Remove the RTP recording related code from webrtc-internals.

The code is dead and we have no plan to implement RTP recording in the near term.
So removing to avoid confusion.

BUG=

Review URL: https://codereview.chromium.org/129533003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@243876 0039d316-1c4b-4281-b951-d872f2087c98

content/browser/media/webrtc_internals.cc

@@ -35,8 +35,7 @@ static base::ListValue* EnsureLogList(base::DictionaryValue* dict) {
 }  // 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);
@@ -230,20 +229,6 @@ void WebRTCInternals::UpdateObserver(WebRTCInternalsUIObserver* observer) {
   }
 }
 
-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);
@@ -354,21 +339,11 @@ void WebRTCInternals::OnRendererExit(int render_process_id) {
   }
 }
 
-// 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;
 }
 

content/browser/media/webrtc_internals.h

@@ -85,10 +85,6 @@ class CONTENT_EXPORT WebRTCInternals : public BrowserChildProcessObserver,
   // Sends all update data to |observer|.
   void UpdateObserver(WebRTCInternalsUIObserver* observer);
 
-  // Tells the renderer processes to start or stop recording RTP packets.
-  void StartRtpRecording();
-  void StopRtpRecording();
-
   // Enables or disables AEC dump (diagnostic echo canceller recording).
   void EnableAecDump(content::WebContents* web_contents);
   void DisableAecDump();
@@ -131,8 +127,6 @@ class CONTENT_EXPORT WebRTCInternals : public BrowserChildProcessObserver,
   // Called when a renderer exits (including crashes).
   void OnRendererExit(int render_process_id);
 
-  void SendRtpRecordingUpdate();
-
   ObserverList<WebRTCInternalsUIObserver> observers_;
 
   // |peer_connection_data_| is a list containing all the PeerConnection
@@ -161,8 +155,6 @@ class CONTENT_EXPORT WebRTCInternals : public BrowserChildProcessObserver,
 
   NotificationRegistrar registrar_;
 
-  bool is_recording_rtp_;
-
   // For managing select file dialog.
   scoped_refptr<ui::SelectFileDialog> select_file_dialog_;
 

content/browser/media/webrtc_internals_message_handler.cc

@@ -27,13 +27,6 @@ void WebRTCInternalsMessageHandler::RegisterMessages() {
       base::Bind(&WebRTCInternalsMessageHandler::OnGetAllStats,
                  base::Unretained(this)));
 
-  web_ui()->RegisterMessageCallback("startRtpRecording",
-      base::Bind(&WebRTCInternalsMessageHandler::OnStartRtpRecording,
-                 base::Unretained(this)));
-
-  web_ui()->RegisterMessageCallback("stopRtpRecording",
-      base::Bind(&WebRTCInternalsMessageHandler::OnStopRtpRecording,
-                 base::Unretained(this)));
   web_ui()->RegisterMessageCallback("enableAecRecording",
       base::Bind(&WebRTCInternalsMessageHandler::OnSetAecRecordingEnabled,
                  base::Unretained(this), true));
@@ -56,16 +49,6 @@ void WebRTCInternalsMessageHandler::OnGetAllStats(
   }
 }
 
-void WebRTCInternalsMessageHandler::OnStartRtpRecording(
-    const base::ListValue* /* unused_list */) {
-  WebRTCInternals::GetInstance()->StartRtpRecording();
-}
-
-void WebRTCInternalsMessageHandler::OnStopRtpRecording(
-    const base::ListValue* /* unused_list */) {
-  WebRTCInternals::GetInstance()->StopRtpRecording();
-}
-
 void WebRTCInternalsMessageHandler::OnSetAecRecordingEnabled(
     bool enable, const base::ListValue* /* unused_list */) {
   if (enable)

content/browser/media/webrtc_internals_message_handler.h

@@ -33,8 +33,6 @@ class WebRTCInternalsMessageHandler : public WebUIMessageHandler,
  private:
   // Javascript message handler.
   void OnGetAllStats(const base::ListValue* list);
-  void OnStartRtpRecording(const base::ListValue* list);
-  void OnStopRtpRecording(const base::ListValue* list);
   void OnSetAecRecordingEnabled(bool enable, const base::ListValue* list);
   void OnDOMLoadDone(const base::ListValue* list);
 

content/browser/resources/media/dump_creator.js

@@ -4,9 +4,7 @@
 
 
 /**
- * Provides the UI to start and stop RTP recording, forwards the start/stop
- * commands to Chrome, and updates the UI based on dump updates. Also provides
- * creating a file containing all PeerConnection updates and stats.
+ * Provides the UI for dump creation.
  */
 var DumpCreator = (function() {
   /**
@@ -15,30 +13,6 @@ var DumpCreator = (function() {
    * @constructor
    */
   function DumpCreator(containerElement) {
-    /**
-     * True if the RTP packets are being recorded.
-     * @type {bool}
-     * @private
-     */
-    this.recording_ = false;
-
-    /**
-     * @type {!Object.<string>}
-     * @private
-     * @const
-     */
-    this.StatusStrings_ = {
-      NOT_STARTED: 'not started.',
-      RECORDING: 'recording...',
-    },
-
-    /**
-     * The status of dump creation.
-     * @type {string}
-     * @private
-     */
-    this.status_ = this.StatusStrings_.NOT_STARTED;
-
     /**
      * The root element of the dump creation UI.
      * @type {Element}
@@ -54,8 +28,7 @@ var DumpCreator = (function() {
     var content = document.createElement('div');
     this.root_.appendChild(content);
 
-    content.innerHTML = '<button disabled></button> Status: <span></span>' +
-        '<div><a><button>' +
+    content.innerHTML = '<div><a><button>' +
         'Download the PeerConnection updates and stats data' +
         '</button></a></div>' +
         '<p><label><input type=checkbox>' +
@@ -67,14 +40,10 @@ var DumpCreator = (function() {
         ' calls. When the box is unchecked or this page is closed, this' +
         ' recording functionality will be disabled.</p>';
 
-    content.getElementsByTagName('button')[0].addEventListener(
-        'click', this.onRtpToggled_.bind(this));
     content.getElementsByTagName('a')[0].addEventListener(
         'click', this.onDownloadData_.bind(this));
     content.getElementsByTagName('input')[0].addEventListener(
         'click', this.onAecRecordingChanged_.bind(this));
-
-    this.updateDisplay_();
   }
 
   DumpCreator.prototype = {
@@ -97,24 +66,6 @@ var DumpCreator = (function() {
       // The default action of the anchor will download the URL.
     },
 
-    /**
-     * Handles the event of toggling the rtp recording state.
-     *
-     * @private
-     */
-    onRtpToggled_: function() {
-      if (this.recording_) {
-        this.recording_ = false;
-        this.status_ = this.StatusStrings_.NOT_STARTED;
-        chrome.send('stopRtpRecording');
-      } else {
-        this.recording_ = true;
-        this.status_ = this.StatusStrings_.RECORDING;
-        chrome.send('startRtpRecording');
-      }
-      this.updateDisplay_();
-    },
-
     /**
      * Handles the event of toggling the AEC recording state.
      *
@@ -128,34 +79,6 @@ var DumpCreator = (function() {
         chrome.send('disableAecRecording');
       }
     },
-
-    /**
-     * Updates the UI based on the recording status.
-     *
-     * @private
-     */
-    updateDisplay_: function() {
-      if (this.recording_) {
-        this.root_.getElementsByTagName('button')[0].textContent =
-            'Stop Recording RTP Packets';
-      } else {
-        this.root_.getElementsByTagName('button')[0].textContent =
-            'Start Recording RTP Packets';
-      }
-
-      this.root_.getElementsByTagName('span')[0].textContent = this.status_;
-    },
-
-    /**
-     * Set the status to the content of the update.
-     * @param {!Object} update
-     */
-    onUpdate: function(update) {
-      if (this.recording_) {
-        this.status_ = JSON.stringify(update);
-        this.updateDisplay_();
-      }
-    },
   };
   return DumpCreator;
 })();

content/browser/resources/media/webrtc_internals.js

@@ -264,16 +264,6 @@ function removeGetUserMediaForRenderer(data) {
   // TODO(jiayl): remove the getUserMedia info from the tabbed UI.
 }
 
-
-/**
- * Delegates to dumpCreator to update the recording status.
- * @param {!Object.<string>} update Key-value pairs describing the status of the
- *     RTP recording.
- */
-function updateDumpStatus(update) {
-  dumpCreator.onUpdate(update);
-}
-
 /**
  * Set
  */