Telemetry: Added a new page set to webrtc.webrtc_smoothness

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 99f5eadae8f2abc0085d46298d1cc734fb772c5a..60f54c62d9d1b6c3d90d4ffab398e2e703f69b2c 100644
--- a/tools/perf/page_sets/webrtc_cases.py
+++ b/tools/perf/page_sets/webrtc_cases.py
@@ -3,11 +3,13 @@
 # found in the LICENSE file.
 import os
 
-from telemetry.page import page as page_module
 from telemetry import story
+from telemetry.page import page as page_module
 
 
 WEBRTC_GITHUB_SAMPLES_URL = 'https://webrtc.github.io/samples/src/content/'
+CANVAS_CAPTURE_URL = ('https://rawgit.com/cricdecyan/mediarecorder/master/'

[phoglund_chromium, 2016/04/08 11:50:22]

Maybe call it MEDIARECORDER_GIT_URL, and assign it
'https://rawgit.com/cricdecyan/mediarecorder/master/'. You can add the rest of
the path in Page8 similar to how the other pages do it.

[cpaulin (no longer in chrome), 2016/04/08 18:01:38]

Done.

+                      'canvascapture/canvas_capture_peerconnection.html')
 
 
 class WebrtcPage(page_module.Page):
@@ -22,13 +24,13 @@ class WebrtcPage(page_module.Page):
 
 
 class Page1(WebrtcPage):
-  """ Why: Acquires a high definition (720p) local stream. """
+  """Why: Acquires a high definition (720p) local stream."""
 
   def __init__(self, page_set):
     super(Page1, self).__init__(
-      url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/resolution/',
-      name='hd_local_stream_10s',
-      page_set=page_set)
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/resolution/',
+        name='hd_local_stream_10s',
+        page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     action_runner.ClickElement('button[id="hd"]')
@@ -36,7 +38,7 @@ class Page1(WebrtcPage):
 
 
 class Page2(WebrtcPage):
-  """ Why: Sets up a local video-only WebRTC 720p call for 45 seconds. """
+  """Why: Sets up a local video-only WebRTC 720p call for 45 seconds."""
 
   def __init__(self, page_set):
     super(Page2, self).__init__(
@@ -58,12 +60,12 @@ class Page2(WebrtcPage):
 
 
 class Page3(WebrtcPage):
-  """ Why: Transfer as much data as possible through a data channel in 20s. """
+  """Why: Transfer as much data as possible through a data channel in 20s."""
 
   def __init__(self, page_set):
     super(Page3, self).__init__(
         url=WEBRTC_GITHUB_SAMPLES_URL + 'datachannel/datatransfer',
-        name="30s_datachannel_transfer",
+        name='30s_datachannel_transfer',
         page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
@@ -74,15 +76,14 @@ class Page3(WebrtcPage):
     action_runner.Wait(30)
 
 
-
 class Page4(WebrtcPage):
-  """ Why: Sets up a WebRTC audio call with Opus. """
+  """Why: Sets up a WebRTC audio call with Opus."""
 
   def __init__(self, page_set):
     super(Page4, self).__init__(
-      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=OPUS',
-      name='audio_call_opus_10s',
-      page_set=page_set)
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=OPUS',
+        name='audio_call_opus_10s',
+        page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     action_runner.ExecuteJavaScript('codecSelector.value="OPUS";')
@@ -91,13 +92,13 @@ class Page4(WebrtcPage):
 
 
 class Page5(WebrtcPage):
-  """ Why: Sets up a WebRTC audio call with G722. """
+  """Why: Sets up a WebRTC audio call with G722."""
 
   def __init__(self, page_set):
     super(Page5, self).__init__(
-      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=G722',
-      name='audio_call_g722_10s',
-      page_set=page_set)
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=G722',
+        name='audio_call_g722_10s',
+        page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     action_runner.ExecuteJavaScript('codecSelector.value="G722";')
@@ -106,13 +107,13 @@ class Page5(WebrtcPage):
 
 
 class Page6(WebrtcPage):
-  """ Why: Sets up a WebRTC audio call with PCMU. """
+  """Why: Sets up a WebRTC audio call with PCMU."""
 
   def __init__(self, page_set):
     super(Page6, self).__init__(
-      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=PCMU',
-      name='audio_call_pcmu_10s',
-      page_set=page_set)
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=PCMU',
+        name='audio_call_pcmu_10s',
+        page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     action_runner.ExecuteJavaScript('codecSelector.value="PCMU";')
@@ -121,13 +122,13 @@ class Page6(WebrtcPage):
 
 
 class Page7(WebrtcPage):
-  """ Why: Sets up a WebRTC audio call with iSAC 16K. """
+  """Why: Sets up a WebRTC audio call with iSAC 16K."""
 
   def __init__(self, page_set):
     super(Page7, self).__init__(
-      url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=ISAC_16K',
-      name='audio_call_isac16k_10s',
-      page_set=page_set)
+        url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=ISAC_16K',
+        name='audio_call_isac16k_10s',
+        page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     action_runner.ExecuteJavaScript('codecSelector.value="ISAC/16000";')
@@ -135,47 +136,77 @@ class Page7(WebrtcPage):
     action_runner.Wait(10)
 
 
+class Page8(WebrtcPage):
+  """Why: Sets up a canvas capture stream connection to a peer connection."""
+
+  def __init__(self, page_set):
+    super(Page8, self).__init__(
+        url=CANVAS_CAPTURE_URL,
+        name='canvas_capture_peer_connection',
+        page_set=page_set)
+
+  def RunPageInteractions(self, action_runner):
+    with action_runner.CreateInteraction('Action_Canvas_PeerConnection',
+                                         repeatable=False):
+      action_runner.ExecuteJavaScript('draw();')
+      action_runner.ExecuteJavaScript('doCanvasCaptureAndPeerConnection();')
+      action_runner.Wait(10)
+
+
 class WebrtcGetusermediaPageSet(story.StorySet):
-  """ WebRTC tests for local getUserMedia: video capture and playback. """
+  """WebRTC tests for local getUserMedia: video capture and playback."""
 
   def __init__(self):
     super(WebrtcGetusermediaPageSet, self).__init__(
-      archive_data_file='data/webrtc_getusermedia_cases.json',
-      cloud_storage_bucket=story.PUBLIC_BUCKET)
+        archive_data_file='data/webrtc_getusermedia_cases.json',
+        cloud_storage_bucket=story.PUBLIC_BUCKET)
 
     self.AddStory(Page1(self))
 
 
 class WebrtcPeerconnectionPageSet(story.StorySet):
-  """ WebRTC tests for Real-time video and audio communication. """
+  """WebRTC tests for Real-time video and audio communication."""
 
   def __init__(self):
     super(WebrtcPeerconnectionPageSet, self).__init__(
-      archive_data_file='data/webrtc_peerconnection_cases.json',
-      cloud_storage_bucket=story.PUBLIC_BUCKET)
+        archive_data_file='data/webrtc_peerconnection_cases.json',
+        cloud_storage_bucket=story.PUBLIC_BUCKET)
 
     self.AddStory(Page2(self))
 
 
 class WebrtcDatachannelPageSet(story.StorySet):
-  """ WebRTC tests for Real-time communication via the data channel. """
+  """WebRTC tests for Real-time communication via the data channel."""
 
   def __init__(self):
     super(WebrtcDatachannelPageSet, self).__init__(
-      archive_data_file='data/webrtc_datachannel_cases.json',
-      cloud_storage_bucket=story.PUBLIC_BUCKET)
+        archive_data_file='data/webrtc_datachannel_cases.json',
+        cloud_storage_bucket=story.PUBLIC_BUCKET)
 
     self.AddStory(Page3(self))
 
+
 class WebrtcAudioPageSet(story.StorySet):
-  """ WebRTC tests for Real-time audio communication. """
+  """WebRTC tests for Real-time audio communication."""
 
   def __init__(self):
     super(WebrtcAudioPageSet, self).__init__(
-      archive_data_file='data/webrtc_audio_cases.json',
-      cloud_storage_bucket=story.PUBLIC_BUCKET)
+        archive_data_file='data/webrtc_audio_cases.json',
+        cloud_storage_bucket=story.PUBLIC_BUCKET)
 
     self.AddStory(Page4(self))
     self.AddStory(Page5(self))
     self.AddStory(Page6(self))
     self.AddStory(Page7(self))
+
+
+class WebrtcRenderingPageSet(story.StorySet):
+  """WebRTC tests for video rendering."""
+
+  def __init__(self):
+    super(WebrtcRenderingPageSet, self).__init__(
+        archive_data_file='data/webrtc_smoothness_cases.json',
+        cloud_storage_bucket=story.PARTNER_BUCKET)
+
+    self.AddStory(Page2(self))
+    self.AddStory(Page8(self))