Support multiple files for AEC dump.

content/browser/media/webrtc_browsertest.cc

 // Copyright (c) 2012 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/command_line.h"
 #include "base/file_util.h"
+#include "base/strings/string_number_conversions.h"
 #include "base/strings/stringprintf.h"
 #include "base/threading/platform_thread.h"
 #include "base/values.h"
 #include "content/browser/media/webrtc_internals.h"
 #include "content/browser/web_contents/web_contents_impl.h"
 #include "content/public/common/content_switches.h"
 #include "content/public/test/browser_test_utils.h"
 #include "content/public/test/content_browser_test_utils.h"
 #include "content/public/test/test_utils.h"
 #include "content/shell/browser/shell.h"
@@ skipping 343 matching lines @@
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioTrackUnmutingWorks(%s);", kUseLenientAudioChecking));
 }
 
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
   MakeTypicalPeerConnectionCall("callAndEnsureVideoTrackMutingWorks();");
 }
 
+#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
 #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
@@ skipping 11 matching lines @@
 
   // 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});");
 
+  // Get the ID for the render process host. There should only be one.
+  RenderProcessHost::iterator it(
+      content::RenderProcessHost::AllHostsIterator());
+  int render_process_host_id = it.GetCurrentValue()->GetID();
+  EXPECT_GE(render_process_host_id, 0);
+
+  // Add file extensions that we expect to be added.
+  static const int kExpectedConsumerId = 0;
+  dump_file = dump_file.AddExtension(IntToStringType(render_process_host_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
 #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_P(WebRtcBrowserTest,
@@ skipping 18 matching lines @@
 
   EXPECT_TRUE(base::PathExists(dump_file));
   int64 file_size = 0;
   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
   EXPECT_EQ(0, file_size);
 
   base::DeleteFile(dump_file, false);
 }
 
 }  // namespace content