content/browser/media/webrtc_aecdump_browsertest.cc

   1 // Copyright (c) 2014 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/process/process_handle.h"
   6 #include "base/strings/string_number_conversions.h"
   7 #include "base/strings/stringprintf.h"
   8 #include "content/browser/media/webrtc_internals.h"
   9 #include "content/browser/web_contents/web_contents_impl.h"
  10 #include "content/public/test/browser_test_utils.h"
  11 #include "content/public/test/content_browser_test_utils.h"
  12 #include "content/shell/browser/shell.h"
  13 #include "content/test/webrtc_content_browsertest_base.h"
  14 #include "media/audio/audio_manager.h"
  15 #include "net/test/embedded_test_server/embedded_test_server.h"
  16
  17 namespace {
  18
  19 const int kExpectedConsumerId = 0;
  20
  21 // Get the ID for the render process host when there should only be one.
  22 bool GetRenderProcessHostId(base::ProcessId* id) {
  23   content::RenderProcessHost::iterator it(
  24       content::RenderProcessHost::AllHostsIterator());
  25   *id = base::GetProcId(it.GetCurrentValue()->GetHandle());
  26   EXPECT_NE(base::kNullProcessId, *id);
  27   if (*id == base::kNullProcessId)
  28     return false;
  29   it.Advance();
  30   EXPECT_TRUE(it.IsAtEnd());
  31   return it.IsAtEnd();
  32 }
  33
  34 }  // namespace
  35
  36 namespace content {
  37
  38 class WebRtcAecDumpBrowserTest : public WebRtcContentBrowserTest {
  39  public:
  40   WebRtcAecDumpBrowserTest() {}
  41   ~WebRtcAecDumpBrowserTest() override {}
  42 };
  43
  44 #if defined(OS_WIN)
  45 #define IntToStringType base::IntToString16
  46 #else
  47 #define IntToStringType base::IntToString
  48 #endif
  49
  50 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
  51 // Timing out on ARM linux bot: http://crbug.com/238490
  52 #define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
  53 #elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
  54 // Renderer crashes under Android ASAN: https://crbug.com/408496.
  55 #define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
  56 #else
  57 #define MAYBE_CallWithAecDump CallWithAecDump
  58 #endif
  59
  60 // This tests will make a complete PeerConnection-based call, verify that
  61 // video is playing for the call, and verify that a non-empty AEC dump file
  62 // exists. The AEC dump is enabled through webrtc-internals. The HTML and
  63 // Javascript is bypassed since it would trigger a file picker dialog. Instead,
  64 // the dialog callback FileSelected() is invoked directly. In fact, there's
  65 // never a webrtc-internals page opened at all since that's not needed.
  66 IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest, MAYBE_CallWithAecDump) {
  67   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
  68     LOG(INFO) << "Missing output devices: skipping test...";
  69     return;
  70   }
  71
  72   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
  73
  74   // We must navigate somewhere first so that the render process is created.
  75   NavigateToURL(shell(), GURL(""));
  76
  77   base::FilePath dump_file;
  78   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
  79   base::DeleteFile(dump_file, false);
  80
  81   // This fakes the behavior of another open tab with webrtc-internals, and
  82   // enabling AEC dump in that tab.
  83   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
  84
  85   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
  86   NavigateToURL(shell(), url);
  87   ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
  88
  89   EXPECT_FALSE(base::PathExists(dump_file));
  90
  91   // Add file extensions that we expect to be added. The dump name will be
  92   // <temporary path>.<render process id>.<consumer id>, for example
  93   // "/tmp/.com.google.Chrome.Z6UC3P.12345.0".
  94   base::ProcessId render_process_id = base::kNullProcessId;
  95   EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
  96   dump_file = dump_file.AddExtension(IntToStringType(render_process_id))
  97                        .AddExtension(IntToStringType(kExpectedConsumerId));
  98
  99   EXPECT_TRUE(base::PathExists(dump_file));
 100   int64 file_size = 0;
 101   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
 102   EXPECT_GT(file_size, 0);
 103
 104   base::DeleteFile(dump_file, false);
 105 }
 106
 107 // TODO(grunell): Add test for multiple dumps when re-use of
 108 // MediaStreamAudioProcessor in AudioCapturer has been removed.
 109
 110 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 111 // Timing out on ARM linux bot: http://crbug.com/238490
 112 #define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
 113 #elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
 114 // Renderer crashes under Android ASAN: https://crbug.com/408496.
 115 #define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
 116 #else
 117 #define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled
 118 #endif
 119
 120 // As above, but enable and disable dump before starting a call. The file should
 121 // be created, but should be empty.
 122 IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest,
 123                        MAYBE_CallWithAecDumpEnabledThenDisabled) {
 124   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
 125     LOG(INFO) << "Missing output devices: skipping test...";
 126     return;
 127   }
 128
 129   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 130
 131   // We must navigate somewhere first so that the render process is created.
 132   NavigateToURL(shell(), GURL(""));
 133
 134   base::FilePath dump_file;
 135   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 136   base::DeleteFile(dump_file, false);
 137
 138   // This fakes the behavior of another open tab with webrtc-internals, and
 139   // enabling AEC dump in that tab, then disabling it.
 140   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
 141   WebRTCInternals::GetInstance()->DisableAecDump();
 142
 143   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
 144   NavigateToURL(shell(), url);
 145   ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 146
 147   // Add file extensions that we expect to be added.
 148   base::ProcessId render_process_id = base::kNullProcessId;
 149   EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
 150   dump_file = dump_file.AddExtension(IntToStringType(render_process_id))
 151                        .AddExtension(IntToStringType(kExpectedConsumerId));
 152
 153   EXPECT_FALSE(base::PathExists(dump_file));
 154
 155   base::DeleteFile(dump_file, false);
 156 }
 157
 158 // Timing out on ARM linux bot: http://crbug.com/238490
 159 // Renderer crashes under Android ASAN: https://crbug.com/408496.
 160 // Flaky on XP and Mac: http://crbug.com/425034.
 161 IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest, DISABLED_TwoCallsWithAecDump) {
 162   if (OnWinXp()) {
 163     LOG(INFO) << "Disabled on Win XP: skipping test...";
 164     return;
 165   }
 166   if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
 167     LOG(INFO) << "Missing output devices: skipping test...";
 168     return;
 169   }
 170
 171   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 172
 173   // We must navigate somewhere first so that the render process is created.
 174   NavigateToURL(shell(), GURL(""));
 175
 176   // Create a second window.
 177   Shell* shell2 = CreateBrowser();
 178   NavigateToURL(shell2, GURL(""));
 179
 180   base::FilePath dump_file;
 181   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 182   base::DeleteFile(dump_file, false);
 183
 184   // This fakes the behavior of another open tab with webrtc-internals, and
 185   // enabling AEC dump in that tab.
 186   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
 187
 188   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
 189
 190   NavigateToURL(shell(), url);
 191   NavigateToURL(shell2, url);
 192   ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 193   std::string result;
 194   EXPECT_TRUE(ExecuteScriptAndExtractString(
 195       shell2->web_contents(),
 196       "call({video: true, audio: true});",
 197       &result));
 198   ASSERT_STREQ("OK", result.c_str());
 199
 200   EXPECT_FALSE(base::PathExists(dump_file));
 201
 202   RenderProcessHost::iterator it =
 203       content::RenderProcessHost::AllHostsIterator();
 204   for (; !it.IsAtEnd(); it.Advance()) {
 205     base::ProcessId render_process_id =
 206         base::GetProcId(it.GetCurrentValue()->GetHandle());
 207     EXPECT_NE(base::kNullProcessId, render_process_id);
 208
 209     // Add file extensions that we expect to be added.
 210     base::FilePath unique_dump_file =
 211         dump_file.AddExtension(IntToStringType(render_process_id))
 212                  .AddExtension(IntToStringType(kExpectedConsumerId));
 213
 214     EXPECT_TRUE(base::PathExists(unique_dump_file));
 215     int64 file_size = 0;
 216     EXPECT_TRUE(base::GetFileSize(unique_dump_file, &file_size));
 217     EXPECT_GT(file_size, 0);
 218
 219     base::DeleteFile(unique_dump_file, false);
 220   }
 221 }
 222
 223 }  // namespace content