| // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 
 | // Use of this source code is governed by a BSD-style license that can be | 
 | // found in the LICENSE file. | 
 |  | 
 | #include "base/command_line.h" | 
 | #include "base/files/file_util.h" | 
 | #include "base/threading/platform_thread.h" | 
 | #include "build/build_config.h" | 
 | #include "content/browser/web_contents/web_contents_impl.h" | 
 | #include "content/browser/webrtc/webrtc_content_browsertest_base.h" | 
 | #include "content/public/common/content_switches.h" | 
 | #include "content/public/common/webrtc_ip_handling_policy.h" | 
 | #include "content/public/test/browser_test_utils.h" | 
 | #include "content/public/test/content_browser_test_utils.h" | 
 | #include "content/public/test/test_utils.h" | 
 | #include "media/audio/audio_manager.h" | 
 | #include "media/base/media_switches.h" | 
 | #include "net/test/embedded_test_server/embedded_test_server.h" | 
 |  | 
 | namespace content { | 
 |  | 
 | #if defined(OS_ANDROID) && defined(ADDRESS_SANITIZER) | 
 | // Renderer crashes under Android ASAN: https://crbug.com/408496. | 
 | #define MAYBE_WebRtcBrowserTest DISABLED_WebRtcBrowserTest | 
 | #else | 
 | #define MAYBE_WebRtcBrowserTest WebRtcBrowserTest | 
 | #endif | 
 |  | 
 | // This class tests the scenario when permission to access mic or camera is | 
 | // granted. | 
 | class MAYBE_WebRtcBrowserTest : public WebRtcContentBrowserTestBase { | 
 |  public: | 
 |   MAYBE_WebRtcBrowserTest() {} | 
 |   ~MAYBE_WebRtcBrowserTest() override {} | 
 |  | 
 |   void SetUpCommandLine(base::CommandLine* command_line) override { | 
 |     WebRtcContentBrowserTestBase::SetUpCommandLine(command_line); | 
 |     // Automatically grant device permission. | 
 |     AppendUseFakeUIForMediaStreamFlag(); | 
 |   } | 
 |  | 
 |  protected: | 
 |   // Convenience function since most peerconnection-call.html tests just load | 
 |   // the page, kick off some javascript and wait for the title to change to OK. | 
 |   void MakeTypicalPeerConnectionCall(const std::string& javascript) { | 
 |     MakeTypicalCall(javascript, "/media/peerconnection-call.html"); | 
 |   } | 
 | }; | 
 |  | 
 | // These tests will make a complete PeerConnection-based call and verify that | 
 | // video is playing for the call. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupDefaultVideoCall) { | 
 |   MakeTypicalPeerConnectionCall( | 
 |       "callAndExpectResolution({video: true}, 640, 480);"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupVideoCallWith1To1AspectRatio) { | 
 |   const std::string javascript = | 
 |       "callAndExpectResolution({video: {mandatory: {minWidth: 320," | 
 |       " maxWidth: 320, minHeight: 320, maxHeight: 320}}}, 320, 320);"; | 
 |   MakeTypicalPeerConnectionCall(javascript); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupVideoCallWith16To9AspectRatio) { | 
 |   const std::string javascript = | 
 |       "callAndExpectResolution({video: {mandatory: {minWidth: 640," | 
 |       " maxWidth: 640, minAspectRatio: 1.777}}}, 640, 360);"; | 
 |   MakeTypicalPeerConnectionCall(javascript); | 
 | } | 
 |  | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupVideoCallWith4To3AspectRatio) { | 
 |   const std::string javascript = | 
 |       "callAndExpectResolution({video: {mandatory: { minWidth: 320," | 
 |       "maxWidth: 320, minAspectRatio: 1.333, maxAspectRatio: 1.333}}}, 320," | 
 |       " 240);"; | 
 |   MakeTypicalPeerConnectionCall(javascript); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupVideoCallAndDisableLocalVideo) { | 
 |   const std::string javascript = | 
 |       "callAndDisableLocalVideo({video: true});"; | 
 |   MakeTypicalPeerConnectionCall(javascript); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupAudioAndVideoCall) { | 
 |   MakeTypicalPeerConnectionCall("call({video: true, audio: true});"); | 
 | } | 
 |  | 
 |  | 
 | #if defined(OS_WIN) && !defined(NVALGRIND) | 
 | // Times out on Dr. Memory bots: https://crbug.com/545740 | 
 | #define MAYBE_CanSetupCallAndSendDtmf DISABLED_CanSetupCallAndSendDtmf | 
 | #else | 
 | #define MAYBE_CanSetupCallAndSendDtmf CanSetupCallAndSendDtmf | 
 | #endif | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        MAYBE_CanSetupCallAndSendDtmf) { | 
 |   MakeTypicalPeerConnectionCall("callAndSendDtmf(\'123,abc\');"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanMakeEmptyCallThenAddStreamsAndRenegotiate) { | 
 |   const char* kJavascript = | 
 |       "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});"; | 
 |   MakeTypicalPeerConnectionCall(kJavascript); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanMakeAudioCallAndThenRenegotiateToVideo) { | 
 |   const char* kJavascript = | 
 |       "callAndRenegotiateToVideo({audio: true}, {audio: true, video:true});"; | 
 |   MakeTypicalPeerConnectionCall(kJavascript); | 
 | } | 
 |  | 
 | // This test makes a call between pc1 and pc2 where a video only stream is sent | 
 | // from pc1 to pc2. The stream sent from pc1 to pc2 is cloned from the stream | 
 | // received on pc2 to test that cloning of remote video and audio tracks works | 
 | // as intended and is sent back to pc1. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CanForwardRemoteStream) { | 
 | #if defined (OS_ANDROID) | 
 |   // This test fails on Nexus 5 devices. | 
 |   // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389 | 
 |   // for details. | 
 |   base::CommandLine::ForCurrentProcess()->AppendSwitch( | 
 |       switches::kDisableWebRtcHWDecoding); | 
 | #endif | 
 |   MakeTypicalPeerConnectionCall( | 
 |       "callAndForwardRemoteStream({video: true, audio: true});"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        NoCrashWhenConnectChromiumSinkToRemoteTrack) { | 
 |   MakeTypicalPeerConnectionCall("ConnectChromiumSinkToRemoteAudioTrack();"); | 
 | } | 
 |  | 
 | // This test will make a complete PeerConnection-based call but remove the | 
 | // MSID and bundle attribute from the initial offer to verify that | 
 | // video is playing for the call even if the initiating client don't support | 
 | // MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02 | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CanSetupAudioAndVideoCallWithoutMsidAndBundle) { | 
 |   MakeTypicalPeerConnectionCall("callWithoutMsidAndBundle();"); | 
 | } | 
 |  | 
 | // This test will modify the SDP offer to an unsupported codec, which should | 
 | // cause SetLocalDescription to fail. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        NegotiateUnsupportedVideoCodec) { | 
 |   MakeTypicalPeerConnectionCall("negotiateUnsupportedVideoCodec();"); | 
 | } | 
 |  | 
 | // This test will modify the SDP offer to use no encryption, which should | 
 | // cause SetLocalDescription to fail. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, NegotiateNonCryptoCall) { | 
 |   MakeTypicalPeerConnectionCall("negotiateNonCryptoCall();"); | 
 | } | 
 |  | 
 | // This test can negotiate an SDP offer that includes a b=AS:xx to control | 
 | // the bandwidth for audio and video | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, NegotiateOfferWithBLine) { | 
 |   MakeTypicalPeerConnectionCall("negotiateOfferWithBLine();"); | 
 | } | 
 |  | 
 | // This test will make a PeerConnection-based call and send a new Video | 
 | // MediaStream that has been created based on a MediaStream created with | 
 | // getUserMedia. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, | 
 |                        CallWithNewVideoMediaStream) { | 
 |   MakeTypicalPeerConnectionCall("callWithNewVideoMediaStream();"); | 
 | } | 
 |  | 
 | // This test will make a PeerConnection-based call and send a new Video | 
 | // MediaStream that has been created based on a MediaStream created with | 
 | // getUserMedia. When video is flowing, the VideoTrack is removed and an | 
 | // AudioTrack is added instead. | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CallAndModifyStream) { | 
 |   MakeTypicalPeerConnectionCall( | 
 |       "callWithNewVideoMediaStreamLaterSwitchToAudio();"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) { | 
 |   MakeTypicalPeerConnectionCall("addTwoMediaStreamsToOneConnection();"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) { | 
 |   MakeTypicalPeerConnectionCall("callAndEnsureVideoTrackMutingWorks();"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CreateOfferWithOfferOptions) { | 
 |   MakeTypicalPeerConnectionCall("testCreateOfferOptions();"); | 
 | } | 
 |  | 
 | IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CallInsideIframe) { | 
 |   MakeTypicalPeerConnectionCall("callInsideIframe({video: true, audio:true});"); | 
 | } | 
 |  | 
 | }  // namespace content |