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 9ea37d81c91878ba5bce15206ae4cf834c80b72c..5d8f3f58a9d808b6fde59bdeb81f3e46e0293aa4 100644 |
--- a/tools/perf/page_sets/webrtc_cases.py |
+++ b/tools/perf/page_sets/webrtc_cases.py |
@@ -7,14 +7,10 @@ from telemetry import story |
from telemetry.page import page as page_module |
-WEBRTC_TEST_PAGES_URL = 'https://test.webrtc.org/manual/' |
-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): |
+ assert url.startswith('file://webrtc_cases/') |
super(WebrtcPage, self).__init__( |
url=url, page_set=page_set, name=name) |
@@ -28,7 +24,7 @@ class Page1(WebrtcPage): |
def __init__(self, page_set): |
super(Page1, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/resolution/', |
+ url='file://webrtc_cases/resolution.html', |
name='hd_local_stream_10s', |
page_set=page_set) |
@@ -42,7 +38,7 @@ class Page2(WebrtcPage): |
def __init__(self, page_set): |
super(Page2, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/constraints/', |
+ url='file://webrtc_cases/constraints.html', |
name='720p_call_45s', |
page_set=page_set) |
@@ -64,7 +60,7 @@ class Page3(WebrtcPage): |
def __init__(self, page_set): |
super(Page3, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'datachannel/datatransfer', |
+ url='file://webrtc_cases/datatransfer.html', |
name='30s_datachannel_transfer', |
page_set=page_set) |
@@ -81,7 +77,7 @@ class Page4(WebrtcPage): |
def __init__(self, page_set): |
super(Page4, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=OPUS', |
+ url='file://webrtc_cases/audio.html?codec=OPUS', |
name='audio_call_opus_10s', |
page_set=page_set) |
@@ -96,7 +92,7 @@ class Page5(WebrtcPage): |
def __init__(self, page_set): |
super(Page5, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=G722', |
+ url='file://webrtc_cases/audio.html?codec=G722', |
name='audio_call_g722_10s', |
page_set=page_set) |
@@ -111,7 +107,7 @@ class Page6(WebrtcPage): |
def __init__(self, page_set): |
super(Page6, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=PCMU', |
+ url='file://webrtc_cases/audio.html?codec=PCMU', |
name='audio_call_pcmu_10s', |
page_set=page_set) |
@@ -126,7 +122,7 @@ class Page7(WebrtcPage): |
def __init__(self, page_set): |
super(Page7, self).__init__( |
- url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=ISAC_16K', |
+ url='file://webrtc_cases/audio.html?codec=ISAC_16K', |
name='audio_call_isac16k_10s', |
page_set=page_set) |
@@ -140,18 +136,15 @@ 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, |
+ url='file://webrtc_cases/canvas-capture.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.WaitForJavaScriptCondition('typeof draw !== "undefined"') |
- action_runner.ExecuteJavaScript('draw();') |
- action_runner.ExecuteJavaScript('doCanvasCaptureAndPeerConnection();') |
+ action_runner.ClickElement('button[id="startButton"]') |
action_runner.Wait(10) |
@@ -160,7 +153,7 @@ class Page9(WebrtcPage): |
def __init__(self, page_set): |
super(Page9, self).__init__( |
- url= WEBRTC_TEST_PAGES_URL + 'multiple-peerconnections/', |
+ url='file://webrtc_cases/multiple-peerconnections.html', |
name='multiple_peerconnections', |
page_set=page_set) |
@@ -181,7 +174,6 @@ class WebrtcGetusermediaPageSet(story.StorySet): |
def __init__(self): |
super(WebrtcGetusermediaPageSet, self).__init__( |
- archive_data_file='data/webrtc_getusermedia_cases.json', |
cloud_storage_bucket=story.PUBLIC_BUCKET) |
self.AddStory(Page1(self)) |
@@ -192,7 +184,6 @@ class WebrtcStresstestPageSet(story.StorySet): |
def __init__(self): |
super(WebrtcStresstestPageSet, self).__init__( |
- archive_data_file='data/webrtc_stresstest_cases.json', |
cloud_storage_bucket=story.PUBLIC_BUCKET) |
self.AddStory(Page9(self)) |
@@ -203,7 +194,6 @@ class WebrtcPeerconnectionPageSet(story.StorySet): |
def __init__(self): |
super(WebrtcPeerconnectionPageSet, self).__init__( |
- archive_data_file='data/webrtc_peerconnection_cases.json', |
cloud_storage_bucket=story.PUBLIC_BUCKET) |
self.AddStory(Page2(self)) |
@@ -214,7 +204,6 @@ class WebrtcDatachannelPageSet(story.StorySet): |
def __init__(self): |
super(WebrtcDatachannelPageSet, self).__init__( |
- archive_data_file='data/webrtc_datachannel_cases.json', |
cloud_storage_bucket=story.PUBLIC_BUCKET) |
self.AddStory(Page3(self)) |
@@ -225,7 +214,6 @@ class WebrtcAudioPageSet(story.StorySet): |
def __init__(self): |
super(WebrtcAudioPageSet, self).__init__( |
- archive_data_file='data/webrtc_audio_cases.json', |
cloud_storage_bucket=story.PUBLIC_BUCKET) |
self.AddStory(Page4(self)) |
@@ -239,7 +227,6 @@ class WebrtcRenderingPageSet(story.StorySet): |
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)) |