Adding peerconnection_unittest.

Review URL: http://webrtc-codereview.appspot.com/226004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@757 4adac7df-926f-26a2-2b94-8c16560cd09d

3 files changed, 346 insertions, 0 deletions

diff --git a/third_party_mods/libjingle/libjingle.gyp b/third_party_mods/libjingle/libjingle.gyp
index a1f3010931..43c308a186 100644
--- a/third_party_mods/libjingle/libjingle.gyp
+++ b/third_party_mods/libjingle/libjingle.gyp
@@ -772,6 +772,7 @@
           'sources': [
             '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamhandler_unittest.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/mediastreamimpl_unittest.cc',           
+            '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnection_unittest.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnection_unittests.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionimpl_unittest.cc',
             '<(libjingle_mods)/source/talk/app/webrtc_dev/peerconnectionmanager_unittest.cc',
diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h
index 22e78365e3..c7acea9a35 100644
--- a/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h
+++ b/third_party_mods/libjingle/source/talk/app/webrtc_dev/mediastream.h
@@ -57,6 +57,8 @@ class Notifier {
  public:
   virtual void RegisterObserver(Observer* observer) = 0;
   virtual void UnregisterObserver(Observer* observer) = 0;
+
+  virtual ~Notifier() {}
 };
 
 // Information about a track.
@@ -100,6 +102,8 @@ class VideoTrackInterface : public MediaStreamTrackInterface {
  public:
   // Set the video renderer for a local or remote stream.
   // This call will start decoding the received video stream and render it.
+  // The VideoRendererInterface is stored as a scoped_refptr. This means that
+  // it is not allowed to call delete renderer after this API has been called.
   virtual void SetRenderer(VideoRendererWrapperInterface* renderer) = 0;
 
   // Get the VideoRenderer associated with this track.
diff --git a/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc
new file mode 100644
index 0000000000..9a2a9805a7
--- /dev/null
+++ b/third_party_mods/libjingle/source/talk/app/webrtc_dev/peerconnection_unittest.cc
@@ -0,0 +1,341 @@
+/*
+ * libjingle
+ * Copyright 2011, 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.
+ */
+
+#include <stdio.h>
+
+#include <list>
+
+#include "gtest/gtest.h"
+#include "modules/video_capture/main/interface/video_capture_factory.h"
+#include "talk/app/webrtc_dev/mediastream.h"
+#include "talk/app/webrtc_dev/peerconnection.h"
+#include "talk/base/thread.h"
+#include "talk/session/phone/videoframe.h"
+#include "talk/session/phone/videorenderer.h"
+
+void GetAllVideoTracks(webrtc::MediaStreamInterface* media_stream,
+                       std::list<webrtc::VideoTrackInterface*>* video_tracks) {
+  webrtc::VideoTracks* track_list = media_stream->video_tracks();
+  for (size_t i = 0; i < track_list->count(); ++i) {
+    webrtc::VideoTrackInterface* track = track_list->at(i);
+    video_tracks->push_back(
+          static_cast<webrtc::VideoTrackInterface*>(track));
+  }
+}
+
+// TODO(henrike): replace with a capture device that reads from a file/buffer.
+scoped_refptr<webrtc::VideoCaptureModule> OpenVideoCaptureDevice() {
+  webrtc::VideoCaptureModule::DeviceInfo* device_info(
+      webrtc::VideoCaptureFactory::CreateDeviceInfo(0));
+  scoped_refptr<webrtc::VideoCaptureModule> video_device;
+
+  const size_t kMaxDeviceNameLength = 128;
+  const size_t kMaxUniqueIdLength = 256;
+  uint8 device_name[kMaxDeviceNameLength];
+  uint8 unique_id[kMaxUniqueIdLength];
+
+  const size_t device_count = device_info->NumberOfDevices();
+  for (size_t i = 0; i < device_count; ++i) {
+    // Get the name of the video capture device.
+    device_info->GetDeviceName(i, device_name, kMaxDeviceNameLength, unique_id,
+        kMaxUniqueIdLength);
+    // Try to open this device.
+    video_device =
+        webrtc::VideoCaptureFactory::Create(0, unique_id);
+    if (video_device.get())
+      break;
+  }
+  delete device_info;
+  return video_device;
+}
+
+class VideoRecorder : public cricket::VideoRenderer {
+ public:
+  static VideoRecorder* CreateVideoRecorder(
+      const char* file_name) {
+    VideoRecorder* renderer = new VideoRecorder();
+    if (!renderer->Init(file_name)) {
+      delete renderer;
+      return NULL;
+    }
+    return renderer;
+  }
+  virtual ~VideoRecorder() {
+    if (output_file_ != NULL) {
+      fclose(output_file_);
+    }
+  }
+
+  // Set up files so that recording can start immediately.
+  bool Init(const char* file_name) {
+    output_file_ = fopen(file_name, "wb");
+    if (output_file_ == NULL) {
+      return false;
+    }
+    return true;
+  }
+
+  virtual bool SetSize(int width, int height, int /*reserved*/) {
+    width_ = width;
+    height_ = height;
+    image_.reset(new uint8[buffersize()]);
+  }
+
+  // |frame| is in I420
+  virtual bool RenderFrame(const cricket::VideoFrame* frame) {
+    const int actual_size = frame->CopyToBuffer(image_.get(),
+                                                buffersize());
+    if (actual_size > buffersize()) {
+      ASSERT(false);
+      // Skip frame.
+      return true;
+    }
+    // Write to file.
+    fwrite(image_.get(), sizeof(uint8), actual_size, output_file_);
+    return true;
+  }
+
+  const uint8* image() const {
+    return image_.get();
+  }
+
+  int buffersize() const {
+    // I420 buffer size
+    return (width_ * height_ * 3) >> 1;
+  }
+
+  int width() const {
+    return width_;
+  }
+
+  int height() const {
+    return height_;
+  }
+
+ protected:
+  VideoRecorder()
+      : width_(0),
+        height_(0),
+        output_file_(NULL) {}
+
+  talk_base::scoped_array<uint8> image_;
+  int width_;
+  int height_;
+
+  // File to record to.
+  FILE* output_file_;
+};
+
+class SignalingMessageReceiver {
+ public:
+  virtual void ReceiveMessage(const std::string& msg) = 0;
+
+ protected:
+  SignalingMessageReceiver() {}
+  virtual ~SignalingMessageReceiver() {}
+};
+
+class PeerConnectionP2PTestClient
+    : public webrtc::PeerConnectionObserver,
+      public SignalingMessageReceiver {
+ public:
+  static PeerConnectionP2PTestClient* CreateClient(int id) {
+    PeerConnectionP2PTestClient* client = new PeerConnectionP2PTestClient(id);
+    if (!client->Init()) {
+      delete client;
+      return NULL;
+    }
+    return client;
+  }
+
+  ~PeerConnectionP2PTestClient() {
+    // Ensure that webrtc::PeerConnection is deleted before
+    // webrtc::PeerConnectionManager or crash will occur
+    webrtc::PeerConnection* temp = peer_connection_.release();
+    temp->Release();
+  }
+
+  void StartSession() {
+    // Audio track doesn't seem to be implemented yet. No need to pass a device
+    // to it.
+    scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(
+        peer_connection_factory_->CreateLocalAudioTrack("audio_track", NULL));
+
+    scoped_refptr<webrtc::LocalVideoTrackInterface> video_track(
+        peer_connection_factory_->CreateLocalVideoTrack(
+            "video_track",
+            OpenVideoCaptureDevice()));
+
+    scoped_refptr<webrtc::LocalMediaStreamInterface> stream =
+        peer_connection_factory_->CreateLocalMediaStream("stream_label");
+
+    stream->AddTrack(audio_track);
+    stream->AddTrack(video_track);
+
+    peer_connection_->AddStream(stream);
+    peer_connection_->CommitStreamChanges();
+  }
+
+  void set_signaling_message_receiver(
+      SignalingMessageReceiver* signaling_message_receiver) {
+    signaling_message_receiver_ = signaling_message_receiver;
+  }
+
+  // SignalingMessageReceiver callback.
+  virtual void ReceiveMessage(const std::string& msg) {
+    peer_connection_->ProcessSignalingMessage(msg);
+  }
+
+  // PeerConnectionObserver callbacks.
+  virtual void OnError() {}
+  virtual void OnMessage(const std::string&) {}
+  virtual void OnSignalingMessage(const std::string& msg)  {
+    if (signaling_message_receiver_ == NULL) {
+      ADD_FAILURE();
+      return;
+    }
+    signaling_message_receiver_->ReceiveMessage(msg);
+  }
+  virtual void OnStateChange(Readiness) {}
+  virtual void OnAddStream(webrtc::MediaStreamInterface* media_stream) {
+    std::list<webrtc::VideoTrackInterface*> video_tracks;
+    GetAllVideoTracks(media_stream, &video_tracks);
+    int track_id = 0;
+    for (std::list<webrtc::VideoTrackInterface*>::iterator iter =
+             video_tracks.begin();
+         iter != video_tracks.end();
+         ++iter) {
+      char file_name[256];
+      GenerateRecordingFileName(track_id, file_name);
+      scoped_refptr<webrtc::VideoRendererWrapperInterface> video_renderer =
+          webrtc::CreateVideoRenderer(
+              VideoRecorder::CreateVideoRecorder(file_name));
+      if (video_renderer == NULL) {
+        ADD_FAILURE();
+        continue;
+      }
+      (*iter)->SetRenderer(video_renderer);
+      track_id++;
+    }
+  }
+  virtual void OnRemoveStream(webrtc::MediaStreamInterface*) {
+  }
+
+ private:
+  explicit PeerConnectionP2PTestClient(int id)
+      : id_(id),
+        peer_connection_(),
+        peer_connection_factory_(),
+        signaling_message_receiver_(NULL) {
+  }
+
+  bool Init() {
+    EXPECT_TRUE(peer_connection_.get() == NULL);
+    EXPECT_TRUE(peer_connection_factory_.get() == NULL);
+    peer_connection_factory_ = webrtc::PeerConnectionManager::Create();
+    if (peer_connection_factory_.get() == NULL) {
+      ADD_FAILURE();
+      return false;
+    }
+
+    const char server_configuration[] = "STUN stun.l.google.com:19302";
+    peer_connection_ = peer_connection_factory_->CreatePeerConnection(
+        server_configuration, this);
+    return peer_connection_.get() != NULL;
+  }
+
+  void GenerateRecordingFileName(int track, char file_name[256]) {
+    if (file_name == NULL) {
+      return;
+    }
+    snprintf(file_name, sizeof(file_name),
+             "p2p_test_client_%d_videotrack_%d.yuv", id_, track);
+  }
+
+  int id_;
+  scoped_refptr<webrtc::PeerConnection> peer_connection_;
+  scoped_refptr<webrtc::PeerConnectionManager> peer_connection_factory_;
+
+  // Remote peer communication.
+  SignalingMessageReceiver* signaling_message_receiver_;
+};
+
+class P2PTestConductor {
+ public:
+  static P2PTestConductor* CreateConductor() {
+    P2PTestConductor* conductor = new P2PTestConductor();
+    if (!conductor->Init()) {
+      delete conductor;
+      return NULL;
+    }
+    return conductor;
+  }
+  ~P2PTestConductor() {
+    for (int i = 0; i < kClients; ++i) {
+      if (clients[i] != NULL) {
+        // TODO(hellner): currently deleting the clients will trigger an assert
+        // in cricket::BaseChannel::DisableMedia_w (not due to the unit test).
+        // Fix that problem and remove the below comment.
+        delete clients[i];
+      }
+    }
+  }
+
+  void StartSession() {
+    PeerConnectionP2PTestClient* initiating_client = clients[0];
+    initiating_client->StartSession();
+  }
+
+ private:
+  static const int kClients = 2;
+  P2PTestConductor() {
+    clients[0] = NULL;
+    clients[1] = NULL;
+  }
+
+  bool Init() {
+    for (int i = 0; i < kClients; ++i) {
+      clients[i] = PeerConnectionP2PTestClient::CreateClient(i);
+      if (clients[i] == NULL) {
+        return false;
+      }
+    }
+    clients[0]->set_signaling_message_receiver(clients[1]);
+    clients[1]->set_signaling_message_receiver(clients[0]);
+    return true;
+  }
+
+  PeerConnectionP2PTestClient* clients[kClients];
+};
+
+TEST(PeerConnection2, LocalP2PTest) {
+  P2PTestConductor* test = P2PTestConductor::CreateConductor();
+  ASSERT_TRUE(test != NULL);
+  test->StartSession();
+  talk_base::Thread::Current()->ProcessMessages(10000);
+  delete test;
+}