Disabling WebRTC call tests on Android ASAN.

We are getting renderer crashes, disable until someone can investigate.

BUG=392224,408496

Review URL: https://codereview.chromium.org/516883002

Cr-Commit-Position: refs/heads/master@{#292611}

content/browser/media/webrtc_aecdump_browsertest.cc

+// Copyright (c) 2014 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/process/process_handle.h"
+#include "base/strings/string_number_conversions.h"
+#include "base/strings/stringprintf.h"
+#include "content/browser/media/webrtc_internals.h"
+#include "content/browser/web_contents/web_contents_impl.h"
+#include "content/public/test/browser_test_utils.h"
+#include "content/public/test/content_browser_test_utils.h"
+#include "content/shell/browser/shell.h"
+#include "content/test/webrtc_content_browsertest_base.h"
+#include "net/test/embedded_test_server/embedded_test_server.h"
+
+namespace {
+
+const int kExpectedConsumerId = 0;
+
+// Get the ID for the render process host when there should only be one.
+bool GetRenderProcessHostId(base::ProcessId* id) {
+  content::RenderProcessHost::iterator it(
+      content::RenderProcessHost::AllHostsIterator());
+  *id = base::GetProcId(it.GetCurrentValue()->GetHandle());
+  EXPECT_NE(base::kNullProcessId, *id);
+  if (*id == base::kNullProcessId)
+    return false;
+  it.Advance();
+  EXPECT_TRUE(it.IsAtEnd());
+  return it.IsAtEnd();
+}
+
+}  // namespace
+
+namespace content {
+
+class WebRtcAecDumpBrowserTest : public WebRtcContentBrowserTest {
+ public:
+  WebRtcAecDumpBrowserTest() {}
+  virtual ~WebRtcAecDumpBrowserTest() {}
+};
+
+#if defined(OS_WIN)
+#define IntToStringType base::IntToString16
+#else
+#define IntToStringType base::IntToString
+#endif
+
+#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
+#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
+// Renderer crashes under Android ASAN: https://crbug.com/408496.
+#define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
+#else
+#define MAYBE_CallWithAecDump CallWithAecDump
+#endif
+
+// 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_F(WebRtcAecDumpBrowserTest, MAYBE_CallWithAecDump) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  // We must navigate somewhere first so that the render process is created.
+  NavigateToURL(shell(), GURL(""));
+
+  base::FilePath dump_file;
+  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
+  base::DeleteFile(dump_file, false);
+
+  // 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);
+  DisableOpusIfOnAndroid();
+  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
+
+  EXPECT_FALSE(base::PathExists(dump_file));
+
+  // Add file extensions that we expect to be added. The dump name will be
+  // <temporary path>.<render process id>.<consumer id>, for example
+  // "/tmp/.com.google.Chrome.Z6UC3P.12345.0".
+  base::ProcessId render_process_id = base::kNullProcessId;
+  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
+  dump_file = dump_file.AddExtension(IntToStringType(render_process_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);
+}
+
+// TODO(grunell): Add test for multiple dumps when re-use of
+// MediaStreamAudioProcessor in AudioCapturer has been removed.
+
+#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
+#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
+// Renderer crashes under Android ASAN: https://crbug.com/408496.
+#define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
+#else
+#define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled
+#endif
+
+// As above, but enable and disable dump before starting a call. The file should
+// be created, but should be empty.
+IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest,
+                       MAYBE_CallWithAecDumpEnabledThenDisabled) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  // We must navigate somewhere first so that the render process is created.
+  NavigateToURL(shell(), GURL(""));
+
+  base::FilePath dump_file;
+  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
+  base::DeleteFile(dump_file, false);
+
+  // 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});");
+
+  // Add file extensions that we expect to be added.
+  base::ProcessId render_process_id = base::kNullProcessId;
+  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
+  dump_file = dump_file.AddExtension(IntToStringType(render_process_id))
+                       .AddExtension(IntToStringType(kExpectedConsumerId));
+
+  EXPECT_FALSE(base::PathExists(dump_file));
+
+  base::DeleteFile(dump_file, false);
+}
+
+#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
+// Timing out on ARM linux bot: http://crbug.com/238490
+#define MAYBE_TwoCallsWithAecDump DISABLED_TwoCallsWithAecDump
+#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
+// Renderer crashes under Android ASAN: https://crbug.com/408496.
+#define MAYBE_TwoCallsWithAecDump DISABLED_TwoCallsWithAecDump
+#else
+#define MAYBE_TwoCallsWithAecDump TwoCallsWithAecDump
+#endif
+
+IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest, MAYBE_TwoCallsWithAecDump) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  // We must navigate somewhere first so that the render process is created.
+  NavigateToURL(shell(), GURL(""));
+
+  // Create a second window.
+  Shell* shell2 = CreateBrowser();
+  NavigateToURL(shell2, GURL(""));
+
+  base::FilePath dump_file;
+  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
+  base::DeleteFile(dump_file, false);
+
+  // 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);
+  NavigateToURL(shell2, url);
+  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
+  std::string result;
+  EXPECT_TRUE(ExecuteScriptAndExtractString(
+      shell2->web_contents(),
+      "call({video: true, audio: true});",
+      &result));
+  ASSERT_STREQ("OK", result.c_str());
+
+  EXPECT_FALSE(base::PathExists(dump_file));
+
+  RenderProcessHost::iterator it =
+      content::RenderProcessHost::AllHostsIterator();
+  for (; !it.IsAtEnd(); it.Advance()) {
+    base::ProcessId render_process_id =
+        base::GetProcId(it.GetCurrentValue()->GetHandle());
+    EXPECT_NE(base::kNullProcessId, render_process_id);
+
+    // Add file extensions that we expect to be added.
+    base::FilePath unique_dump_file =
+        dump_file.AddExtension(IntToStringType(render_process_id))
+                 .AddExtension(IntToStringType(kExpectedConsumerId));
+
+    EXPECT_TRUE(base::PathExists(unique_dump_file));
+    int64 file_size = 0;
+    EXPECT_TRUE(base::GetFileSize(unique_dump_file, &file_size));
+    EXPECT_GT(file_size, 0);
+
+    base::DeleteFile(unique_dump_file, false);
+  }
+}
+
+}  // namespace content

content/browser/media/webrtc_internals.h

@@ -103,10 +103,10 @@ class CONTENT_EXPORT WebRTCInternals : public NotificationObserver,
 
  private:
   friend struct DefaultSingletonTraits<WebRTCInternals>;
-  FRIEND_TEST_ALL_PREFIXES(WebRtcBrowserTest, CallWithAecDump);
-  FRIEND_TEST_ALL_PREFIXES(WebRtcBrowserTest,
+  FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest, CallWithAecDump);
+  FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest,
                            CallWithAecDumpEnabledThenDisabled);
-  FRIEND_TEST_ALL_PREFIXES(WebRtcBrowserTest, TwoCallsWithAecDump);
+  FRIEND_TEST_ALL_PREFIXES(WebRtcAecDumpBrowserTest, TwoCallsWithAecDump);
   FRIEND_TEST_ALL_PREFIXES(WebRTCInternalsTest,
                            AecRecordingFileSelectionCanceled);
 

content/content_tests.gypi

@@ -1317,6 +1317,7 @@
             }],
             ['enable_webrtc==1', {
               'sources': [
+                'browser/media/webrtc_aecdump_browsertest.cc',
                 'browser/media/webrtc_browsertest.cc',
                 'browser/media/webrtc_getusermedia_browsertest.cc',
                 'browser/media/webrtc_internals_browsertest.cc',

content/test/webrtc_content_browsertest_base.cc

@@ -75,4 +75,12 @@ std::string WebRtcContentBrowserTest::GenerateGetUserMediaCall(
       max_frame_rate);
 }
 
+void WebRtcContentBrowserTest::DisableOpusIfOnAndroid() {
+#if defined(OS_ANDROID)
+  // Always force iSAC 16K on Android for now (Opus is broken).
+  EXPECT_EQ("isac-forced",
+            ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
+#endif
+}
+
 }  // namespace content

content/test/webrtc_content_browsertest_base.h

@@ -26,6 +26,8 @@ class WebRtcContentBrowserTest: public ContentBrowserTest {
   // If the javascript returns != OK or times out, we fail the test.
   void ExecuteJavascriptAndWaitForOk(const std::string& javascript);
 
+  void DisableOpusIfOnAndroid();
+
   // Generates javascript code for a getUserMedia call.
   std::string GenerateGetUserMediaCall(const char* function_name,
                                        int min_width,