Cleanup: move content/browser/media/webrtc/ to content/browser/webrtc

content/browser/media/webrtc/webrtc_audio_debug_recordings_browsertest.cc

Index: content/browser/media/webrtc/webrtc_audio_debug_recordings_browsertest.cc
diff --git a/content/browser/media/webrtc/webrtc_audio_debug_recordings_browsertest.cc b/content/browser/media/webrtc/webrtc_audio_debug_recordings_browsertest.cc
deleted file mode 100644
index 0a9de6f34c0250a7f7c8c4502f749d53f3f0ac23..0000000000000000000000000000000000000000
--- a/content/browser/media/webrtc/webrtc_audio_debug_recordings_browsertest.cc
+++ /dev/null
@@ -1,279 +0,0 @@
-// Copyright 2015 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 <stdint.h>
-
-#include "base/process/process_handle.h"
-#include "base/strings/string_number_conversions.h"
-#include "base/strings/stringprintf.h"
-#include "build/build_config.h"
-#include "content/browser/media/webrtc/webrtc_content_browsertest_base.h"
-#include "content/browser/media/webrtc/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 "media/audio/audio_manager.h"
-#include "net/test/embedded_test_server/embedded_test_server.h"
-
-#if defined(OS_WIN)
-#define IntToStringType base::IntToString16
-#else
-#define IntToStringType base::IntToString
-#endif
-
-namespace {
-
-const int kExpectedConsumerId = 1;
-const int kExpectedStreamId = 1;
-
-// 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();
-}
-
-// Get the expected AEC dump file name. The name will be
-// <temporary path>.<render process id>.aec_dump.<consumer id>, for example
-// "/tmp/.com.google.Chrome.Z6UC3P.12345.aec_dump.1".
-base::FilePath GetExpectedAecDumpFileName(const base::FilePath& base_file,
-                                          int render_process_id) {
-  return base_file.AddExtension(IntToStringType(render_process_id))
-                  .AddExtension(FILE_PATH_LITERAL("aec_dump"))
-                  .AddExtension(IntToStringType(kExpectedConsumerId));
-}
-
-// Get the expected input audio file name. The name will be
-// <temporary path>.<render process id>.source_input.<stream id>.pcm, for
-// example "/tmp/.com.google.Chrome.Z6UC3P.12345.source_input.1.pcm".
-base::FilePath GetExpectedInputAudioFileName(const base::FilePath& base_file,
-                                             int render_process_id) {
-  return base_file.AddExtension(IntToStringType(render_process_id))
-                  .AddExtension(FILE_PATH_LITERAL("source_input"))
-                  .AddExtension(IntToStringType(kExpectedStreamId))
-                  .AddExtension(FILE_PATH_LITERAL("wav"));
-}
-
-}  // namespace
-
-namespace content {
-
-class WebRtcAudioDebugRecordingsBrowserTest : public WebRtcContentBrowserTest {
- public:
-  WebRtcAudioDebugRecordingsBrowserTest() {
-    // Automatically grant device permission.
-    AppendUseFakeUIForMediaStreamFlag();
-  }
-  ~WebRtcAudioDebugRecordingsBrowserTest() override {}
-};
-
-#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
-// Timing out on ARM linux bot: http://crbug.com/238490
-#define MAYBE_CallWithAudioDebugRecordings DISABLED_CallWithAudioDebugRecordings
-#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
-// Renderer crashes under Android ASAN: https://crbug.com/408496.
-#define MAYBE_CallWithAudioDebugRecordings DISABLED_CallWithAudioDebugRecordings
-#elif defined(OS_ANDROID)
-// Renderer crashes on Android M. https://crbug.com/535728.
-#define MAYBE_CallWithAudioDebugRecordings DISABLED_CallWithAudioDebugRecordings
-#else
-#define MAYBE_CallWithAudioDebugRecordings CallWithAudioDebugRecordings
-#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(WebRtcAudioDebugRecordingsBrowserTest,
-                       MAYBE_CallWithAudioDebugRecordings) {
-  bool prev = base::ThreadRestrictions::SetIOAllowed(true);
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(embedded_test_server()->Start());
-
-  // We must navigate somewhere first so that the render process is created.
-  NavigateToURL(shell(), GURL(""));
-
-  base::FilePath base_file;
-  ASSERT_TRUE(CreateTemporaryFile(&base_file));
-  base::DeleteFile(base_file, false);
-
-  // This fakes the behavior of another open tab with webrtc-internals, and
-  // enabling AEC dump in that tab.
-  WebRTCInternals::GetInstance()->FileSelected(base_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(base_file));
-
-  // Verify that the expected AEC dump file exists and contains some data.
-  base::ProcessId render_process_id = base::kNullProcessId;
-  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
-  base::FilePath aec_dump_file = GetExpectedAecDumpFileName(base_file,
-                                                            render_process_id);
-
-  EXPECT_TRUE(base::PathExists(aec_dump_file));
-  int64_t file_size = 0;
-  EXPECT_TRUE(base::GetFileSize(aec_dump_file, &file_size));
-  EXPECT_GT(file_size, 0);
-
-  base::DeleteFile(aec_dump_file, false);
-
-  // Verify that the expected input audio file exists and contains some data.
-  base::FilePath input_audio_file =
-      GetExpectedInputAudioFileName(base_file, render_process_id);
-
-  EXPECT_TRUE(base::PathExists(input_audio_file));
-  file_size = 0;
-  EXPECT_TRUE(base::GetFileSize(input_audio_file, &file_size));
-  EXPECT_GT(file_size, 0);
-
-  base::DeleteFile(input_audio_file, false);
-  base::ThreadRestrictions::SetIOAllowed(prev);
-}
-
-// 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_CallWithAudioDebugRecordingsEnabledThenDisabled DISABLED_CallWithAudioDebugRecordingsEnabledThenDisabled
-#elif defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
-// Renderer crashes under Android ASAN: https://crbug.com/408496.
-#define MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled DISABLED_CallWithAudioDebugRecordingsEnabledThenDisabled
-#else
-#define MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled CallWithAudioDebugRecordingsEnabledThenDisabled
-#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(WebRtcAudioDebugRecordingsBrowserTest,
-                       MAYBE_CallWithAudioDebugRecordingsEnabledThenDisabled) {
-  bool prev = base::ThreadRestrictions::SetIOAllowed(true);
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(embedded_test_server()->Start());
-
-  // We must navigate somewhere first so that the render process is created.
-  NavigateToURL(shell(), GURL(""));
-
-  base::FilePath base_file;
-  ASSERT_TRUE(CreateTemporaryFile(&base_file));
-  base::DeleteFile(base_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(base_file, -1, NULL);
-  WebRTCInternals::GetInstance()->DisableAudioDebugRecordings();
-
-  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
-  NavigateToURL(shell(), url);
-  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
-
-  // Verify that the expected AEC dump file doesn't exist.
-  base::ProcessId render_process_id = base::kNullProcessId;
-  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
-  base::FilePath aec_dump_file = GetExpectedAecDumpFileName(base_file,
-                                                            render_process_id);
-  EXPECT_FALSE(base::PathExists(aec_dump_file));
-  base::DeleteFile(aec_dump_file, false);
-
-  // Verify that the expected input audio file doesn't exist.
-  base::FilePath input_audio_file =
-      GetExpectedInputAudioFileName(base_file, render_process_id);
-  EXPECT_FALSE(base::PathExists(input_audio_file));
-  base::DeleteFile(input_audio_file, false);
-  base::ThreadRestrictions::SetIOAllowed(prev);
-}
-
-// Timing out on ARM linux bot: http://crbug.com/238490
-// Renderer crashes under Android ASAN: https://crbug.com/408496.
-IN_PROC_BROWSER_TEST_F(WebRtcAudioDebugRecordingsBrowserTest,
-                       DISABLED_TwoCallsWithAudioDebugRecordings) {
-  bool prev = base::ThreadRestrictions::SetIOAllowed(true);
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(embedded_test_server()->Start());
-
-  // 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 base_file;
-  ASSERT_TRUE(CreateTemporaryFile(&base_file));
-  base::DeleteFile(base_file, false);
-
-  // This fakes the behavior of another open tab with webrtc-internals, and
-  // enabling AEC dump in that tab.
-  WebRTCInternals::GetInstance()->FileSelected(base_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, "call({video: true, audio: true});", &result));
-  ASSERT_STREQ("OK", result.c_str());
-
-  EXPECT_FALSE(base::PathExists(base_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);
-
-    // Verify that the expected AEC dump file exists and contains some data.
-    base::FilePath aec_dump_file =
-        GetExpectedAecDumpFileName(base_file, render_process_id);
-
-    EXPECT_TRUE(base::PathExists(aec_dump_file));
-    int64_t file_size = 0;
-    EXPECT_TRUE(base::GetFileSize(aec_dump_file, &file_size));
-    EXPECT_GT(file_size, 0);
-
-    base::DeleteFile(aec_dump_file, false);
-
-    // Verify that the expected input audio file exists and contains some data.
-    base::FilePath input_audio_file =
-        GetExpectedInputAudioFileName(base_file, render_process_id);
-
-    EXPECT_TRUE(base::PathExists(input_audio_file));
-    file_size = 0;
-    EXPECT_TRUE(base::GetFileSize(input_audio_file, &file_size));
-    EXPECT_GT(file_size, 0);
-
-    base::DeleteFile(input_audio_file, false);
-  }
-  base::ThreadRestrictions::SetIOAllowed(prev);
-}
-
-}  // namespace content