diff --git a/video_engine/internal/video_call.cc b/video_engine/internal/call.cc
similarity index 81%
rename from video_engine/internal/video_call.cc
rename to video_engine/internal/call.cc
index 6e50395..acb65bf 100644
--- a/video_engine/internal/video_call.cc
+++ b/video_engine/internal/call.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/video_engine/internal/video_call.h"
+#include "webrtc/video_engine/internal/call.h"
 
 #include <assert.h>
 #include <string.h>
@@ -24,17 +24,16 @@
 
 namespace webrtc {
 
-VideoCall* VideoCall::Create(const VideoCall::Config& config) {
-  webrtc::VideoEngine* video_engine = webrtc::VideoEngine::Create();
+Call* Call::Create(const Call::Config& config) {
+  VideoEngine* video_engine = VideoEngine::Create();
   assert(video_engine != NULL);
 
-  return new internal::VideoCall(video_engine, config);
+  return new internal::Call(video_engine, config);
 }
 
 namespace internal {
 
-VideoCall::VideoCall(webrtc::VideoEngine* video_engine,
-                     const VideoCall::Config& config)
+Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
     : config_(config),
       receive_lock_(RWLockWrapper::CreateRWLock()),
       send_lock_(RWLockWrapper::CreateRWLock()),
@@ -50,15 +49,15 @@
   assert(codec_ != NULL);
 }
 
-VideoCall::~VideoCall() {
+Call::~Call() {
   codec_->Release();
   rtp_rtcp_->Release();
   webrtc::VideoEngine::Delete(video_engine_);
 }
 
-PacketReceiver* VideoCall::Receiver() { return this; }
+PacketReceiver* Call::Receiver() { return this; }
 
-std::vector<VideoCodec> VideoCall::GetVideoCodecs() {
+std::vector<VideoCodec> Call::GetVideoCodecs() {
   std::vector<VideoCodec> codecs;
 
   VideoCodec codec;
@@ -70,14 +69,13 @@
   return codecs;
 }
 
-VideoSendStream::Config VideoCall::GetDefaultSendConfig() {
+VideoSendStream::Config Call::GetDefaultSendConfig() {
   VideoSendStream::Config config;
   codec_->GetCodec(0, config.codec);
   return config;
 }
 
-VideoSendStream* VideoCall::CreateSendStream(
-    const VideoSendStream::Config& config) {
+VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
   assert(config.rtp.ssrcs.size() > 0);
   assert(config.codec.numberOfSimulcastStreams == 0 ||
          config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
@@ -93,8 +91,7 @@
   return send_stream;
 }
 
-SendStreamState* VideoCall::DestroySendStream(
-    webrtc::VideoSendStream* send_stream) {
+SendStreamState* Call::DestroySendStream(webrtc::VideoSendStream* send_stream) {
   assert(send_stream != NULL);
 
   VideoSendStream* send_stream_impl = NULL;
@@ -119,11 +116,11 @@
   return NULL;
 }
 
-VideoReceiveStream::Config VideoCall::GetDefaultReceiveConfig() {
+VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
   return VideoReceiveStream::Config();
 }
 
-VideoReceiveStream* VideoCall::CreateReceiveStream(
+VideoReceiveStream* Call::CreateReceiveStream(
     const VideoReceiveStream::Config& config) {
   VideoReceiveStream* receive_stream =
       new VideoReceiveStream(video_engine_, config, config_.send_transport);
@@ -134,8 +131,7 @@
   return receive_stream;
 }
 
-void VideoCall::DestroyReceiveStream(
-    webrtc::VideoReceiveStream* receive_stream) {
+void Call::DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream) {
   assert(receive_stream != NULL);
 
   VideoReceiveStream* receive_stream_impl = NULL;
@@ -157,17 +153,17 @@
   delete receive_stream_impl;
 }
 
-uint32_t VideoCall::SendBitrateEstimate() {
+uint32_t Call::SendBitrateEstimate() {
   // TODO(pbos): Return send-bitrate estimate
   return 0;
 }
 
-uint32_t VideoCall::ReceiveBitrateEstimate() {
+uint32_t Call::ReceiveBitrateEstimate() {
   // TODO(pbos): Return receive-bitrate estimate
   return 0;
 }
 
-bool VideoCall::DeliverRtcp(const uint8_t* packet, size_t length) {
+bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(pbos): Figure out what channel needs it actually.
   //             Do NOT broadcast! Also make sure it's a valid packet.
   bool rtcp_delivered = false;
@@ -199,9 +195,9 @@
   return rtcp_delivered;
 }
 
-bool VideoCall::DeliverRtp(const RTPHeader& header,
-                           const uint8_t* packet,
-                           size_t length) {
+bool Call::DeliverRtp(const RTPHeader& header,
+                      const uint8_t* packet,
+                      size_t length) {
   VideoReceiveStream* receiver;
   {
     ReadLockScoped read_lock(*receive_lock_);
@@ -217,7 +213,7 @@
   return receiver->DeliverRtp(static_cast<const uint8_t*>(packet), length);
 }
 
-bool VideoCall::DeliverPacket(const uint8_t* packet, size_t length) {
+bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
   // TODO(pbos): ExtensionMap if there are extensions.
   if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
     return DeliverRtcp(packet, length);
diff --git a/video_engine/internal/video_call.h b/video_engine/internal/call.h
similarity index 80%
rename from video_engine/internal/video_call.h
rename to video_engine/internal/call.h
index a271d48..13cdd89 100644
--- a/video_engine/internal/video_call.h
+++ b/video_engine/internal/call.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIDEO_CALL_IMPL_H_
+#ifndef WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
+#define WEBRTC_VIDEO_ENGINE_CALL_IMPL_H_
 
 #include <map>
 #include <vector>
@@ -18,7 +18,7 @@
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video_engine/internal/video_receive_stream.h"
 #include "webrtc/video_engine/internal/video_send_stream.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 
 namespace webrtc {
 
@@ -30,11 +30,10 @@
 
 // TODO(pbos): Split out the packet receiver, should be sharable between
 //             VideoEngine and VoiceEngine.
-class VideoCall : public webrtc::VideoCall, public PacketReceiver {
+class Call : public webrtc::Call, public PacketReceiver {
  public:
-  VideoCall(webrtc::VideoEngine* video_engine,
-            const VideoCall::Config& config);
-  virtual ~VideoCall();
+  Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
+  virtual ~Call();
 
   virtual PacketReceiver* Receiver() OVERRIDE;
   virtual std::vector<VideoCodec> GetVideoCodecs() OVERRIDE;
@@ -52,8 +51,8 @@
   virtual VideoReceiveStream* CreateReceiveStream(
       const VideoReceiveStream::Config& config) OVERRIDE;
 
-  virtual void DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream)
-      OVERRIDE;
+  virtual void DestroyReceiveStream(
+      webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
 
   virtual uint32_t SendBitrateEstimate() OVERRIDE;
   virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
@@ -66,7 +65,7 @@
                   const uint8_t* packet,
                   size_t length);
 
-  VideoCall::Config config_;
+  Call::Config config_;
 
   std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_;
   scoped_ptr<RWLockWrapper> receive_lock_;
@@ -80,9 +79,9 @@
   ViERTP_RTCP* rtp_rtcp_;
   ViECodec* codec_;
 
-  DISALLOW_COPY_AND_ASSIGN(VideoCall);
+  DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_INTERNAL_VIDEO_CALL_H_
+#endif  // WEBRTC_VIDEO_ENGINE_INTERNAL_CALL_H_
diff --git a/video_engine/new_include/video_call.h b/video_engine/new_include/call.h
similarity index 84%
rename from video_engine/new_include/video_call.h
rename to video_engine/new_include/call.h
index 96a93e5..6650216 100644
--- a/video_engine/new_include/video_call.h
+++ b/video_engine/new_include/call.h
@@ -7,8 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
-#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
 
 #include <string>
 #include <vector>
@@ -31,10 +31,10 @@
   virtual ~PacketReceiver() {}
 };
 
-// A VideoCall instance can contain several send and/or receive streams. All
-// streams are assumed to have the same remote endpoint and will share bitrate
-// estimates etc.
-class VideoCall {
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+class Call {
  public:
   struct Config {
     explicit Config(newapi::Transport* send_transport)
@@ -47,14 +47,14 @@
     newapi::Transport* send_transport;
     bool overuse_detection;
 
-    // VoiceEngine used for audio/video synchronization for this VideoCall.
+    // VoiceEngine used for audio/video synchronization for this Call.
     VoiceEngine* voice_engine;
 
     TraceCallback* trace_callback;
     uint32_t trace_filter;
   };
 
-  static VideoCall* Create(const VideoCall::Config& config);
+  static Call* Create(const Call::Config& config);
 
   virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
 
@@ -76,7 +76,7 @@
 
   // All received RTP and RTCP packets for the call should be inserted to this
   // PacketReceiver. The PacketReceiver pointer is valid as long as the
-  // VideoCall instance exists.
+  // Call instance exists.
   virtual PacketReceiver* Receiver() = 0;
 
   // Returns the estimated total send bandwidth. Note: this can differ from the
@@ -87,8 +87,8 @@
   // differ from the actual receive bitrate.
   virtual uint32_t ReceiveBitrateEstimate() = 0;
 
-  virtual ~VideoCall() {}
+  virtual ~Call() {}
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_VIDEO_CALL_H_
+#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
diff --git a/video_engine/test/common/direct_transport.cc b/video_engine/test/common/direct_transport.cc
index 7324401..f4f364b 100644
--- a/video_engine/test/common/direct_transport.cc
+++ b/video_engine/test/common/direct_transport.cc
@@ -10,7 +10,7 @@
 #include "webrtc/video_engine/test/common/direct_transport.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 
 namespace webrtc {
 namespace test {
diff --git a/video_engine/test/engine_tests.cc b/video_engine/test/engine_tests.cc
index 3978c52..4ba659a 100644
--- a/video_engine/test/engine_tests.cc
+++ b/video_engine/test/engine_tests.cc
@@ -21,7 +21,7 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 #include "webrtc/video_engine/test/common/direct_transport.h"
 #include "webrtc/video_engine/test/common/fake_decoder.h"
 #include "webrtc/video_engine/test/common/fake_encoder.h"
@@ -35,7 +35,8 @@
 class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
  public:
   typedef std::map<uint32_t, int> BytesSentMap;
-  StreamObserver(int num_expected_ssrcs, newapi::Transport* feedback_transport,
+  StreamObserver(int num_expected_ssrcs,
+                 newapi::Transport* feedback_transport,
                  Clock* clock)
       : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
         all_ssrcs_sent_(EventWrapper::Create()),
@@ -65,21 +66,19 @@
     CriticalSectionScoped lock(critical_section_.get());
     if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps)
       all_ssrcs_sent_->Set();
-    rtp_rtcp_->SetREMBData(bitrate, static_cast<uint8_t>(ssrcs.size()),
-                           &ssrcs[0]);
+    rtp_rtcp_->SetREMBData(
+        bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
     rtp_rtcp_->Process();
   }
 
   virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE {
     CriticalSectionScoped lock(critical_section_.get());
     RTPHeader header;
-    EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length),
-                                   &header));
+    EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
     receive_stats_->IncomingPacket(header, length, false);
     rtp_rtcp_->SetRemoteSSRC(header.ssrc);
-    remote_bitrate_estimator_->IncomingPacket(clock_->TimeInMilliseconds(),
-                                              static_cast<int>(length - 12),
-                                              header);
+    remote_bitrate_estimator_->IncomingPacket(
+        clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header);
     if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
       remote_bitrate_estimator_->Process();
     }
@@ -90,9 +89,7 @@
     return true;
   }
 
-  EventTypeWrapper Wait() {
-    return all_ssrcs_sent_->Wait(120 * 1000);
-  }
+  EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); }
 
  private:
   class TransportWrapper : public webrtc::Transport {
@@ -100,15 +97,18 @@
     explicit TransportWrapper(newapi::Transport* new_transport)
         : new_transport_(new_transport) {}
 
-    virtual int SendPacket(int channel, const void *data, int len) OVERRIDE {
-      return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len) ?
-          len : -1;
+    virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
+      return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len)
+                 ? len
+                 : -1;
     }
 
-    virtual int SendRTCPPacket(int channel, const void *data,
+    virtual int SendRTCPPacket(int channel,
+                               const void* data,
                                int len) OVERRIDE {
-      return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len) ?
-          len : -1;
+      return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len)
+                 ? len
+                 : -1;
     }
 
    private:
@@ -130,9 +130,7 @@
 
 class RampUpTest : public ::testing::TestWithParam<bool> {
  public:
-  virtual void SetUp() {
-    reserved_ssrcs_.clear();
-  }
+  virtual void SetUp() { reserved_ssrcs_.clear(); }
 
  protected:
   std::map<uint32_t, bool> reserved_ssrcs_;
@@ -140,12 +138,11 @@
 
 TEST_P(RampUpTest, RampUpWithPadding) {
   test::DirectTransport receiver_transport;
-  StreamObserver stream_observer(3, &receiver_transport,
-                                 Clock::GetRealTimeClock());
-  VideoCall::Config call_config(&stream_observer);
-  scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
-  VideoSendStream::Config send_config =
-      call->GetDefaultSendConfig();
+  StreamObserver stream_observer(
+      3, &receiver_transport, Clock::GetRealTimeClock());
+  Call::Config call_config(&stream_observer);
+  scoped_ptr<Call> call(Call::Create(call_config));
+  VideoSendStream::Config send_config = call->GetDefaultSendConfig();
 
   receiver_transport.SetReceiver(call->Receiver());
 
@@ -157,22 +154,20 @@
 
   test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
 
-  VideoSendStream* send_stream =
-      call->CreateSendStream(send_config);
+  VideoSendStream* send_stream = call->CreateSendStream(send_config);
 
   VideoReceiveStream::Config receive_config;
   receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
-  receive_config.rtp.nack.rtp_history_ms =
-      send_config.rtp.nack.rtp_history_ms;
-  VideoReceiveStream* receive_stream = call->CreateReceiveStream(
-      receive_config);
+  receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
+  VideoReceiveStream* receive_stream =
+      call->CreateReceiveStream(receive_config);
 
   scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
       test::FrameGeneratorCapturer::Create(
           send_stream->Input(),
-          test::FrameGenerator::Create(
-              send_config.codec.width, send_config.codec.height,
-              Clock::GetRealTimeClock()),
+          test::FrameGenerator::Create(send_config.codec.width,
+                                       send_config.codec.height,
+                                       Clock::GetRealTimeClock()),
           30));
 
   receive_stream->StartReceive();
@@ -213,10 +208,10 @@
  protected:
   void CreateCalls(newapi::Transport* sender_transport,
                    newapi::Transport* receiver_transport) {
-    VideoCall::Config sender_config(sender_transport);
-    VideoCall::Config receiver_config(receiver_transport);
-    sender_call_.reset(VideoCall::Create(sender_config));
-    receiver_call_.reset(VideoCall::Create(receiver_config));
+    Call::Config sender_config(sender_transport);
+    Call::Config receiver_config(receiver_transport);
+    sender_call_.reset(Call::Create(sender_config));
+    receiver_call_.reset(Call::Create(receiver_config));
   }
 
   void CreateTestConfigs() {
@@ -273,14 +268,14 @@
       sender_call_->DestroySendStream(send_stream_);
     if (receive_stream_ != NULL)
       receiver_call_->DestroyReceiveStream(receive_stream_);
-    send_stream_= NULL;
+    send_stream_ = NULL;
     receive_stream_ = NULL;
   }
 
   void ReceivesPliAndRecovers(int rtp_history_ms);
 
-  scoped_ptr<VideoCall> sender_call_;
-  scoped_ptr<VideoCall> receiver_call_;
+  scoped_ptr<Call> sender_call_;
+  scoped_ptr<Call> receiver_call_;
 
   VideoSendStream::Config send_config_;
   VideoReceiveStream::Config receive_config_;
@@ -310,6 +305,7 @@
   static const int kNumberOfNacksToObserve = 4;
   static const int kInverseProbabilityToStartLossBurst = 20;
   static const int kMaxLossBurst = 10;
+
  public:
   NackObserver()
       : received_all_retransmissions_(EventWrapper::Create()),
@@ -444,9 +440,10 @@
 
 class PliObserver : public test::RtpRtcpObserver {
   static const int kInverseDropProbability = 16;
+
  public:
-  PliObserver(bool nack_enabled) :
-        renderer_(this),
+  PliObserver(bool nack_enabled)
+      : renderer_(this),
         rtp_header_parser_(RtpHeaderParser::Create()),
         nack_enabled_(nack_enabled),
         first_retransmitted_timestamp_(0),
@@ -572,9 +569,7 @@
     explicit PacketInputObserver(PacketReceiver* receiver)
         : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {}
 
-    EventTypeWrapper Wait() {
-      return delivered_packet_->Wait(30 * 1000);
-    }
+    EventTypeWrapper Wait() { return delivered_packet_->Wait(30 * 1000); }
 
    private:
     virtual bool DeliverPacket(const uint8_t* packet, size_t length) {
diff --git a/video_engine/test/full_stack.cc b/video_engine/test/full_stack.cc
index 8c9e755..365423f 100644
--- a/video_engine/test/full_stack.cc
+++ b/video_engine/test/full_stack.cc
@@ -23,7 +23,7 @@
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 #include "webrtc/video_engine/test/common/direct_transport.h"
 #include "webrtc/video_engine/test/common/file_capturer.h"
 #include "webrtc/video_engine/test/common/frame_generator_capturer.h"
@@ -47,14 +47,16 @@
   double avg_ssim_threshold;
 };
 
-FullStackTestParams paris_qcif = {"net_delay_0_0_plr_0",
-                                  {"paris_qcif", 176, 144, 30}, 300, 36.0,
-                                  0.96};
+FullStackTestParams paris_qcif = {
+    "net_delay_0_0_plr_0", {"paris_qcif", 176, 144, 30}, 300, 36.0, 0.96};
 
 // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
-FullStackTestParams foreman_cif = {"foreman_cif_net_delay_0_0_plr_0",
-                                   {"foreman_cif", 352, 288, 30}, 700, 0.0,
-                                   0.0};
+FullStackTestParams foreman_cif = {
+    "foreman_cif_net_delay_0_0_plr_0",
+    {"foreman_cif", 352, 288, 30},
+    700,
+    0.0,
+    0.0};
 
 class FullStackTest : public ::testing::TestWithParam<FullStackTestParams> {
  protected:
@@ -280,9 +282,9 @@
       params.avg_ssim_threshold,
       static_cast<uint64_t>(FLAGS_seconds * params.clip.fps));
 
-  VideoCall::Config call_config(&analyzer);
+  Call::Config call_config(&analyzer);
 
-  scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+  scoped_ptr<Call> call(Call::Create(call_config));
   analyzer.receiver_ = call->Receiver();
   transport.SetReceiver(&analyzer);
 
@@ -314,8 +316,7 @@
               test_clock),
           params.clip.fps));
 
-  VideoReceiveStream::Config receive_config =
-      call->GetDefaultReceiveConfig();
+  VideoReceiveStream::Config receive_config = call->GetDefaultReceiveConfig();
   receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
   receive_config.renderer = &analyzer;
 
diff --git a/video_engine/test/loopback.cc b/video_engine/test/loopback.cc
index 19d99c1..93f2d83 100644
--- a/video_engine/test/loopback.cc
+++ b/video_engine/test/loopback.cc
@@ -17,7 +17,7 @@
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 #include "webrtc/video_engine/test/common/direct_transport.h"
 #include "webrtc/video_engine/test/common/flags.h"
 #include "webrtc/video_engine/test/common/generate_ssrcs.h"
@@ -40,9 +40,9 @@
       "Loopback Video", test::flags::Width(), test::flags::Height()));
 
   test::DirectTransport transport;
-  VideoCall::Config call_config(&transport);
+  Call::Config call_config(&transport);
   call_config.overuse_detection = true;
-  scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+  scoped_ptr<Call> call(Call::Create(call_config));
 
   // Loopback, call sends to itself.
   transport.SetReceiver(call->Receiver());
@@ -74,8 +74,7 @@
                                   test::flags::Fps(),
                                   test_clock));
 
-  VideoReceiveStream::Config receive_config =
-      call->GetDefaultReceiveConfig();
+  VideoReceiveStream::Config receive_config = call->GetDefaultReceiveConfig();
   receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
   receive_config.renderer = loopback_video.get();
 
diff --git a/video_engine/test/send_stream_tests.cc b/video_engine/test/send_stream_tests.cc
index 1e1ac27..dcf2ec1 100644
--- a/video_engine/test/send_stream_tests.cc
+++ b/video_engine/test/send_stream_tests.cc
@@ -16,7 +16,7 @@
 #include "webrtc/video_engine/test/common/frame_generator.h"
 #include "webrtc/video_engine/test/common/frame_generator_capturer.h"
 #include "webrtc/video_engine/test/common/null_transport.h"
-#include "webrtc/video_engine/new_include/video_call.h"
+#include "webrtc/video_engine/new_include/call.h"
 #include "webrtc/video_engine/new_include/video_send_stream.h"
 
 namespace webrtc {
@@ -28,9 +28,7 @@
         send_test_complete_(EventWrapper::Create()),
         timeout_ms_(timeout_ms) {}
 
-  EventTypeWrapper Wait() {
-    return send_test_complete_->Wait(timeout_ms_);
-  }
+  EventTypeWrapper Wait() { return send_test_complete_->Wait(timeout_ms_); }
 
  protected:
   scoped_ptr<RtpHeaderParser> rtp_header_parser_;
@@ -43,9 +41,10 @@
 class VideoSendStreamTest : public ::testing::Test {
  public:
   VideoSendStreamTest() : fake_encoder_(Clock::GetRealTimeClock()) {}
+
  protected:
   static const uint32_t kSendSsrc;
-  void RunSendTest(VideoCall* call,
+  void RunSendTest(Call* call,
                    const VideoSendStream::Config& config,
                    SendTransportObserver* observer) {
     VideoSendStream* send_stream = call->CreateSendStream(config);
@@ -64,7 +63,7 @@
     call->DestroySendStream(send_stream);
   }
 
-  VideoSendStream::Config GetSendTestConfig(VideoCall* call) {
+  VideoSendStream::Config GetSendTestConfig(Call* call) {
     VideoSendStream::Config config = call->GetDefaultSendConfig();
     config.encoder = &fake_encoder_;
     config.internal_source = false;
@@ -94,8 +93,8 @@
     }
   } observer;
 
-  VideoCall::Config call_config(&observer);
-  scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+  Call::Config call_config(&observer);
+  scoped_ptr<Call> call(Call::Create(call_config));
 
   VideoSendStream::Config send_config = GetSendTestConfig(call.get());
   send_config.rtp.ssrcs.push_back(kSendSsrc);
@@ -127,8 +126,8 @@
     }
   } observer;
 
-  VideoCall::Config call_config(&observer);
-  scoped_ptr<VideoCall> call(VideoCall::Create(call_config));
+  Call::Config call_config(&observer);
+  scoped_ptr<Call> call(Call::Create(call_config));
 
   VideoSendStream::Config send_config = GetSendTestConfig(call.get());
   send_config.rtp.ssrcs.push_back(kSendSsrc);
diff --git a/video_engine/video_engine_core.gypi b/video_engine/video_engine_core.gypi
index 29849fa..5c7876a 100644
--- a/video_engine/video_engine_core.gypi
+++ b/video_engine/video_engine_core.gypi
@@ -115,16 +115,16 @@
         'vie_sync_module.cc',
 
         # New VideoEngine API
-        'internal/video_call.cc',
-        'internal/video_call.h',
+        'internal/call.cc',
+        'internal/call.h',
         'internal/video_receive_stream.cc',
         'internal/video_receive_stream.h',
         'internal/video_send_stream.cc',
         'internal/video_send_stream.h',
+        'new_include/call.h',
         'new_include/config.h',
         'new_include/frame_callback.h',
         'new_include/transport.h',
-        'new_include/video_call.h',
         'new_include/video_receive_stream.h',
         'new_include/video_renderer.h',
         'new_include/video_send_stream.h',