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Side by Side Diff: tools/perf/page_sets/webrtc_cases.py

Issue 962113003: Adding HD peerconnection WebRTC call 45-second performance test. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Adding page sets Created 5 years, 9 months ago
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1 # Copyright 2014 The Chromium Authors. All rights reserved. 1 # Copyright 2014 The Chromium Authors. All rights reserved.
2 # Use of this source code is governed by a BSD-style license that can be 2 # Use of this source code is governed by a BSD-style license that can be
3 # found in the LICENSE file. 3 # found in the LICENSE file.
4 import os 4 import os
5 5
6 from telemetry.page import page as page_module 6 from telemetry.page import page as page_module
7 from telemetry.page import page_set as page_set_module 7 from telemetry.page import page_set as page_set_module
8 8
9 9
10 WEBRTC_GITHUB_SAMPLES_URL = 'http://webrtc.github.io/samples/src/content/'
11
12
10 class WebrtcCasesPage(page_module.Page): 13 class WebrtcCasesPage(page_module.Page):
11 14
12 def __init__(self, url, page_set): 15 def __init__(self, url, page_set):
13 super(WebrtcCasesPage, self).__init__(url=url, page_set=page_set) 16 super(WebrtcCasesPage, self).__init__(url=url, page_set=page_set)
14 17
15 with open(os.path.join(os.path.dirname(__file__), 18 with open(os.path.join(os.path.dirname(__file__),
16 'webrtc_track_peerconnections.js')) as javascript: 19 'webrtc_track_peerconnections.js')) as javascript:
17 self.script_to_evaluate_on_commit = javascript.read() 20 self.script_to_evaluate_on_commit = javascript.read()
18 21
19 22
20 class Page1(WebrtcCasesPage): 23 class Page1(WebrtcCasesPage):
21 24
22 """ Why: Acquires a vga local stream. """ 25 """ Why: Acquires a vga local stream. """
23 26
24 def __init__(self, page_set): 27 def __init__(self, page_set):
25 super(Page1, self).__init__( 28 super(Page1, self).__init__(
26 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 29 url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/gum/',
27 'getusermedia/gum/'),
28 page_set=page_set) 30 page_set=page_set)
29 31
30 def RunPageInteractions(self, action_runner): 32 def RunPageInteractions(self, action_runner):
31 action_runner.Wait(10) 33 action_runner.Wait(10)
32 34
33 35
34 class Page2(WebrtcCasesPage): 36 class Page2(WebrtcCasesPage):
35 37
36 """ Why: Sets up a local WebRTC call. """ 38 """ Why: Sets up a local WebRTC call. """
37 39
38 def __init__(self, page_set): 40 def __init__(self, page_set):
39 super(Page2, self).__init__( 41 super(Page2, self).__init__(
40 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 42 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/pc1/',
41 'peerconnection/pc1/'),
42 page_set=page_set) 43 page_set=page_set)
43 44
44 def RunPageInteractions(self, action_runner): 45 def RunPageInteractions(self, action_runner):
45 action_runner.ClickElement('button[id="startButton"]') 46 action_runner.ClickElement('button[id="startButton"]')
46 action_runner.Wait(2) 47 action_runner.Wait(2)
47 action_runner.ClickElement('button[id="callButton"]') 48 action_runner.ClickElement('button[id="callButton"]')
48 action_runner.Wait(10) 49 action_runner.Wait(10)
49 action_runner.ClickElement('button[id="hangupButton"]') 50 action_runner.ClickElement('button[id="hangupButton"]')
50 51
51 52
52 class Page3(WebrtcCasesPage): 53 class Page3(WebrtcCasesPage):
53 54
54 """ Why: Acquires a high definition local stream. """ 55 """ Why: Acquires a high definition local stream. """
55 56
56 def __init__(self, page_set): 57 def __init__(self, page_set):
57 super(Page3, self).__init__( 58 super(Page3, self).__init__(
58 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 59 url=WEBRTC_GITHUB_SAMPLES_URL + 'getusermedia/resolution/',
59 'getusermedia/resolution/'),
60 page_set=page_set) 60 page_set=page_set)
61 61
62 def RunPageInteractions(self, action_runner): 62 def RunPageInteractions(self, action_runner):
63 action_runner.ClickElement('button[id="hd"]') 63 action_runner.ClickElement('button[id="hd"]')
64 action_runner.Wait(10) 64 action_runner.Wait(10)
65 65
66 66
67 class Page4(WebrtcCasesPage): 67 class Page4(WebrtcCasesPage):
68 68
69 """ Why: Sets up a WebRTC audio call with Opus. """ 69 """ Why: Sets up a WebRTC audio call with Opus. """
70 70
71 def __init__(self, page_set): 71 def __init__(self, page_set):
72 super(Page4, self).__init__( 72 super(Page4, self).__init__(
73 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 73 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=OPUS',
74 'peerconnection/audio/?codec=OPUS'),
75 page_set=page_set) 74 page_set=page_set)
76 75
77 def RunPageInteractions(self, action_runner): 76 def RunPageInteractions(self, action_runner):
78 action_runner.ExecuteJavaScript('codecSelector.value="OPUS";') 77 action_runner.ExecuteJavaScript('codecSelector.value="OPUS";')
79 action_runner.ClickElement('button[id="callButton"]') 78 action_runner.ClickElement('button[id="callButton"]')
80 action_runner.Wait(10) 79 action_runner.Wait(10)
81 80
82 81
83 class Page5(WebrtcCasesPage): 82 class Page5(WebrtcCasesPage):
84 83
85 """ Why: Sets up a WebRTC audio call with G722. """ 84 """ Why: Sets up a WebRTC audio call with G722. """
86 85
87 def __init__(self, page_set): 86 def __init__(self, page_set):
88 super(Page5, self).__init__( 87 super(Page5, self).__init__(
89 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 88 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=G722',
90 'peerconnection/audio/?codec=G722'),
91 page_set=page_set) 89 page_set=page_set)
92 90
93 def RunPageInteractions(self, action_runner): 91 def RunPageInteractions(self, action_runner):
94 action_runner.ExecuteJavaScript('codecSelector.value="G722";') 92 action_runner.ExecuteJavaScript('codecSelector.value="G722";')
95 action_runner.ClickElement('button[id="callButton"]') 93 action_runner.ClickElement('button[id="callButton"]')
96 action_runner.Wait(10) 94 action_runner.Wait(10)
97 95
98 96
99 class Page6(WebrtcCasesPage): 97 class Page6(WebrtcCasesPage):
100 98
101 """ Why: Sets up a WebRTC audio call with PCMU. """ 99 """ Why: Sets up a WebRTC audio call with PCMU. """
102 100
103 def __init__(self, page_set): 101 def __init__(self, page_set):
104 super(Page6, self).__init__( 102 super(Page6, self).__init__(
105 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 103 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=PCMU',
106 'peerconnection/audio/?codec=PCMU'),
107 page_set=page_set) 104 page_set=page_set)
108 105
109 def RunPageInteractions(self, action_runner): 106 def RunPageInteractions(self, action_runner):
110 action_runner.ExecuteJavaScript('codecSelector.value="PCMU";') 107 action_runner.ExecuteJavaScript('codecSelector.value="PCMU";')
111 action_runner.ClickElement('button[id="callButton"]') 108 action_runner.ClickElement('button[id="callButton"]')
112 action_runner.Wait(10) 109 action_runner.Wait(10)
113 110
114 111
115 class Page7(WebrtcCasesPage): 112 class Page7(WebrtcCasesPage):
116 113
117 """ Why: Sets up a WebRTC audio call with iSAC 16K. """ 114 """ Why: Sets up a WebRTC audio call with iSAC 16K. """
118 115
119 def __init__(self, page_set): 116 def __init__(self, page_set):
120 super(Page7, self).__init__( 117 super(Page7, self).__init__(
121 url=('http://googlechrome.github.io/webrtc/samples/web/content/' 118 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/audio/?codec=ISAC_16K',
122 'peerconnection/audio/?codec=ISAC_16K'),
123 page_set=page_set) 119 page_set=page_set)
124 120
125 def RunPageInteractions(self, action_runner): 121 def RunPageInteractions(self, action_runner):
126 action_runner.ExecuteJavaScript('codecSelector.value="ISAC/16000";') 122 action_runner.ExecuteJavaScript('codecSelector.value="ISAC/16000";')
127 action_runner.ClickElement('button[id="callButton"]') 123 action_runner.ClickElement('button[id="callButton"]')
128 action_runner.Wait(10) 124 action_runner.Wait(10)
129 125
130 126
127 class Page8(WebrtcCasesPage):
128
129 """ Why: Sets up a WebRTC 1080p call for 45 seconds. """
130
131 def __init__(self, page_set):
132 super(Page8, self).__init__(
133 url=WEBRTC_GITHUB_SAMPLES_URL + 'peerconnection/constraints/',
134 page_set=page_set)
135
136 def RunPageInteractions(self, action_runner):
137 action_runner.ExecuteJavaScript('minWidthInput.value = 1920')
138 action_runner.ExecuteJavaScript('maxWidthInput.value = 1920')
139 action_runner.ExecuteJavaScript('minHeightInput.value = 1080')
140 action_runner.ExecuteJavaScript('maxHeightInput.value = 1080')
141 action_runner.ClickElement('button[id="getMedia"]')
142 action_runner.Wait(2)
143 action_runner.ClickElement('button[id="connect"]')
144 action_runner.Wait(45)
145
131 class WebrtcCasesPageSet(page_set_module.PageSet): 146 class WebrtcCasesPageSet(page_set_module.PageSet):
132 147
133 """ WebRTC tests for Real-time audio and video communication. """ 148 """ WebRTC tests for Real-time audio and video communication. """
134 149
135 def __init__(self): 150 def __init__(self):
136 super(WebrtcCasesPageSet, self).__init__( 151 super(WebrtcCasesPageSet, self).__init__(
137 archive_data_file='data/webrtc_cases.json', 152 archive_data_file='data/webrtc_cases.json',
138 bucket=page_set_module.PUBLIC_BUCKET) 153 bucket=page_set_module.PUBLIC_BUCKET)
139 154
140 self.AddUserStory(Page1(self)) 155 self.AddUserStory(Page1(self))
141 self.AddUserStory(Page2(self)) 156 self.AddUserStory(Page2(self))
142 self.AddUserStory(Page3(self)) 157 self.AddUserStory(Page3(self))
143 self.AddUserStory(Page1(self)) 158 self.AddUserStory(Page1(self))
144 self.AddUserStory(Page2(self)) 159 self.AddUserStory(Page2(self))
145 self.AddUserStory(Page3(self)) 160 self.AddUserStory(Page3(self))
146 self.AddUserStory(Page4(self)) 161 self.AddUserStory(Page4(self))
147 self.AddUserStory(Page5(self)) 162 self.AddUserStory(Page5(self))
148 self.AddUserStory(Page6(self)) 163 self.AddUserStory(Page6(self))
149 self.AddUserStory(Page7(self)) 164 self.AddUserStory(Page7(self))
165 self.AddUserStory(Page8(self))
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