Java WebRtcEndpoint.connect方法代码示例

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
private void connectAccordingToProfile(WebRtcEndpoint webRtcEndpoint, RecorderEndpoint recorder,
    MediaProfileSpecType profile) {
  switch (profile) {
    case WEBM:
      webRtcEndpoint.connect(recorder, MediaType.AUDIO);
      webRtcEndpoint.connect(recorder, MediaType.VIDEO);
      break;
    case WEBM_AUDIO_ONLY:
      webRtcEndpoint.connect(recorder, MediaType.AUDIO);
      break;
    case WEBM_VIDEO_ONLY:
      webRtcEndpoint.connect(recorder, MediaType.VIDEO);
      break;
    default:
      throw new UnsupportedOperationException("Unsupported profile for this tutorial: " + profile);
  }
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
private void start(WebSocketSession session, JsonObject jsonMessage) {
	try {
		// Media Logic (Media Pipeline and Elements)
		MediaPipeline pipeline = kurento.createMediaPipeline();
		pipelines.put(session.getId(), pipeline);

		WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline)
				.build();
		FaceOverlayFilter faceOverlayFilter = new FaceOverlayFilter.Builder(
				pipeline).build();
		faceOverlayFilter.setOverlayedImage(
				"http://files.kurento.org/imgs/mario-wings.png", -0.35F,
				-1.2F, 1.6F, 1.6F);

		webRtcEndpoint.connect(faceOverlayFilter);
		faceOverlayFilter.connect(webRtcEndpoint);

		// SDP negotiation (offer and answer)
		String sdpOffer = jsonMessage.get("sdpOffer").getAsString();
		String sdpAnswer = webRtcEndpoint.processOffer(sdpOffer);

		// Sending response back to client
		JsonObject response = new JsonObject();
		response.addProperty("id", "startResponse");
		response.addProperty("sdpAnswer", sdpAnswer);
		session.sendMessage(new TextMessage(response.toString()));
	} catch (Throwable t) {
		sendError(session, t.getMessage());
	}
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@RequestMapping(value = "/helloworld", method = RequestMethod.POST)
private String processRequest(@RequestBody String sdpOffer)
		throws IOException {

	// Media Logic
	MediaPipeline pipeline = kurento.createMediaPipeline();
	WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline)
			.build();
	webRtcEndpoint.connect(webRtcEndpoint);

	// SDP negotiation (offer and answer)
	String responseSdp = webRtcEndpoint.processOffer(sdpOffer);
	return responseSdp;
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testWebRtcStabilityLoopback() throws Exception {
  final int playTime = Integer.parseInt(
      System.getProperty("test.webrtcstability.playtime", String.valueOf(DEFAULT_PLAYTIME)));

  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(mp).build();
  webRtcEndpoint.connect(webRtcEndpoint);

  // WebRTC
  getPage().subscribeEvents("playing");
  getPage().initWebRtc(webRtcEndpoint, WebRtcChannel.VIDEO_ONLY, WebRtcMode.SEND_RCV);

  // Latency assessment
  LatencyController cs = new LatencyController("WebRTC in loopback");
  getPage().activateLatencyControl(VideoTagType.LOCAL.getId(), VideoTagType.REMOTE.getId());
  cs.checkLatency(playTime, TimeUnit.MINUTES, getPage());

  // Release Media Pipeline
  mp.release();

  // Draw latency results (PNG chart and CSV file)
  cs.drawChart(getDefaultOutputFile(".png"), 500, 270);
  cs.writeCsv(getDefaultOutputFile(".csv"));
  cs.logLatencyErrorrs();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testWebRtcStabilityBack2Back() throws Exception {
  final int playTime = Integer.parseInt(System
      .getProperty("test.webrtc.stability.back2back.playtime", String.valueOf(DEFAULT_PLAYTIME)));

  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEndpoint1 = new WebRtcEndpoint.Builder(mp).build();
  WebRtcEndpoint webRtcEndpoint2 = new WebRtcEndpoint.Builder(mp).build();
  webRtcEndpoint1.connect(webRtcEndpoint2);
  webRtcEndpoint2.connect(webRtcEndpoint1);

  // Latency control
  LatencyController cs = new LatencyController("WebRTC latency control");

  // WebRTC
  getPresenter().subscribeLocalEvents("playing");
  getPresenter().initWebRtc(webRtcEndpoint1, WebRtcChannel.VIDEO_ONLY, WebRtcMode.SEND_ONLY);
  getViewer().subscribeEvents("playing");
  getViewer().initWebRtc(webRtcEndpoint2, WebRtcChannel.VIDEO_ONLY, WebRtcMode.RCV_ONLY);

  // Latency assessment
  cs.checkLatency(playTime, TimeUnit.MINUTES, getPresenter(), getViewer());

  // Release Media Pipeline
  mp.release();

  // Draw latency results (PNG chart and CSV file)
  cs.drawChart(getDefaultOutputFile(".png"), 500, 270);
  cs.writeCsv(getDefaultOutputFile(".csv"));
  cs.logLatencyErrorrs();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testDispatcherPlayer() throws Exception {
  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();

  WebRtcEndpoint webRtcEp = new WebRtcEndpoint.Builder(mp).useDataChannels().build();
  WebRtcEndpoint webRtcEp2 = new WebRtcEndpoint.Builder(mp).useDataChannels().build();

  webRtcEp.connect(webRtcEp2);
  webRtcEp2.connect(webRtcEp);

  // Test execution
  getPage(0).initWebRtc(webRtcEp, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.RCV_ONLY, true);
  getPage(1).initWebRtc(webRtcEp2, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.RCV_ONLY, true);

  Thread.sleep(8000);

  for (int i = 0; i < TIMES; i++) {
    String messageSentBrower0 = "Data sent from the browser0. Message" + i;
    String messageSentBrower1 = "Data sent from the browser1. Message" + i;

    getPage(0).sendDataByDataChannel(messageSentBrower0);
    getPage(1).sendDataByDataChannel(messageSentBrower1);

    Assert.assertTrue("The message should be: " + messageSentBrower1,
        getPage(0).compareDataChannelMessage(messageSentBrower1));

    Assert.assertTrue("The message should be: " + messageSentBrower0,
        getPage(1).compareDataChannelMessage(messageSentBrower0));
  }

  // Release Media Pipeline
  mp.release();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testWebRtcFaceOverlay() throws InterruptedException {

  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(mp).build();
  FaceOverlayFilter faceOverlayFilter = new FaceOverlayFilter.Builder(mp).build();
  webRtcEndpoint.connect(faceOverlayFilter);
  faceOverlayFilter.connect(webRtcEndpoint);

  // Start WebRTC and wait for playing event
  getPage().subscribeEvents("playing");
  getPage().initWebRtc(webRtcEndpoint, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_RCV);
  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage().waitForEvent("playing"));

  // Guard time to play the video
  int playTime = Integer.parseInt(
      System.getProperty("test.webrtcfaceoverlay.playtime", String.valueOf(DEFAULT_PLAYTIME)));
  waitSeconds(playTime);

  // Assertions
  double currentTime = getPage().getCurrentTime();
  Assert.assertTrue(
      "Error in play time (expected: " + playTime + " sec, real: " + currentTime + " sec)",
      getPage().compare(playTime, currentTime));
  Assert.assertTrue("The color of the video should be green",
      getPage().similarColor(CHROME_VIDEOTEST_COLOR));

  // Release Media Pipeline
  mp.release();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
public void doTest(BrowserType browserType, String videoPath, String audioUrl, Color color)
    throws InterruptedException {
  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(mp).build();
  webRtcEndpoint.connect(webRtcEndpoint);

  getPage().subscribeEvents("playing");
  getPage().initWebRtc(webRtcEndpoint, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_RCV);

  // Wait until event playing in the remote stream
  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage().waitForEvent("playing"));

  // Guard time to play the video
  Thread.sleep(TimeUnit.SECONDS.toMillis(PLAYTIME));

  // Assert play time
  double currentTime = getPage().getCurrentTime();
  Assert.assertTrue("Error in play time of player (expected: " + PLAYTIME + " sec, real: "
      + currentTime + " sec)", getPage().compare(PLAYTIME, currentTime));

  // Assert color
  if (color != null) {
    Assert.assertTrue("The color of the video should be " + color, getPage().similarColor(color));
  }

  // Assert audio quality
  if (audioUrl != null) {
    float realPesqMos = Ffmpeg.getPesqMos(audioUrl, AUDIO_SAMPLE_RATE);
    Assert.assertTrue("Bad perceived audio quality: PESQ MOS too low (expected=" + MIN_PESQ_MOS
        + ", real=" + realPesqMos + ")", realPesqMos >= MIN_PESQ_MOS);
  }

  // Release Media Pipeline
  mp.release();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
private String createSdpResponseForUser(RoomParticipant sender, String sdpOffer) {

    WebRtcEndpoint receivingEndpoint = sender.getReceivingEndpoint();
    if (receivingEndpoint == null) {
      log.warn("PARTICIPANT {}: Trying to connect to a user without receiving endpoint "
          + "(it seems is not yet fully connected)", this.name);
      return null;
    }

    if (sender.getName().equals(name)) {
      // FIXME: Use another message type for receiving sdp offer
      log.debug("PARTICIPANT {}: configuring loopback", this.name);
      return receivingEndpoint.processOffer(sdpOffer);
    }

    if (sendingEndpoints.get(sender.getName()) != null) {
      log.warn("PARTICIPANT {}: There is a sending endpoint to user {} "
          + "when trying to create another one", this.name, sender.getName());
      return null;
    }

    log.debug("PARTICIPANT {}: Creating a sending endpoint to user {}", this.name,
        sender.getName());

    WebRtcEndpoint sendingEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
    WebRtcEndpoint oldSendingEndpoint =
        sendingEndpoints.putIfAbsent(sender.getName(), sendingEndpoint);

    if (oldSendingEndpoint != null) {
      log.warn(
          "PARTICIPANT {}: 2 threads have simultaneously created a sending endpoint for user {}",
          this.name, sender.getName());
      return null;
    }

    log.debug("PARTICIPANT {}: Created sending endpoint for user {}", this.name, sender.getName());
    try {
      receivingEndpoint = sender.getReceivingEndpoint();
      if (receivingEndpoint != null) {
        receivingEndpoint.connect(sendingEndpoint);
        return sendingEndpoint.processOffer(sdpOffer);
      }

    } catch (KurentoServerException e) {

      // TODO Check object status when KurentoClient set this info in the
      // object
      if (e.getCode() == 40101) {
        log.warn("Receiving endpoint is released when trying to connect a sending endpoint to it",
            e);
      } else {
        log.error("Exception connecting receiving endpoint to sending endpoint", e);
        sendingEndpoint.release(new Continuation<Void>() {
          @Override
          public void onSuccess(Void result) throws Exception {

          }

          @Override
          public void onError(Throwable cause) throws Exception {
            log.error("Exception releasing WebRtcEndpoint", cause);
          }
        });
      }

      sendingEndpoints.remove(sender.getName());

      releaseEndpoint(sender.getName(), sendingEndpoint);
    }

    return null;
  }
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testAlphaBlendingWebRtc() throws Exception {
  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEpRed = new WebRtcEndpoint.Builder(mp).build();
  WebRtcEndpoint webRtcEpGreen = new WebRtcEndpoint.Builder(mp).build();
  WebRtcEndpoint webRtcEpBlue = new WebRtcEndpoint.Builder(mp).build();

  AlphaBlending alphaBlending = new AlphaBlending.Builder(mp).build();
  HubPort hubPort1 = new HubPort.Builder(alphaBlending).build();
  HubPort hubPort2 = new HubPort.Builder(alphaBlending).build();
  HubPort hubPort3 = new HubPort.Builder(alphaBlending).build();

  webRtcEpRed.connect(hubPort1);
  webRtcEpGreen.connect(hubPort2);
  webRtcEpBlue.connect(hubPort3);

  WebRtcEndpoint webRtcEpAlphabaBlending = new WebRtcEndpoint.Builder(mp).build();
  HubPort hubPort4 = new HubPort.Builder(alphaBlending).build();
  hubPort4.connect(webRtcEpAlphabaBlending);

  alphaBlending.setMaster(hubPort1, 1);

  alphaBlending.setPortProperties(0F, 0F, 8, 0.2F, 0.2F, hubPort2);
  alphaBlending.setPortProperties(0.4F, 0.4F, 7, 0.2F, 0.2F, hubPort3);

  getPage(BROWSER1).subscribeLocalEvents("playing");
  getPage(BROWSER1).initWebRtc(webRtcEpRed, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);

  getPage(BROWSER2).subscribeLocalEvents("playing");
  getPage(BROWSER2).initWebRtc(webRtcEpGreen, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.SEND_ONLY);

  getPage(BROWSER3).subscribeLocalEvents("playing");
  getPage(BROWSER3).initWebRtc(webRtcEpBlue, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);

  getPage(BROWSER4).subscribeEvents("playing");
  getPage(BROWSER4).initWebRtc(webRtcEpAlphabaBlending, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.RCV_ONLY);

  // Assertions
  Assert.assertTrue("Upper left part of the video must be blue",
      getPage(BROWSER4).similarColorAt(Color.GREEN, 0, 0));
  Assert.assertTrue("Lower right part of the video must be red",
      getPage(BROWSER4).similarColorAt(Color.RED, 315, 235));
  Assert.assertTrue("Center of the video must be blue",
      getPage(BROWSER4).similarColorAt(Color.BLUE, 160, 120));

  // alphaBlending.setMaster(hubPort3, 1);
  alphaBlending.setPortProperties(0.8F, 0.8F, 7, 0.2F, 0.2F, hubPort3);

  Assert.assertTrue("Lower right part of the video must be blue",
      getPage(BROWSER4).similarColorAt(Color.BLUE, 315, 235));
  Assert.assertTrue("Center of the video must be red",
      getPage(BROWSER4).similarColorAt(Color.RED, 160, 120));
  Thread.sleep(PLAYTIME * 1000);
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testWebRtcStabilityRtpH264() throws Exception {
  final int playTime =
      Integer.parseInt(System.getProperty("test.webrtc.stability.switch.webrtc2rtp.playtime",
          String.valueOf(DEFAULT_PLAYTIME)));

  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(mp).build();
  RtpEndpoint rtpEndpoint1 = new RtpEndpoint.Builder(mp).build();
  RtpEndpoint rtpEndpoint2 = new RtpEndpoint.Builder(mp).build();
  webRtcEndpoint.connect(rtpEndpoint1);
  rtpEndpoint2.connect(webRtcEndpoint);

  // RTP session (rtpEndpoint1 --> rtpEndpoint2)
  String sdpOffer = rtpEndpoint1.generateOffer();
  log.debug("SDP offer in rtpEndpoint1\n{}", sdpOffer);

  // SDP mangling
  sdpOffer = SdpUtils.mangleSdp(sdpOffer, REMOVE_CODECS);
  log.debug("SDP offer in rtpEndpoint1 after mangling\n{}", sdpOffer);

  String sdpAnswer1 = rtpEndpoint2.processOffer(sdpOffer);
  log.debug("SDP answer in rtpEndpoint2\n{}", sdpAnswer1);
  String sdpAnswer2 = rtpEndpoint1.processAnswer(sdpAnswer1);
  log.debug("SDP answer in rtpEndpoint1\n{}", sdpAnswer2);

  // Latency controller
  LatencyController cs = new LatencyController();

  // WebRTC
  getPage().subscribeEvents("playing");
  getPage().initWebRtc(webRtcEndpoint, WebRtcChannel.VIDEO_ONLY, WebRtcMode.SEND_RCV);

  // Assertion: wait to playing event in browser
  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage().waitForEvent("playing"));

  // Latency assessment
  getPage().activateLatencyControl(VideoTagType.LOCAL.getId(), VideoTagType.REMOTE.getId());
  cs.checkLatency(playTime, TimeUnit.MINUTES, getPage());

  // Release Media Pipeline
  mp.release();

  // Draw latency results (PNG chart and CSV file)
  cs.drawChart(getDefaultOutputFile(".png"), 500, 270);
  cs.writeCsv(getDefaultOutputFile(".csv"));
  cs.logLatencyErrorrs();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
public void doTestWithPlayer(MediaProfileSpecType mediaProfileSpecType, String expectedVideoCodec,
    String expectedAudioCodec, String extension, String mediaUrlPlayer) throws Exception {
  // Media Pipeline #1
  getPage(BROWSER2).close();
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  final CountDownLatch errorPipelinelatch = new CountDownLatch(1);

  mp.addErrorListener(new EventListener<ErrorEvent>() {

    @Override
    public void onEvent(ErrorEvent event) {
      msgError = "Description:" + event.getDescription() + "; Error code:" + event.getType();
      errorPipelinelatch.countDown();
    }
  });

  WebRtcEndpoint webRtcEpRed = new WebRtcEndpoint.Builder(mp).build();
  PlayerEndpoint playerEp = new PlayerEndpoint.Builder(mp, mediaUrlPlayer).build();

  String recordingFile = getRecordUrl(extension);
  RecorderEndpoint recorderEp = new RecorderEndpoint.Builder(mp, recordingFile)
      .withMediaProfile(mediaProfileSpecType).build();

  // Test execution
  getPage(BROWSER1).subscribeLocalEvents("playing");
  long startWebrtc = System.currentTimeMillis();
  getPage(BROWSER1).initWebRtc(webRtcEpRed, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);

  webRtcEpRed.connect(recorderEp);
  recorderEp.record();

  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage(BROWSER1).waitForEvent("playing"));
  long webrtcRedConnectionTime = System.currentTimeMillis() - startWebrtc;
  Thread.sleep(TimeUnit.SECONDS.toMillis(PLAYTIME) / N_PLAYER);

  startWebrtc = System.currentTimeMillis();

  playerEp.play();
  playerEp.connect(recorderEp);
  long playerEpConnectionTime = System.currentTimeMillis() - startWebrtc;
  Thread.sleep(TimeUnit.SECONDS.toMillis(PLAYTIME) / N_PLAYER);

  webRtcEpRed.connect(recorderEp);
  Thread.sleep(TimeUnit.SECONDS.toMillis(PLAYTIME) / N_PLAYER);

  // Release Media Pipeline #1
  saveGstreamerDot(mp);

  final CountDownLatch recorderLatch = new CountDownLatch(1);
  recorderEp.stopAndWait(new Continuation<Void>() {

    @Override
    public void onSuccess(Void result) throws Exception {
      recorderLatch.countDown();
    }

    @Override
    public void onError(Throwable cause) throws Exception {
      recorderLatch.countDown();
    }
  });

  Assert.assertTrue("Not stop properly",
      recorderLatch.await(getPage(BROWSER1).getTimeout(), TimeUnit.SECONDS));
  mp.release();

  Assert.assertTrue(msgError, errorPipelinelatch.getCount() == 1);

  final long playtime = PLAYTIME
      + TimeUnit.MILLISECONDS.toSeconds((2 * webrtcRedConnectionTime) + playerEpConnectionTime);

  checkRecordingFile(recordingFile, BROWSER3, EXPECTED_COLORS, playtime, expectedVideoCodec,
      expectedAudioCodec);
  success = true;
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
public void doTest(MediaProfileSpecType mediaProfileSpecType, String expectedVideoCodec,
    String expectedAudioCodec, String extension) throws Exception {

  String multiSlashses = File.separator + File.separator + File.separator;
  final CountDownLatch recorderLatch = new CountDownLatch(1);

  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEp = new WebRtcEndpoint.Builder(mp).build();

  String recordingFile = getRecordUrl(extension).replace(getSimpleTestName(),
      new Date().getTime() + File.separator + getSimpleTestName());

  String recordingFileWithMultiSlashes = recordingFile.replace(File.separator, multiSlashses);

  log.debug("The path with multi slash is {} ", recordingFileWithMultiSlashes);

  RecorderEndpoint recorderEp = new RecorderEndpoint.Builder(mp, recordingFileWithMultiSlashes)
      .withMediaProfile(mediaProfileSpecType).build();
  webRtcEp.connect(webRtcEp);
  webRtcEp.connect(recorderEp);

  getPage().subscribeEvents("playing");
  getPage().initWebRtc(webRtcEp, AUDIO_AND_VIDEO, WebRtcMode.SEND_RCV);
  recorderEp.record();

  // Wait until event playing in the remote stream
  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage().waitForEvent("playing"));

  Thread.sleep(SECONDS.toMillis(PLAYTIME));

  recorderEp.stopAndWait(new Continuation<Void>() {

    @Override
    public void onSuccess(Void result) throws Exception {
      recorderLatch.countDown();
    }

    @Override
    public void onError(Throwable cause) throws Exception {
      recorderLatch.countDown();
    }
  });

  Assert.assertTrue("Not stop properly",
      recorderLatch.await(getPage().getTimeout(), TimeUnit.SECONDS));

  // Wait until file exists
  waitForFileExists(recordingFile);

  AssertMedia.assertCodecs(recordingFile, expectedVideoCodec, expectedAudioCodec);
  mp.release();
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testCompositeRecorder() throws Exception {

  // MediaPipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();

  Composite composite = new Composite.Builder(mp).build();

  HubPort hubPort1 = new HubPort.Builder(composite).build();
  WebRtcEndpoint webRtcEpRed = new WebRtcEndpoint.Builder(mp).build();
  webRtcEpRed.connect(hubPort1);

  HubPort hubPort2 = new HubPort.Builder(composite).build();
  WebRtcEndpoint webRtcEpGreen = new WebRtcEndpoint.Builder(mp).build();
  webRtcEpGreen.connect(hubPort2, MediaType.AUDIO);

  HubPort hubPort3 = new HubPort.Builder(composite).build();
  WebRtcEndpoint webRtcEpBlue = new WebRtcEndpoint.Builder(mp).build();
  webRtcEpBlue.connect(hubPort3, MediaType.AUDIO);

  HubPort hubPort4 = new HubPort.Builder(composite).build();
  WebRtcEndpoint webRtcEpWhite = new WebRtcEndpoint.Builder(mp).build();
  webRtcEpWhite.connect(hubPort4, MediaType.AUDIO);

  String recordingFile = getDefaultOutputFile(EXTENSION_WEBM);
  RecorderEndpoint recorderEp =
      new RecorderEndpoint.Builder(mp, Protocol.FILE + recordingFile).build();
  HubPort hubPort5 = new HubPort.Builder(composite).build();
  hubPort5.connect(recorderEp);

  // WebRTC browsers
  getPage(BROWSER2).initWebRtc(webRtcEpRed, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);
  getPage(BROWSER3).initWebRtc(webRtcEpGreen, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.SEND_ONLY);
  getPage(BROWSER4).initWebRtc(webRtcEpBlue, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);
  getPage(BROWSER5).initWebRtc(webRtcEpWhite, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.SEND_ONLY);

  recorderEp.record();

  Thread.sleep(PLAYTIME * 1000);

  final CountDownLatch recorderLatch = new CountDownLatch(1);
  recorderEp.stopAndWait(new Continuation<Void>() {

    @Override
    public void onSuccess(Void result) throws Exception {
      recorderLatch.countDown();
    }

    @Override
    public void onError(Throwable cause) throws Exception {
      recorderLatch.countDown();
    }
  });

  Assert.assertTrue("Not stop properly",
      recorderLatch.await(getPage(BROWSER1).getTimeout(), TimeUnit.SECONDS));

  mp.release();

  // Media Pipeline #2
  MediaPipeline mp2 = kurentoClient.createMediaPipeline();
  PlayerEndpoint playerEp2 =
      new PlayerEndpoint.Builder(mp2, Protocol.FILE + recordingFile).build();
  WebRtcEndpoint webRtcEp2 = new WebRtcEndpoint.Builder(mp2).build();
  playerEp2.connect(webRtcEp2);

  // Playing the recorded file
  launchBrowser(mp2, webRtcEp2, playerEp2, null, EXPECTED_VIDEO_CODEC_WEBM,
      EXPECTED_AUDIO_CODEC_WEBM, recordingFile, Color.RED, 0, 0, PLAYTIME);

  // Release Media Pipeline #2
  mp2.release();

  success = true;
}
 

import org.kurento.client.WebRtcEndpoint; //导入方法依赖的package包/类
@Test
public void testCompositeWebRtc() throws Exception {
  // Media Pipeline
  MediaPipeline mp = kurentoClient.createMediaPipeline();
  WebRtcEndpoint webRtcEpRed = new WebRtcEndpoint.Builder(mp).build();
  WebRtcEndpoint webRtcEpGreen = new WebRtcEndpoint.Builder(mp).build();
  WebRtcEndpoint webRtcEpBlue = new WebRtcEndpoint.Builder(mp).build();

  Composite composite = new Composite.Builder(mp).build();
  HubPort hubPort1 = new HubPort.Builder(composite).build();
  HubPort hubPort2 = new HubPort.Builder(composite).build();
  HubPort hubPort3 = new HubPort.Builder(composite).build();

  webRtcEpRed.connect(hubPort1);
  webRtcEpGreen.connect(hubPort2);
  webRtcEpBlue.connect(hubPort3);

  WebRtcEndpoint webRtcEpWhite = new WebRtcEndpoint.Builder(mp).build();
  HubPort hubPort4 = new HubPort.Builder(composite).build();
  webRtcEpWhite.connect(hubPort4);

  WebRtcEndpoint webRtcEpComposite = new WebRtcEndpoint.Builder(mp).build();
  HubPort hubPort5 = new HubPort.Builder(composite).build();
  hubPort5.connect(webRtcEpComposite);

  // WebRTC browsers
  getPage(BROWSER2).initWebRtc(webRtcEpRed, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);
  getPage(BROWSER3).initWebRtc(webRtcEpGreen, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.SEND_ONLY);
  getPage(BROWSER4).initWebRtc(webRtcEpBlue, WebRtcChannel.AUDIO_AND_VIDEO, WebRtcMode.SEND_ONLY);
  getPage(BROWSER5).initWebRtc(webRtcEpWhite, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.SEND_ONLY);

  getPage(BROWSER1).subscribeEvents("playing");
  getPage(BROWSER1).initWebRtc(webRtcEpComposite, WebRtcChannel.AUDIO_AND_VIDEO,
      WebRtcMode.RCV_ONLY);

  // Assertions
  Assert.assertTrue("Not received media (timeout waiting playing event)",
      getPage(BROWSER1).waitForEvent("playing"));
  Assert.assertTrue("Upper left part of the video must be red",
      getPage(BROWSER1).similarColorAt(Color.RED, 0, 0));
  Assert.assertTrue("Upper right part of the video must be green",
      getPage(BROWSER1).similarColorAt(Color.GREEN, 450, 0));
  Assert.assertTrue("Lower left part of the video must be blue",
      getPage(BROWSER1).similarColorAt(Color.BLUE, 0, 450));
  Assert.assertTrue("Lower right part of the video must be white",
      getPage(BROWSER1).similarColorAt(Color.WHITE, 450, 450));

  // Finally, a black & white filter is connected to one WebRTC
  GStreamerFilter bwFilter =
      new GStreamerFilter.Builder(mp, "videobalance saturation=0.0").build();
  webRtcEpRed.connect(bwFilter);
  bwFilter.connect(hubPort1);
  Thread.sleep(TimeUnit.SECONDS.toMillis(PLAYTIME));
  Assert.assertTrue("When connecting the filter, the upper left part of the video must be gray",
      getPage(BROWSER1).similarColorAt(new Color(75, 75, 75), 0, 0));

  // Release Media Pipeline
  mp.release();
}