Implement writing mic audio input data to file for debugging purposes.

content/browser/media/webrtc_aecdump_browsertest.cc

Index: content/browser/media/webrtc_aecdump_browsertest.cc
diff --git a/content/browser/media/webrtc_aecdump_browsertest.cc b/content/browser/media/webrtc_aecdump_browsertest.cc
deleted file mode 100644
index 212cc880b9bcfff4b43d6d65e93714afd4d7be45..0000000000000000000000000000000000000000
--- a/content/browser/media/webrtc_aecdump_browsertest.cc
+++ /dev/null
@@ -1,223 +0,0 @@
-// 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 "media/audio/audio_manager.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() {}
-  ~WebRtcAecDumpBrowserTest() override {}
-};
-
-#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) {
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  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);
-  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) {
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  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);
-  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);
-}
-
-// Timing out on ARM linux bot: http://crbug.com/238490
-// Renderer crashes under Android ASAN: https://crbug.com/408496.
-// Flaky on XP and Mac: http://crbug.com/425034.
-IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest, DISABLED_TwoCallsWithAecDump) {
-  if (OnWinXp()) {
-    LOG(INFO) << "Disabled on Win XP: skipping test...";
-    return;
-  }
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  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