Telemetry: Added a new page set to webrtc.webrtc_smoothness

tools/perf/benchmarks/webrtc.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.
 
 from core import perf_benchmark
 
 from measurements import webrtc
 import page_sets
 from telemetry import benchmark
 from telemetry.timeline import tracing_category_filter
 from telemetry.web_perf import timeline_based_measurement
 from telemetry.web_perf.metrics import webrtc_rendering_timeline
 
 RENDERING_VALUE_PREFIX = 'WebRTCRendering_'
 
 # TODO(qyearsley, mcasas): Add webrtc.audio when http://crbug.com/468732
 # is fixed, or revert https://codereview.chromium.org/1544573002/ when
 # http://crbug.com/568333 is fixed.
 
+
 # Disabled because the reference set becomes flaky with the new
 # https:// page set introduced in http://crbug.com/523517.
 # Try removing once the Chrome used for ref builds advances
 # past blink commit pos 200986.
 @benchmark.Disabled('reference')
 class _Webrtc(perf_benchmark.PerfBenchmark):
   """Base class for WebRTC metrics for real-time communications tests."""
   test = webrtc.WebRTC
 
 
 class WebrtcGetusermedia(_Webrtc):
-  """Measures WebRtc GetUserMedia for video capture and local playback"""
+  """Measures WebRtc GetUserMedia for video capture and local playback."""
   page_set = page_sets.WebrtcGetusermediaPageSet
 
   @classmethod
   def Name(cls):
     return 'webrtc.getusermedia'
 
 
 class WebrtcPeerConnection(_Webrtc):
-  """Measures WebRtc Peerconnection for remote video and audio communication """
+  """Measures WebRtc Peerconnection for remote video and audio communication."""
   page_set = page_sets.WebrtcPeerconnectionPageSet
 
   @classmethod
   def Name(cls):
     return 'webrtc.peerconnection'
 
 
 class WebrtcDataChannel(_Webrtc):
-  """Measures WebRtc DataChannel loopback """
+  """Measures WebRtc DataChannel loopback."""
   page_set = page_sets.WebrtcDatachannelPageSet
 
   @classmethod
   def Name(cls):
     return 'webrtc.datachannel'
 
 
 # WebrtcRendering must be a PerfBenchmark, and not a _Webrtc, because it is a
 # timeline-based.
+# Disabled on reference builds because they crash and don't support tab
+# capture. See http://crbug.com/603232.
+@benchmark.Disabled('reference')
 class WebrtcRendering(perf_benchmark.PerfBenchmark):
   """Specific time measurements (e.g. fps, smoothness) for WebRtc rendering."""
 
-  page_set = page_sets.WebrtcPeerconnectionPageSet
+  page_set = page_sets.WebrtcRenderingPageSet
 
   def CreateTimelineBasedMeasurementOptions(self):
     category_filter = tracing_category_filter.TracingCategoryFilter(
         filter_string='webrtc,webkit.console,blink.console')
     options = timeline_based_measurement.Options(category_filter)
     options.SetLegacyTimelineBasedMetrics(
         [webrtc_rendering_timeline.WebRtcRenderingTimelineMetric()])
     return options
 
   def SetExtraBrowserOptions(self, options):
     options.AppendExtraBrowserArgs('--use-fake-device-for-media-stream')
     options.AppendExtraBrowserArgs('--use-fake-ui-for-media-stream')
 
   @classmethod
   def Name(cls):
     return 'webrtc.webrtc_smoothness'