Start and stop RTC event logs from private extension API.

chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_apitest.cc

 // Copyright 2013 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 <stddef.h>
 
 #include "base/json/json_writer.h"
 #include "base/strings/string_split.h"
 #include "base/strings/stringprintf.h"
 #include "chrome/browser/browser_process.h"
@@ skipping 15 matching lines @@
 using extensions::WebrtcLoggingPrivateSetMetaDataFunction;
 using extensions::WebrtcLoggingPrivateStartFunction;
 using extensions::WebrtcLoggingPrivateStartRtpDumpFunction;
 using extensions::WebrtcLoggingPrivateStopFunction;
 using extensions::WebrtcLoggingPrivateStopRtpDumpFunction;
 using extensions::WebrtcLoggingPrivateStoreFunction;
 using extensions::WebrtcLoggingPrivateUploadFunction;
 using extensions::WebrtcLoggingPrivateUploadStoredFunction;
 using extensions::WebrtcLoggingPrivateStartAudioDebugRecordingsFunction;
 using extensions::WebrtcLoggingPrivateStopAudioDebugRecordingsFunction;
+using extensions::WebrtcLoggingPrivateStartRtcEventLoggingFunction;
+using extensions::WebrtcLoggingPrivateStopRtcEventLoggingFunction;
 
 namespace utils = extension_function_test_utils;
 
 namespace {
 
 static const char kTestLoggingSessionIdKey[] = "app_session_id";
 static const char kTestLoggingSessionIdValue[] = "0123456789abcdef";
 static const char kTestLoggingUrl[] = "dummy url string";
 
 std::string ParamsToString(const base::ListValue& parameters) {
@@ skipping 128 matching lines @@
         params, true);
   }
 
   bool StopAudioDebugRecordings() {
     base::ListValue params;
     AppendTabIdAndUrl(&params);
     return RunFunction<WebrtcLoggingPrivateStopAudioDebugRecordingsFunction>(
         params, true);
   }
 
+  bool StartRtcEventLogging(int seconds) {
+    base::ListValue params;
+    AppendTabIdAndUrl(&params);
+    params.AppendInteger(seconds);
+    return RunFunction<WebrtcLoggingPrivateStartRtcEventLoggingFunction>(params,
+                                                                         true);
+  }
+
+  bool StopRtcEventLogging() {
+    base::ListValue params;
+    AppendTabIdAndUrl(&params);
+    return RunFunction<WebrtcLoggingPrivateStopRtcEventLoggingFunction>(params,
+                                                                        true);
+  }
+
  private:
   scoped_refptr<Extension> extension_;
 };
 
 // Helper class to temporarily tell the uploader to save the multipart buffer to
 // a test string instead of uploading.
 class ScopedOverrideUploadBuffer {
  public:
   ScopedOverrideUploadBuffer() {
     g_browser_process->webrtc_log_uploader()->
@@ skipping 242 matching lines @@
   ASSERT_TRUE(StartAudioDebugRecordings(0));
   ASSERT_TRUE(StopAudioDebugRecordings());
 }
 
 IN_PROC_BROWSER_TEST_F(WebrtcLoggingPrivateApiTest,
                        TestStartTimedAudioDebugRecordings) {
   base::CommandLine::ForCurrentProcess()->AppendSwitch(
       switches::kEnableAudioDebugRecordingsFromExtension);
   ASSERT_TRUE(StartAudioDebugRecordings(1));
 }
+
+IN_PROC_BROWSER_TEST_F(WebrtcLoggingPrivateApiTest,
+                       TestStartStopRtcEventLogging) {
+  // TODO(terelius): These tests are missing verification of the actual RTC
+  // event log data. Maybe this could be incorporated into crbug.com/569957,
+  // and solved at the same time?
+  base::CommandLine::ForCurrentProcess()->AppendSwitch(
+      switches::kEnableRtcEventLoggingFromExtension);
+  ASSERT_TRUE(StartRtcEventLogging(0));
+  ASSERT_TRUE(StopRtcEventLogging());
+}
+
+IN_PROC_BROWSER_TEST_F(WebrtcLoggingPrivateApiTest,
+                       TestStartTimedRtcEventLogging) {
+  base::CommandLine::ForCurrentProcess()->AppendSwitch(
+      switches::kEnableRtcEventLoggingFromExtension);
+  ASSERT_TRUE(StartRtcEventLogging(1));
+}