Split WebRTC browser tests in getusermedia and peerconnection tests.

content/browser/media/webrtc_browsertest.cc

 // 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/json/json_reader.h"
 #include "base/strings/stringprintf.h"
-#include "base/strings/utf_string_conversions.h"
-#include "base/test/trace_event_analyzer.h"
 #include "base/values.h"
+#include "content/browser/media/webrtc_content_browsertest_base.h"
 #include "content/browser/media/webrtc_internals.h"
 #include "content/browser/web_contents/web_contents_impl.h"
 #include "content/public/common/content_switches.h"
 #include "content/public/test/browser_test_utils.h"
 #include "content/public/test/test_utils.h"
 #include "content/shell/browser/shell.h"
-#include "content/test/content_browser_test.h"
 #include "content/test/content_browser_test_utils.h"
 #include "media/audio/audio_manager.h"
 #include "net/test/embedded_test_server/embedded_test_server.h"
-#include "testing/perf/perf_test.h"
 
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #endif
 
-using trace_analyzer::TraceAnalyzer;
-using trace_analyzer::Query;
-using trace_analyzer::TraceEventVector;
-
-namespace {
-
-static const char kGetUserMediaAndStop[] = "getUserMediaAndStop";
-static const char kGetUserMediaAndWaitAndStop[] = "getUserMediaAndWaitAndStop";
-static const char kGetUserMediaAndAnalyseAndStop[] =
-    "getUserMediaAndAnalyseAndStop";
-
-// Results returned by JS.
-static const char kOK[] = "OK";
-static const char kGetUserMediaFailed[] =
-    "GetUserMedia call failed with code undefined";
-
-std::string GenerateGetUserMediaCall(const char* function_name,
-                                     int min_width,
-                                     int max_width,
-                                     int min_height,
-                                     int max_height,
-                                     int min_frame_rate,
-                                     int max_frame_rate) {
-  return base::StringPrintf(
-      "%s({video: {mandatory: {minWidth: %d, maxWidth: %d, "
-      "minHeight: %d, maxHeight: %d, minFrameRate: %d, maxFrameRate: %d}, "
-      "optional: []}});",
-      function_name,
-      min_width,
-      max_width,
-      min_height,
-      max_height,
-      min_frame_rate,
-      max_frame_rate);
-}
-
-std::string GenerateGetUserMediaWithMandatorySourceID(
-    const std::string& function_name,
-    const std::string& audio_source_id,
-    const std::string& video_source_id) {
-  const std::string audio_constraint =
-      "audio: {mandatory: { sourceId:\"" + audio_source_id + "\"}}, ";
-
-  const std::string video_constraint =
-      "video: {mandatory: { sourceId:\"" + video_source_id + "\"}}";
-  return function_name + "({" + audio_constraint + video_constraint + "});";
-}
-
-std::string GenerateGetUserMediaWithOptionalSourceID(
-    const std::string& function_name,
-    const std::string& audio_source_id,
-    const std::string& video_source_id) {
-  const std::string audio_constraint =
-      "audio: {optional: [{sourceId:\"" + audio_source_id + "\"}]}, ";
-
-  const std::string video_constraint =
-      "video: {optional: [{ sourceId:\"" + video_source_id + "\"}]}";
-  return function_name + "({" + audio_constraint + video_constraint + "});";
-}
-
-}
-
 namespace content {
 
-class WebrtcBrowserTest: public ContentBrowserTest {
+class WebRtcBrowserTest: public WebRtcContentBrowserTest {
  public:
-  WebrtcBrowserTest() : trace_log_(NULL) {}
-  virtual ~WebrtcBrowserTest() {}
-
-  virtual void SetUp() OVERRIDE {
-    // These tests require pixel output.
-    UseRealGLContexts();
-    ContentBrowserTest::SetUp();
-  }
-
-  virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE {
-    // We need fake devices in this test since we want to run on naked VMs. We
-    // assume these switches are set by default in content_browsertests.
-    ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
-        switches::kUseFakeDeviceForMediaStream));
-    ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
-        switches::kUseFakeUIForMediaStream));
-  }
-
-  void StartTracing() {
-    CHECK(trace_log_ == NULL) << "Can only can start tracing once";
-    trace_log_ = base::debug::TraceLog::GetInstance();
-    trace_log_->SetEnabled(base::debug::CategoryFilter("video"),
-                           base::debug::TraceLog::RECORDING_MODE,
-                           base::debug::TraceLog::ENABLE_SAMPLING);
-    // Check that we are indeed recording.
-    EXPECT_EQ(trace_log_->GetNumTracesRecorded(), 1);
-  }
-
-  void StopTracing() {
-    CHECK(message_loop_runner_ == NULL) << "Calling StopTracing more than once";
-    trace_log_->SetDisabled();
-    message_loop_runner_ = new MessageLoopRunner;
-    trace_log_->Flush(base::Bind(&WebrtcBrowserTest::OnTraceDataCollected,
-                                 base::Unretained(this)));
-    message_loop_runner_->Run();
-  }
-
-  void OnTraceDataCollected(
-      const scoped_refptr<base::RefCountedString>& events_str_ptr,
-      bool has_more_events) {
-    CHECK(!has_more_events);
-    recorded_trace_data_ = events_str_ptr;
-    message_loop_runner_->Quit();
-  }
-
-  TraceAnalyzer* CreateTraceAnalyzer() {
-    return TraceAnalyzer::Create("[" + recorded_trace_data_->data() + "]");
-  }
-
-  void GetSources(std::vector<std::string>* audio_ids,
-                  std::vector<std::string>* video_ids) {
-    GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-    NavigateToURL(shell(), url);
-
-    std::string sources_as_json = ExecuteJavascriptAndReturnResult(
-        "getSources()");
-    EXPECT_FALSE(sources_as_json.empty());
-
-    int error_code;
-    std::string error_message;
-    scoped_ptr<base::Value> value(
-        base::JSONReader::ReadAndReturnError(sources_as_json,
-                                             base::JSON_ALLOW_TRAILING_COMMAS,
-                                             &error_code,
-                                             &error_message));
-
-    ASSERT_TRUE(value.get() != NULL) << error_message;
-    EXPECT_EQ(value->GetType(), base::Value::TYPE_LIST);
-
-    base::ListValue* values;
-    ASSERT_TRUE(value->GetAsList(&values));
-
-    for (base::ListValue::iterator it = values->begin();
-         it != values->end(); ++it) {
-      const base::DictionaryValue* dict;
-      std::string kind;
-      std::string id;
-      ASSERT_TRUE((*it)->GetAsDictionary(&dict));
-      ASSERT_TRUE(dict->GetString("kind", &kind));
-      ASSERT_TRUE(dict->GetString("id", &id));
-      ASSERT_FALSE(id.empty());
-      EXPECT_TRUE(kind == "audio" || kind == "video");
-      if (kind == "audio") {
-        audio_ids->push_back(id);
-      } else if (kind == "video") {
-        video_ids->push_back(id);
-      }
-    }
-    ASSERT_FALSE(audio_ids->empty());
-    ASSERT_FALSE(video_ids->empty());
-  }
-
- protected:
-  bool ExecuteJavascript(const std::string& javascript) {
-    return ExecuteScript(shell()->web_contents(), javascript);
-  }
-
-  // Executes |javascript|. The script is required to use
-  // window.domAutomationController.send to send a string value back to here.
-  std::string ExecuteJavascriptAndReturnResult(const std::string& javascript) {
-    std::string result;
-    EXPECT_TRUE(ExecuteScriptAndExtractString(shell()->web_contents(),
-                                              javascript,
-                                              &result));
-    return result;
-  }
-
-  void ExpectTitle(const std::string& expected_title) const {
-    base::string16 expected_title16(base::ASCIIToUTF16(expected_title));
-    TitleWatcher title_watcher(shell()->web_contents(), expected_title16);
-    EXPECT_EQ(expected_title16, title_watcher.WaitAndGetTitle());
-  }
+  WebRtcBrowserTest() {}
+  virtual ~WebRtcBrowserTest() {}
 
   // 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) {
     ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
     GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
     NavigateToURL(shell(), url);
     ExecuteTestAndWaitForOk(javascript);
   }
 
   void ExecuteTestAndWaitForOk(const std::string& javascript) {
 #if defined (OS_ANDROID)
     // Always force iSAC 16K on Android for now (Opus is broken).
     ASSERT_TRUE(ExecuteJavascript("forceIsac16KInSdp();"));
 #endif
 
     ASSERT_TRUE(ExecuteJavascript(javascript));
     ExpectTitle("OK");
   }
-
- private:
-  base::debug::TraceLog* trace_log_;
-  scoped_refptr<base::RefCountedString> recorded_trace_data_;
-  scoped_refptr<MessageLoopRunner> message_loop_runner_;
 };
 
-// These tests will all make a getUserMedia call with different constraints and
-// see that the success callback is called. If the error callback is called or
-// none of the callbacks are called the tests will simply time out and fail.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetVideoStreamAndStop) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  ASSERT_TRUE(ExecuteJavascript(
-      base::StringPrintf("%s({video: true});", kGetUserMediaAndStop)));
-
-  ExpectTitle("OK");
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndStop) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  ASSERT_TRUE(ExecuteJavascript(base::StringPrintf(
-      "%s({video: true, audio: true});", kGetUserMediaAndStop)));
-
-  ExpectTitle("OK");
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndClone) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  ASSERT_TRUE(ExecuteJavascript("getUserMediaAndClone();"));
-
-  ExpectTitle("OK");
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetUserMediaWithMandatorySourceID_1) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  for (size_t j = 0; j < video_ids.size() / 2; ++j) {
-    for (size_t i = 0; i < audio_ids.size(); ++i) {
-      EXPECT_EQ(kOK,
-                ExecuteJavascriptAndReturnResult(
-                    GenerateGetUserMediaWithMandatorySourceID(
-                        kGetUserMediaAndStop, audio_ids[i], video_ids[j])));
-    }
-  }
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetUserMediaWithMandatorySourceID_2) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  for (size_t j = video_ids.size() / 2; j < video_ids.size(); ++j) {
-    for (size_t i = 0; i < audio_ids.size(); ++i) {
-      EXPECT_EQ(kOK,
-                ExecuteJavascriptAndReturnResult(
-                    GenerateGetUserMediaWithMandatorySourceID(
-                        kGetUserMediaAndStop, audio_ids[i], video_ids[j])));
-    }
-  }
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
-                       GetUserMediaWithInvalidMandatorySourceID) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-
-  // Test with invalid mandatory audio sourceID.
-  NavigateToURL(shell(), url);
-  EXPECT_EQ(kGetUserMediaFailed,
-            ExecuteJavascriptAndReturnResult(
-                GenerateGetUserMediaWithMandatorySourceID(
-                    kGetUserMediaAndStop, "something invalid", video_ids[0])));
-
-  // Test with invalid mandatory video sourceID.
-  EXPECT_EQ(kGetUserMediaFailed,
-            ExecuteJavascriptAndReturnResult(
-                GenerateGetUserMediaWithMandatorySourceID(
-                    kGetUserMediaAndStop, audio_ids[0], "something invalid")));
-
-  // Test with empty mandatory audio sourceID.
-  EXPECT_EQ(kGetUserMediaFailed,
-            ExecuteJavascriptAndReturnResult(
-                GenerateGetUserMediaWithMandatorySourceID(
-                    kGetUserMediaAndStop, "", video_ids[0])));
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetUserMediaWithOptionalSourceID_1) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  // Test all combinations of mandatory sourceID.
-  for (size_t j = 0; j < video_ids.size() / 2; ++j) {
-    for (size_t i = 0; i < audio_ids.size(); ++i) {
-      EXPECT_EQ(kOK,
-                ExecuteJavascriptAndReturnResult(
-                    GenerateGetUserMediaWithOptionalSourceID(
-                        kGetUserMediaAndStop, audio_ids[i], video_ids[j])));
-    }
-  }
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetUserMediaWithOptionalSourceID_2) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-
-  // Test all combinations of mandatory sourceID.
-  for (size_t j = video_ids.size() / 2; j < video_ids.size(); ++j) {
-    for (size_t i = 0; i < audio_ids.size(); ++i) {
-      EXPECT_EQ(kOK,
-                ExecuteJavascriptAndReturnResult(
-                    GenerateGetUserMediaWithOptionalSourceID(
-                        kGetUserMediaAndStop, audio_ids[i], video_ids[j])));
-    }
-  }
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
-                       GetUserMediaWithInvalidOptionalSourceID) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  std::vector<std::string> audio_ids;
-  std::vector<std::string> video_ids;
-  GetSources(&audio_ids, &video_ids);
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-
-  // Test with invalid optional audio sourceID.
-  NavigateToURL(shell(), url);
-  EXPECT_EQ(
-      kOK,
-      ExecuteJavascriptAndReturnResult(GenerateGetUserMediaWithOptionalSourceID(
-          kGetUserMediaAndStop, "something invalid", video_ids[0])));
-
-  // Test with invalid optional video sourceID.
-  EXPECT_EQ(
-      kOK,
-      ExecuteJavascriptAndReturnResult(GenerateGetUserMediaWithOptionalSourceID(
-          kGetUserMediaAndStop, audio_ids[0], "something invalid")));
-
-  // Test with empty optional audio sourceID.
-  EXPECT_EQ(
-      kOK,
-      ExecuteJavascriptAndReturnResult(GenerateGetUserMediaWithOptionalSourceID(
-          kGetUserMediaAndStop, "", video_ids[0])));
-}
-
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CanSetupVideoCall DISABLED_CanSetupVideoCall
 #else
 #define MAYBE_CanSetupVideoCall CanSetupVideoCall
 #endif
 
 // These tests will make a complete PeerConnection-based call and verify that
 // video is playing for the call.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupVideoCall) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupVideoCall) {
   MakeTypicalPeerConnectionCall("call({video: true});");
 }
 
-// This test will make a simple getUserMedia page, verify that video is playing
-// in a simple local <video>, and for a couple of seconds, collect some
-// performance traces.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TracePerformanceDuringGetUserMedia) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-  NavigateToURL(shell(), url);
-  // Put getUserMedia to work and let it run for a couple of seconds.
-  ASSERT_TRUE(ExecuteJavascript(base::StringPrintf(
-      "%s({video: true}, 10);", kGetUserMediaAndWaitAndStop)));
-
-  // Make sure the stream is up and running, then start collecting traces.
-  ExpectTitle("Running...");
-  StartTracing();
-
-  // Wait until the page title changes to "OK". Do not sleep() here since that
-  // would stop both this code and the browser underneath.
-  ExpectTitle("OK");
-  StopTracing();
-
-  scoped_ptr<TraceAnalyzer> analyzer(CreateTraceAnalyzer());
-  analyzer->AssociateBeginEndEvents();
-  trace_analyzer::TraceEventVector events;
-  analyzer->FindEvents(
-      Query::EventNameIs("VideoCaptureController::OnIncomingCapturedFrame"),
-      &events);
-  ASSERT_GT(events.size(), 0u)
-      << "Could not collect any samples during test, this is bad";
-
-  std::string duration_us;
-  std::string interarrival_us;
-  for (size_t i = 0; i != events.size(); ++i) {
-    duration_us.append(
-        base::StringPrintf("%d,", static_cast<int>(events[i]->duration)));
-  }
-
-  for (size_t i = 1; i < events.size(); ++i) {
-    interarrival_us.append(base::StringPrintf(
-        "%d,",
-        static_cast<int>(events[i]->timestamp - events[i - 1]->timestamp)));
-  }
-
-  perf_test::PrintResultList(
-      "video_capture", "", "sample_duration", duration_us, "us", true);
-
-  perf_test::PrintResultList(
-      "video_capture", "", "interarrival_time", interarrival_us, "us", true);
-}
-
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux, see http://crbug.com/240376
 #define MAYBE_CanSetupAudioAndVideoCall DISABLED_CanSetupAudioAndVideoCall
 #else
 #define MAYBE_CanSetupAudioAndVideoCall CanSetupAudioAndVideoCall
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) {
   MakeTypicalPeerConnectionCall("call({video: true, audio: true});");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) {
   MakeTypicalPeerConnectionCall("callAndSendDtmf(\'123,abc\');");
 }
 
 // TODO(phoglund): this test fails because the peer connection state will be
 // stable in the second negotiation round rather than have-local-offer.
 // http://crbug.com/293125.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        DISABLED_CanMakeEmptyCallThenAddStreamsAndRenegotiate) {
   const char* kJavascript =
       "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});";
   MakeTypicalPeerConnectionCall(kJavascript);
 }
 
 // Below 2 test will make a complete PeerConnection-based call between pc1 and
 // pc2, and then use the remote stream to setup a call between pc3 and pc4, and
 // then verify that video is received on pc3 and pc4.
 // Flaky on win xp. http://crbug.com/304775
 #if defined(OS_WIN)
 #define MAYBE_CanForwardRemoteStream DISABLED_CanForwardRemoteStream
 #define MAYBE_CanForwardRemoteStream720p DISABLED_CanForwardRemoteStream720p
 #else
 #define MAYBE_CanForwardRemoteStream CanForwardRemoteStream
 #define MAYBE_CanForwardRemoteStream720p CanForwardRemoteStream720p
 #endif
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanForwardRemoteStream) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
   MakeTypicalPeerConnectionCall(
       "callAndForwardRemoteStream({video: true, audio: true});");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
   const std::string javascript = GenerateGetUserMediaCall(
       "callAndForwardRemoteStream", 1280, 1280, 720, 720, 30, 30);
   MakeTypicalPeerConnectionCall(javascript);
 }
 
 // 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
 #if defined(OS_WIN)
 // Disabled for win, see http://crbug.com/235089.
 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
         DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle
 #elif defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux, see http://crbug.com/240373
 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
         DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle
 #else
 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
         CanSetupAudioAndVideoCallWithoutMsidAndBundle
 #endif
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        MAYBE_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(WebrtcBrowserTest, NegotiateUnsupportedVideoCodec) {
+IN_PROC_BROWSER_TEST_F(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(WebrtcBrowserTest, NegotiateNonCryptoCall) {
+IN_PROC_BROWSER_TEST_F(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(WebrtcBrowserTest, NegotiateOfferWithBLine) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, NegotiateOfferWithBLine) {
   MakeTypicalPeerConnectionCall("negotiateOfferWithBLine();");
 }
 
 // This test will make a complete PeerConnection-based call using legacy SDP
 // settings: GIce, external SDES, and no BUNDLE.
 #if defined(OS_WIN)
 // Disabled for win7, see http://crbug.com/235089.
 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall
 #elif defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux, see http://crbug.com/240373
 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall
 #else
 #define MAYBE_CanSetupLegacyCall CanSetupLegacyCall
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupLegacyCall) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupLegacyCall) {
   MakeTypicalPeerConnectionCall("callWithLegacySdp();");
 }
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel.
 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataOnly) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallWithDataOnly) {
   MakeTypicalPeerConnectionCall("callWithDataOnly();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithSctpDataOnly) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallWithSctpDataOnly) {
   MakeTypicalPeerConnectionCall("callWithSctpDataOnly();");
 }
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithDataAndMedia DISABLED_CallWithDataAndMedia
 #else
 #define MAYBE_CallWithDataAndMedia CallWithDataAndMedia
 #endif
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and audio and video tracks.
 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndMedia) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithDataAndMedia) {
   MakeTypicalPeerConnectionCall("callWithDataAndMedia();");
 }
 
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithSctpDataAndMedia DISABLED_CallWithSctpDataAndMedia
 #else
 #define MAYBE_CallWithSctpDataAndMedia CallWithSctpDataAndMedia
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        MAYBE_CallWithSctpDataAndMedia) {
   MakeTypicalPeerConnectionCall("callWithSctpDataAndMedia();");
 }
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithDataAndLaterAddMedia DISABLED_CallWithDataAndLaterAddMedia
 #else
 // Temporarily disable the test on all platforms. http://crbug.com/293252
 #define MAYBE_CallWithDataAndLaterAddMedia DISABLED_CallWithDataAndLaterAddMedia
 #endif
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and later add an audio and video track.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) {
   MakeTypicalPeerConnectionCall("callWithDataAndLaterAddMedia();");
 }
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithNewVideoMediaStream DISABLED_CallWithNewVideoMediaStream
 #else
 #define MAYBE_CallWithNewVideoMediaStream CallWithNewVideoMediaStream
 #endif
 
 // 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(WebrtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_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.
 // TODO(phoglund): This test is manual since not all buildbots has an audio
 // input.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MANUAL_CallAndModifyStream) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MANUAL_CallAndModifyStream) {
   MakeTypicalPeerConnectionCall(
       "callWithNewVideoMediaStreamLaterSwitchToAudio();");
 }
 
-namespace {
-
-struct UserMediaSizes {
-  int min_width;
-  int max_width;
-  int min_height;
-  int max_height;
-  int min_frame_rate;
-  int max_frame_rate;
-};
-
-}  // namespace
-
-class WebrtcUserMediaBrowserTest
-    : public WebrtcBrowserTest,
-      public testing::WithParamInterface<UserMediaSizes> {
- public:
-  WebrtcUserMediaBrowserTest() : user_media_(GetParam()) {}
-  const UserMediaSizes& user_media() const { return user_media_; }
-
- private:
-  UserMediaSizes user_media_;
-};
-
-// This test calls getUserMedia in sequence with different constraints.
-IN_PROC_BROWSER_TEST_P(WebrtcUserMediaBrowserTest, GetUserMediaConstraints) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-
-  std::string call = GenerateGetUserMediaCall(kGetUserMediaAndStop,
-                                              user_media().min_width,
-                                              user_media().max_width,
-                                              user_media().min_height,
-                                              user_media().max_height,
-                                              user_media().min_frame_rate,
-                                              user_media().max_frame_rate);
-  DVLOG(1) << "Calling getUserMedia: " << call;
-  NavigateToURL(shell(), url);
-  ASSERT_TRUE(ExecuteJavascript(call));
-  ExpectTitle("OK");
-}
-
-static const UserMediaSizes kAllUserMediaSizes[] = {
-    {320, 320, 180, 180, 30, 30},
-    {320, 320, 240, 240, 30, 30},
-    {640, 640, 360, 360, 30, 30},
-    {640, 640, 480, 480, 30, 30},
-    {960, 960, 720, 720, 30, 30},
-    {1280, 1280, 720, 720, 30, 30},
-    {1920, 1920, 1080, 1080, 30, 30}};
-
-INSTANTIATE_TEST_CASE_P(UserMedia,
-                        WebrtcUserMediaBrowserTest,
-                        testing::ValuesIn(kAllUserMediaSizes));
-
-// This test calls getUserMedia and checks for aspect ratio behavior.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TestGetUserMediaAspectRatio) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
-
-  std::string constraints_4_3 = GenerateGetUserMediaCall(
-      kGetUserMediaAndAnalyseAndStop, 640, 640, 480, 480, 30, 30);
-  std::string constraints_16_9 = GenerateGetUserMediaCall(
-      kGetUserMediaAndAnalyseAndStop, 640, 640, 360, 360, 30, 30);
-
-  // TODO(mcasas): add more aspect ratios, in particular 16:10 crbug.com/275594.
-
-  NavigateToURL(shell(), url);
-  ASSERT_TRUE(ExecuteJavascript(constraints_4_3));
-  ExpectTitle("4:3 letterbox");
-
-  NavigateToURL(shell(), url);
-  ASSERT_TRUE(ExecuteJavascript(constraints_16_9));
-  ExpectTitle("16:9 letterbox");
-}
-
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, AddTwoMediaStreamsToOnePC) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) {
   MakeTypicalPeerConnectionCall("addTwoMediaStreamsToOneConnection();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        EstablishAudioVideoCallAndMeasureOutputLevel) {
   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
     // Bots with no output devices will force the audio code into a different
     // path where it doesn't manage to set either the low or high latency path.
     // This test will compute useless values in that case, so skip running on
     // such bots (see crbug.com/326338).
     LOG(INFO) << "Missing output devices: skipping test...";
     return;
   }
 
   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kUseFakeDeviceForMediaStream))
           << "Must run with fake devices since the test will explicitly look "
           << "for the fake device signal.";
 
   MakeTypicalPeerConnectionCall("callAndEnsureAudioIsPlaying();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        EstablishAudioVideoCallAndVerifyMutingWorks) {
   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
     // Bots with no output devices will force the audio code into a different
     // path where it doesn't manage to set either the low or high latency path.
     // This test will compute useless values in that case, so skip running on
     // such bots (see crbug.com/326338).
     LOG(INFO) << "Missing output devices: skipping test...";
     return;
   }
 
   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kUseFakeDeviceForMediaStream))
           << "Must run with fake devices since the test will explicitly look "
           << "for the fake device signal.";
 
   MakeTypicalPeerConnectionCall("callAndEnsureAudioMutingWorks();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallAndVerifyVideoMutingWorks) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
   MakeTypicalPeerConnectionCall("callAndEnsureVideoMutingWorks();");
 }
 
 #if defined(OS_WIN) || (defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY))
 // Timing out on ARM linux bot: http://crbug.com/238490
 // Failing on Windows: http://crbug.com/331035
 #define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
 #else
 #define MAYBE_CallWithAecDump CallWithAecDump
 #endif
 
 // This tests will make a complete PeerConnection-based call, verify that
 // video is playing for the call, and verify that a non-empty AEC dump file
 // exists. The AEC dump is enabled through webrtc-internals, in contrast to
 // using a command line flag (tested in webrtc_aecdump_browsertest.cc). The HTML
 // and Javascript is bypassed since it would trigger a file picker dialog.
 // Instead, the dialog callback FileSelected() is invoked directly. In fact,
 // there's never a webrtc-internals page opened at all since that's not needed.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithAecDump) {
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithAecDump) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
   // We must navigate somewhere first so that the render process is created.
   NavigateToURL(shell(), GURL(""));
 
   base::FilePath dump_file;
   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 
   // This fakes the behavior of another open tab with webrtc-internals, and
   // enabling AEC dump in that tab.
@@ skipping 13 matching lines @@
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
 #else
 #define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled
 #endif
 
 // As above, but enable and disable dump before starting a call. The file should
 // be created, but should be empty.
-IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
                        MAYBE_CallWithAecDumpEnabledThenDisabled) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
   // We must navigate somewhere first so that the render process is created.
   NavigateToURL(shell(), GURL(""));
 
   base::FilePath dump_file;
   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 
   // This fakes the behavior of another open tab with webrtc-internals, and
   // enabling AEC dump in that tab, then disabling it.
   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
   WebRTCInternals::GetInstance()->DisableAecDump();
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
   ExecuteTestAndWaitForOk("call({video: true, audio: true});");
 
   EXPECT_TRUE(base::PathExists(dump_file));
   int64 file_size = 0;
   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
   EXPECT_EQ(0, file_size);
 
   base::DeleteFile(dump_file, false);
 }
 
-
 }  // namespace content