Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 122 insertions, 0 deletions

diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.h b/talk/app/webrtc/test/peerconnectiontestwrapper.h
new file mode 100644
index 0000000000..b65426326f
--- /dev/null
+++ b/talk/app/webrtc/test/peerconnectiontestwrapper.h
@@ -0,0 +1,122 @@
+/*
+ * libjingle
+ * Copyright 2013 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
+#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
+#include "talk/app/webrtc/test/fakeconstraints.h"
+#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
+#include "webrtc/base/sigslot.h"
+
+namespace webrtc {
+class DtlsIdentityStoreInterface;
+class PortAllocatorFactoryInterface;
+}
+
+class PeerConnectionTestWrapper
+    : public webrtc::PeerConnectionObserver,
+      public webrtc::CreateSessionDescriptionObserver,
+      public sigslot::has_slots<> {
+ public:
+  static void Connect(PeerConnectionTestWrapper* caller,
+                      PeerConnectionTestWrapper* callee);
+
+  explicit PeerConnectionTestWrapper(const std::string& name);
+  virtual ~PeerConnectionTestWrapper();
+
+  bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
+
+  rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
+      const std::string& label,
+      const webrtc::DataChannelInit& init);
+
+  // Implements PeerConnectionObserver.
+  virtual void OnSignalingChange(
+     webrtc::PeerConnectionInterface::SignalingState new_state) {}
+  virtual void OnStateChange(
+      webrtc::PeerConnectionObserver::StateType state_changed) {}
+  virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
+  virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {}
+  virtual void OnDataChannel(webrtc::DataChannelInterface* data_channel);
+  virtual void OnRenegotiationNeeded() {}
+  virtual void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) {}
+  virtual void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) {}
+  virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
+  virtual void OnIceComplete() {}
+
+  // Implements CreateSessionDescriptionObserver.
+  virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
+  virtual void OnFailure(const std::string& error) {}
+
+  void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
+  void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
+  void ReceiveOfferSdp(const std::string& sdp);
+  void ReceiveAnswerSdp(const std::string& sdp);
+  void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
+                       const std::string& candidate);
+  void WaitForCallEstablished();
+  void WaitForConnection();
+  void WaitForAudio();
+  void WaitForVideo();
+  void GetAndAddUserMedia(
+    bool audio, const webrtc::FakeConstraints& audio_constraints,
+    bool video, const webrtc::FakeConstraints& video_constraints);
+
+  // sigslots
+  sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
+  sigslot::signal3<const std::string&,
+                   int,
+                   const std::string&> SignalOnIceCandidateReady;
+  sigslot::signal1<std::string*> SignalOnSdpCreated;
+  sigslot::signal1<const std::string&> SignalOnSdpReady;
+  sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
+
+ private:
+  void SetLocalDescription(const std::string& type, const std::string& sdp);
+  void SetRemoteDescription(const std::string& type, const std::string& sdp);
+  bool CheckForConnection();
+  bool CheckForAudio();
+  bool CheckForVideo();
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
+      bool audio, const webrtc::FakeConstraints& audio_constraints,
+      bool video, const webrtc::FakeConstraints& video_constraints);
+
+  std::string name_;
+  rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
+      allocator_factory_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
+      peer_connection_factory_;
+  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
+  rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
+};
+
+#endif  // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_