content/browser/media/webrtc_browsertest.cc

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "base/command_line.h"
   6 #include "base/strings/stringprintf.h"
   7 #include "base/values.h"
   8 #include "content/browser/media/webrtc_internals.h"
   9 #include "content/browser/web_contents/web_contents_impl.h"
  10 #include "content/public/common/content_switches.h"
  11 #include "content/public/test/browser_test_utils.h"
  12 #include "content/public/test/content_browser_test_utils.h"
  13 #include "content/public/test/test_utils.h"
  14 #include "content/shell/browser/shell.h"
  15 #include "content/test/webrtc_content_browsertest_base.h"
  16 #include "media/audio/audio_manager.h"
  17 #include "net/test/embedded_test_server/embedded_test_server.h"
  18
  19 #if defined(OS_WIN)
  20 #include "base/win/windows_version.h"
  21 #endif
  22
  23 namespace content {
  24
  25 class WebRtcBrowserTest : public WebRtcContentBrowserTest,
  26                           public testing::WithParamInterface<bool> {
  27  public:
  28   WebRtcBrowserTest() {}
  29   virtual ~WebRtcBrowserTest() {}
  30
  31   virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE {
  32     WebRtcContentBrowserTest::SetUpCommandLine(command_line);
  33
  34     bool enable_audio_track_processing = GetParam();
  35     if (enable_audio_track_processing)
  36       command_line->AppendSwitch(switches::kEnableAudioTrackProcessing);
  37   }
  38
  39   // Convenience function since most peerconnection-call.html tests just load
  40   // the page, kick off some javascript and wait for the title to change to OK.
  41   void MakeTypicalPeerConnectionCall(const std::string& javascript) {
  42     ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
  43
  44     GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
  45     NavigateToURL(shell(), url);
  46
  47     DisableOpusIfOnAndroid();
  48     ExecuteJavascriptAndWaitForOk(javascript);
  49   }
  50
  51   void DisableOpusIfOnAndroid() {
  52 #if defined (OS_ANDROID)
  53     // Always force iSAC 16K on Android for now (Opus is broken).
  54     EXPECT_EQ("isac-forced",
  55               ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
  56 #endif
  57   }
  58 };
  59
  60 static const bool kRunTestsWithFlag[] = { false, true };
  61 INSTANTIATE_TEST_CASE_P(WebRtcBrowserTests,
  62                         WebRtcBrowserTest,
  63                         testing::ValuesIn(kRunTestsWithFlag));
  64
  65 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
  66 // Timing out on ARM linux bot: http://crbug.com/238490
  67 #define MAYBE_CanSetupDefaultVideoCall DISABLED_CanSetupDefaultVideoCall
  68 #else
  69 #define MAYBE_CanSetupDefaultVideoCall CanSetupDefaultVideoCall
  70 #endif
  71
  72 // These tests will make a complete PeerConnection-based call and verify that
  73 // video is playing for the call.
  74 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupDefaultVideoCall) {
  75   MakeTypicalPeerConnectionCall(
  76       "callAndExpectResolution({video: true}, 640, 480);");
  77 }
  78
  79 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CanSetupVideoCallWith1To1AspecRatio) {
  80   const std::string javascript =
  81       "callAndExpectResolution({video: {mandatory: {minWidth: 320,"
  82       " maxWidth: 320, minHeight: 320, maxHeight: 320}}}, 320, 320);";
  83   MakeTypicalPeerConnectionCall(javascript);
  84 }
  85
  86 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
  87 // Timing out on ARM linux, see http://crbug.com/240376
  88 #define MAYBE_CanSetupAudioAndVideoCall DISABLED_CanSetupAudioAndVideoCall
  89 #else
  90 #define MAYBE_CanSetupAudioAndVideoCall CanSetupAudioAndVideoCall
  91 #endif
  92
  93 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) {
  94   MakeTypicalPeerConnectionCall("call({video: true, audio: true});");
  95 }
  96
  97 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) {
  98   MakeTypicalPeerConnectionCall("callAndSendDtmf(\'123,abc\');");
  99 }
 100
 101 // TODO(phoglund): this test fails because the peer connection state will be
 102 // stable in the second negotiation round rather than have-local-offer.
 103 // http://crbug.com/293125.
 104 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 105                        DISABLED_CanMakeEmptyCallThenAddStreamsAndRenegotiate) {
 106   const char* kJavascript =
 107       "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});";
 108   MakeTypicalPeerConnectionCall(kJavascript);
 109 }
 110
 111 // Below 2 test will make a complete PeerConnection-based call between pc1 and
 112 // pc2, and then use the remote stream to setup a call between pc3 and pc4, and
 113 // then verify that video is received on pc3 and pc4.
 114 // Flaky on win xp. http://crbug.com/304775
 115 #if defined(OS_WIN)
 116 #define MAYBE_CanForwardRemoteStream DISABLED_CanForwardRemoteStream
 117 #define MAYBE_CanForwardRemoteStream720p DISABLED_CanForwardRemoteStream720p
 118 #else
 119 #define MAYBE_CanForwardRemoteStream CanForwardRemoteStream
 120 #define MAYBE_CanForwardRemoteStream720p CanForwardRemoteStream720p
 121 #endif
 122 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
 123   MakeTypicalPeerConnectionCall(
 124       "callAndForwardRemoteStream({video: true, audio: false});");
 125 }
 126
 127 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
 128   const std::string javascript = GenerateGetUserMediaCall(
 129       "callAndForwardRemoteStream", 1280, 1280, 720, 720, 30, 30);
 130   MakeTypicalPeerConnectionCall(javascript);
 131 }
 132
 133 // This test will make a complete PeerConnection-based call but remove the
 134 // MSID and bundle attribute from the initial offer to verify that
 135 // video is playing for the call even if the initiating client don't support
 136 // MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02
 137 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 138 // Timing out on ARM linux, see http://crbug.com/240373
 139 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
 140         DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle
 141 #else
 142 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
 143         CanSetupAudioAndVideoCallWithoutMsidAndBundle
 144 #endif
 145 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 146                        MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle) {
 147   MakeTypicalPeerConnectionCall("callWithoutMsidAndBundle();");
 148 }
 149
 150 // This test will modify the SDP offer to an unsupported codec, which should
 151 // cause SetLocalDescription to fail.
 152 #if defined(USE_OZONE)
 153 // Disabled for Ozone, see http://crbug.com/315392#c15
 154 #define MAYBE_NegotiateUnsupportedVideoCodec\
 155     DISABLED_NegotiateUnsupportedVideoCodec
 156 #else
 157 #define MAYBE_NegotiateUnsupportedVideoCodec NegotiateUnsupportedVideoCodec
 158 #endif
 159
 160 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 161                        MAYBE_NegotiateUnsupportedVideoCodec) {
 162   MakeTypicalPeerConnectionCall("negotiateUnsupportedVideoCodec();");
 163 }
 164
 165 // This test will modify the SDP offer to use no encryption, which should
 166 // cause SetLocalDescription to fail.
 167 #if defined(USE_OZONE)
 168 // Disabled for Ozone, see http://crbug.com/315392#c15
 169 #define MAYBE_NegotiateNonCryptoCall DISABLED_NegotiateNonCryptoCall
 170 #else
 171 #define MAYBE_NegotiateNonCryptoCall NegotiateNonCryptoCall
 172 #endif
 173
 174 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_NegotiateNonCryptoCall) {
 175   MakeTypicalPeerConnectionCall("negotiateNonCryptoCall();");
 176 }
 177
 178 // This test can negotiate an SDP offer that includes a b=AS:xx to control
 179 // the bandwidth for audio and video
 180 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, NegotiateOfferWithBLine) {
 181   MakeTypicalPeerConnectionCall("negotiateOfferWithBLine();");
 182 }
 183
 184 // This test will make a complete PeerConnection-based call using legacy SDP
 185 // settings: GIce, external SDES, and no BUNDLE.
 186 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 187 // Timing out on ARM linux, see http://crbug.com/240373
 188 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall
 189 #else
 190 #define MAYBE_CanSetupLegacyCall CanSetupLegacyCall
 191 #endif
 192
 193 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupLegacyCall) {
 194   MakeTypicalPeerConnectionCall("callWithLegacySdp();");
 195 }
 196
 197 // This test will make a PeerConnection-based call and test an unreliable text
 198 // dataChannel.
 199 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
 200 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallWithDataOnly) {
 201   MakeTypicalPeerConnectionCall("callWithDataOnly();");
 202 }
 203
 204 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallWithSctpDataOnly) {
 205   MakeTypicalPeerConnectionCall("callWithSctpDataOnly();");
 206 }
 207
 208 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 209 // Timing out on ARM linux bot: http://crbug.com/238490
 210 #define MAYBE_CallWithDataAndMedia DISABLED_CallWithDataAndMedia
 211 #else
 212 #define MAYBE_CallWithDataAndMedia CallWithDataAndMedia
 213 #endif
 214
 215 // This test will make a PeerConnection-based call and test an unreliable text
 216 // dataChannel and audio and video tracks.
 217 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
 218 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithDataAndMedia) {
 219   MakeTypicalPeerConnectionCall("callWithDataAndMedia();");
 220 }
 221
 222
 223 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 224 // Timing out on ARM linux bot: http://crbug.com/238490
 225 #define MAYBE_CallWithSctpDataAndMedia DISABLED_CallWithSctpDataAndMedia
 226 #else
 227 #define MAYBE_CallWithSctpDataAndMedia CallWithSctpDataAndMedia
 228 #endif
 229
 230 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 231                        MAYBE_CallWithSctpDataAndMedia) {
 232   MakeTypicalPeerConnectionCall("callWithSctpDataAndMedia();");
 233 }
 234
 235 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 236 // Timing out on ARM linux bot: http://crbug.com/238490
 237 #define MAYBE_CallWithDataAndLaterAddMedia DISABLED_CallWithDataAndLaterAddMedia
 238 #else
 239 // Temporarily disable the test on all platforms. http://crbug.com/293252
 240 #define MAYBE_CallWithDataAndLaterAddMedia DISABLED_CallWithDataAndLaterAddMedia
 241 #endif
 242
 243 // This test will make a PeerConnection-based call and test an unreliable text
 244 // dataChannel and later add an audio and video track.
 245 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) {
 246   MakeTypicalPeerConnectionCall("callWithDataAndLaterAddMedia();");
 247 }
 248
 249 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 250 // Timing out on ARM linux bot: http://crbug.com/238490
 251 #define MAYBE_CallWithNewVideoMediaStream DISABLED_CallWithNewVideoMediaStream
 252 #else
 253 #define MAYBE_CallWithNewVideoMediaStream CallWithNewVideoMediaStream
 254 #endif
 255
 256 // This test will make a PeerConnection-based call and send a new Video
 257 // MediaStream that has been created based on a MediaStream created with
 258 // getUserMedia.
 259 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) {
 260   MakeTypicalPeerConnectionCall("callWithNewVideoMediaStream();");
 261 }
 262
 263 // This test will make a PeerConnection-based call and send a new Video
 264 // MediaStream that has been created based on a MediaStream created with
 265 // getUserMedia. When video is flowing, the VideoTrack is removed and an
 266 // AudioTrack is added instead.
 267 // TODO(phoglund): This test is manual since not all buildbots has an audio
 268 // input.
 269 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CallAndModifyStream) {
 270   MakeTypicalPeerConnectionCall(
 271       "callWithNewVideoMediaStreamLaterSwitchToAudio();");
 272 }
 273
 274 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) {
 275   MakeTypicalPeerConnectionCall("addTwoMediaStreamsToOneConnection();");
 276 }
 277
 278 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 279                        EstablishAudioVideoCallAndMeasureOutputLevel) {
 280   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
 281     // Bots with no output devices will force the audio code into a different
 282     // path where it doesn't manage to set either the low or high latency path.
 283     // This test will compute useless values in that case, so skip running on
 284     // such bots (see crbug.com/326338).
 285     LOG(INFO) << "Missing output devices: skipping test...";
 286     return;
 287   }
 288
 289   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
 290       switches::kUseFakeDeviceForMediaStream))
 291           << "Must run with fake devices since the test will explicitly look "
 292           << "for the fake device signal.";
 293
 294   MakeTypicalPeerConnectionCall("callAndEnsureAudioIsPlaying();");
 295 }
 296
 297 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 298                        EstablishAudioVideoCallAndVerifyMutingWorks) {
 299   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
 300     // Bots with no output devices will force the audio code into a different
 301     // path where it doesn't manage to set either the low or high latency path.
 302     // This test will compute useless values in that case, so skip running on
 303     // such bots (see crbug.com/326338).
 304     LOG(INFO) << "Missing output devices: skipping test...";
 305     return;
 306   }
 307
 308   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
 309       switches::kUseFakeDeviceForMediaStream))
 310           << "Must run with fake devices since the test will explicitly look "
 311           << "for the fake device signal.";
 312
 313   MakeTypicalPeerConnectionCall("callAndEnsureAudioTrackMutingWorks();");
 314 }
 315
 316 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 317                        EstablishAudioVideoCallAndVerifyUnmutingWorks) {
 318   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
 319     // See comment on EstablishAudioVideoCallAndVerifyMutingWorks.
 320     LOG(INFO) << "Missing output devices: skipping test...";
 321     return;
 322   }
 323
 324   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
 325       switches::kUseFakeDeviceForMediaStream))
 326           << "Must run with fake devices since the test will explicitly look "
 327           << "for the fake device signal.";
 328
 329   MakeTypicalPeerConnectionCall("callAndEnsureAudioTrackUnmutingWorks();");
 330 }
 331
 332 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
 333   MakeTypicalPeerConnectionCall("callAndEnsureVideoTrackMutingWorks();");
 334 }
 335
 336 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 337 // Timing out on ARM linux bot: http://crbug.com/238490
 338 #define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
 339 #else
 340 #define MAYBE_CallWithAecDump CallWithAecDump
 341 #endif
 342
 343 // This tests will make a complete PeerConnection-based call, verify that
 344 // video is playing for the call, and verify that a non-empty AEC dump file
 345 // exists. The AEC dump is enabled through webrtc-internals, in contrast to
 346 // using a command line flag (tested in webrtc_aecdump_browsertest.cc). The HTML
 347 // and Javascript is bypassed since it would trigger a file picker dialog.
 348 // Instead, the dialog callback FileSelected() is invoked directly. In fact,
 349 // there's never a webrtc-internals page opened at all since that's not needed.
 350 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithAecDump) {
 351   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 352
 353   // We must navigate somewhere first so that the render process is created.
 354   NavigateToURL(shell(), GURL(""));
 355
 356   base::FilePath dump_file;
 357   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 358
 359   // This fakes the behavior of another open tab with webrtc-internals, and
 360   // enabling AEC dump in that tab.
 361   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
 362
 363   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
 364   NavigateToURL(shell(), url);
 365   DisableOpusIfOnAndroid();
 366   ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 367
 368   EXPECT_TRUE(base::PathExists(dump_file));
 369   int64 file_size = 0;
 370   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
 371   EXPECT_GT(file_size, 0);
 372
 373   base::DeleteFile(dump_file, false);
 374 }
 375
 376 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 377 // Timing out on ARM linux bot: http://crbug.com/238490
 378 #define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
 379 #else
 380 #define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled
 381 #endif
 382
 383 // As above, but enable and disable dump before starting a call. The file should
 384 // be created, but should be empty.
 385 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 386                        MAYBE_CallWithAecDumpEnabledThenDisabled) {
 387   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 388
 389   // We must navigate somewhere first so that the render process is created.
 390   NavigateToURL(shell(), GURL(""));
 391
 392   base::FilePath dump_file;
 393   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 394
 395   // This fakes the behavior of another open tab with webrtc-internals, and
 396   // enabling AEC dump in that tab, then disabling it.
 397   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
 398   WebRTCInternals::GetInstance()->DisableAecDump();
 399
 400   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
 401   NavigateToURL(shell(), url);
 402   DisableOpusIfOnAndroid();
 403   ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 404
 405   EXPECT_TRUE(base::PathExists(dump_file));
 406   int64 file_size = 0;
 407   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
 408   EXPECT_EQ(0, file_size);
 409
 410   base::DeleteFile(dump_file, false);
 411 }
 412
 413 }  // namespace content