Move Ownership of RtpModules to VideoSendStream.

webrtc/video/payload_router.cc

Index: webrtc/video/payload_router.cc
diff --git a/webrtc/video/payload_router.cc b/webrtc/video/payload_router.cc
index 968d82df62a0fa2c7153f49ca70202ea6c05090a..d466e41bb01bb44f0621fb2844b99ac2751c83cf 100644
--- a/webrtc/video/payload_router.cc
+++ b/webrtc/video/payload_router.cc
@@ -16,8 +16,10 @@
 
 namespace webrtc {
 
-PayloadRouter::PayloadRouter()
-    : active_(false), num_sending_modules_(0) {}
+PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules)
+    : active_(false), num_sending_modules_(1), rtp_modules_(rtp_modules) {
+  UpdateModuleSendingState();

[mflodman, 2016/04/15 08:25:42]

This means the RTP module will be in a sending state from the start, but I
assume in practice there will always be a call to SetSendingModules very soon
after anyway and this is fine.

+}
 
 PayloadRouter::~PayloadRouter() {}
 
@@ -26,12 +28,6 @@ size_t PayloadRouter::DefaultMaxPayloadLength() {
   return IP_PACKET_SIZE - kIpUdpSrtpLength;
 }
 
-void PayloadRouter::Init(
-    const std::vector<RtpRtcp*>& rtp_modules) {
-  RTC_DCHECK(rtp_modules_.empty());
-  rtp_modules_ = rtp_modules;
-}
-
 void PayloadRouter::set_active(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)