Allow locally rendered video to be downscaled in end-to-end tests.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 1276 matching lines @@
                          receiving_client_->AudioFramesReceivedCheck(
                              pc2_audio_received + kEndAudioFrameCount),
                      kMaxWaitForFramesMs);
 
     // During this time, we shouldn't have received any additional video frames
     // for the rejected video tracks.
     EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
     EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
   }
 
-  void VerifyRenderedSize(int width, int height) {
-    VerifyRenderedSize(width, height, webrtc::kVideoRotation_0);
+  void VerifyRenderedAspectRatio(int width, int height) {
+    VerifyRenderedAspectRatio(width, height, webrtc::kVideoRotation_0);
   }
 
-  void VerifyRenderedSize(int width,
-                          int height,
-                          webrtc::VideoRotation rotation) {
+  void VerifyRenderedAspectRatio(int width,
+                                 int height,
+                                 webrtc::VideoRotation rotation) {
     double expected_aspect_ratio = static_cast<double>(width) / height;
     double receiving_client_rendered_aspect_ratio =
         static_cast<double>(receiving_client()->rendered_width()) /
         receiving_client()->rendered_height();
     double initializing_client_rendered_aspect_ratio =
         static_cast<double>(initializing_client()->rendered_width()) /
         initializing_client()->rendered_height();
+    double initializing_client_local_rendered_aspect_ratio =
+        static_cast<double>(initializing_client()->local_rendered_width()) /
+        initializing_client()->local_rendered_height();
+    // Verify end-to-end rendered aspect ratio.
     EXPECT_EQ(expected_aspect_ratio, receiving_client_rendered_aspect_ratio);
     EXPECT_EQ(expected_aspect_ratio, initializing_client_rendered_aspect_ratio);
+    // Verify aspect ratio of the local preview.
+    EXPECT_EQ(expected_aspect_ratio,
+              initializing_client_local_rendered_aspect_ratio);
 
+    // Verify rotation.
     EXPECT_EQ(rotation, receiving_client()->rendered_rotation());
     EXPECT_EQ(rotation, initializing_client()->rendered_rotation());
-
-    // Verify size of the local preview.
-    EXPECT_EQ(width, initializing_client()->local_rendered_width());
-    EXPECT_EQ(height, initializing_client()->local_rendered_height());
   }
 
   void VerifySessionDescriptions() {
     initiating_client_->VerifyRejectedMediaInSessionDescription();
     receiving_client_->VerifyRejectedMediaInSessionDescription();
     initiating_client_->VerifyLocalIceUfragAndPassword();
     receiving_client_->VerifyLocalIceUfragAndPassword();
   }
 
   ~P2PTestConductor() {
@@ skipping 142 matching lines @@
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
     // Disable resolution adaptation, we don't want it interfering with the
     // test results.
     webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
     rtc_config.set_cpu_adaptation(false);
 
     ASSERT_TRUE(CreateTestClients(&setup_constraints, nullptr, &rtc_config,
                                   &setup_constraints, nullptr, &rtc_config));
     LocalP2PTest();
-    VerifyRenderedSize(640, 480);
+    VerifyRenderedAspectRatio(640, 480);
   }
 
   PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
     FakeConstraints setup_constraints;
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
     // Disable resolution adaptation, we don't want it interfering with the
     // test results.
     webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
     rtc_config.set_cpu_adaptation(false);
@@ skipping 163 matching lines @@
 // received video has a resolution of 1280*720.
 // TODO(mallinath): Enable when
 // http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
 TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
   ASSERT_TRUE(CreateTestClients());
   FakeConstraints constraint;
   constraint.SetMandatoryMinWidth(1280);
   constraint.SetMandatoryMinHeight(720);
   SetVideoConstraints(constraint, constraint);
   LocalP2PTest();
-  VerifyRenderedSize(1280, 720);
+  VerifyRenderedAspectRatio(1280, 720);
 }
 
 // This test sets up a call between two endpoints that are configured to use
 // DTLS key agreement. As a result, DTLS is negotiated and used for transport.
 TEST_F(P2PTestConductor, LocalP2PTestDtls) {
   SetupAndVerifyDtlsCall();
 }
 
 // This test sets up an one-way call, with media only from initiator to
 // responder.
@@ skipping 31 matching lines @@
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
   std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_initializing_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();
 
   SetSignalingReceivers();
   receiving_client()->SetExpectIceRestart(true);
   LocalP2PTest();
-  VerifyRenderedSize(640, 480);
+  VerifyRenderedAspectRatio(640, 480);
 }
 
 // This test sets up a non-bundle call and apply bundle during ICE restart. When
 // bundle is in effect in the restart, the channel can successfully reset its
 // DTLS-SRTP context.
 #if defined(MEMORY_SANITIZER)
 // Fails under MemorySanitizer:
 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=6811
 #define MAYBE_LocalP2PTestDtlsBundleInIceRestart \
   DISABLED_LocalP2PTestDtlsBundleInIceRestart
 #else
 #define MAYBE_LocalP2PTestDtlsBundleInIceRestart \
   LocalP2PTestDtlsBundleInIceRestart
 #endif
 TEST_F(P2PTestConductor, MAYBE_LocalP2PTestDtlsBundleInIceRestart) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                  true);
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   receiving_client()->RemoveBundleFromReceivedSdp(true);
   LocalP2PTest();
-  VerifyRenderedSize(640, 480);
+  VerifyRenderedAspectRatio(640, 480);
 
   initializing_client()->IceRestart();
   receiving_client()->SetExpectIceRestart(true);
   receiving_client()->RemoveBundleFromReceivedSdp(false);
   LocalP2PTest();
-  VerifyRenderedSize(640, 480);
+  VerifyRenderedAspectRatio(640, 480);
 }
 
 // This test sets up a call transfer to a new callee with a different DTLS
 // fingerprint.
 TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   SetupAndVerifyDtlsCall();
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
   std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_receiving_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();
 
   SetSignalingReceivers();
   initializing_client()->IceRestart();
   LocalP2PTest();
-  VerifyRenderedSize(640, 480);
+  VerifyRenderedAspectRatio(640, 480);
 }
 
 TEST_F(P2PTestConductor, LocalP2PTestCVO) {
   ASSERT_TRUE(CreateTestClients());
   SetCaptureRotation(webrtc::kVideoRotation_90);
   LocalP2PTest();
-  VerifyRenderedSize(640, 480, webrtc::kVideoRotation_90);
+  VerifyRenderedAspectRatio(640, 480, webrtc::kVideoRotation_90);
 }
 
 TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) {
   ASSERT_TRUE(CreateTestClients());
   SetCaptureRotation(webrtc::kVideoRotation_90);
   receiving_client()->RemoveCvoFromReceivedSdp(true);
   LocalP2PTest();
-  VerifyRenderedSize(480, 640, webrtc::kVideoRotation_0);
+  VerifyRenderedAspectRatio(480, 640, webrtc::kVideoRotation_0);
 }
 
 // This test sets up a call between two endpoints that are configured to use
 // DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
 // negotiated and used for transport.
 TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                  true);
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
   LocalP2PTest();
-  VerifyRenderedSize(640, 480);
+  VerifyRenderedAspectRatio(640, 480);
 }
 
 // This test verifies that the negotiation will succeed with data channel only
 // in max-bundle mode.
 TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
   webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
   rtc_config.bundle_policy =
       webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
   ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config));
   initializing_client()->CreateDataChannel();
@@ skipping 994 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace