Remove the RTP recording related code from webrtc-internals.

content/browser/media/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_internals.h"
 
 #include "base/command_line.h"
 #include "content/browser/media/webrtc_internals_ui_observer.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/browser/child_process_data.h"
@@ skipping 17 matching lines @@
     log = new base::ListValue();
     if (log)
       dict->Set("log", log);
   }
   return log;
 }
 
 }  // namespace
 
 WebRTCInternals::WebRTCInternals()
-    : is_recording_rtp_(false),
-      aec_dump_enabled_(false) {
+    : aec_dump_enabled_(false) {
   registrar_.Add(this, NOTIFICATION_RENDERER_PROCESS_TERMINATED,
                  NotificationService::AllBrowserContextsAndSources());
   BrowserChildProcessObserver::Add(this);
 // TODO(grunell): Shouldn't all the webrtc_internals* files be excluded from the
 // build if WebRTC is disabled?
 #if defined(ENABLE_WEBRTC)
   if (CommandLine::ForCurrentProcess()->HasSwitch(
           switches::kEnableWebRtcAecRecordings)) {
     aec_dump_enabled_ = true;
     aec_dump_file_path_ = CommandLine::ForCurrentProcess()->GetSwitchValuePath(
@@ skipping 173 matching lines @@
   if (peer_connection_data_.GetSize() > 0)
     observer->OnUpdate("updateAllPeerConnections", &peer_connection_data_);
 
   for (base::ListValue::iterator it = get_user_media_requests_.begin();
        it != get_user_media_requests_.end();
        ++it) {
     observer->OnUpdate("addGetUserMedia", *it);
   }
 }
 
-void WebRTCInternals::StartRtpRecording() {
-  if (!is_recording_rtp_) {
-    is_recording_rtp_ = true;
-    // TODO(justinlin): start RTP recording.
-  }
-}
-
-void WebRTCInternals::StopRtpRecording() {
-  if (is_recording_rtp_) {
-    is_recording_rtp_ = false;
-    // TODO(justinlin): stop RTP recording.
-  }
-}
-
 void WebRTCInternals::EnableAecDump(content::WebContents* web_contents) {
 #if defined(ENABLE_WEBRTC)
   select_file_dialog_ = ui::SelectFileDialog::Create(this, NULL);
   select_file_dialog_->SelectFile(
       ui::SelectFileDialog::SELECT_SAVEAS_FILE,
       base::string16(),
       aec_dump_file_path_,
       NULL,
       0,
       FILE_PATH_LITERAL(""),
@@ skipping 90 matching lines @@
     }
   }
 
   if (found_any && observers_.might_have_observers()) {
     base::DictionaryValue update;
     update.SetInteger("rid", render_process_id);
     SendUpdate("removeGetUserMediaForRenderer", &update);
   }
 }
 
-// TODO(justlin): Calls this method as necessary to update the recording status
-// UI.
-void WebRTCInternals::SendRtpRecordingUpdate() {
-  DCHECK(is_recording_rtp_);
-  base::DictionaryValue update;
-  // TODO(justinlin): Fill in |update| with values as appropriate.
-  SendUpdate("updateDumpStatus", &update);
-}
-
 void WebRTCInternals::ResetForTesting() {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::UI));
   observers_.Clear();
   peer_connection_data_.Clear();
   get_user_media_requests_.Clear();
-  is_recording_rtp_ = false;
   aec_dump_enabled_ = false;
 }
 
 }  // namespace content