronizer;
    }
  }

  public void connect(Recorder recorder) {
    if (!(recorder instanceof RecorderRtpImpl)) return;

    ((RecorderRtpImpl) recorder).setSynchronizer(getSynchronizer());
  }

  private void emptyPacketBuffer(long ssrc) {
    RawPacket[] pkts = rtpConnector.packetBuffer.emptyBuffer(ssrc);
    RTPConnectorImpl.OutputDataStreamImpl dataStream;

    try {
      dataStream = rtpConnector.getDataOutputStream();
    } catch (IOException ioe) {
      logger.error("Failed to empty packet buffer for SSRC=" + ssrc + ": " + ioe);
      return;
    }
    for (RawPacket pkt : pkts)
      dataStream.write(
          pkt.getBuffer(), pkt.getOffset(), pkt.getLength(), false /* already transformed */);
  }
  /** The <tt>RTPConnector</tt> implementation used by this <tt>RecorderRtpImpl</tt>. */
  private class RTPConnectorImpl implements RTPConnector {
    private PushSourceStreamImpl controlInputStream;
    private OutputDataStreamImpl controlOutputStream;

    private PushSourceStreamImpl dataInputStream;
    private OutputDataStreamImpl dataOutputStream;

    private SourceTransferHandler dataTransferHandler;
    private SourceTransferHandler controlTransferHandler;

    private RawPacket pendingDataPacket = new RawPacket();
    private RawPacket pendingControlPacket = new RawPacket();

    private PacketTransformer rtpPacketTransformer = null;
    private PacketTransformer rtcpPacketTransformer = null;

    /** The PacketBuffer instance which we use as a jitter buffer. */
    private PacketBuffer packetBuffer;

    private RTPConnectorImpl(byte redPT, byte ulpfecPT) {
      packetBuffer = new PacketBuffer();
      // The chain of transformers will be applied in reverse order for
      // incoming packets.
      TransformEngine transformEngine =
          new TransformEngineChain(
              new TransformEngine[] {
                packetBuffer,
                new TransformEngineImpl(),
                new CompoundPacketEngine(),
                new FECTransformEngine(ulpfecPT, (byte) -1),
                new REDTransformEngine(redPT, (byte) -1)
              });

      rtpPacketTransformer = transformEngine.getRTPTransformer();
      rtcpPacketTransformer = transformEngine.getRTCPTransformer();
    }

    private RTPConnectorImpl() {}

    @Override
    public void close() {
      try {
        if (dataOutputStream != null) dataOutputStream.close();
        if (controlOutputStream != null) controlOutputStream.close();
      } catch (IOException ioe) {
        throw new UndeclaredThrowableException(ioe);
      }
    }

    @Override
    public PushSourceStream getControlInputStream() throws IOException {
      if (controlInputStream == null) {
        controlInputStream = new PushSourceStreamImpl(true);
      }

      return controlInputStream;
    }

    @Override
    public OutputDataStream getControlOutputStream() throws IOException {
      if (controlOutputStream == null) {
        controlOutputStream = new OutputDataStreamImpl(true);
      }

      return controlOutputStream;
    }

    @Override
    public PushSourceStream getDataInputStream() throws IOException {
      if (dataInputStream == null) {
        dataInputStream = new PushSourceStreamImpl(false);
      }

      return dataInputStream;
    }

    @Override
    public OutputDataStreamImpl getDataOutputStream() throws IOException {
      if (dataOutputStream == null) {
        dataOutputStream = new OutputDataStreamImpl(false);
      }

      return dataOutputStream;
    }

    @Override
    public double getRTCPBandwidthFraction() {
      return -1;
    }

    @Override
    public double getRTCPSenderBandwidthFraction() {
      return -1;
    }

    @Override
    public int getReceiveBufferSize() {
      // TODO Auto-generated method stub
      return 0;
    }

    @Override
    public int getSendBufferSize() {
      // TODO Auto-generated method stub
      return 0;
    }

    @Override
    public void setReceiveBufferSize(int arg0) throws IOException {
      // TODO Auto-generated method stub

    }

    @Override
    public void setSendBufferSize(int arg0) throws IOException {
      // TODO Auto-generated method stub
    }

    private class OutputDataStreamImpl implements OutputDataStream {
      boolean isControlStream;
      private RawPacket[] rawPacketArray = new RawPacket[1];

      public OutputDataStreamImpl(boolean isControlStream) {
        this.isControlStream = isControlStream;
      }

      public int write(byte[] buffer, int offset, int length) {
        return write(buffer, offset, length, true);
      }

      public int write(byte[] buffer, int offset, int length, boolean transform) {
        RawPacket pkt = rawPacketArray[0];
        if (pkt == null) pkt = new RawPacket();
        rawPacketArray[0] = pkt;

        byte[] pktBuf = pkt.getBuffer();
        if (pktBuf == null || pktBuf.length < length) {
          pktBuf = new byte[length];
          pkt.setBuffer(pktBuf);
        }
        System.arraycopy(buffer, offset, pktBuf, 0, length);
        pkt.setOffset(0);
        pkt.setLength(length);

        if (transform) {
          PacketTransformer packetTransformer =
              isControlStream ? rtcpPacketTransformer : rtpPacketTransformer;

          if (packetTransformer != null)
            rawPacketArray = packetTransformer.reverseTransform(rawPacketArray);
        }

        SourceTransferHandler transferHandler;
        PushSourceStream pushSourceStream;

        try {
          if (isControlStream) {
            transferHandler = controlTransferHandler;
            pushSourceStream = getControlInputStream();
          } else {
            transferHandler = dataTransferHandler;
            pushSourceStream = getDataInputStream();
          }
        } catch (IOException ioe) {
          throw new UndeclaredThrowableException(ioe);
        }

        for (int i = 0; i < rawPacketArray.length; i++) {
          RawPacket packet = rawPacketArray[i];

          // keep the first element for reuse
          if (i != 0) rawPacketArray[i] = null;

          if (packet != null) {
            if (isControlStream) pendingControlPacket = packet;
            else pendingDataPacket = packet;

            if (transferHandler != null) {
              transferHandler.transferData(pushSourceStream);
            }
          }
        }

        return length;
      }

      public void close() throws IOException {}
    }

    /**
     * A dummy implementation of {@link PushSourceStream}.
     *
     * @author Vladimir Marinov
     */
    private class PushSourceStreamImpl implements PushSourceStream {

      private boolean isControlStream = false;

      public PushSourceStreamImpl(boolean isControlStream) {
        this.isControlStream = isControlStream;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public boolean endOfStream() {
        return false;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public ContentDescriptor getContentDescriptor() {
        return null;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public long getContentLength() {
        return 0;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public Object getControl(String arg0) {
        return null;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public Object[] getControls() {
        return null;
      }

      /** Not implemented because there are currently no uses of the underlying functionality. */
      @Override
      public int getMinimumTransferSize() {
        if (isControlStream) {
          if (pendingControlPacket.getBuffer() != null) {
            return pendingControlPacket.getLength();
          }
        } else {
          if (pendingDataPacket.getBuffer() != null) {
            return pendingDataPacket.getLength();
          }
        }

        return 0;
      }

      @Override
      public int read(byte[] buffer, int offset, int length) throws IOException {

        RawPacket pendingPacket;
        if (isControlStream) {
          pendingPacket = pendingControlPacket;
        } else {
          pendingPacket = pendingDataPacket;
        }
        int bytesToRead = 0;
        byte[] pendingPacketBuffer = pendingPacket.getBuffer();
        if (pendingPacketBuffer != null) {
          int pendingPacketLength = pendingPacket.getLength();
          bytesToRead = length > pendingPacketLength ? pendingPacketLength : length;
          System.arraycopy(
              pendingPacketBuffer, pendingPacket.getOffset(), buffer, offset, bytesToRead);
        }
        return bytesToRead;
      }

      /**
       * {@inheritDoc}
       *
       * <p>We keep the first non-null <tt>SourceTransferHandler</tt> that was set, because we don't
       * want it to be overwritten when we initialize a second <tt>RTPManager</tt> with this
       * <tt>RTPConnector</tt>.
       *
       * <p>See {@link RecorderRtpImpl#start(String, String)}
       */
      @Override
      public void setTransferHandler(SourceTransferHandler transferHandler) {
        if (isControlStream) {
          if (RTPConnectorImpl.this.controlTransferHandler == null) {
            RTPConnectorImpl.this.controlTransferHandler = transferHandler;
          }
        } else {
          if (RTPConnectorImpl.this.dataTransferHandler == null) {
            RTPConnectorImpl.this.dataTransferHandler = transferHandler;
          }
        }
      }
    }

    /**
     * A transform engine implementation which allows <tt>RecorderRtpImpl</tt> to intercept RTP and
     * RTCP packets in.
     */
    private class TransformEngineImpl implements TransformEngine {
      SinglePacketTransformer rtpTransformer =
          new SinglePacketTransformer() {
            @Override
            public RawPacket transform(RawPacket pkt) {
              return pkt;
            }

            @Override
            public RawPacket reverseTransform(RawPacket pkt) {
              RecorderRtpImpl.this.handleRtpPacket(pkt);
              return pkt;
            }

            @Override
            public void close() {}
          };

      SinglePacketTransformer rtcpTransformer =
          new SinglePacketTransformer() {
            @Override
            public RawPacket transform(RawPacket pkt) {
              return pkt;
            }

            @Override
            public RawPacket reverseTransform(RawPacket pkt) {
              RecorderRtpImpl.this.handleRtcpPacket(pkt);
              if (pkt != null && pkt.getRTCPPayloadType() == 203) {
                // An RTCP BYE packet. Remove the receive stream before
                // it gets to FMJ, because we want to, for example,
                // flush the packet buffer before that.

                long ssrc = pkt.getRTCPSSRC() & 0xffffffffl;
                if (logger.isInfoEnabled()) logger.info("RTCP BYE for SSRC=" + ssrc);

                ReceiveStreamDesc receiveStream = findReceiveStream(ssrc);
                if (receiveStream != null) removeReceiveStream(receiveStream, false);
              }

              return pkt;
            }

            @Override
            public void close() {}
          };

      @Override
      public PacketTransformer getRTPTransformer() {
        return rtpTransformer;
      }

      @Override
      public PacketTransformer getRTCPTransformer() {
        return rtcpTransformer;
      }
    }
  }

  private class RecorderEventHandlerImpl implements RecorderEventHandler {
    private RecorderEventHandler handler;
    private final Set<RecorderEvent> pendingEvents = new HashSet<RecorderEvent>();

    private RecorderEventHandlerImpl(RecorderEventHandler handler) {
      this.handler = handler;
    }

    @Override
    public boolean handleEvent(RecorderEvent ev) {
      if (ev == null) return true;
      if (RecorderEvent.Type.RECORDING_STARTED.equals(ev.getType())) {
        long instant = getSynchronizer().getLocalTime(ev.getSsrc(), ev.getRtpTimestamp());
        if (instant != -1) {
          ev.setInstant(instant);
          return handler.handleEvent(ev);
        } else {
          pendingEvents.add(ev);
          return true;
        }
      }
      return handler.handleEvent(ev);
    }

    private void nudge() {
      for (Iterator<RecorderEvent> iter = pendingEvents.iterator(); iter.hasNext(); ) {
        RecorderEvent ev = iter.next();
        long instant = getSynchronizer().getLocalTime(ev.getSsrc(), ev.getRtpTimestamp());
        if (instant != -1) {
          iter.remove();
          ev.setInstant(instant);
          handler.handleEvent(ev);
        }
      }
    }

    @Override
    public void close() {
      for (RecorderEvent ev : pendingEvents) handler.handleEvent(ev);
    }
  }

  /** Represents a <tt>ReceiveStream</tt> for the purposes of this <tt>RecorderRtpImpl</tt>. */
  private class ReceiveStreamDesc {
    /**
     * The actual <tt>ReceiveStream</tt> which is represented by this <tt>ReceiveStreamDesc</tt>.
     */
    private ReceiveStream receiveStream;

    /** The SSRC of the stream. */
    long ssrc;

    /**
     * The <tt>Processor</tt> used to transcode this receive stream into a format appropriate for
     * saving to a file.
     */
    private Processor processor;

    /** The <tt>DataSink</tt> which saves the <tt>this.dataSource</tt> to a file. */
    private DataSink dataSink;

    /**
     * The <tt>DataSource</tt> for this receive stream which is to be saved using a
     * <tt>DataSink</tt> (i.e. the <tt>DataSource</tt> "after" all needed transcoding is done).
     */
    private DataSource dataSource;

    /** The name of the file into which this stream is being saved. */
    private String filename;

    /** The (original) format of this receive stream. */
    private Format format;

    /** The <tt>SilenceEffect</tt> used for this stream (for audio streams only). */
    private SilenceEffect silenceEffect;

    private ReceiveStreamDesc(ReceiveStream receiveStream) {
      this.receiveStream = receiveStream;
      this.ssrc = getReceiveStreamSSRC(receiveStream);
    }
  }
}
コード例 #2
0
/**
 * The <tt>BasicRTCPTerminationStrategy</tt> "gateways" PLIs, FIRs, NACKs, etc, in the sense that it
 * replaces the packet sender information in the PLIs, FIRs, NACKs, etc and it generates its own
 * SRs/RRs/REMBs based on information that it collects and from information found in FMJ.
 *
 * @author George Politis
 */
public class BasicRTCPTerminationStrategy extends MediaStreamRTCPTerminationStrategy {
  /**
   * The <tt>Logger</tt> used by the <tt>BasicRTCPTerminationStrategy</tt> class and its instances
   * to print debug information.
   */
  private static final Logger logger = Logger.getLogger(BasicRTCPTerminationStrategy.class);

  /** The maximum number of RTCP report blocks that an RR or an SR can contain. */
  private static final int MAX_RTCP_REPORT_BLOCKS = 31;

  /** The minimum number of RTCP report blocks that an RR or an SR can contain. */
  private static final int MIN_RTCP_REPORT_BLOCKS = 0;

  /**
   * A reusable array that can be used to hold up to <tt>MAX_RTCP_REPORT_BLOCKS</tt>
   * <tt>RTCPReportBlock</tt>s. It is assumed that a single thread is accessing this field at a
   * given time.
   */
  private final RTCPReportBlock[] MAX_RTCP_REPORT_BLOCKS_ARRAY =
      new RTCPReportBlock[MAX_RTCP_REPORT_BLOCKS];

  /** A reusable array that holds 0 <tt>RTCPReportBlock</tt>s. */
  private static final RTCPReportBlock[] MIN_RTCP_REPORTS_BLOCKS_ARRAY =
      new RTCPReportBlock[MIN_RTCP_REPORT_BLOCKS];

  /**
   * The RTP stats map that holds RTP statistics about all the streams that this
   * <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
   */
  private final RTPStatsMap rtpStatsMap = new RTPStatsMap();

  /**
   * The RTCP stats map that holds RTCP statistics about all the streams that this
   * <tt>BasicRTCPTerminationStrategy</tt> (as a <tt>TransformEngine</tt>) has observed.
   */
  private final RemoteClockEstimator remoteClockEstimator = new RemoteClockEstimator();

  /**
   * The <tt>CNameRegistry</tt> holds the CNAMEs that this RTCP termination, seen as a
   * TransformEngine, has seen.
   */
  private final CNAMERegistry cnameRegistry = new CNAMERegistry();

  /** The parser that parses <tt>RawPacket</tt>s to <tt>RTCPCompoundPacket</tt>s. */
  private final RTCPPacketParserEx parser = new RTCPPacketParserEx();

  /** The generator that generates <tt>RawPacket</tt>s from <tt>RTCPCompoundPacket</tt>s. */
  private final RTCPGenerator generator = new RTCPGenerator();

  /**
   * The RTCP feedback gateway responsible for dropping all the stuff that we support in this RTCP
   * termination strategy.
   */
  private final FeedbackGateway feedbackGateway = new FeedbackGateway();

  /** The garbage collector that cleans-up the state of this RTCP termination strategy. */
  private final GarbageCollector garbageCollector = new GarbageCollector();

  /** The RTP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
  private final PacketTransformer rtpTransformer =
      new SinglePacketTransformer() {
        /** {@inheritDoc} */
        @Override
        public RawPacket transform(RawPacket pkt) {
          // Update our RTP stats map (packets/octet sent).
          rtpStatsMap.apply(pkt);

          return pkt;
        }

        /** {@inheritDoc} */
        @Override
        public RawPacket reverseTransform(RawPacket pkt) {
          // Let everything pass through.
          return pkt;
        }
      };

  /** The RTCP <tt>PacketTransformer</tt> of this <tt>BasicRTCPTerminationStrategy</tt>. */
  private final PacketTransformer rtcpTransformer =
      new SinglePacketTransformer() {
        /** {@inheritDoc} */
        @Override
        public RawPacket transform(RawPacket pkt) {
          if (pkt == null) {
            return pkt;
          }

          RTCPCompoundPacket inPacket;
          try {
            inPacket =
                (RTCPCompoundPacket)
                    parser.parse(pkt.getBuffer(), pkt.getOffset(), pkt.getLength());
          } catch (BadFormatException e) {
            logger.warn("Failed to terminate an RTCP packet. " + "Dropping packet.");
            return null;
          }

          // Update our RTCP stats map (timestamps). This operation is
          // read-only.
          remoteClockEstimator.apply(inPacket);

          cnameRegistry.update(inPacket);

          // Remove SRs and RRs from the RTCP packet.
          pkt = feedbackGateway.gateway(inPacket);

          return pkt;
        }

        /** {@inheritDoc} */
        @Override
        public RawPacket reverseTransform(RawPacket pkt) {
          // Let everything pass through.
          return pkt;
        }
      };

  /** A counter that counts the number of times we've sent "full-blown" SDES. */
  private int sdesCounter = 0;

  /** {@inheritDoc} */
  @Override
  public PacketTransformer getRTPTransformer() {
    return rtpTransformer;
  }

  /** {@inheritDoc} */
  @Override
  public PacketTransformer getRTCPTransformer() {
    return rtcpTransformer;
  }

  /** {@inheritDoc} */
  @Override
  public RawPacket report() {
    garbageCollector.cleanup();

    // TODO Compound RTCP packets should not exceed the MTU of the network
    // path.
    //
    // An individual RTP participant should send only one compound RTCP
    // packet per report interval in order for the RTCP bandwidth per
    // participant to be estimated correctly, except when the compound
    // RTCP packet is split for partial encryption.
    //
    // If there are too many sources to fit all the necessary RR packets
    // into one compound RTCP packet without exceeding the maximum
    // transmission unit (MTU) of the network path, then only the subset
    // that will fit into one MTU should be included in each interval. The
    // subsets should be selected round-robin across multiple intervals so
    // that all sources are reported.
    //
    // It is impossible to know in advance what the MTU of path will be.
    // There are various algorithms for experimenting to find out, but many
    // devices do not properly implement (or deliberately ignore) the
    // necessary standards so it all comes down to trial and error. For that
    // reason, we can just guess 1200 or 1500 bytes per message.
    long time = System.currentTimeMillis();

    Collection<RTCPPacket> packets = new ArrayList<RTCPPacket>();

    // First, we build the RRs.
    Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(time);
    if (rrPackets != null && rrPackets.size() != 0) {
      packets.addAll(rrPackets);
    }

    // Next, we build the SRs.
    Collection<RTCPSRPacket> srPackets = makeRTCPSRPackets(time);
    if (srPackets != null && srPackets.size() != 0) {
      packets.addAll(srPackets);
    }

    // Bail out if we have nothing to report.
    if (packets.size() == 0) {
      return null;
    }

    // Next, we build the REMB.
    RTCPREMBPacket rembPacket = makeRTCPREMBPacket();
    if (rembPacket != null) {
      packets.add(rembPacket);
    }

    // Finally, we add an SDES packet.
    RTCPSDESPacket sdesPacket = makeSDESPacket();
    if (sdesPacket != null) {
      packets.add(sdesPacket);
    }

    // Prepare the <tt>RTCPCompoundPacket</tt> to return.
    RTCPPacket rtcpPackets[] = packets.toArray(new RTCPPacket[packets.size()]);

    RTCPCompoundPacket cp = new RTCPCompoundPacket(rtcpPackets);

    // Build the <tt>RTCPCompoundPacket</tt> and return the
    // <tt>RawPacket</tt> to inject to the <tt>MediaStream</tt>.
    return generator.apply(cp);
  }

  /**
   * (attempts) to get the local SSRC that will be used in the media sender SSRC field of the RTCP
   * reports. TAG(cat4-local-ssrc-hurricane)
   *
   * @return
   */
  private long getLocalSSRC() {
    return getStream().getStreamRTPManager().getLocalSSRC();
  }

  /**
   * Makes <tt>RTCPRRPacket</tt>s using information in FMJ.
   *
   * @param time
   * @return A <tt>Collection</tt> of <tt>RTCPRRPacket</tt>s to inject to the <tt>MediaStream</tt>.
   */
  private Collection<RTCPRRPacket> makeRTCPRRPackets(long time) {
    RTCPReportBlock[] reportBlocks = makeRTCPReportBlocks(time);
    if (reportBlocks == null || reportBlocks.length == 0) {
      return null;
    }

    Collection<RTCPRRPacket> rrPackets = new ArrayList<RTCPRRPacket>();

    // We use the stream's local source ID (SSRC) as the SSRC of packet
    // sender.
    long streamSSRC = getLocalSSRC();

    // Since a maximum of 31 reception report blocks will fit in an SR
    // or RR packet, additional RR packets SHOULD be stacked after the
    // initial SR or RR packet as needed to contain the reception
    // reports for all sources heard during the interval since the last
    // report.
    if (reportBlocks.length > MAX_RTCP_REPORT_BLOCKS) {
      for (int offset = 0; offset < reportBlocks.length; offset += MAX_RTCP_REPORT_BLOCKS) {
        RTCPReportBlock[] blocks =
            (reportBlocks.length - offset < MAX_RTCP_REPORT_BLOCKS)
                ? new RTCPReportBlock[reportBlocks.length - offset]
                : MAX_RTCP_REPORT_BLOCKS_ARRAY;

        System.arraycopy(reportBlocks, offset, blocks, 0, blocks.length);

        RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, blocks);
        rrPackets.add(rr);
      }
    } else {
      RTCPRRPacket rr = new RTCPRRPacket((int) streamSSRC, reportBlocks);
      rrPackets.add(rr);
    }

    return rrPackets;
  }

  /**
   * Iterate through all the <tt>ReceiveStream</tt>s that this <tt>MediaStream</tt> has and make
   * <tt>RTCPReportBlock</tt>s for all of them.
   *
   * @param time
   * @return
   */
  private RTCPReportBlock[] makeRTCPReportBlocks(long time) {
    MediaStream stream = getStream();
    // State validation.
    if (stream == null) {
      logger.warn("stream is null.");
      return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
    }

    StreamRTPManager streamRTPManager = stream.getStreamRTPManager();
    if (streamRTPManager == null) {
      logger.warn("streamRTPManager is null.");
      return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
    }

    Collection<ReceiveStream> receiveStreams = streamRTPManager.getReceiveStreams();

    if (receiveStreams == null || receiveStreams.size() == 0) {
      logger.info("There are no receive streams to build report " + "blocks for.");
      return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
    }

    SSRCCache cache = streamRTPManager.getSSRCCache();
    if (cache == null) {
      logger.info("cache is null.");
      return MIN_RTCP_REPORTS_BLOCKS_ARRAY;
    }

    // Create the return object.
    Collection<RTCPReportBlock> rtcpReportBlocks = new ArrayList<RTCPReportBlock>();

    // Populate the return object.
    for (ReceiveStream receiveStream : receiveStreams) {
      // Dig into the guts of FMJ and get the stats for the current
      // receiveStream.
      SSRCInfo info = cache.cache.get((int) receiveStream.getSSRC());

      if (!info.ours && info.sender) {
        RTCPReportBlock rtcpReportBlock = info.makeReceiverReport(time);
        rtcpReportBlocks.add(rtcpReportBlock);
      }
    }

    return rtcpReportBlocks.toArray(new RTCPReportBlock[rtcpReportBlocks.size()]);
  }

  /**
   * Makes an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which we
   * receive.
   *
   * @return an <tt>RTCPREMBPacket</tt> that provides receiver feedback to the endpoint from which
   *     we receive.
   */
  private RTCPREMBPacket makeRTCPREMBPacket() {
    // TODO we should only make REMBs if REMB support has been advertised.
    // Destination
    RemoteBitrateEstimator remoteBitrateEstimator =
        ((VideoMediaStream) getStream()).getRemoteBitrateEstimator();

    Collection<Integer> ssrcs = remoteBitrateEstimator.getSsrcs();

    // TODO(gp) intersect with SSRCs from signaled simulcast layers
    // NOTE(gp) The Google Congestion Control algorithm (sender side)
    // doesn't seem to care about the SSRCs in the dest field.
    long[] dest = new long[ssrcs.size()];
    int i = 0;

    for (Integer ssrc : ssrcs) dest[i++] = ssrc & 0xFFFFFFFFL;

    // Exp & mantissa
    long bitrate = remoteBitrateEstimator.getLatestEstimate();

    if (bitrate == -1) return null;

    if (logger.isDebugEnabled()) logger.debug("Estimated bitrate: " + bitrate);

    // Create and return the packet.
    // We use the stream's local source ID (SSRC) as the SSRC of packet
    // sender.
    long streamSSRC = getLocalSSRC();

    return new RTCPREMBPacket(streamSSRC, /* mediaSSRC */ 0L, bitrate, dest);
  }

  /**
   * Makes <tt>RTCPSRPacket</tt>s for all the RTP streams that we're sending.
   *
   * @return a <tt>List</tt> of <tt>RTCPSRPacket</tt> for all the RTP streams that we're sending.
   */
  private Collection<RTCPSRPacket> makeRTCPSRPackets(long time) {
    Collection<RTCPSRPacket> srPackets = new ArrayList<RTCPSRPacket>();

    for (RTPStatsEntry rtpStatsEntry : rtpStatsMap.values()) {
      int ssrc = rtpStatsEntry.getSsrc();
      RemoteClock estimate = remoteClockEstimator.estimate(ssrc, time);
      if (estimate == null) {
        // We're not going to go far without an estimate..
        continue;
      }

      RTCPSRPacket srPacket = new RTCPSRPacket(ssrc, MIN_RTCP_REPORTS_BLOCKS_ARRAY);

      // Set the NTP timestamp for this SR.
      long estimatedRemoteTime = estimate.getRemoteTime();
      long secs = estimatedRemoteTime / 1000L;
      double fraction = (estimatedRemoteTime - secs * 1000L) / 1000D;
      srPacket.ntptimestamplsw = (int) (fraction * 4294967296D);
      srPacket.ntptimestampmsw = secs;

      // Set the RTP timestamp.
      srPacket.rtptimestamp = estimate.getRtpTimestamp();

      // Fill-in packet and octet send count.
      srPacket.packetcount = rtpStatsEntry.getPacketsSent();
      srPacket.octetcount = rtpStatsEntry.getBytesSent();

      srPackets.add(srPacket);
    }

    return srPackets;
  }

  /**
   * Makes <tt>RTCPSDES</tt> packets for all the RTP streams that we're sending.
   *
   * @return a <tt>List</tt> of <tt>RTCPSDES</tt> packets for all the RTP streams that we're
   *     sending.
   */
  private RTCPSDESPacket makeSDESPacket() {
    Collection<RTCPSDES> sdesChunks = new ArrayList<RTCPSDES>();

    // Create an SDES for our own SSRC.
    RTCPSDES ownSDES = new RTCPSDES();

    SSRCInfo ourinfo = getStream().getStreamRTPManager().getSSRCCache().ourssrc;
    ownSDES.ssrc = (int) getLocalSSRC();
    Collection<RTCPSDESItem> ownItems = new ArrayList<RTCPSDESItem>();
    ownItems.add(new RTCPSDESItem(RTCPSDESItem.CNAME, ourinfo.sourceInfo.getCNAME()));

    // Throttle the source description bandwidth. See RFC3550#6.3.9
    // Allocation of Source Description Bandwidth.

    if (sdesCounter % 3 == 0) {
      if (ourinfo.name != null && ourinfo.name.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.NAME, ourinfo.name.getDescription()));
      if (ourinfo.email != null && ourinfo.email.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.EMAIL, ourinfo.email.getDescription()));
      if (ourinfo.phone != null && ourinfo.phone.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.PHONE, ourinfo.phone.getDescription()));
      if (ourinfo.loc != null && ourinfo.loc.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.LOC, ourinfo.loc.getDescription()));
      if (ourinfo.tool != null && ourinfo.tool.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.TOOL, ourinfo.tool.getDescription()));
      if (ourinfo.note != null && ourinfo.note.getDescription() != null)
        ownItems.add(new RTCPSDESItem(RTCPSDESItem.NOTE, ourinfo.note.getDescription()));
    }

    sdesCounter++;

    ownSDES.items = ownItems.toArray(new RTCPSDESItem[ownItems.size()]);

    sdesChunks.add(ownSDES);

    for (Map.Entry<Integer, byte[]> entry : cnameRegistry.entrySet()) {
      RTCPSDES sdes = new RTCPSDES();
      sdes.ssrc = entry.getKey();
      sdes.items = new RTCPSDESItem[] {new RTCPSDESItem(RTCPSDESItem.CNAME, entry.getValue())};
    }

    RTCPSDES[] sps = sdesChunks.toArray(new RTCPSDES[sdesChunks.size()]);
    RTCPSDESPacket sp = new RTCPSDESPacket(sps);

    return sp;
  }

  /**
   * The garbage collector runs at each reporting interval and cleans up the data structures of this
   * RTCP termination strategy based on the SSRCs that the owner <tt>MediaStream</tt> is still
   * sending.
   */
  class GarbageCollector {
    public void cleanup() {
      // TODO We need to fix TAG(cat4-local-ssrc-hurricane) and
      // TAG(cat4-remote-ssrc-hurricane) first. The idea is to remove
      // from our data structures everything that is not listed in as
      // a remote SSRC.
    }
  }

  /**
   * Removes receiver and sender feedback from RTCP packets. Typically this means dropping SRs, RR
   * report blocks and REMBs. It needs to pass through PLIs, FIRs, NACKs, etc.
   */
  class FeedbackGateway {
    /**
     * Removes receiver and sender feedback from RTCP packets.
     *
     * @param inPacket the <tt>RTCPCompoundPacket</tt> to filter.
     * @return the filtered <tt>RawPacket</tt>.
     */
    public RawPacket gateway(RTCPCompoundPacket inPacket) {
      if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
        logger.info("Ignoring empty RTCP packet.");
        return null;
      }

      ArrayList<RTCPPacket> outPackets = new ArrayList<RTCPPacket>(inPacket.packets.length);

      for (RTCPPacket p : inPacket.packets) {
        switch (p.type) {
          case RTCPPacket.RR:
          case RTCPPacket.SR:
          case RTCPPacket.SDES:
            // We generate our own RR/SR/SDES packets. We only want
            // to forward NACKs/PLIs/etc.
            break;
          case RTCPFBPacket.PSFB:
            RTCPFBPacket psfb = (RTCPFBPacket) p;
            switch (psfb.fmt) {
              case RTCPREMBPacket.FMT:
                // We generate its own REMB packets.
                break;
              default:
                // We let through everything else, like NACK
                // packets.
                outPackets.add(psfb);
                break;
            }
            break;
          default:
            // We let through everything else, like BYE and APP
            // packets.
            outPackets.add(p);
            break;
        }
      }

      if (outPackets.size() == 0) {
        return null;
      }

      // We have feedback messages to send. Pack them in a compound
      // RR and send them. TODO Use RFC5506 Reduced-Size RTCP, if the
      // receiver supports it.
      Collection<RTCPRRPacket> rrPackets = makeRTCPRRPackets(System.currentTimeMillis());

      if (rrPackets != null && rrPackets.size() != 0) {
        outPackets.addAll(0, rrPackets);
      } else {
        logger.warn("We might be sending invalid RTCPs.");
      }

      RTCPPacket[] pkts = outPackets.toArray(new RTCPPacket[outPackets.size()]);
      RTCPCompoundPacket outPacket = new RTCPCompoundPacket(pkts);

      return generator.apply(outPacket);
    }
  }

  /** Holds the NTP timestamp and the associated RTP timestamp for a given RTP stream. */
  class RemoteClock {
    /**
     * Ctor.
     *
     * @param remoteTime
     * @param rtpTimestamp
     */
    public RemoteClock(long remoteTime, int rtpTimestamp) {
      this.remoteTime = remoteTime;
      this.rtpTimestamp = rtpTimestamp;
    }

    /**
     * The last NTP timestamp that we received for {@link this.ssrc} expressed in millis. Should be
     * treated a signed long.
     */
    private final long remoteTime;

    /**
     * The RTP timestamp associated to {@link this.ntpTimestamp}. The RTP timestamp is an unsigned
     * int.
     */
    private final int rtpTimestamp;

    /** @return */
    public int getRtpTimestamp() {
      return rtpTimestamp;
    }

    /** @return */
    public long getRemoteTime() {
      return remoteTime;
    }
  }

  /** */
  class ReceivedRemoteClock {
    /** The SSRC. */
    private final int ssrc;

    /**
     * The <tt>RemoteClock</tt> which was received at {@link this.receivedTime} for this RTP stream.
     */
    private final RemoteClock remoteClock;

    /**
     * The local time in millis when we received the RTCP report with the RTP/NTP timestamps. It's a
     * signed long.
     */
    private final long receivedTime;

    /**
     * The clock rate for {@link.ssrc}. We need to have received at least two SRs in order to be
     * able to calculate this. Unsigned short.
     */
    private final int frequencyHz;

    /**
     * Ctor.
     *
     * @param ssrc
     * @param remoteTime
     * @param rtpTimestamp
     * @param frequencyHz
     */
    ReceivedRemoteClock(int ssrc, long remoteTime, int rtpTimestamp, int frequencyHz) {
      this.ssrc = ssrc;
      this.remoteClock = new RemoteClock(remoteTime, rtpTimestamp);
      this.frequencyHz = frequencyHz;
      this.receivedTime = System.currentTimeMillis();
    }

    /** @return */
    public RemoteClock getRemoteClock() {
      return remoteClock;
    }

    /** @return */
    public long getReceivedTime() {
      return receivedTime;
    }

    /** @return */
    public int getSsrc() {
      return ssrc;
    }

    /** @return */
    public int getFrequencyHz() {
      return frequencyHz;
    }
  }

  /** The <tt>RTPStatsEntry</tt> class contains information about an outgoing SSRC. */
  class RTPStatsEntry {
    /** The SSRC of the stream that this instance tracks. */
    private final int ssrc;

    /**
     * The total number of _payload_ octets (i.e., not including header or padding) transmitted in
     * RTP data packets by the sender since starting transmission up until the time this SR packet
     * was generated. This should be treated as an unsigned int.
     */
    private final int bytesSent;

    /**
     * The total number of RTP data packets transmitted by the sender (including re-transmissions)
     * since starting transmission up until the time this SR packet was generated. Re-transmissions
     * using an RTX stream are tracked in the RTX SSRC. This should be treated as an unsigned int.
     */
    private final int packetsSent;

    /** @return */
    public int getSsrc() {
      return ssrc;
    }

    /** @return */
    public int getBytesSent() {
      return bytesSent;
    }

    /** @return */
    public int getPacketsSent() {
      return packetsSent;
    }

    /**
     * Ctor.
     *
     * @param ssrc
     * @param bytesSent
     */
    RTPStatsEntry(int ssrc, int bytesSent, int packetsSent) {
      this.ssrc = ssrc;
      this.bytesSent = bytesSent;
      this.packetsSent = packetsSent;
    }
  }

  /**
   * The <tt>RtpStatsMap</tt> gathers stats from RTP packets that the <tt>RTCPReportBuilder</tt>
   * uses to build its reports.
   */
  class RTPStatsMap extends ConcurrentHashMap<Integer, RTPStatsEntry> {
    /**
     * Updates this <tt>RTPStatsMap</tt> with information it gets from the <tt>RawPacket</tt>.
     *
     * @param pkt the <tt>RawPacket</tt> that is being transmitted.
     */
    public void apply(RawPacket pkt) {
      int ssrc = pkt.getSSRC();
      if (this.containsKey(ssrc)) {
        RTPStatsEntry oldRtpStatsEntry = this.get(ssrc);

        // Replace whatever was in there before. A feature of the two's
        // complement encoding (which is used by Java integers) is that
        // the bitwise results for add, subtract, and multiply are the
        // same if both inputs are interpreted as signed values or both
        // inputs are interpreted as unsigned values. (Other encodings
        // like one's complement and signed magnitude don't have this
        // properly.)
        this.put(
            ssrc,
            new RTPStatsEntry(
                ssrc,
                oldRtpStatsEntry.getBytesSent()
                    + pkt.getLength()
                    - pkt.getHeaderLength()
                    - pkt.getPaddingSize(),
                oldRtpStatsEntry.getPacketsSent() + 1));
      } else {
        // Add a new <tt>RTPStatsEntry</tt> in this map.
        this.put(
            ssrc,
            new RTPStatsEntry(
                ssrc, pkt.getLength() - pkt.getHeaderLength() - pkt.getPaddingSize(), 1));
      }
    }
  }

  /** A class that can be used to estimate the remote time at a given local time. */
  class RemoteClockEstimator {
    /** base: 7-Feb-2036 @ 06:28:16 UTC */
    private static final long msb0baseTime = 2085978496000L;

    /** base: 1-Jan-1900 @ 01:00:00 UTC */
    private static final long msb1baseTime = -2208988800000L;

    /** A map holding the received remote clocks. */
    private Map<Integer, ReceivedRemoteClock> receivedClocks =
        new ConcurrentHashMap<Integer, ReceivedRemoteClock>();

    /**
     * Inspect an <tt>RTCPCompoundPacket</tt> and build-up the state for future estimations.
     *
     * @param pkt
     */
    public void apply(RTCPCompoundPacket pkt) {
      if (pkt == null || pkt.packets == null || pkt.packets.length == 0) {
        return;
      }

      for (RTCPPacket rtcpPacket : pkt.packets) {
        switch (rtcpPacket.type) {
          case RTCPPacket.SR:
            RTCPSRPacket srPacket = (RTCPSRPacket) rtcpPacket;

            // The media sender SSRC.
            int ssrc = srPacket.ssrc;

            // Convert 64-bit NTP timestamp to Java standard time.
            // Note that java time (milliseconds) by definition has
            // less precision then NTP time (picoseconds) so
            // converting NTP timestamp to java time and back to NTP
            // timestamp loses precision. For example, Tue, Dec 17
            // 2002 09:07:24.810 EST is represented by a single
            // Java-based time value of f22cd1fc8a, but its NTP
            // equivalent are all values ranging from
            // c1a9ae1c.cf5c28f5 to c1a9ae1c.cf9db22c.

            // Use round-off on fractional part to preserve going to
            // lower precision
            long fraction = Math.round(1000D * srPacket.ntptimestamplsw / 0x100000000L);
            /*
             * If the most significant bit (MSB) on the seconds
             * field is set we use a different time base. The
             * following text is a quote from RFC-2030 (SNTP v4):
             *
             * If bit 0 is set, the UTC time is in the range
             * 1968-2036 and UTC time is reckoned from 0h 0m 0s UTC
             * on 1 January 1900. If bit 0 is not set, the time is
             * in the range 2036-2104 and UTC time is reckoned from
             * 6h 28m 16s UTC on 7 February 2036.
             */
            long msb = srPacket.ntptimestampmsw & 0x80000000L;
            long remoteTime =
                (msb == 0)
                    // use base: 7-Feb-2036 @ 06:28:16 UTC
                    ? msb0baseTime + (srPacket.ntptimestampmsw * 1000) + fraction
                    // use base: 1-Jan-1900 @ 01:00:00 UTC
                    : msb1baseTime + (srPacket.ntptimestampmsw * 1000) + fraction;

            // Estimate the clock rate of the sender.
            int frequencyHz = -1;
            if (receivedClocks.containsKey(ssrc)) {
              // Calculate the clock rate.
              ReceivedRemoteClock oldStats = receivedClocks.get(ssrc);
              RemoteClock oldRemoteClock = oldStats.getRemoteClock();
              frequencyHz =
                  Math.round(
                      (float)
                              (((int) srPacket.rtptimestamp - oldRemoteClock.getRtpTimestamp())
                                  & 0xffffffffl)
                          / (remoteTime - oldRemoteClock.getRemoteTime()));
            }

            // Replace whatever was in there before.
            receivedClocks.put(
                ssrc,
                new ReceivedRemoteClock(
                    ssrc, remoteTime, (int) srPacket.rtptimestamp, frequencyHz));
            break;
          case RTCPPacket.SDES:
            break;
        }
      }
    }

    /**
     * Estimate the <tt>RemoteClock</tt> of a given RTP stream (identified by its SSRC) at a given
     * time.
     *
     * @param ssrc the SSRC of the RTP stream whose <tt>RemoteClock</tt> we want to estimate.
     * @param time the local time that will be mapped to a remote time.
     * @return An estimation of the <tt>RemoteClock</tt> at time "time".
     */
    public RemoteClock estimate(int ssrc, long time) {
      ReceivedRemoteClock receivedRemoteClock = receivedClocks.get(ssrc);
      if (receivedRemoteClock == null || receivedRemoteClock.getFrequencyHz() == -1) {
        // We can't continue if we don't have NTP and RTP timestamps
        // and/or the original sender frequency, so move to the next
        // one.
        return null;
      }

      long delayMillis = time - receivedRemoteClock.getReceivedTime();

      // Estimate the remote wall clock.
      long remoteTime = receivedRemoteClock.getRemoteClock().getRemoteTime();
      long estimatedRemoteTime = remoteTime + delayMillis;

      // Drift the RTP timestamp.
      int rtpTimestamp =
          receivedRemoteClock.getRemoteClock().getRtpTimestamp()
              + ((int) delayMillis) * (receivedRemoteClock.getFrequencyHz() / 1000);
      return new RemoteClock(estimatedRemoteTime, rtpTimestamp);
    }
  }

  /** Keeps track of the CNAMEs of the RTP streams that we've seen. */
  class CNAMERegistry extends ConcurrentHashMap<Integer, byte[]> {
    /** @param inPacket */
    public void update(RTCPCompoundPacket inPacket) {
      // Update CNAMEs.
      if (inPacket == null || inPacket.packets == null || inPacket.packets.length == 0) {
        return;
      }

      for (RTCPPacket p : inPacket.packets) {
        switch (p.type) {
          case RTCPPacket.SDES:
            RTCPSDESPacket sdesPacket = (RTCPSDESPacket) p;
            if (sdesPacket.sdes == null || sdesPacket.sdes.length == 0) {
              continue;
            }

            for (RTCPSDES chunk : sdesPacket.sdes) {
              if (chunk.items == null || chunk.items.length == 0) {
                continue;
              }

              for (RTCPSDESItem sdesItm : chunk.items) {
                if (sdesItm.type != RTCPSDESItem.CNAME) {
                  continue;
                }

                this.put(chunk.ssrc, sdesItm.data);
              }
            }
            break;
        }
      }
    }
  }
}