From e4736eee200873837bf66ff757004971f377b712 Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Thu, 11 Jul 2013 08:28:35 +0000
Subject: [PATCH] Fixes a crash when sending SR reports from a sender only
 module.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 .../modules/rtp_rtcp/interface/receive_statistics.h | 13 +++++++++++++
 webrtc/modules/rtp_rtcp/source/rtcp_sender.cc       | 13 +++++--------
 webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc     |  2 +-
 webrtc/modules/rtp_rtcp/source/rtp_utility.cc       |  5 +++++
 webrtc/modules/rtp_rtcp/source/rtp_utility.h        |  2 ++
 5 files changed, 26 insertions(+), 9 deletions(-)

diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
index 4cc069a93f..07288c4dbf 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
@@ -69,5 +69,18 @@ class ReceiveStatistics : public Module {
   virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
 };
 
+class NullReceiveStatistics : public ReceiveStatistics {
+ public:
+  virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
+                                bool retransmitted, bool in_order) {}
+  virtual void GetActiveStatisticians(
+      StatisticianMap* statisticians) const { statisticians->clear(); }
+  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const {
+    return NULL;
+  }
+  virtual int32_t TimeUntilNextProcess() { return 0; }
+  virtual int32_t Process() { return 0; }
+};
+
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index c3bc8f5e46..de6eb0045f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -1689,14 +1689,12 @@ int RTCPSender::PrepareRTCP(
   // If the data does not fit in the packet we fill it as much as possible.
   int32_t buildVal = 0;
 
+  // We need to send our NTP even if we haven't received any reports.
+  _clock->CurrentNtp(NTPsec, NTPfrac);
   if (ShouldSendReportBlocks(rtcpPacketTypeFlags)) {
     ReceiveStatistics::StatisticianMap statisticians;
     receive_statistics_->GetActiveStatisticians(&statisticians);
-    if (statisticians.empty()) {
-      // We need to send our NTP even if we dont have received any
-      // reports.
-      _clock->CurrentNtp(NTPsec, NTPfrac);
-    } else {
+    if (!statisticians.empty()) {
       ReceiveStatistics::StatisticianMap::const_iterator it;
       int i;
       for (it = statisticians.begin(), i = 0; it != statisticians.end();
@@ -1705,9 +1703,8 @@ int RTCPSender::PrepareRTCP(
         if (PrepareReport(it->second, &report_block, &NTPsec, &NTPfrac))
           AddReportBlock(it->first, &internal_report_blocks_, &report_block);
       }
-      if (_IJ && !statisticians.empty())
-      {
-          rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
+      if (_IJ && !statisticians.empty()) {
+        rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
       }
       _lastRTCPTime[0] = Clock::NtpToMs(NTPsec, NTPfrac);
     }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 2a1c3d52e4..ea4bfcba34 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -41,7 +41,7 @@ RtpRtcp::Configuration::Configuration()
       audio(false),
       clock(NULL),
       default_module(NULL),
-      receive_statistics(),
+      receive_statistics(NullObjectReceiveStatistics()),
       outgoing_transport(NULL),
       rtcp_feedback(NULL),
       intra_frame_callback(NULL),
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index 25afd73d70..3fdbfe0718 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -61,6 +61,11 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback() {
   return &null_rtp_audio_feedback;
 }
 
+ReceiveStatistics* NullObjectReceiveStatistics() {
+  static NullReceiveStatistics null_receive_statistics;
+  return &null_receive_statistics;
+}
+
 namespace ModuleRTPUtility {
 
 enum {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index e95830effe..baf70fae20 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -14,6 +14,7 @@
 #include <cstddef> // size_t, ptrdiff_t
 
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -25,6 +26,7 @@ const uint8_t kRtpMarkerBitMask = 0x80;
 RtpData* NullObjectRtpData();
 RtpFeedback* NullObjectRtpFeedback();
 RtpAudioFeedback* NullObjectRtpAudioFeedback();
+ReceiveStatistics* NullObjectReceiveStatistics();
 
 namespace ModuleRTPUtility
 {