Added a basic WebRTC peerconnection browser test with video verification.

content/browser/webrtc_browsertest.cc

Index: content/browser/webrtc_browsertest.cc
diff --git a/content/browser/webrtc_browsertest.cc b/content/browser/webrtc_browsertest.cc
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+++ b/content/browser/webrtc_browsertest.cc
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+// 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/utf_string_conversions.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/shell/shell.h"
+#include "content/test/content_browser_test.h"
+#include "content/test/content_browser_test_utils.h"
+#include "net/test/test_server.h"
+
+namespace content {
+
+class WebrtcBrowserTest: public ContentBrowserTest {
+ public:
+  WebrtcBrowserTest() {}
+  virtual ~WebrtcBrowserTest() {}
+
+  virtual void SetUp() OVERRIDE {
+    // We need fake devices in this test since we want to run on naked VMs. We
+    // assume this switch is set by default in content_browsertests.
+    ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
+        switches::kUseFakeDeviceForMediaStream));
+
+    ASSERT_TRUE(test_server()->Start());
+    ContentBrowserTest::SetUp();
+  }
+ protected:
+  bool ExecuteJavascript(const std::string& javascript) {
+    RenderViewHost* render_view_host =
+        shell()->web_contents()->GetRenderViewHost();
+
+    return ExecuteJavaScript(render_view_host, L"", ASCIIToWide(javascript));
+  }
+
+  void ExpectTitle(const std::string& expected_title) const {
+    string16 expected_title16(ASCIIToUTF16(expected_title));
+    TitleWatcher title_watcher(shell()->web_contents(), expected_title16);
+    EXPECT_EQ(expected_title16, title_watcher.WaitAndGetTitle());
+  }
+};
+
+// 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) {

[perkj_chrome, 2012/12/04 10:51:43]

Why are these tests called stop? you never call stop in the media stream?

Can you add that?

[phoglund_chromium, 2012/12/04 13:03:27]

Yes, we do call stop in getusermedia_and_stop.html. So we request user media,
get it, and then stop the local stream immediately.

+  GURL url(test_server()->GetURL("files/media/getusermedia_and_stop.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("getUserMedia({video: true});"));
+
+  ExpectTitle("OK");
+}
+
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndStop) {
+  GURL url(test_server()->GetURL("files/media/getusermedia_and_stop.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("getUserMedia({video: true, audio: true});"));
+
+  ExpectTitle("OK");
+}
+
+// These tests will make a complete PeerConnection-based call and verify that
+// video is playing for the call.
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CanSetupVideoCall) {
+  GURL url(test_server()->GetURL("files/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("call({video: true});"));
+  ExpectTitle("OK");
+}
+
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CanSetupAudioAndVideoCall) {
+  GURL url(test_server()->GetURL("files/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("call({video: true, audio: true});"));
+  ExpectTitle("OK");
+}
+
+}  // namespace content
+