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Unified Diff: tools/perf/page_sets/webrtc_cases.py

Issue 1873503002: Telemetry: Added a new page set to webrtc.webrtc_smoothness (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Disabled reference run on perf cq bots Created 4 years, 8 months ago
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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..f157d576047d6c108272aa1caf33868fa586720d 100644
--- a/tools/perf/page_sets/webrtc_cases.py
+++ b/tools/perf/page_sets/webrtc_cases.py
@@ -3,11 +3,12 @@
# 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/'
+MEDIARECORDER_GITHUB_URL = 'https://rawgit.com/cricdecyan/mediarecorder/master/'
class WebrtcPage(page_module.Page):
@@ -22,13 +23,13 @@ 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 +37,7 @@ 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 +59,12 @@ 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 +75,14 @@ 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 +91,13 @@ 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 +106,13 @@ 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 +121,13 @@ 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 +135,78 @@ 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):
+ 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))
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