Add interface to signal a network down event.

- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -987,15 +987,25 @@ void ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
         (*it)->rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms);
         return;
       }
-      it++;
+      ++it;
     }
   } else {
-    bool have_child_modules(child_modules_.empty() ? false : true);
+    bool have_child_modules = !child_modules_.empty();
     if (!have_child_modules) {
       // Don't send from default module.
       if (SendingMedia() && ssrc == rtp_sender_.SSRC()) {
         rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms);
       }
+    } else {
+      CriticalSectionScoped lock(critical_section_module_ptrs_.get());
+      std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
+      while (it != child_modules_.end()) {
+        if ((*it)->SendingMedia() && ssrc == (*it)->rtp_sender_.SSRC()) {
+          (*it)->rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms);
+          return;
+        }
+        ++it;
+      }
     }
   }
 }