// found in the LICENSE file.
#include "base/command_line.h"
+#include "base/file_util.h"
#include "base/strings/stringprintf.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 "base/win/windows_version.h"
#endif
+#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
+static const char kUseLenientAudioChecking[] = "false";
+#endif
+
namespace content {
class WebRtcBrowserTest : public WebRtcContentBrowserTest,
command_line->AppendSwitch(switches::kEnableAudioTrackProcessing);
}
+ virtual void TearDownOnMainThread() OVERRIDE {
+#if defined(OS_ANDROID)
+ // TODO(phoglund): this is a ugly workaround to let the IO thread
+ // finish its work. The reason we need this on Android is that
+ // content_browsertests tearDown logic is broken with respect
+ // to threading, which causes the IO thread to compete with the
+ // teardown. See http://crbug.com/362852. I also tried with 2
+ // seconds, but that isn't enough.
+ base::PlatformThread::Sleep(base::TimeDelta::FromSeconds(5));
+#endif
+ }
+
// 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) {
}
void DisableOpusIfOnAndroid() {
-#if defined (OS_ANDROID)
+#if defined(OS_ANDROID)
// Always force iSAC 16K on Android for now (Opus is broken).
EXPECT_EQ("isac-forced",
ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
#define MAYBE_CanForwardRemoteStream720p CanForwardRemoteStream720p
#endif
IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
+#if defined (OS_ANDROID)
+ // This test fails on Nexus 5 devices.
+ // TODO: 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_P(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
+#if defined (OS_ANDROID)
+ // This test fails on Nexus 5 devices.
+ // TODO: 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, 30, 30);
+ "callAndForwardRemoteStream", 1280, 1280, 720, 720, 10, 30);
MakeTypicalPeerConnectionCall(javascript);
}
// This test will modify the SDP offer to an unsupported codec, which should
// cause SetLocalDescription to fail.
-#if defined(USE_OZONE)
-// Disabled for Ozone, see http://crbug.com/315392#c15
-#define MAYBE_NegotiateUnsupportedVideoCodec\
- DISABLED_NegotiateUnsupportedVideoCodec
-#else
-#define MAYBE_NegotiateUnsupportedVideoCodec NegotiateUnsupportedVideoCodec
-#endif
-
-IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
- MAYBE_NegotiateUnsupportedVideoCodec) {
+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.
-#if defined(USE_OZONE)
-// Disabled for Ozone, see http://crbug.com/315392#c15
-#define MAYBE_NegotiateNonCryptoCall DISABLED_NegotiateNonCryptoCall
-#else
-#define MAYBE_NegotiateNonCryptoCall NegotiateNonCryptoCall
-#endif
-
-IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MAYBE_NegotiateNonCryptoCall) {
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, NegotiateNonCryptoCall) {
MakeTypicalPeerConnectionCall("negotiateNonCryptoCall();");
}
<< "Must run with fake devices since the test will explicitly look "
<< "for the fake device signal.";
- MakeTypicalPeerConnectionCall("callAndEnsureAudioIsPlaying();");
+ MakeTypicalPeerConnectionCall(base::StringPrintf(
+ "callAndEnsureAudioIsPlaying(%s);", kUseLenientAudioChecking));
}
IN_PROC_BROWSER_TEST_P(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).
+ // See comment on EstablishAudioVideoCallAndMeasureOutputLevel.
LOG(INFO) << "Missing output devices: skipping test...";
return;
}
<< "Must run with fake devices since the test will explicitly look "
<< "for the fake device signal.";
- MakeTypicalPeerConnectionCall("callAndEnsureAudioTrackMutingWorks();");
+ MakeTypicalPeerConnectionCall(base::StringPrintf(
+ "callAndEnsureAudioTrackMutingWorks(%s);", kUseLenientAudioChecking));
}
IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
EstablishAudioVideoCallAndVerifyUnmutingWorks) {
if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
- // See comment on EstablishAudioVideoCallAndVerifyMutingWorks.
+ // See comment on EstablishAudioVideoCallAndMeasureOutputLevel.
LOG(INFO) << "Missing output devices: skipping test...";
return;
}
<< "Must run with fake devices since the test will explicitly look "
<< "for the fake device signal.";
- MakeTypicalPeerConnectionCall("callAndEnsureAudioTrackUnmutingWorks();");
+ MakeTypicalPeerConnectionCall(base::StringPrintf(
+ "callAndEnsureAudioTrackUnmutingWorks(%s);", kUseLenientAudioChecking));
}
IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
// 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, in contrast to
-// using a command line flag (tested in webrtc_aecdump_browsertest.cc). 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.
+// 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());
projects / platform / framework / web / crosswalk.git / blobdiff