| Index: content/browser/media/webrtc/webrtc_browsertest_audio.cc
|
| diff --git a/content/browser/media/webrtc/webrtc_browsertest_audio.cc b/content/browser/media/webrtc/webrtc_browsertest_audio.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..7b6557ddc88fc871f83b353a11c9e23038fa8c3d
|
| --- /dev/null
|
| +++ b/content/browser/media/webrtc/webrtc_browsertest_audio.cc
|
| @@ -0,0 +1,120 @@
|
| +// 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/threading/platform_thread.h"
|
| +#include "build/build_config.h"
|
| +#include "content/browser/web_contents/web_contents_impl.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 "content/test/webrtc_content_browsertest_base.h"
|
| +#include "media/audio/audio_manager.h"
|
| +#include "media/base/media_switches.h"
|
| +#include "net/test/embedded_test_server/embedded_test_server.h"
|
| +
|
| +namespace content {
|
| +
|
| +#if defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
|
| +// Renderer crashes under Android ASAN: https://crbug.com/408496.
|
| +#define MAYBE_WebRtcBrowserAudioTest DISABLED_WebRtcBrowserAudioTest
|
| +#else
|
| +#define MAYBE_WebRtcBrowserAudioTest WebRtcBrowserAudioTest
|
| +#endif
|
| +
|
| +// This class tests the scenario when permission to access mic or camera is
|
| +// granted.
|
| +class MAYBE_WebRtcBrowserAudioTest : public WebRtcContentBrowserTest {
|
| + public:
|
| + MAYBE_WebRtcBrowserAudioTest() {}
|
| + ~MAYBE_WebRtcBrowserAudioTest() override {}
|
| +
|
| + void SetUpCommandLine(base::CommandLine* command_line) override {
|
| + WebRtcContentBrowserTest::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 (!media::AudioManager::Get()->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");
|
| + }
|
| +};
|
| +
|
| +// Causes asserts in libjingle: http://crbug.com/484826.
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + DISABLED_CanMakeVideoCallAndThenRenegotiateToAudio) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndRenegotiateToAudio({audio: true, video:true}, {audio: true});");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishAudioVideoCallAndEnsureAudioIsPlaying) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureAudioIsPlaying({audio:true, video:true});");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishAudioOnlyCallAndEnsureAudioIsPlaying) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureAudioIsPlaying({audio:true});");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishIsac16KCallAndEnsureAudioIsPlaying) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callWithIsac16KAndEnsureAudioIsPlaying({audio:true});");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishAudioVideoCallAndVerifyRemoteMutingWorks) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureRemoteAudioTrackMutingWorks();");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishAudioVideoCallAndVerifyLocalMutingWorks) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureLocalAudioTrackMutingWorks();");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EnsureLocalVideoMuteDoesntMuteAudio) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureLocalVideoMutingDoesntMuteAudio();");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EnsureRemoteVideoMuteDoesntMuteAudio) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureRemoteVideoMutingDoesntMuteAudio();");
|
| +}
|
| +
|
| +IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserAudioTest,
|
| + EstablishAudioVideoCallAndVerifyUnmutingWorks) {
|
| + MakeAudioDetectingPeerConnectionCall(
|
| + "callAndEnsureAudioTrackUnmutingWorks();");
|
| +}
|
| +
|
| +} // namespace content
|
|
|