Adding telemetry WebRTC measurements for the audio codecs we support.

tools/perf/page_sets/webrtc_cases.py

Index: tools/perf/page_sets/webrtc_cases.py
diff --git a/tools/perf/page_sets/webrtc_cases.py b/tools/perf/page_sets/webrtc_cases.py
index e5db1bd60944344540dd8822c62f40252b7f67c1..1bd0f9cceb34f39320676d39caf599d7d99403df 100644
--- a/tools/perf/page_sets/webrtc_cases.py
+++ b/tools/perf/page_sets/webrtc_cases.py
@@ -64,6 +64,70 @@ class Page3(WebrtcCasesPage):
     action_runner.Wait(10)
 
 
+class Page4(WebrtcCasesPage):
+
+  """ Why: Sets up a WebRTC audio call with Opus. """
+
+  def __init__(self, page_set):
+    super(Page4, self).__init__(
+      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
+           'peerconnection/audio/?codec=OPUS'),
+      page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    action_runner.ExecuteJavaScript('codecSelector.value="OPUS";')
+    action_runner.ClickElement('button[id="callButton"]')
+    action_runner.Wait(10)
+
+
+class Page5(WebrtcCasesPage):
+
+  """ Why: Sets up a WebRTC audio call with G722. """
+
+  def __init__(self, page_set):
+    super(Page5, self).__init__(
+      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
+           'peerconnection/audio/?codec=G722'),
+      page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    action_runner.ExecuteJavaScript('codecSelector.value="G722";')
+    action_runner.ClickElement('button[id="callButton"]')
+    action_runner.Wait(10)
+
+
+class Page6(WebrtcCasesPage):
+
+  """ Why: Sets up a WebRTC audio call with PCMU. """
+
+  def __init__(self, page_set):
+    super(Page6, self).__init__(
+      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
+           'peerconnection/audio/?codec=PCMU'),
+      page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    action_runner.ExecuteJavaScript('codecSelector.value="PCMU";')
+    action_runner.ClickElement('button[id="callButton"]')
+    action_runner.Wait(10)
+
+
+class Page7(WebrtcCasesPage):
+
+  """ Why: Sets up a WebRTC audio call with iSAC 16K. """
+
+  def __init__(self, page_set):
+    super(Page7, self).__init__(
+      url=('http://googlechrome.github.io/webrtc/samples/web/content/'
+           'peerconnection/audio/?codec=ISAC_16K'),
+      page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    action_runner.ExecuteJavaScript('codecSelector.value="ISAC/16000";')
+    action_runner.ClickElement('button[id="callButton"]')
+    action_runner.Wait(10)
+
+
 class WebrtcCasesPageSet(page_set_module.PageSet):
 
   """ WebRTC tests for Real-time audio and video communication. """
@@ -76,3 +140,10 @@ class WebrtcCasesPageSet(page_set_module.PageSet):
     self.AddUserStory(Page1(self))
     self.AddUserStory(Page2(self))
     self.AddUserStory(Page3(self))
+    self.AddUserStory(Page1(self))
+    self.AddUserStory(Page2(self))
+    self.AddUserStory(Page3(self))
+    self.AddUserStory(Page4(self))
+    self.AddUserStory(Page5(self))
+    self.AddUserStory(Page6(self))
+    self.AddUserStory(Page7(self))