Add an example app for Android native API.

The app is a simple loopback demo demonstrating the usage of Android
native API. This is an initial version and I will add support for
HW codecs etc. in the future.

Bug: webrtc:8769
Change-Id: Ifb6209769dabeb8ca3185b969a1ef8afd6d84390
Reviewed-on: https://webrtc-review.googlesource.com/60540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22385}
diff --git a/examples/androidnativeapi/AndroidManifest.xml b/examples/androidnativeapi/AndroidManifest.xml
new file mode 100644
index 0000000..19e4dc0
--- /dev/null
+++ b/examples/androidnativeapi/AndroidManifest.xml
@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+  package="org.webrtc.examples.androidnativeapi">
+
+  <uses-sdk android:minSdkVersion="19" android:targetSdkVersion="27" />
+
+  <uses-permission android:name="android.permission.INTERNET" />
+
+  <application
+    android:allowBackup="true"
+    android:label="@string/app_name"
+    android:supportsRtl="true">
+    <activity android:name=".MainActivity">
+      <intent-filter>
+        <action android:name="android.intent.action.MAIN" />
+
+        <category android:name="android.intent.category.LAUNCHER" />
+      </intent-filter>
+    </activity>
+  </application>
+
+</manifest>
diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn
new file mode 100644
index 0000000..a98d63d
--- /dev/null
+++ b/examples/androidnativeapi/BUILD.gn
@@ -0,0 +1,76 @@
+import("//webrtc.gni")
+
+rtc_android_apk("androidnativeapi") {
+  testonly = true
+  apk_name = "androidnativeapi"
+  android_manifest = "AndroidManifest.xml"
+
+  java_files = [
+    "java/org/webrtc/examples/androidnativeapi/MainActivity.java",
+    "java/org/webrtc/examples/androidnativeapi/CallClient.java",
+  ]
+
+  deps = [
+    ":resources",
+    "//sdk/android:libjingle_peerconnection_java",
+  ]
+
+  shared_libraries = [ ":examples_androidnativeapi_jni" ]
+}
+
+generate_jni("generated_jni") {
+  testonly = true
+  sources = [
+    "java/org/webrtc/examples/androidnativeapi/CallClient.java",
+  ]
+  jni_package = ""
+  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
+}
+
+rtc_shared_library("examples_androidnativeapi_jni") {
+  testonly = true
+  sources = [
+    "jni/androidcallclient.cc",
+    "jni/androidcallclient.h",
+    "jni/onload.cc",
+  ]
+
+  suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+  configs += [ "//build/config/android:hide_all_but_jni" ]
+
+  if (is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [
+      "//build/config/clang:extra_warnings",
+      "//build/config/clang:find_bad_constructs",
+    ]
+  }
+
+  deps = [
+    ":generated_jni",
+    "//api:libjingle_peerconnection_api",
+    "//api/audio_codecs:builtin_audio_decoder_factory",
+    "//api/audio_codecs:builtin_audio_encoder_factory",
+    "//logging:rtc_event_log_impl_base",
+    "//media:rtc_audio_video",
+    "//media:rtc_internal_video_codecs",
+    "//modules/audio_processing",
+    "//modules/utility:utility",
+    "//pc:libjingle_peerconnection",
+    "//pc:pc_test_utils",
+    "//rtc_base:rtc_base",
+    "//rtc_base:rtc_base_approved",
+    "//sdk/android:native_api_base",
+    "//sdk/android:native_api_jni",
+    "//sdk/android:native_api_video",
+    "//system_wrappers:field_trial_default",
+    "//system_wrappers:metrics_default",
+    "//system_wrappers:runtime_enabled_features_default",
+  ]
+}
+
+android_resources("resources") {
+  testonly = true
+  resource_dirs = [ "res" ]
+  custom_package = "org.webrtc.examples.androidnativeapi"
+}
diff --git a/examples/androidnativeapi/DEPS b/examples/androidnativeapi/DEPS
new file mode 100644
index 0000000..2d4c0d8
--- /dev/null
+++ b/examples/androidnativeapi/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+  "+modules/utility/include",
+  "+sdk/android/native_api",
+]
diff --git a/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java b/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
new file mode 100644
index 0000000..5c18cb7
--- /dev/null
+++ b/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2018 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.webrtc.examples.androidnativeapi;
+
+import android.os.Handler;
+import android.os.HandlerThread;
+import org.webrtc.NativeClassQualifiedName;
+import org.webrtc.VideoSink;
+
+public class CallClient {
+  private static final String TAG = "CallClient";
+
+  private final HandlerThread thread;
+  private final Handler handler;
+
+  private long nativeClient;
+
+  public CallClient() {
+    thread = new HandlerThread(TAG + "Thread");
+    thread.start();
+    handler = new Handler(thread.getLooper());
+    handler.post(() -> { nativeClient = nativeCreateClient(); });
+  }
+
+  public void call(VideoSink localSink, VideoSink remoteSink) {
+    handler.post(() -> { nativeCall(nativeClient, localSink, remoteSink); });
+  }
+
+  public void hangup() {
+    handler.post(() -> { nativeHangup(nativeClient); });
+  }
+
+  public void close() {
+    handler.post(() -> {
+      nativeDelete(nativeClient);
+      nativeClient = 0;
+    });
+    thread.quitSafely();
+  }
+
+  private static native long nativeCreateClient();
+  @NativeClassQualifiedName("webrtc_examples::AndroidCallClient")
+  private static native void nativeCall(long nativePtr, VideoSink localSink, VideoSink remoteSink);
+  @NativeClassQualifiedName("webrtc_examples::AndroidCallClient")
+  private static native void nativeHangup(long nativePtr);
+  @NativeClassQualifiedName("webrtc_examples::AndroidCallClient")
+  private static native void nativeDelete(long nativePtr);
+}
diff --git a/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java b/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
new file mode 100644
index 0000000..cb68c55
--- /dev/null
+++ b/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2018 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.webrtc.examples.androidnativeapi;
+
+import android.app.Activity;
+import android.os.Bundle;
+import android.widget.Button;
+import org.webrtc.ContextUtils;
+import org.webrtc.EglBase;
+import org.webrtc.GlRectDrawer;
+import org.webrtc.SurfaceViewRenderer;
+
+public class MainActivity extends Activity {
+  private CallClient callClient;
+  private EglBase eglBase;
+  private SurfaceViewRenderer localRenderer;
+  private SurfaceViewRenderer remoteRenderer;
+
+  @Override
+  protected void onCreate(Bundle savedInstance) {
+    ContextUtils.initialize(getApplicationContext());
+
+    super.onCreate(savedInstance);
+    setContentView(R.layout.activity_main);
+
+    System.loadLibrary("examples_androidnativeapi_jni");
+    callClient = new CallClient();
+
+    Button callButton = (Button) findViewById(R.id.call_button);
+    callButton.setOnClickListener((view) -> { callClient.call(localRenderer, remoteRenderer); });
+
+    Button hangupButton = (Button) findViewById(R.id.hangup_button);
+    hangupButton.setOnClickListener((view) -> { callClient.hangup(); });
+  }
+
+  @Override
+  protected void onStart() {
+    super.onStart();
+
+    eglBase = EglBase.create(null /* sharedContext */, EglBase.CONFIG_PLAIN);
+    localRenderer = (SurfaceViewRenderer) findViewById(R.id.local_renderer);
+    remoteRenderer = (SurfaceViewRenderer) findViewById(R.id.remote_renderer);
+
+    localRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */, EglBase.CONFIG_PLAIN,
+        new GlRectDrawer());
+    remoteRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */,
+        EglBase.CONFIG_PLAIN, new GlRectDrawer());
+  }
+
+  @Override
+  protected void onStop() {
+    callClient.hangup();
+
+    localRenderer.release();
+    remoteRenderer.release();
+    eglBase.release();
+
+    localRenderer = null;
+    remoteRenderer = null;
+    eglBase = null;
+
+    super.onStop();
+  }
+
+  @Override
+  protected void onDestroy() {
+    callClient.close();
+    callClient = null;
+
+    super.onDestroy();
+  }
+}
diff --git a/examples/androidnativeapi/jni/androidcallclient.cc b/examples/androidnativeapi/jni/androidcallclient.cc
new file mode 100644
index 0000000..657bce2
--- /dev/null
+++ b/examples/androidnativeapi/jni/androidcallclient.cc
@@ -0,0 +1,286 @@
+/*
+ *  Copyright 2018 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.
+ */
+
+#include "examples/androidnativeapi/jni/androidcallclient.h"
+
+#include <utility>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/peerconnectioninterface.h"
+#include "examples/androidnativeapi/generated_jni/jni/CallClient_jni.h"
+#include "media/engine/internaldecoderfactory.h"
+#include "media/engine/internalencoderfactory.h"
+#include "media/engine/webrtcmediaengine.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "pc/test/fakeperiodicvideocapturer.h"
+#include "rtc_base/ptr_util.h"
+#include "sdk/android/native_api/jni/java_types.h"
+#include "sdk/android/native_api/video/wrapper.h"
+
+namespace webrtc_examples {
+
+class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver {
+ public:
+  explicit PCObserver(AndroidCallClient* client);
+
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override;
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
+  void OnRenegotiationNeeded() override;
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+
+ private:
+  const AndroidCallClient* client_;
+};
+
+namespace {
+
+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  explicit CreateOfferObserver(
+      rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
+
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(const std::string& error) override;
+
+ private:
+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
+};
+
+class SetRemoteSessionDescriptionObserver
+    : public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
+};
+
+class SetLocalSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  void OnSuccess() override;
+  void OnFailure(const std::string& error) override;
+};
+
+}  // namespace
+
+AndroidCallClient::AndroidCallClient()
+    : call_started_(false), pc_observer_(rtc::MakeUnique<PCObserver>(this)) {
+  thread_checker_.DetachFromThread();
+  CreatePeerConnectionFactory();
+}
+
+void AndroidCallClient::Call(JNIEnv* env,
+                             const webrtc::JavaRef<jobject>& cls,
+                             const webrtc::JavaRef<jobject>& local_sink,
+                             const webrtc::JavaRef<jobject>& remote_sink) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  rtc::CritScope lock(&pc_mutex_);
+  if (call_started_) {
+    RTC_LOG(LS_WARNING) << "Call already started.";
+    return;
+  }
+  call_started_ = true;
+
+  local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj());
+  remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj());
+
+  // The fake video source wants to be created on the same thread as it is
+  // destroyed. It is destroyed on the signaling thread so we have to invoke
+  // here.
+  // TODO(sakal): Get picture from camera?
+  video_source_ = pcf_->CreateVideoSource(
+      signaling_thread_
+          ->Invoke<std::unique_ptr<webrtc::FakePeriodicVideoCapturer>>(
+              RTC_FROM_HERE, [&] {
+                return rtc::MakeUnique<webrtc::FakePeriodicVideoCapturer>();
+              }));
+
+  CreatePeerConnection();
+  Connect();
+}
+
+void AndroidCallClient::Hangup(JNIEnv* env,
+                               const webrtc::JavaRef<jobject>& cls) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  call_started_ = false;
+
+  {
+    rtc::CritScope lock(&pc_mutex_);
+    if (pc_ != nullptr) {
+      pc_->Close();
+      pc_ = nullptr;
+    }
+  }
+
+  local_sink_ = nullptr;
+  remote_sink_ = nullptr;
+  video_source_ = nullptr;
+}
+
+void AndroidCallClient::Delete(JNIEnv* env,
+                               const webrtc::JavaRef<jobject>& cls) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  delete this;
+}
+
+void AndroidCallClient::CreatePeerConnectionFactory() {
+  network_thread_ = rtc::Thread::CreateWithSocketServer();
+  network_thread_->SetName("network_thread", nullptr);
+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
+
+  worker_thread_ = rtc::Thread::Create();
+  worker_thread_->SetName("worker_thread", nullptr);
+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
+
+  signaling_thread_ = rtc::Thread::Create();
+  signaling_thread_->SetName("signaling_thread", nullptr);
+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
+
+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
+      cricket::WebRtcMediaEngineFactory::Create(
+          nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
+          webrtc::CreateBuiltinAudioDecoderFactory(),
+          rtc::MakeUnique<webrtc::InternalEncoderFactory>(),
+          rtc::MakeUnique<webrtc::InternalDecoderFactory>(),
+          nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
+  RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
+
+  pcf_ = CreateModularPeerConnectionFactory(
+      network_thread_.get(), worker_thread_.get(), signaling_thread_.get(),
+      std::move(media_engine), webrtc::CreateCallFactory(),
+      webrtc::CreateRtcEventLogFactory());
+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_;
+}
+
+void AndroidCallClient::CreatePeerConnection() {
+  rtc::CritScope lock(&pc_mutex_);
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  // DTLS SRTP has to be disabled for loopback to work.
+  config.enable_dtls_srtp = false;
+  pc_ = pcf_->CreatePeerConnection(config, nullptr /* port_allocator */,
+                                   nullptr /* cert_generator */,
+                                   pc_observer_.get());
+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
+      pcf_->CreateVideoTrack("video", video_source_);
+  local_video_track->AddOrUpdateSink(local_sink_.get(), rtc::VideoSinkWants());
+  pc_->AddTransceiver(local_video_track);
+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track;
+
+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
+       pc_->GetTransceivers()) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track =
+        tranceiver->receiver()->track();
+    if (track &&
+        track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+      static_cast<webrtc::VideoTrackInterface*>(track.get())
+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track;
+      break;
+    }
+  }
+}
+
+void AndroidCallClient::Connect() {
+  rtc::CritScope lock(&pc_mutex_);
+  pc_->CreateOffer(new rtc::RefCountedObject<CreateOfferObserver>(pc_),
+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+}
+
+AndroidCallClient::PCObserver::PCObserver(AndroidCallClient* client)
+    : client_(client) {}
+
+void AndroidCallClient::PCObserver::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnDataChannel(
+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
+  RTC_LOG(LS_INFO) << "OnDataChannel";
+}
+
+void AndroidCallClient::PCObserver::OnRenegotiationNeeded() {
+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
+}
+
+void AndroidCallClient::PCObserver::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
+  rtc::CritScope lock(&client_->pc_mutex_);
+  RTC_DCHECK(client_->pc_ != nullptr);
+  client_->pc_->AddIceCandidate(candidate);
+}
+
+CreateOfferObserver::CreateOfferObserver(
+    rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
+    : pc_(pc) {}
+
+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
+  std::string sdp;
+  desc->ToString(&sdp);
+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
+
+  // Ownership of desc was transferred to us, now we transfer it forward.
+  pc_->SetLocalDescription(
+      new rtc::RefCountedObject<SetLocalSessionDescriptionObserver>(), desc);
+
+  // Generate a fake answer.
+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
+  pc_->SetRemoteDescription(
+      std::move(answer),
+      new rtc::RefCountedObject<SetRemoteSessionDescriptionObserver>());
+}
+
+void CreateOfferObserver::OnFailure(const std::string& error) {
+  RTC_LOG(LS_INFO) << "Failed to create offer: " << error;
+}
+
+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(
+    webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
+}
+
+void SetLocalSessionDescriptionObserver::OnSuccess() {
+  RTC_LOG(LS_INFO) << "Set local description success!";
+}
+
+void SetLocalSessionDescriptionObserver::OnFailure(const std::string& error) {
+  RTC_LOG(LS_INFO) << "Set local description failure: " << error;
+}
+
+}  // namespace webrtc_examples
+
+static jlong JNI_CallClient_CreateClient(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jclass>& cls) {
+  return webrtc::NativeToJavaPointer(new webrtc_examples::AndroidCallClient());
+}
diff --git a/examples/androidnativeapi/jni/androidcallclient.h b/examples/androidnativeapi/jni/androidcallclient.h
new file mode 100644
index 0000000..2815b9d
--- /dev/null
+++ b/examples/androidnativeapi/jni/androidcallclient.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2018 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.
+ */
+
+#ifndef EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
+#define EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+
+#include "api/peerconnectioninterface.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "rtc_base/thread_checker.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+
+namespace webrtc_examples {
+
+class AndroidCallClient {
+ public:
+  AndroidCallClient();
+
+  void Call(JNIEnv* env,
+            const webrtc::JavaRef<jobject>& cls,
+            const webrtc::JavaRef<jobject>& local_sink,
+            const webrtc::JavaRef<jobject>& remote_sink);
+  void Hangup(JNIEnv* env, const webrtc::JavaRef<jobject>& cls);
+  // A helper method for Java code to delete this object. Calls delete this.
+  void Delete(JNIEnv* env, const webrtc::JavaRef<jobject>& cls);
+
+ private:
+  class PCObserver;
+
+  void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_);
+  void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
+  void Connect() RTC_RUN_ON(thread_checker_);
+
+  rtc::ThreadChecker thread_checker_;
+
+  bool call_started_ RTC_GUARDED_BY(thread_checker_);
+
+  const std::unique_ptr<PCObserver> pc_observer_;
+
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_
+      RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> signaling_thread_
+      RTC_GUARDED_BY(thread_checker_);
+
+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> local_sink_
+      RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_
+      RTC_GUARDED_BY(thread_checker_);
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source_
+      RTC_GUARDED_BY(thread_checker_);
+
+  rtc::CriticalSection pc_mutex_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_
+      RTC_GUARDED_BY(pc_mutex_);
+};
+
+}  // namespace webrtc_examples
+
+#endif  // EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
diff --git a/examples/androidnativeapi/jni/onload.cc b/examples/androidnativeapi/jni/onload.cc
new file mode 100644
index 0000000..4b4b5d9
--- /dev/null
+++ b/examples/androidnativeapi/jni/onload.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 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.
+ */
+
+#include <jni.h>
+
+#include "modules/utility/include/jvm_android.h"
+#include "rtc_base/ssladapter.h"
+#include "sdk/android/native_api/base/init.h"
+
+namespace webrtc_examples {
+
+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
+  webrtc::InitAndroid(jvm);
+  webrtc::JVM::Initialize(jvm);
+  RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+  return JNI_VERSION_1_6;
+}
+
+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
+  RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
+}
+
+}  // namespace webrtc_examples
diff --git a/examples/androidnativeapi/res/layout/activity_main.xml b/examples/androidnativeapi/res/layout/activity_main.xml
new file mode 100644
index 0000000..ac80373
--- /dev/null
+++ b/examples/androidnativeapi/res/layout/activity_main.xml
@@ -0,0 +1,52 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+  xmlns:app="http://schemas.android.com/apk/res-auto"
+  xmlns:tools="http://schemas.android.com/tools"
+  android:orientation="vertical"
+  android:layout_width="match_parent"
+  android:layout_height="match_parent"
+  android:padding="8dp"
+  tools:context="org.webrtc.examples.androidnativeapi.MainActivity">
+
+  <org.webrtc.SurfaceViewRenderer
+    android:id="@+id/local_renderer"
+    android:layout_width="match_parent"
+    android:layout_height="0dp"
+    android:layout_weight="1"
+    android:layout_margin="8dp"/>
+
+  <org.webrtc.SurfaceViewRenderer
+    android:id="@+id/remote_renderer"
+    android:layout_width="match_parent"
+    android:layout_height="0dp"
+    android:layout_weight="1"
+    android:layout_margin="8dp"/>
+
+
+  <LinearLayout
+    android:orientation="horizontal"
+    android:layout_width="match_parent"
+    android:layout_height="48dp"
+    style="?android:attr/buttonBarStyle">
+
+    <Button
+      android:id="@+id/call_button"
+      android:text="@string/call_button"
+      style="?android:attr/buttonBarButtonStyle"
+      android:layout_width="0dp"
+      android:layout_height="48dp"
+      android:layout_weight="1"
+      android:layout_margin="8dp"/>
+
+    <Button
+      android:id="@+id/hangup_button"
+      android:text="@string/hangup_button"
+      style="?android:attr/buttonBarButtonStyle"
+      android:layout_width="0dp"
+      android:layout_height="48dp"
+      android:layout_weight="1"
+      android:layout_margin="8dp"/>
+
+  </LinearLayout>
+
+</LinearLayout>
diff --git a/examples/androidnativeapi/res/values/strings.xml b/examples/androidnativeapi/res/values/strings.xml
new file mode 100644
index 0000000..a00920c
--- /dev/null
+++ b/examples/androidnativeapi/res/values/strings.xml
@@ -0,0 +1,5 @@
+<resources>
+  <string name="app_name">androidnativeapi</string>
+  <string name="call_button">Call</string>
+  <string name="hangup_button">Hangup</string>
+</resources>