All WebRTC content browsertests will now force iSAC 16K on Android.

All WebRTC content browsertests will now force iSAC 16K on Android.

R=wjia@chromium.org
BUG=181069
Committed: https://src.chromium.org/viewvc/chrome?view=rev&revision=242807

[wjia(left Chromium), 2013-12-18 18:10:02]

lgtm

[commit-bot: I haz the power, 2013-12-20 13:53:16]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/phoglund@chromium.org/118163002/40001

[phoglund_chromium, 2013-12-20 13:55:03]

So Wei, I did two changes after your lgtm: enabled the tests you intended to
re-enable in our patch (https://codereview.chromium.org/117843002/), and
loosened the audio test's values a bit since it seems they could be responsible
for the flakes I see on the chromium bots.

Let me know if you have any objections and I'll fix them in a later patch.

[commit-bot: I haz the power, 2013-12-20 16:06:43]

Retried try job too often on win_rel for step(s) browser_tests
http://build.chromium.org/p/tryserver.chromium/buildstatus?builder=win_rel&nu...

[commit-bot: I haz the power, 2013-12-20 17:10:02]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/phoglund@chromium.org/118163002/40001

[commit-bot: I haz the power, 2013-12-20 20:21:27]

Retried try job too often on win_rel for step(s) unit_tests
http://build.chromium.org/p/tryserver.chromium/buildstatus?builder=win_rel&nu...

[commit-bot: I haz the power, 2013-12-22 20:56:38]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/phoglund@chromium.org/118163002/40001

[commit-bot: I haz the power, 2013-12-22 23:53:54]

Retried try job too often on win_rel for step(s) unit_tests
http://build.chromium.org/p/tryserver.chromium/buildstatus?builder=win_rel&nu...

[commit-bot: I haz the power, 2014-01-01 17:10:50]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/phoglund@chromium.org/118163002/40001

[commit-bot: I haz the power, 2014-01-01 17:10:54]

Failed to apply patch for content/browser/media/webrtc_browsertest.cc:
While running patch -p1 --forward --force --no-backup-if-mismatch;
  patching file content/browser/media/webrtc_browsertest.cc
  Hunk #1 succeeded at 23 (offset 1 line).
  Hunk #2 succeeded at 116 (offset 31 lines).
  Hunk #3 FAILED at 97.
  Hunk #4 succeeded at 220 (offset 85 lines).
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  Hunk #22 succeeded at 674 (offset 198 lines).
  Hunk #23 succeeded at 693 with fuzz 1 (offset 198 lines).
  1 out of 23 hunks FAILED -- saving rejects to file
content/browser/media/webrtc_browsertest.cc.rej

Patch:       content/browser/media/webrtc_browsertest.cc
Index: content/browser/media/webrtc_browsertest.cc
diff --git a/content/browser/media/webrtc_browsertest.cc
b/content/browser/media/webrtc_browsertest.cc
index
47847a7d53bac2b7b1776d4602123880539f273f..c9eea10bd0c66e75ee762efe2bacab4516393381
100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -22,14 +22,6 @@
 #include "base/win/windows_version.h"
 #endif
 
-const char kForceIsac16K[] =
-#ifdef OS_ANDROID
-  // The default audio codec, Opus, doesn't work on Android.
-  "true";
-#else
-  "false";
-#endif
-
 namespace {
 
 static const char kGetUserMediaAndStop[] = "getUserMediaAndStop";
@@ -93,7 +85,7 @@ class WebrtcBrowserTest: public ContentBrowserTest {
                                              &error_message));
 
     ASSERT_TRUE(value.get() != NULL) << error_message;
-    EXPECT_EQ(value->GetType(), base::Value::TYPE_LIST);
+    ASSERT_EQ(value->GetType(), base::Value::TYPE_LIST);
 
     base::ListValue* values;
     ASSERT_TRUE(value->GetAsList(&values));
@@ -105,7 +97,7 @@ class WebrtcBrowserTest: public ContentBrowserTest {
     std::string samples_interarrival_ns;
     for (ListValue::iterator it = values->begin(); it != values->end(); ++it) {
       const DictionaryValue* dict;
-      EXPECT_TRUE((*it)->GetAsDictionary(&dict));
+      ASSERT_TRUE((*it)->GetAsDictionary(&dict));
 
       if (dict->GetInteger("dur", &duration_ns))
         samples_duration.append(base::StringPrintf("%d,", duration_ns));
@@ -143,6 +135,23 @@ class WebrtcBrowserTest: public ContentBrowserTest {
     TitleWatcher title_watcher(shell()->web_contents(), expected_title16);
     EXPECT_EQ(expected_title16, title_watcher.WaitAndGetTitle());
   }
+
+  // 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());
+
+    GURL
url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
+    NavigateToURL(shell(), url);
+
+#if defined (OS_ANDROID)
+    // Always force iSAC 16K on Android for now (Opus is broken).
+    ASSERT_TRUE(ExecuteJavascript("forceIsac16KInSdp();"));
+#endif
+
+    ASSERT_TRUE(ExecuteJavascript(javascript));
+    ExpectTitle("OK");
+  }
 };
 
 // These tests will all make a getUserMedia call with different constraints and
@@ -154,7 +163,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
GetVideoStreamAndStop) {
   GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
   NavigateToURL(shell(), url);
 
-  EXPECT_TRUE(ExecuteJavascript(
+  ASSERT_TRUE(ExecuteJavascript(
       base::StringPrintf("%s({video: true});", kGetUserMediaAndStop)));
 
   ExpectTitle("OK");
@@ -166,7 +175,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
GetAudioAndVideoStreamAndStop) {
   GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
   NavigateToURL(shell(), url);
 
-  EXPECT_TRUE(ExecuteJavascript(base::StringPrintf(
+  ASSERT_TRUE(ExecuteJavascript(base::StringPrintf(
       "%s({video: true, audio: true});", kGetUserMediaAndStop)));
 
   ExpectTitle("OK");
@@ -178,7 +187,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
GetAudioAndVideoStreamAndClone) {
   GURL url(embedded_test_server()->GetURL("/media/getusermedia.html"));
   NavigateToURL(shell(), url);
 
-  EXPECT_TRUE(ExecuteJavascript("getUserMediaAndClone();"));
+  ASSERT_TRUE(ExecuteJavascript("getUserMediaAndClone();"));
 
   ExpectTitle("OK");
 }
@@ -194,13 +203,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
GetAudioAndVideoStreamAndClone) {
 // 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);
-
-  EXPECT_TRUE(ExecuteJavascript("call({video: true});"));
-  ExpectTitle("OK");
+  MakeTypicalPeerConnectionCall("call({video: true});");
 }
 
 // This test will make a simple getUserMedia page, verify that video is playing
@@ -212,7 +215,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
TracePerformanceDuringGetUserMedia) {
   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(
+  ASSERT_TRUE(ExecuteJavascript(base::StringPrintf(
       "%s({video: true}, 10);", kGetUserMediaAndWaitAndStop)));
 
   // Make sure the stream is up and running, then start collecting traces.
@@ -221,7 +224,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
TracePerformanceDuringGetUserMedia) {
   trace_log->SetEnabled(base::debug::CategoryFilter("video"),
                         base::debug::TraceLog::ENABLE_SAMPLING);
   // Check that we are indeed recording.
-  EXPECT_EQ(trace_log->GetNumTracesRecorded(), 1);
+  ASSERT_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.
@@ -241,36 +244,18 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
TracePerformanceDuringGetUserMedia) {
 #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");
+  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');"));
+  MakeTypicalPeerConnectionCall("callAndSendDtmf('123,abc');");
 }
 
 IN_PROC_BROWSER_TEST_F(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
@@ -285,27 +270,14 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
 #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");
+  MakeTypicalPeerConnectionCall(
+      "callAndForwardRemoteStream({video: true, audio: true});");
 }
 
 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");
+  const std::string javascript = GenerateGetUserMediaCall(
+      "callAndForwardRemoteStream", 1280, 1280, 720, 720, 30, 30);
+  MakeTypicalPeerConnectionCall(javascript);
 }
 
 // This test will make a complete PeerConnection-based call but remove the
@@ -326,38 +298,19 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
MAYBE_CanForwardRemoteStream720p) {
 #endif
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
                        MAYBE_CanSetupAudioAndVideoCa…
(message too large)

[commit-bot: I haz the power, 2014-01-02 09:47:08]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/phoglund@chromium.org/118163002/410001

[commit-bot: I haz the power, 2014-01-02 21:49:28]

Change committed as 242807