Bug 1358224 - pt 3 - fix leak in RTPHeaderExtension's rid char buffer. r=drno

Turns out since Firefox doesn't receive simulcast streams, we never
noticed this leak.  Convert RTPHeaderExtension.rid from a char* to
rtc::scoped_ptr<char[]> so it gets deleted properly.  This also
requires a new copy constructor and assignment operator.

MozReview-Commit-ID: Jh4Gp4dAl9g

--HG--
extra : rebase_source : 8c1081fecd6e56a8f932af54fbd294adb85866f5

media/webrtc/signaling/src/mediapipeline/MediaPipelineFilter.cpp

@@ -33,7 +33,7 @@ bool MediaPipelineFilter::Filter(const webrtc::RTPHeader& header,
 
   if (header.extension.hasRID &&
       remote_rid_set_.size() &&
-      remote_rid_set_.count(header.extension.rid)) {
+      remote_rid_set_.count(header.extension.rid.get())) {
     return true;
   }
 

media/webrtc/trunk/webrtc/common_types.cc

@@ -33,7 +33,35 @@ RTPHeaderExtension::RTPHeaderExtension()
       hasVideoRotation(false),
       videoRotation(0),
       hasRID(false),
-      rid(NULL) {
+      rid(nullptr) {
+}
+
+RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& rhs) {
+  *this = rhs;
+}
+
+RTPHeaderExtension&
+RTPHeaderExtension::operator=(const RTPHeaderExtension& rhs) {
+  hasTransmissionTimeOffset = rhs.hasTransmissionTimeOffset;
+  transmissionTimeOffset = rhs.transmissionTimeOffset;
+  hasAbsoluteSendTime = rhs.hasAbsoluteSendTime;
+  absoluteSendTime = rhs.absoluteSendTime;
+  hasTransportSequenceNumber = rhs.hasTransportSequenceNumber;
+  transportSequenceNumber = rhs.transportSequenceNumber;
+
+  hasAudioLevel = rhs.hasAudioLevel;
+  voiceActivity = rhs.voiceActivity;
+  audioLevel = rhs.audioLevel;
+
+  hasVideoRotation = rhs.hasVideoRotation;
+  videoRotation = rhs.videoRotation;
+
+  hasRID = rhs.hasRID;
+  if (rhs.rid) {
+    rid.reset(new char[strlen(rhs.rid.get())+1]);
+    strcpy(rid.get(), rhs.rid.get());
+  }
+  return *this;
 }
 
 RTPHeader::RTPHeader()

media/webrtc/trunk/webrtc/common_types.h

@@ -17,6 +17,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 #if defined(_MSC_VER)
@@ -830,6 +831,8 @@ struct PacketTime {
 
 struct RTPHeaderExtension {
   RTPHeaderExtension();
+  RTPHeaderExtension(const RTPHeaderExtension& rhs);
+  RTPHeaderExtension& operator=(const RTPHeaderExtension& rhs);
 
   bool hasTransmissionTimeOffset;
   int32_t transmissionTimeOffset;
@@ -852,7 +855,7 @@ struct RTPHeaderExtension {
 
   // RID values for simulcast; see draft-roach-avtext-rid
   bool hasRID;
-  char *rid; // UTF8 string
+  rtc::scoped_ptr<char[]> rid; // UTF8 string
 };
 
 struct RTPHeader {

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

@@ -223,10 +223,10 @@ bool RtpReceiverImpl::IncomingRtpPacket(
     last_received_payload_length_ = payload_data_length;
     // RID rarely if ever changes
     if (rtp_header.extension.hasRID &&
-        (!rid_ || strcmp(rtp_header.extension.rid, rid_) != 0)) {
+        (!rid_ || strcmp(rtp_header.extension.rid.get(), rid_) != 0)) {
       delete [] rid_;
-      rid_ = new char[strlen(rtp_header.extension.rid)+1];
-      strcpy(rid_, rtp_header.extension.rid);
+      rid_ = new char[strlen(rtp_header.extension.rid.get())+1];
+      strcpy(rid_, rtp_header.extension.rid.get());
     }
     if (in_order) {
       if (last_received_timestamp_ != rtp_header.timestamp) {

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -444,12 +444,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
             return;
           }
 
-          // TODO(jesup) - avoid allocating on each packet - high watermark the
-          // RID buffer?
-          char* ptrRID = new char[len + 2];
-          memcpy(ptrRID, ptr, len + 1);
-          ptrRID[len + 1] = '\0';
-          header->extension.rid = ptrRID;
+          header->extension.rid.reset(new char[len + 2]);
+          memcpy(header->extension.rid.get(), ptr, len + 1);
+          header->extension.rid.get()[len + 1] = '\0';
           header->extension.hasRID = true;
           break;
         }