diff options
Diffstat (limited to 'chromium/content/browser/media/webrtc_browsertest.cc')
| -rw-r--r-- | chromium/content/browser/media/webrtc_browsertest.cc | 678 |
1 files changed, 262 insertions, 416 deletions
diff --git a/chromium/content/browser/media/webrtc_browsertest.cc b/chromium/content/browser/media/webrtc_browsertest.cc index 47847a7d53b..f5bb216ead0 100644 --- a/chromium/content/browser/media/webrtc_browsertest.cc +++ b/chromium/content/browser/media/webrtc_browsertest.cc @@ -3,234 +3,133 @@ // found in the LICENSE file. #include "base/command_line.h" -#include "base/debug/trace_event_impl.h" -#include "base/json/json_reader.h" +#include "base/file_util.h" +#include "base/strings/string_number_conversions.h" #include "base/strings/stringprintf.h" -#include "base/strings/utf_string_conversions.h" +#include "base/threading/platform_thread.h" #include "base/values.h" +#include "content/browser/media/webrtc_internals.h" #include "content/browser/web_contents/web_contents_impl.h" #include "content/public/common/content_switches.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/shell/browser/shell.h" -#include "content/test/content_browser_test.h" -#include "content/test/content_browser_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" -#include "testing/perf/perf_test.h" #if defined(OS_WIN) #include "base/win/windows_version.h" #endif -const char kForceIsac16K[] = -#ifdef OS_ANDROID - // The default audio codec, Opus, doesn't work on Android. - "true"; +#if defined (OS_ANDROID) || defined(THREAD_SANITIZER) +// Just do the bare minimum of audio checking on Android and under TSAN since +// it's a bit sensitive to device performance. +static const char kUseLenientAudioChecking[] = "true"; #else - "false"; +static const char kUseLenientAudioChecking[] = "false"; #endif -namespace { - -static const char kGetUserMediaAndStop[] = "getUserMediaAndStop"; -static const char kGetUserMediaAndWaitAndStop[] = "getUserMediaAndWaitAndStop"; -static const char kGetUserMediaAndAnalyseAndStop[] = - "getUserMediaAndAnalyseAndStop"; - -std::string GenerateGetUserMediaCall(const char* function_name, - int min_width, - int max_width, - int min_height, - int max_height, - int min_frame_rate, - int max_frame_rate) { - return base::StringPrintf( - "%s({video: {mandatory: {minWidth: %d, maxWidth: %d, " - "minHeight: %d, maxHeight: %d, minFrameRate: %d, maxFrameRate: %d}, " - "optional: []}});", - function_name, - min_width, - max_width, - min_height, - max_height, - min_frame_rate, - max_frame_rate); -} -} - namespace content { -class WebrtcBrowserTest: public ContentBrowserTest { +class WebRtcBrowserTest : public WebRtcContentBrowserTest, + public testing::WithParamInterface<bool> { public: - WebrtcBrowserTest() {} - virtual ~WebrtcBrowserTest() {} + WebRtcBrowserTest() {} + virtual ~WebRtcBrowserTest() {} virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE { - // We need fake devices in this test since we want to run on naked VMs. We - // assume these switches are set by default in content_browsertests. - ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch( - switches::kUseFakeDeviceForMediaStream)); - ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch( - switches::kUseFakeUIForMediaStream)); + WebRtcContentBrowserTest::SetUpCommandLine(command_line); - // The video playback will not work without a GPU, so force its use here. - // This may not be available on all VMs though. - command_line->AppendSwitch(switches::kUseGpuInTests); + bool enable_audio_track_processing = GetParam(); + if (!enable_audio_track_processing) + command_line->AppendSwitch(switches::kDisableAudioTrackProcessing); } - void DumpChromeTraceCallback( - const scoped_refptr<base::RefCountedString>& events, - bool has_more_events) { - // Convert the dump output into a correct JSON List. - std::string contents = "[" + events->data() + "]"; - - int error_code; - std::string error_message; - scoped_ptr<base::Value> value( - base::JSONReader::ReadAndReturnError(contents, - base::JSON_ALLOW_TRAILING_COMMAS, - &error_code, - &error_message)); - - ASSERT_TRUE(value.get() != NULL) << error_message; - EXPECT_EQ(value->GetType(), base::Value::TYPE_LIST); - - base::ListValue* values; - ASSERT_TRUE(value->GetAsList(&values)); - - int duration_ns = 0; - std::string samples_duration; - double timestamp_ns = 0.0; - double previous_timestamp_ns = 0.0; - std::string samples_interarrival_ns; - for (ListValue::iterator it = values->begin(); it != values->end(); ++it) { - const DictionaryValue* dict; - EXPECT_TRUE((*it)->GetAsDictionary(&dict)); - - if (dict->GetInteger("dur", &duration_ns)) - samples_duration.append(base::StringPrintf("%d,", duration_ns)); - if (dict->GetDouble("ts", ×tamp_ns)) { - if (previous_timestamp_ns) { - samples_interarrival_ns.append( - base::StringPrintf("%f,", timestamp_ns - previous_timestamp_ns)); - } - previous_timestamp_ns = timestamp_ns; - } - } - ASSERT_GT(samples_duration.size(), 0u) - << "Could not collect any samples during test, this is bad"; - perf_test::PrintResultList("video_capture", - "", - "sample_duration", - samples_duration, - "ns", - true); - perf_test::PrintResultList("video_capture", - "", - "interarrival_time", - samples_interarrival_ns, - "ns", - true); - } + // Convenience function since most peerconnection-call.html tests just load + // the page, kick off some javascript and wait for the title to change to OK. + void MakeTypicalPeerConnectionCall(const std::string& javascript) { + ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - protected: - bool ExecuteJavascript(const std::string& javascript) { - return ExecuteScript(shell()->web_contents(), javascript); - } + GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); + NavigateToURL(shell(), url); - void ExpectTitle(const std::string& expected_title) const { - base::string16 expected_title16(ASCIIToUTF16(expected_title)); - TitleWatcher title_watcher(shell()->web_contents(), expected_title16); - EXPECT_EQ(expected_title16, title_watcher.WaitAndGetTitle()); + DisableOpusIfOnAndroid(); + ExecuteJavascriptAndWaitForOk(javascript); } -}; - -// These tests will all make a getUserMedia call with different constraints and -// see that the success callback is called. If the error callback is called or -// none of the callbacks are called the tests will simply time out and fail. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetVideoStreamAndStop) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript( - base::StringPrintf("%s({video: true});", kGetUserMediaAndStop))); - - ExpectTitle("OK"); -} - -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndStop) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript(base::StringPrintf( - "%s({video: true, audio: true});", kGetUserMediaAndStop))); - ExpectTitle("OK"); -} - -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndClone) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); + // 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; + } - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); - NavigateToURL(shell(), url); + ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch( + switches::kUseFakeDeviceForMediaStream)) + << "Must run with fake devices since the test will explicitly look " + << "for the fake device signal."; - EXPECT_TRUE(ExecuteJavascript("getUserMediaAndClone();")); + MakeTypicalPeerConnectionCall(javascript); + } - ExpectTitle("OK"); -} + void DisableOpusIfOnAndroid() { +#if defined(OS_ANDROID) + // Always force iSAC 16K on Android for now (Opus is broken). + EXPECT_EQ("isac-forced", + ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();")); +#endif + } +}; +static const bool kRunTestsWithFlag[] = { false, true }; +INSTANTIATE_TEST_CASE_P(WebRtcBrowserTests, + WebRtcBrowserTest, + testing::ValuesIn(kRunTestsWithFlag)); #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) // Timing out on ARM linux bot: http://crbug.com/238490 -#define MAYBE_CanSetupVideoCall DISABLED_CanSetupVideoCall +#define MAYBE_CanSetupDefaultVideoCall DISABLED_CanSetupDefaultVideoCall #else -#define MAYBE_CanSetupVideoCall CanSetupVideoCall +#define MAYBE_CanSetupDefaultVideoCall CanSetupDefaultVideoCall #endif // These tests will make a complete PeerConnection-based call and verify that // video is playing for the call. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupVideoCall) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupDefaultVideoCall) { + MakeTypicalPeerConnectionCall( + "callAndExpectResolution({video: true}, 640, 480);"); +} - EXPECT_TRUE(ExecuteJavascript("call({video: true});")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CanSetupVideoCallWith1To1AspecRatio) { + const std::string javascript = + "callAndExpectResolution({video: {mandatory: {minWidth: 320," + " maxWidth: 320, minHeight: 320, maxHeight: 320}}}, 320, 320);"; + MakeTypicalPeerConnectionCall(javascript); } -// This test will make a simple getUserMedia page, verify that video is playing -// in a simple local <video>, and for a couple of seconds, collect some -// performance traces. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TracePerformanceDuringGetUserMedia) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + CanSetupVideoCallWith16To9AspecRatio) { + const std::string javascript = + "callAndExpectResolution({video: {mandatory: {minWidth: 640," + " maxWidth: 640, minAspectRatio: 1.777}}}, 640, 360);"; + MakeTypicalPeerConnectionCall(javascript); +} - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); - NavigateToURL(shell(), url); - // Put getUserMedia to work and let it run for a couple of seconds. - EXPECT_TRUE(ExecuteJavascript(base::StringPrintf( - "%s({video: true}, 10);", kGetUserMediaAndWaitAndStop))); - - // Make sure the stream is up and running, then start collecting traces. - ExpectTitle("Running..."); - base::debug::TraceLog* trace_log = base::debug::TraceLog::GetInstance(); - trace_log->SetEnabled(base::debug::CategoryFilter("video"), - base::debug::TraceLog::ENABLE_SAMPLING); - // Check that we are indeed recording. - EXPECT_EQ(trace_log->GetNumTracesRecorded(), 1); - - // Wait until the page title changes to "OK". Do not sleep() here since that - // would stop both this code and the browser underneath. - ExpectTitle("OK"); - - // Note that we need to stop the trace recording before flushing the data. - trace_log->SetDisabled(); - trace_log->Flush(base::Bind(&WebrtcBrowserTest::DumpChromeTraceCallback, - base::Unretained(this))); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + CanSetupVideoCallWith4To3AspecRatio) { + const std::string javascript = + "callAndExpectResolution({video: {mandatory: {minWidth: 960," + "maxAspectRatio: 1.333}}}, 960, 720);"; + MakeTypicalPeerConnectionCall(javascript); } #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) @@ -240,83 +139,73 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TracePerformanceDuringGetUserMedia) { #define MAYBE_CanSetupAudioAndVideoCall CanSetupAudioAndVideoCall #endif -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("call({video: true, audio: true});")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) { + MakeTypicalPeerConnectionCall("call({video: true, audio: true});"); } -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE( - ExecuteJavascript("callAndSendDtmf('123,abc');")); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) { + MakeTypicalPeerConnectionCall("callAndSendDtmf(\'123,abc\');"); } -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, +// TODO(phoglund): this test fails because the peer connection state will be +// stable in the second negotiation round rather than have-local-offer. +// http://crbug.com/293125. +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, DISABLED_CanMakeEmptyCallThenAddStreamsAndRenegotiate) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - const char* kJavascript = "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});"; - EXPECT_TRUE(ExecuteJavascript(kJavascript)); - ExpectTitle("OK"); + MakeTypicalPeerConnectionCall(kJavascript); } // Below 2 test will make a complete PeerConnection-based call between pc1 and // pc2, and then use the remote stream to setup a call between pc3 and pc4, and // then verify that video is received on pc3 and pc4. +// The stream sent from pc3 to pc4 is the stream received on pc1. +// The stream sent from pc4 to pc3 is cloned from stream the stream received +// on pc2. // Flaky on win xp. http://crbug.com/304775 #if defined(OS_WIN) #define MAYBE_CanForwardRemoteStream DISABLED_CanForwardRemoteStream #define MAYBE_CanForwardRemoteStream720p DISABLED_CanForwardRemoteStream720p #else #define MAYBE_CanForwardRemoteStream CanForwardRemoteStream +// Flaky on TSAN v2. http://crbug.com/373637 +#if defined(THREAD_SANITIZER) +#define MAYBE_CanForwardRemoteStream720p DISABLED_CanForwardRemoteStream720p +#else #define MAYBE_CanForwardRemoteStream720p CanForwardRemoteStream720p #endif -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanForwardRemoteStream) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript( - "callAndForwardRemoteStream({video: true, audio: true});")); - ExpectTitle("OK"); +#endif +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) { +#if defined (OS_ANDROID) + // This test fails on Nexus 5 devices. + // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389 + // for details. + CommandLine::ForCurrentProcess()->AppendSwitch( + switches::kDisableWebRtcHWDecoding); +#endif + MakeTypicalPeerConnectionCall( + "callAndForwardRemoteStream({video: true, audio: false});"); } -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanForwardRemoteStream720p) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - const std::string cmd = GenerateGetUserMediaCall("callAndForwardRemoteStream", - 1280, 1280, - 720, 720, 30, 30); - EXPECT_TRUE(ExecuteJavascript(cmd)); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) { +#if defined (OS_ANDROID) + // This test fails on Nexus 5 devices. + // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389 + // for details. + CommandLine::ForCurrentProcess()->AppendSwitch( + switches::kDisableWebRtcHWDecoding); +#endif + const std::string javascript = GenerateGetUserMediaCall( + "callAndForwardRemoteStream", 1280, 1280, 720, 720, 10, 30); + MakeTypicalPeerConnectionCall(javascript); } // This test will make a complete PeerConnection-based call but remove the // MSID and bundle attribute from the initial offer to verify that // video is playing for the call even if the initiating client don't support // MSID. http://tools.ietf.org/html/draft-alvestrand-rtcweb-msid-02 -#if defined(OS_WIN) && defined(USE_AURA) -// Disabled for win7_aura, see http://crbug.com/235089. -#define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\ - DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle -#elif defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) +#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) // Timing out on ARM linux, see http://crbug.com/240373 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\ DISABLED_CanSetupAudioAndVideoCallWithoutMsidAndBundle @@ -324,85 +213,51 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanForwardRemoteStream720p) { #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\ CanSetupAudioAndVideoCallWithoutMsidAndBundle #endif -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithoutMsidAndBundle();")); - ExpectTitle("OK"); + MakeTypicalPeerConnectionCall("callWithoutMsidAndBundle();"); } // This test will modify the SDP offer to an unsupported codec, which should // cause SetLocalDescription to fail. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, - NegotiateUnsupportedVideoCodec) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("negotiateUnsupportedVideoCodec();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, NegotiateUnsupportedVideoCodec) { + MakeTypicalPeerConnectionCall("negotiateUnsupportedVideoCodec();"); } // This test will modify the SDP offer to use no encryption, which should // cause SetLocalDescription to fail. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, NegotiateNonCryptoCall) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, NegotiateNonCryptoCall) { + MakeTypicalPeerConnectionCall("negotiateNonCryptoCall();"); +} - EXPECT_TRUE(ExecuteJavascript("negotiateNonCryptoCall();")); - ExpectTitle("OK"); +// This test can negotiate an SDP offer that includes a b=AS:xx to control +// the bandwidth for audio and video +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, NegotiateOfferWithBLine) { + MakeTypicalPeerConnectionCall("negotiateOfferWithBLine();"); } // This test will make a complete PeerConnection-based call using legacy SDP // settings: GIce, external SDES, and no BUNDLE. -#if defined(OS_WIN) && defined(USE_AURA) -// Disabled for win7_aura, see http://crbug.com/235089. -#define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall -#elif defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) +#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) // Timing out on ARM linux, see http://crbug.com/240373 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall #else #define MAYBE_CanSetupLegacyCall CanSetupLegacyCall #endif -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupLegacyCall) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithLegacySdp();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanSetupLegacyCall) { + MakeTypicalPeerConnectionCall("callWithLegacySdp();"); } // This test will make a PeerConnection-based call and test an unreliable text // dataChannel. // TODO(mallinath) - Remove this test after rtp based data channel is disabled. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithDataOnly) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithDataOnly();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallWithDataOnly) { + MakeTypicalPeerConnectionCall("callWithDataOnly();"); } -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithSctpDataOnly) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithSctpDataOnly();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallWithSctpDataOnly) { + MakeTypicalPeerConnectionCall("callWithSctpDataOnly();"); } #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) @@ -415,14 +270,8 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithSctpDataOnly) { // This test will make a PeerConnection-based call and test an unreliable text // dataChannel and audio and video tracks. // TODO(mallinath) - Remove this test after rtp based data channel is disabled. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndMedia) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithDataAndMedia();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, DISABLED_CallWithDataAndMedia) { + MakeTypicalPeerConnectionCall("callWithDataAndMedia();"); } @@ -433,15 +282,9 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndMedia) { #define MAYBE_CallWithSctpDataAndMedia CallWithSctpDataAndMedia #endif -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithSctpDataAndMedia) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithSctpDataAndMedia();")); - ExpectTitle("OK"); + MakeTypicalPeerConnectionCall("callWithSctpDataAndMedia();"); } #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) @@ -454,14 +297,8 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, // This test will make a PeerConnection-based call and test an unreliable text // dataChannel and later add an audio and video track. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithDataAndLaterAddMedia();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) { + MakeTypicalPeerConnectionCall("callWithDataAndLaterAddMedia();"); } #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) @@ -474,14 +311,8 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) { // This test will make a PeerConnection-based call and send a new Video // MediaStream that has been created based on a MediaStream created with // getUserMedia. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); - - EXPECT_TRUE(ExecuteJavascript("callWithNewVideoMediaStream();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) { + MakeTypicalPeerConnectionCall("callWithNewVideoMediaStream();"); } // This test will make a PeerConnection-based call and send a new Video @@ -490,132 +321,147 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) { // AudioTrack is added instead. // TODO(phoglund): This test is manual since not all buildbots has an audio // input. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MANUAL_CallAndModifyStream) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CallAndModifyStream) { + MakeTypicalPeerConnectionCall( + "callWithNewVideoMediaStreamLaterSwitchToAudio();"); +} - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) { + MakeTypicalPeerConnectionCall("addTwoMediaStreamsToOneConnection();"); +} - EXPECT_TRUE( - ExecuteJavascript("callWithNewVideoMediaStreamLaterSwitchToAudio();")); - ExpectTitle("OK"); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + EstablishAudioVideoCallAndEnsureAudioIsPlaying) { + MakeAudioDetectingPeerConnectionCall(base::StringPrintf( + "callAndEnsureAudioIsPlaying(%s, {audio:true, video:true});", + kUseLenientAudioChecking)); } -// This test calls getUserMedia in sequence with different constraints. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TestGetUserMediaConstraints) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + EstablishAudioOnlyCallAndEnsureAudioIsPlaying) { + MakeAudioDetectingPeerConnectionCall(base::StringPrintf( + "callAndEnsureAudioIsPlaying(%s, {audio:true});", + kUseLenientAudioChecking)); +} - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); - - std::vector<std::string> list_of_get_user_media_calls; - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 320, 320, 180, 180, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 320, 320, 240, 240, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 640, 640, 360, 360, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 640, 640, 480, 480, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 960, 960, 720, 720, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 1280, 1280, 720, 720, 30, 30)); - list_of_get_user_media_calls.push_back(GenerateGetUserMediaCall( - kGetUserMediaAndStop, 1920, 1920, 1080, 1080, 30, 30)); - - for (std::vector<std::string>::iterator const_iterator = - list_of_get_user_media_calls.begin(); - const_iterator != list_of_get_user_media_calls.end(); - ++const_iterator) { - DVLOG(1) << "Calling getUserMedia: " << *const_iterator; - NavigateToURL(shell(), url); - EXPECT_TRUE(ExecuteJavascript(*const_iterator)); - ExpectTitle("OK"); - } +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + EstablishAudioVideoCallAndVerifyMutingWorks) { + MakeAudioDetectingPeerConnectionCall(base::StringPrintf( + "callAndEnsureAudioTrackMutingWorks(%s);", kUseLenientAudioChecking)); } -// This test calls getUserMedia and checks for aspect ratio behavior. -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, TestGetUserMediaAspectRatio) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); +// Flaky on TSAN v2: http://crbug.com/373637 +#if defined(THREAD_SANITIZER) +#define MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks\ + DISABLED_EstablishAudioVideoCallAndVerifyUnmutingWorks +#else +#define MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks\ + EstablishAudioVideoCallAndVerifyUnmutingWorks +#endif +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks) { + MakeAudioDetectingPeerConnectionCall(base::StringPrintf( + "callAndEnsureAudioTrackUnmutingWorks(%s);", kUseLenientAudioChecking)); +} - GURL url(embedded_test_server()->GetURL("/media/getusermedia.html")); +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) { + MakeTypicalPeerConnectionCall("callAndEnsureVideoTrackMutingWorks();"); +} + +#if defined(OS_WIN) +#define IntToStringType base::IntToString16 +#else +#define IntToStringType base::IntToString +#endif - std::string constraints_4_3 = GenerateGetUserMediaCall( - kGetUserMediaAndAnalyseAndStop, 640, 640, 480, 480, 30, 30); - std::string constraints_16_9 = GenerateGetUserMediaCall( - kGetUserMediaAndAnalyseAndStop, 640, 640, 360, 360, 30, 30); +#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) +// Timing out on ARM linux bot: http://crbug.com/238490 +#define MAYBE_CallWithAecDump DISABLED_CallWithAecDump +#else +#define MAYBE_CallWithAecDump CallWithAecDump +#endif - // TODO(mcasas): add more aspect ratios, in particular 16:10 crbug.com/275594. +// This tests will make a complete PeerConnection-based call, verify that +// video is playing for the call, and verify that a non-empty AEC dump file +// exists. The AEC dump is enabled through webrtc-internals. The HTML and +// Javascript is bypassed since it would trigger a file picker dialog. Instead, +// the dialog callback FileSelected() is invoked directly. In fact, there's +// never a webrtc-internals page opened at all since that's not needed. +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CallWithAecDump) { + ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - NavigateToURL(shell(), url); - EXPECT_TRUE(ExecuteJavascript(constraints_4_3)); - ExpectTitle("4:3 letterbox"); + // We must navigate somewhere first so that the render process is created. + NavigateToURL(shell(), GURL("")); - NavigateToURL(shell(), url); - EXPECT_TRUE(ExecuteJavascript(constraints_16_9)); - ExpectTitle("16:9 letterbox"); -} + base::FilePath dump_file; + ASSERT_TRUE(CreateTemporaryFile(&dump_file)); -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, AddTwoMediaStreamsToOnePC) { - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); + // This fakes the behavior of another open tab with webrtc-internals, and + // enabling AEC dump in that tab. + WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL); GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); NavigateToURL(shell(), url); - - EXPECT_TRUE( - ExecuteJavascript("addTwoMediaStreamsToOneConnection();")); - ExpectTitle("OK"); + DisableOpusIfOnAndroid(); + ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});"); + + // Get the ID for the render process host. There should only be one. + RenderProcessHost::iterator it( + content::RenderProcessHost::AllHostsIterator()); + int render_process_host_id = it.GetCurrentValue()->GetID(); + EXPECT_GE(render_process_host_id, 0); + + // Add file extensions that we expect to be added. + static const int kExpectedConsumerId = 0; + dump_file = dump_file.AddExtension(IntToStringType(render_process_host_id)) + .AddExtension(IntToStringType(kExpectedConsumerId)); + + EXPECT_TRUE(base::PathExists(dump_file)); + int64 file_size = 0; + EXPECT_TRUE(base::GetFileSize(dump_file, &file_size)); + EXPECT_GT(file_size, 0); + + base::DeleteFile(dump_file, false); } -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, - EstablishAudioVideoCallAndMeasureOutputLevel) { - if (!media::AudioManager::Get()->HasAudioOutputDevices()) { - // Bots with no output devices will force the audio code into a different - // path 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; - } +// TODO(grunell): Add test for multiple dumps when re-use of +// MediaStreamAudioProcessor in AudioCapturer has been removed. - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch( - switches::kUseFakeDeviceForMediaStream)) - << "Must run with fake devices since the test will explicitly look " - << "for the fake device signal."; +#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY) +// Timing out on ARM linux bot: http://crbug.com/238490 +#define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled +#else +#define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled +#endif - GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); - NavigateToURL(shell(), url); +// As above, but enable and disable dump before starting a call. The file should +// be created, but should be empty. +IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, + MAYBE_CallWithAecDumpEnabledThenDisabled) { + ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - EXPECT_TRUE(ExecuteJavascript( - base::StringPrintf("callAndEnsureAudioIsPlaying(%s);", kForceIsac16K))); - ExpectTitle("OK"); -} + // We must navigate somewhere first so that the render process is created. + NavigateToURL(shell(), GURL("")); -IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, - EstablishAudioVideoCallAndVerifyMutingWorks) { - if (!media::AudioManager::Get()->HasAudioOutputDevices()) { - // Bots with no output devices will force the audio code into a different - // path 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; - } + base::FilePath dump_file; + ASSERT_TRUE(CreateTemporaryFile(&dump_file)); - ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady()); - ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch( - switches::kUseFakeDeviceForMediaStream)) - << "Must run with fake devices since the test will explicitly look " - << "for the fake device signal."; + // This fakes the behavior of another open tab with webrtc-internals, and + // enabling AEC dump in that tab, then disabling it. + WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL); + WebRTCInternals::GetInstance()->DisableAecDump(); GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html")); NavigateToURL(shell(), url); + DisableOpusIfOnAndroid(); + ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});"); + + EXPECT_TRUE(base::PathExists(dump_file)); + int64 file_size = 0; + EXPECT_TRUE(base::GetFileSize(dump_file, &file_size)); + EXPECT_EQ(0, file_size); - EXPECT_TRUE(ExecuteJavascript( - base::StringPrintf("callAndEnsureAudioMutingWorks(%s);", - kForceIsac16K))); - ExpectTitle("OK"); + base::DeleteFile(dump_file, false); } } // namespace content |