1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
|
// Copyright (c) 2016 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.
#include "base/command_line.h"
#include "base/files/file_util.h"
#include "base/test/scoped_feature_list.h"
#include "base/threading/platform_thread.h"
#include "build/build_config.h"
#include "content/browser/web_contents/web_contents_impl.h"
#include "content/browser/webrtc/webrtc_content_browsertest_base.h"
#include "content/public/common/content_features.h"
#include "content/public/common/content_switches.h"
#include "content/public/common/webrtc_ip_handling_policy.h"
#include "content/public/test/browser_test_utils.h"
#include "content/public/test/content_browser_test_utils.h"
#include "content/public/test/test_utils.h"
#include "media/base/media_switches.h"
#include "net/test/embedded_test_server/embedded_test_server.h"
#include "testing/gtest/include/gtest/gtest-param-test.h"
namespace content {
// This class tests the scenario when permission to access mic or camera is
// granted.
class WebRtcAudioBrowserTest : public WebRtcContentBrowserTestBase,
public testing::WithParamInterface<bool> {
public:
WebRtcAudioBrowserTest() {
std::vector<base::Feature> audio_service_oop_features = {
features::kAudioServiceAudioStreams,
features::kAudioServiceOutOfProcess};
if (GetParam()) {
// Force audio service out of process to enabled.
audio_service_features_.InitWithFeatures(audio_service_oop_features, {});
} else {
// Force audio service out of process to disabled.
audio_service_features_.InitWithFeatures({}, audio_service_oop_features);
}
}
~WebRtcAudioBrowserTest() override {}
void SetUpCommandLine(base::CommandLine* command_line) override {
WebRtcContentBrowserTestBase::SetUpCommandLine(command_line);
// Automatically grant device permission.
AppendUseFakeUIForMediaStreamFlag();
}
protected:
// Convenience method for making calls that detect if audio os playing (which
// has some special prerequisites, such that there needs to be an audio output
// device on the executing machine).
void MakeAudioDetectingPeerConnectionCall(const std::string& javascript) {
if (!HasAudioOutputDevices()) {
// Bots with no output devices will force the audio code into a state
// where it doesn't manage to set either the low or high latency path.
// This test will compute useless values in that case, so skip running on
// such bots (see crbug.com/326338).
LOG(INFO) << "Missing output devices: skipping test...";
return;
}
ASSERT_TRUE(base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kUseFakeDeviceForMediaStream))
<< "Must run with fake devices since the test will explicitly look "
<< "for the fake device signal.";
MakeTypicalCall(javascript, "/media/peerconnection-call-audio.html");
}
private:
base::test::ScopedFeatureList audio_service_features_;
};
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
CanMakeVideoCallAndThenRenegotiateToAudio) {
MakeAudioDetectingPeerConnectionCall(
"callAndRenegotiateToAudio({audio: true, video:true}, {audio: true});");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishAudioVideoCallAndEnsureAudioIsPlaying) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureAudioIsPlaying({audio:true, video:true});");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishAudioOnlyCallAndEnsureAudioIsPlaying) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureAudioIsPlaying({audio:true});");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishIsac16KCallAndEnsureAudioIsPlaying) {
MakeAudioDetectingPeerConnectionCall(
"callWithIsac16KAndEnsureAudioIsPlaying({audio:true});");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishAudioVideoCallAndVerifyRemoteMutingWorks) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureRemoteAudioTrackMutingWorks();");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishAudioVideoCallAndVerifyLocalMutingWorks) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureLocalAudioTrackMutingWorks();");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EnsureLocalVideoMuteDoesntMuteAudio) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureLocalVideoMutingDoesntMuteAudio();");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EnsureRemoteVideoMuteDoesntMuteAudio) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureRemoteVideoMutingDoesntMuteAudio();");
}
IN_PROC_BROWSER_TEST_P(WebRtcAudioBrowserTest,
EstablishAudioVideoCallAndVerifyUnmutingWorks) {
MakeAudioDetectingPeerConnectionCall(
"callAndEnsureAudioTrackUnmutingWorks();");
}
// We run these tests with the audio service both in and out of the the browser
// process to have waterfall coverage while the feature rolls out. It should be
// removed after launch.
#if defined(OS_LINUX) || defined(OS_MACOSX) || defined(OS_WIN)
// Supported platforms.
INSTANTIATE_TEST_CASE_P(, WebRtcAudioBrowserTest, ::testing::Bool());
#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
// Renderer crashes under Android ASAN: https://crbug.com/408496.
#else
// Platforms where the out of process audio service isn't supported
INSTANTIATE_TEST_CASE_P(, WebRtcAudioBrowserTest, ::testing::Values(false));
#endif
} // namespace content
|
