Telemetry: Added a new page set to webrtc.webrtc_smoothness

tools/perf/page_sets/webrtc_cases.py

 # Copyright 2014 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.
 import os
 
+from telemetry import story
 from telemetry.page import page as page_module
-from telemetry import story
 
 
 WEBRTC_GITHUB_SAMPLES_URL = 'https://webrtc.github.io/samples/src/content/'
+MEDIARECORDER_GITHUB_URL = 'https://rawgit.com/cricdecyan/mediarecorder/master/'
 
 
 class WebrtcPage(page_module.Page):
 
   def __init__(self, url, page_set, name):
     super(WebrtcPage, self).__init__(
         url=url, page_set=page_set, name=name)
 
     with open(os.path.join(os.path.dirname(__file__),
                            'webrtc_track_peerconnections.js')) as javascript:
       self.script_to_evaluate_on_commit = javascript.read()
 
 
 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"]')
     action_runner.Wait(10)
 
 
 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__(
         url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/constraints/',
         name='720p_call_45s',
         page_set=page_set)
 
   def RunPageInteractions(self, action_runner):
     with action_runner.CreateInteraction('Action_Create_PeerConnection',
                                          repeatable=False):
       action_runner.ExecuteJavaScript('minWidthInput.value = 1280')
       action_runner.ExecuteJavaScript('maxWidthInput.value = 1280')
       action_runner.ExecuteJavaScript('minHeightInput.value = 720')
       action_runner.ExecuteJavaScript('maxHeightInput.value = 720')
       action_runner.ClickElement('button[id="getMedia"]')
       action_runner.Wait(2)
       action_runner.ClickElement('button[id="connect"]')
       action_runner.Wait(45)
 
 
 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):
     # It won't have time to finish the 512 MB, but we're only interested in
     # cpu + memory anyway rather than how much data we manage to transfer.
     action_runner.ExecuteJavaScript('megsToSend.value = 512;')
     action_runner.ClickElement('button[id="sendTheData"]')
     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";')
     action_runner.ClickElement('button[id="callButton"]')
     action_runner.Wait(10)
 
 
 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";')
     action_runner.ClickElement('button[id="callButton"]')
     action_runner.Wait(10)
 
 
 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";')
     action_runner.ClickElement('button[id="callButton"]')
     action_runner.Wait(10)
 
 
 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";')
     action_runner.ClickElement('button[id="callButton"]')
     action_runner.Wait(10)
 
 
+class Page8(WebrtcPage):
+  """Why: Sets up a canvas capture stream connection to a peer connection."""
+
+  def __init__(self, page_set):
+    canvas_capure_html = 'canvascapture/canvas_capture_peerconnection.html'
+    super(Page8, self).__init__(
+        url=MEDIARECORDER_GITHUB_URL + canvas_capure_html,
+        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))