Add unit test for WebRTCInternalsMessageHandler.

content/browser/media/webrtc/webrtc_internals.cc

 // Copyright (c) 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 "content/browser/media/webrtc/webrtc_internals.h"
 
 #include <stddef.h>
 
 #include "base/strings/string_number_conversions.h"
 #include "build/build_config.h"
@@ skipping 51 matching lines @@
 const std::string& WebRTCInternals::PendingUpdate::command() const {
   DCHECK(thread_checker_.CalledOnValidThread());
   return command_;
 }
 
 const base::Value* WebRTCInternals::PendingUpdate::value() const {
   DCHECK(thread_checker_.CalledOnValidThread());
   return value_.get();
 }
 
-WebRTCInternals::WebRTCInternals() : WebRTCInternals(500) {}
+WebRTCInternals::WebRTCInternals() : WebRTCInternals(500, true) {}
 
-WebRTCInternals::WebRTCInternals(int aggregate_updates_ms)
+WebRTCInternals::WebRTCInternals(int aggregate_updates_ms,
+                                 bool should_block_power_saving)
     : audio_debug_recordings_(false),
       event_log_recordings_(false),
       selecting_event_log_(false),
+      should_block_power_saving_(should_block_power_saving),
       aggregate_updates_ms_(aggregate_updates_ms),
       weak_factory_(this) {
 // TODO(grunell): Shouldn't all the webrtc_internals* files be excluded from the
 // build if WebRTC is disabled?
 #if defined(ENABLE_WEBRTC)
   audio_debug_recordings_file_path_ =
       GetContentClient()->browser()->GetDefaultDownloadDirectory();
   event_log_recordings_file_path_ = audio_debug_recordings_file_path_;
 
   if (audio_debug_recordings_file_path_.empty()) {
@@ skipping 402 matching lines @@
   event_log_recordings_ = true;
   for (RenderProcessHost::iterator i(
            content::RenderProcessHost::AllHostsIterator());
        !i.IsAtEnd(); i.Advance())
     i.GetCurrentValue()->StartWebRTCEventLog(event_log_recordings_file_path_);
 }
 #endif
 
 void WebRTCInternals::CreateOrReleasePowerSaveBlocker() {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
+  if (!should_block_power_saving_)
+    return;
 
   if (peer_connection_data_.empty() && power_save_blocker_) {
     DVLOG(1) << ("Releasing the block on application suspension since no "
                  "PeerConnections are active anymore.");
     power_save_blocker_.reset();
   } else if (!peer_connection_data_.empty() && !power_save_blocker_) {
     DVLOG(1) << ("Preventing the application from being suspended while one or "
                  "more PeerConnections are active.");
     power_save_blocker_.reset(new device::PowerSaveBlocker(
         device::PowerSaveBlocker::kPowerSaveBlockPreventAppSuspension,
         device::PowerSaveBlocker::kReasonOther,
         "WebRTC has active PeerConnections",
         BrowserThread::GetTaskRunnerForThread(BrowserThread::UI),
         BrowserThread::GetTaskRunnerForThread(BrowserThread::FILE)));
   }
 }
 
 void WebRTCInternals::ProcessPendingUpdates() {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   while (!pending_updates_.empty()) {
     const auto& update = pending_updates_.front();
     FOR_EACH_OBSERVER(WebRTCInternalsUIObserver,
                       observers_,
                       OnUpdate(update.command(), update.value()));
     pending_updates_.pop();
   }
 }
 
 }  // namespace content