Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/statscollector.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 12 matching lines @@
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/app/webrtc/statscollector.h"
 
 #include <utility>
 #include <vector>
 
+#include "talk/app/webrtc/peerconnection.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/timing.h"
 
 using rtc::scoped_ptr;
 
 namespace webrtc {
 namespace {
@@ skipping 306 matching lines @@
     case rtc::ADAPTER_TYPE_VPN:
       return STATSREPORT_ADAPTER_TYPE_VPN;
     case rtc::ADAPTER_TYPE_LOOPBACK:
       return STATSREPORT_ADAPTER_TYPE_LOOPBACK;
     default:
       RTC_DCHECK(false);
       return "";
   }
 }
 
-StatsCollector::StatsCollector(WebRtcSession* session)
-    : session_(session),
-      stats_gathering_started_(0) {
+StatsCollector::StatsCollector(PeerConnection* pc, WebRtcSession* session)
+    : pc_(pc), session_(session), stats_gathering_started_(0) {
+  RTC_DCHECK(pc_);
   RTC_DCHECK(session_);
 }
 
 StatsCollector::~StatsCollector() {
   RTC_DCHECK(session_->signaling_thread()->IsCurrent());
 }
 
 double StatsCollector::GetTimeNow() {
   return rtc::Timing::WallTimeNow() * rtc::kNumMillisecsPerSec;
 }
@@ skipping 452 matching lines @@
     ExtractStats(
         video_info.bw_estimations[0], stats_gathering_started_, level, report);
   }
 }
 
 void StatsCollector::ExtractDataInfo() {
   RTC_DCHECK(session_->signaling_thread()->IsCurrent());
 
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
-  for (const auto& dc :
-           session_->mediastream_signaling()->sctp_data_channels()) {
+  for (const auto& dc : pc_->sctp_data_channels()) {
     StatsReport::Id id(StatsReport::NewTypedIntId(
         StatsReport::kStatsReportTypeDataChannel, dc->id()));
     StatsReport* report = reports_.ReplaceOrAddNew(id);
     report->set_timestamp(stats_gathering_started_);
     report->AddString(StatsReport::kStatsValueNameLabel, dc->label());
     report->AddInt(StatsReport::kStatsValueNameDataChannelId, dc->id());
     report->AddString(StatsReport::kStatsValueNameProtocol, dc->protocol());
     report->AddString(StatsReport::kStatsValueNameState,
                       DataChannelInterface::DataStateString(dc->state()));
   }
@@ skipping 87 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc