diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index eef20d4bb8..70d3600cea 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -657,6 +657,8 @@ if (is_win || is_android) {
     deps = [
       "../api:libjingle_peerconnection_test_api",
       "../api:video_frame_api",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
+      "../api/audio_codecs:builtin_audio_encoder_factory",
       "../media:rtc_media",
       "../media:rtc_media_base",
       "../modules/video_capture:video_capture_module",
diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc
index 9332a83a27..2ea8227b12 100644
--- a/examples/unityplugin/simple_peer_connection.cc
+++ b/examples/unityplugin/simple_peer_connection.cc
@@ -12,6 +12,8 @@
 
 #include <utility>
 
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/test/fakeconstraints.h"
 #include "api/videosourceproxy.h"
 #include "media/engine/webrtcvideocapturerfactory.h"
@@ -91,7 +93,8 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
 
     g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
         g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
-        nullptr, nullptr, nullptr);
+        nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+        webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
   }
   if (!g_peer_connection_factory.get()) {
     DeletePeerConnection();
@@ -99,19 +102,19 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
   }
 
   g_peer_count++;
-  if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential,
-                            is_receiver)) {
+  if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) {
     DeletePeerConnection();
     return false;
   }
+
+  mandatory_receive_ = is_receiver;
   return peer_connection_.get() != nullptr;
 }
 
 bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
                                                 const int no_of_urls,
                                                 const char* username,
-                                                const char* credential,
-                                                bool is_receiver) {
+                                                const char* credential) {
   RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
   RTC_DCHECK(peer_connection_.get() == nullptr);
 
@@ -148,11 +151,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
   webrtc::FakeConstraints constraints;
   constraints.SetAllowDtlsSctpDataChannels();
 
-  if (is_receiver) {
-    constraints.SetMandatoryReceiveAudio(true);
-    constraints.SetMandatoryReceiveVideo(true);
-  }
-
   peer_connection_ = g_peer_connection_factory->CreatePeerConnection(
       config_, &constraints, nullptr, nullptr, this);
 
@@ -192,7 +190,12 @@ bool SimplePeerConnection::CreateOffer() {
   if (!peer_connection_.get())
     return false;
 
-  peer_connection_->CreateOffer(this, nullptr);
+  webrtc::FakeConstraints constraints;
+  if (mandatory_receive_) {
+    constraints.SetMandatoryReceiveAudio(true);
+    constraints.SetMandatoryReceiveVideo(true);
+  }
+  peer_connection_->CreateOffer(this, &constraints);
   return true;
 }
 
@@ -200,7 +203,12 @@ bool SimplePeerConnection::CreateAnswer() {
   if (!peer_connection_.get())
     return false;
 
-  peer_connection_->CreateAnswer(this, nullptr);
+  webrtc::FakeConstraints constraints;
+  if (mandatory_receive_) {
+    constraints.SetMandatoryReceiveAudio(true);
+    constraints.SetMandatoryReceiveVideo(true);
+  }
+  peer_connection_->CreateAnswer(this, &constraints);
   return true;
 }
 
@@ -421,8 +429,8 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
     RTC_DCHECK(texture_helper != nullptr)
         << "Cannot get the Surface Texture Helper.";
 
-    rtc::scoped_refptr<AndroidVideoTrackSource> source(
-        new rtc::RefCountedObject<AndroidVideoTrackSource>(
+    rtc::scoped_refptr<webrtc::jni::AndroidVideoTrackSource> source(
+        new rtc::RefCountedObject<webrtc::jni::AndroidVideoTrackSource>(
             g_signaling_thread.get(), env, texture_helper, false));
     rtc::scoped_refptr<webrtc::VideoTrackSourceProxy> proxy_source =
         webrtc::VideoTrackSourceProxy::Create(g_signaling_thread.get(),
diff --git a/examples/unityplugin/simple_peer_connection.h b/examples/unityplugin/simple_peer_connection.h
index 0c7a2615d7..0c490253ee 100644
--- a/examples/unityplugin/simple_peer_connection.h
+++ b/examples/unityplugin/simple_peer_connection.h
@@ -64,8 +64,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
   bool CreatePeerConnection(const char** turn_urls,
                             const int no_of_urls,
                             const char* username,
-                            const char* credential,
-                            bool is_receiver);
+                            const char* credential);
   void CloseDataChannel();
   std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
   void SetAudioControl();
@@ -127,6 +126,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
 
   bool is_mute_audio_ = false;
   bool is_record_audio_ = false;
+  bool mandatory_receive_ = false;
 
   // disallow copy-and-assign
   SimplePeerConnection(const SimplePeerConnection&) = delete;