Adding HD peerconnection WebRTC call 45-second performance test.

The purpose is to test CPU usage and vital WebRTC stats over a
longer, more taxing call. Also updates all pages since the GitHub
location has moved again.

BUG=436361

Review URL: https://codereview.chromium.org/962113003

Cr-Commit-Position: refs/heads/master@{#319623}

tools/perf/page_sets/data/webrtc_cases.json

@@ -9,6 +9,16 @@
             "third_party/webrtc/samples/js/demos/html/pc1.html", 
             "webrtc/local-video.html"
         ], 
+        "webrtc_cases_010.wpr": [
+            "http://webrtc.github.io/samples/src/content/peerconnection/audio/?codec=PCMU", 
+            "http://webrtc.github.io/samples/src/content/peerconnection/audio/?codec=G722", 
+            "http://webrtc.github.io/samples/src/content/peerconnection/constraints/", 
+            "http://webrtc.github.io/samples/src/content/getusermedia/gum/", 
+            "http://webrtc.github.io/samples/src/content/peerconnection/pc1/", 
+            "http://webrtc.github.io/samples/src/content/peerconnection/audio/?codec=ISAC_16K", 
+            "http://webrtc.github.io/samples/src/content/getusermedia/resolution/", 
+            "http://webrtc.github.io/samples/src/content/peerconnection/audio/?codec=OPUS"
+        ], 
         "webrtc_cases_009.wpr": [
             "http://googlechrome.github.io/webrtc/samples/web/content/peerconnection/audio/?codec=PCMU", 
             "http://googlechrome.github.io/webrtc/samples/web/content/getusermedia/resolution/", 

tools/perf/page_sets/data/webrtc_cases_010.wpr.sha1

+8a10731694606d02ad2da04fc7a151e46fe6c32f
\ No newline at end of file

tools/perf/page_sets/webrtc_cases.py

@@ -7,6 +7,9 @@ from telemetry.page import page as page_module
 from telemetry.page import page_set as page_set_module
 
 
+WEBRTC_GITHUB_SAMPLES_URL = 'http://webrtc.github.io/samples/src/content/'
+
+
 class WebrtcCasesPage(page_module.Page):
 
   def __init__(self, url, page_set):
@@ -23,8 +26,7 @@ class Page1(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page1, self).__init__(
-        url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-             'getusermedia/gum/'),
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/gum/',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -37,8 +39,7 @@ class Page2(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page2, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'peerconnection/pc1/'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/pc1/',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -55,8 +56,7 @@ class Page3(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page3, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'getusermedia/resolution/'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/resolution/',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -70,8 +70,7 @@ class Page4(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page4, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'peerconnection/audio/?codec=OPUS'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=OPUS',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -86,8 +85,7 @@ class Page5(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page5, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'peerconnection/audio/?codec=G722'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=G722',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -102,8 +100,7 @@ class Page6(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page6, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'peerconnection/audio/?codec=PCMU'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=PCMU',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -118,8 +115,7 @@ class Page7(WebrtcCasesPage):
 
   def __init__(self, page_set):
     super(Page7, self).__init__(
-      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
-           'peerconnection/audio/?codec=ISAC_16K'),
+      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=ISAC_16K',
       page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -128,6 +124,25 @@ class Page7(WebrtcCasesPage):
     action_runner.Wait(10)
 
 
+class Page8(WebrtcCasesPage):
+
+  """ Why: Sets up a WebRTC 1080p call for 45 seconds. """
+
+  def __init__(self, page_set):
+    super(Page8, self).__init__(
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/constraints/',
+        page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    action_runner.ExecuteJavaScript('minWidthInput.value = 1920')
+    action_runner.ExecuteJavaScript('maxWidthInput.value = 1920')
+    action_runner.ExecuteJavaScript('minHeightInput.value = 1080')
+    action_runner.ExecuteJavaScript('maxHeightInput.value = 1080')
+    action_runner.ClickElement('button[id="getMedia"]')
+    action_runner.Wait(2)
+    action_runner.ClickElement('button[id="connect"]')
+    action_runner.Wait(45)
+
 class WebrtcCasesPageSet(page_set_module.PageSet):
 
   """ WebRTC tests for Real-time audio and video communication. """
@@ -147,3 +162,4 @@ class WebrtcCasesPageSet(page_set_module.PageSet):
     self.AddUserStory(Page5(self))
     self.AddUserStory(Page6(self))
     self.AddUserStory(Page7(self))
+    self.AddUserStory(Page8(self))