Index: content/browser/media/webrtc_internals.cc |
diff --git a/content/browser/media/webrtc_internals.cc b/content/browser/media/webrtc_internals.cc |
index ed0b6b53aa476883295e54d661a8704b63c993d0..713da32507f879fb08c0e447027b89026882c875 100644 |
--- a/content/browser/media/webrtc_internals.cc |
+++ b/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; |
} |