Add a field trial to produce VideoFrames in camera capturers.

webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java

 /*
  *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 package org.appspot.apprtc;
 
 import android.content.Context;
 import android.os.Environment;
 import android.os.ParcelFileDescriptor;
 import android.util.Log;
 import java.io.File;
 import java.io.IOException;
 import java.nio.ByteBuffer;
 import java.util.ArrayList;
 import java.util.Arrays;
 import java.util.Collections;
 import java.util.EnumSet;
 import java.util.Iterator;
 import java.util.LinkedList;
 import java.util.List;
 import java.util.Timer;
 import java.util.TimerTask;
+import java.util.concurrent.ExecutorService;
 import java.util.concurrent.Executors;
-import java.util.concurrent.ExecutorService;
 import java.util.regex.Matcher;
 import java.util.regex.Pattern;
 import org.appspot.apprtc.AppRTCClient.SignalingParameters;
 import org.webrtc.AudioSource;
 import org.webrtc.AudioTrack;
 import org.webrtc.CameraVideoCapturer;
 import org.webrtc.DataChannel;
 import org.webrtc.EglBase;
 import org.webrtc.IceCandidate;
 import org.webrtc.Logging;
 import org.webrtc.MediaConstraints;
 import org.webrtc.MediaStream;
 import org.webrtc.PeerConnection;
 import org.webrtc.PeerConnection.IceConnectionState;
 import org.webrtc.PeerConnectionFactory;
 import org.webrtc.RtpParameters;
 import org.webrtc.RtpReceiver;
 import org.webrtc.RtpSender;
 import org.webrtc.SdpObserver;
 import org.webrtc.SessionDescription;
 import org.webrtc.StatsObserver;
 import org.webrtc.StatsReport;
 import org.webrtc.VideoCapturer;
 import org.webrtc.VideoRenderer;
 import org.webrtc.VideoSource;
 import org.webrtc.VideoTrack;
 import org.webrtc.voiceengine.WebRtcAudioManager;
 import org.webrtc.voiceengine.WebRtcAudioRecord;
-import org.webrtc.voiceengine.WebRtcAudioTrack;
 import org.webrtc.voiceengine.WebRtcAudioRecord.AudioRecordStartErrorCode;
 import org.webrtc.voiceengine.WebRtcAudioRecord.WebRtcAudioRecordErrorCallback;
+import org.webrtc.voiceengine.WebRtcAudioTrack;
 import org.webrtc.voiceengine.WebRtcAudioTrack.WebRtcAudioTrackErrorCallback;
 import org.webrtc.voiceengine.WebRtcAudioUtils;
 
 /**
  * Peer connection client implementation.
  *
  * <p>All public methods are routed to local looper thread.
  * All PeerConnectionEvents callbacks are invoked from the same looper thread.
  * This class is a singleton.
  */
@@ skipping 10 matching lines @@
   private static final String AUDIO_CODEC_OPUS = "opus";
   private static final String AUDIO_CODEC_ISAC = "ISAC";
   private static final String VIDEO_CODEC_PARAM_START_BITRATE = "x-google-start-bitrate";
   private static final String VIDEO_FLEXFEC_FIELDTRIAL =
       "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
   private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/";
   private static final String VIDEO_H264_HIGH_PROFILE_FIELDTRIAL =
       "WebRTC-H264HighProfile/Enabled/";
   private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL =
       "WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/";
+  private static final String VIDEO_FRAME_EMIT_FIELDTRIAL =
+      PeerConnectionFactory.VIDEO_FRAME_EMIT_TRIAL + "/" + PeerConnectionFactory.TRIAL_ENABLED
+      + "/";
   private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate";
   private static final String AUDIO_ECHO_CANCELLATION_CONSTRAINT = "googEchoCancellation";
   private static final String AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT = "googAutoGainControl";
   private static final String AUDIO_HIGH_PASS_FILTER_CONSTRAINT = "googHighpassFilter";
   private static final String AUDIO_NOISE_SUPPRESSION_CONSTRAINT = "googNoiseSuppression";
   private static final String AUDIO_LEVEL_CONTROL_CONSTRAINT = "levelControl";
   private static final String DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT = "DtlsSrtpKeyAgreement";
   private static final int HD_VIDEO_WIDTH = 1280;
   private static final int HD_VIDEO_HEIGHT = 720;
   private static final int BPS_IN_KBPS = 1000;
@@ skipping 289 matching lines @@
     String fieldTrials = "";
     if (peerConnectionParameters.videoFlexfecEnabled) {
       fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL;
       Log.d(TAG, "Enable FlexFEC field trial.");
     }
     fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL;
     if (peerConnectionParameters.disableWebRtcAGCAndHPF) {
       fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL;
       Log.d(TAG, "Disable WebRTC AGC field trial.");
     }
+    fieldTrials += VIDEO_FRAME_EMIT_FIELDTRIAL;
 
     // Check preferred video codec.
     preferredVideoCodec = VIDEO_CODEC_VP8;
     if (videoCallEnabled && peerConnectionParameters.videoCodec != null) {
       switch (peerConnectionParameters.videoCodec) {
         case VIDEO_CODEC_VP8:
           preferredVideoCodec = VIDEO_CODEC_VP8;
           break;
         case VIDEO_CODEC_VP9:
           preferredVideoCodec = VIDEO_CODEC_VP9;
@@ skipping 941 matching lines @@
     public void onCreateFailure(final String error) {
       reportError("createSDP error: " + error);
     }
 
     @Override
     public void onSetFailure(final String error) {
       reportError("setSDP error: " + error);
     }
   }
 }