Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.

Clean some logs and add asserts in the way.

BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/interface/fec_receiver.h

@@ -18,7 +18,7 @@ namespace webrtc {
 
 class FecReceiver {
  public:
-  static FecReceiver* Create(int32_t id, RtpData* callback);
+  static FecReceiver* Create(RtpData* callback);
 
   virtual ~FecReceiver() {}
 

webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h

@@ -54,8 +54,7 @@ class RTPPayloadStrategy {
 class RTPPayloadRegistry {
  public:
   // The registry takes ownership of the strategy.
-  RTPPayloadRegistry(const int32_t id,
+  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
-                     RTPPayloadStrategy* rtp_payload_strategy);
   ~RTPPayloadRegistry();
 
   int32_t RegisterReceivePayload(
@@ -153,7 +152,6 @@ class RTPPayloadRegistry {
 
   scoped_ptr<CriticalSectionWrapper> crit_sect_;
   ModuleRTPUtility::PayloadTypeMap payload_type_map_;
-  int32_t id_;
   scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
   int8_t  red_payload_type_;
   int8_t ulpfec_payload_type_;

webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc

@@ -16,20 +16,19 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 // RFC 5109
 namespace webrtc {
 
-FecReceiver* FecReceiver::Create(int32_t id, RtpData* callback) {
+FecReceiver* FecReceiver::Create(RtpData* callback) {
-  return new FecReceiverImpl(id, callback);
+  return new FecReceiverImpl(callback);
 }
 
-FecReceiverImpl::FecReceiverImpl(const int32_t id, RtpData* callback)
+FecReceiverImpl::FecReceiverImpl(RtpData* callback)
-    : id_(id),
+    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       recovered_packet_callback_(callback),
-      fec_(new ForwardErrorCorrection(id)) {}
+      fec_(new ForwardErrorCorrection()) {}
 
 FecReceiverImpl::~FecReceiverImpl() {
   while (!received_packet_list_.empty()) {
@@ -103,8 +102,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
     if (timestamp_offset != 0) {
       // |timestampOffset| should be 0. However, it's possible this is the first
       // location a corrupt payload can be caught, so don't assert.
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+      LOG(LS_WARNING) << "Corrupt payload found.";
-                   "Corrupt payload found in %s", __FUNCTION__);
       delete received_packet;
       return -1;
     }

webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h

@@ -25,7 +25,7 @@ class CriticalSectionWrapper;
 
 class FecReceiverImpl : public FecReceiver {
  public:
-  FecReceiverImpl(const int32_t id, RtpData* callback);
+  FecReceiverImpl(RtpData* callback);
   virtual ~FecReceiverImpl();
 
   virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
@@ -36,7 +36,6 @@ class FecReceiverImpl : public FecReceiver {
   virtual int32_t ProcessReceivedFec() OVERRIDE;
 
  private:
-  int id_;
   scoped_ptr<CriticalSectionWrapper> crit_sect_;
   RtpData* recovered_packet_callback_;
   ForwardErrorCorrection* fec_;

webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc

@@ -39,8 +39,8 @@ class MockRtpData : public RtpData {
 class ReceiverFecTest : public ::testing::Test {
  protected:
   virtual void SetUp() {
-    fec_ = new ForwardErrorCorrection(0);
+    fec_ = new ForwardErrorCorrection();
-    receiver_fec_ = FecReceiver::Create(0, &rtp_data_callback_);
+    receiver_fec_ = FecReceiver::Create(&rtp_data_callback_);
     generator_ = new FrameGenerator();
   }
 

webrtc/modules/rtp_rtcp/source/forward_error_correction.cc

@@ -20,7 +20,7 @@
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
@@ -82,9 +82,8 @@ ForwardErrorCorrection::ReceivedPacket::~ReceivedPacket() {}
 ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() {}
 ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {}
 
-ForwardErrorCorrection::ForwardErrorCorrection(int32_t id)
+ForwardErrorCorrection::ForwardErrorCorrection()
-    : id_(id),
+    : generated_fec_packets_(kMaxMediaPackets),
-      generated_fec_packets_(kMaxMediaPackets),
       fec_packet_received_(false) {}
 
 ForwardErrorCorrection::~ForwardErrorCorrection() {}
@@ -112,43 +111,23 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
                                             bool use_unequal_protection,
                                             FecMaskType fec_mask_type,
                                             PacketList* fec_packet_list) {
-  if (media_packet_list.empty()) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "%s media packet list is empty", __FUNCTION__);
-    return -1;
-  }
-  if (!fec_packet_list->empty()) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "%s FEC packet list is not empty", __FUNCTION__);
-    return -1;
-  }
   const uint16_t num_media_packets = media_packet_list.size();
+
+  // Sanity check arguments.
+  assert(num_media_packets > 0);
+  assert(num_important_packets >= 0 &&
+         num_important_packets <= num_media_packets);
+  assert(fec_packet_list->empty());
+
+  if (num_media_packets > kMaxMediaPackets) {
+    LOG(LS_WARNING) << "Can't protect " << num_media_packets
+                    << " media packets per frame. Max is " << kMaxMediaPackets;
+    return -1;
+  }
+
   bool l_bit = (num_media_packets > 8 * kMaskSizeLBitClear);
   int num_maskBytes = l_bit ? kMaskSizeLBitSet : kMaskSizeLBitClear;
 
-  if (num_media_packets > kMaxMediaPackets) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "%s can only protect %d media packets per frame; %d requested",
-                 __FUNCTION__, kMaxMediaPackets, num_media_packets);
-    return -1;
-  }
-
-  // Error checking on the number of important packets.
-  // Can't have more important packets than media packets.
-  if (num_important_packets > num_media_packets) {
-    WEBRTC_TRACE(
-        kTraceError, kTraceRtpRtcp, id_,
-        "Number of important packets (%d) greater than number of media "
-        "packets (%d)",
-        num_important_packets, num_media_packets);
-    return -1;
-  }
-  if (num_important_packets < 0) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "Number of important packets (%d) less than zero",
-                 num_important_packets);
-    return -1;
-  }
   // Do some error checking on the media packets.
   PacketList::const_iterator media_list_it = media_packet_list.begin();
   while (media_list_it != media_packet_list.end()) {
@@ -156,20 +135,16 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
     assert(media_packet);
 
     if (media_packet->length < kRtpHeaderSize) {
-      WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+      LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
-                   "%s media packet (%d bytes) is smaller than RTP header",
+                      << "is smaller than RTP header.";
-                   __FUNCTION__, media_packet->length);
       return -1;
     }
 
     // Ensure our FEC packets will fit in a typical MTU.
     if (media_packet->length + PacketOverhead() + kTransportOverhead >
         IP_PACKET_SIZE) {
-      WEBRTC_TRACE(
+      LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
-          kTraceError, kTraceRtpRtcp, id_,
+                      << "with overhead is larger than " << IP_PACKET_SIZE;
-          "%s media packet (%d bytes) with overhead is larger than MTU(%d)",
-          __FUNCTION__, media_packet->length, IP_PACKET_SIZE);
-      return -1;
     }
     media_list_it++;
   }
@@ -582,9 +557,7 @@ void ForwardErrorCorrection::InsertFECPacket(
   }
   if (fec_packet->protected_pkt_list.empty()) {
     // All-zero packet mask; we can discard this FEC packet.
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "FEC packet has an all-zero packet mask.";
-                 "FEC packet %u has an all-zero packet mask.",
-                 fec_packet->seq_num, __FUNCTION__);
     delete fec_packet;
   } else {
     AssignRecoveredPackets(fec_packet, recovered_packet_list);

webrtc/modules/rtp_rtcp/source/forward_error_correction.h

@@ -117,8 +117,7 @@ class ForwardErrorCorrection {
   typedef std::list<ReceivedPacket*> ReceivedPacketList;
   typedef std::list<RecoveredPacket*> RecoveredPacketList;
 
-  // \param[in] id Module ID
+  ForwardErrorCorrection();
-  ForwardErrorCorrection(int32_t id);
 
   virtual ~ForwardErrorCorrection();
 
@@ -304,7 +303,6 @@ class ForwardErrorCorrection {
   static void DiscardOldPackets(RecoveredPacketList* recovered_packet_list);
   static uint16_t ParseSequenceNumber(uint8_t* packet);
 
-  int32_t id_;
   std::vector<Packet> generated_fec_packets_;
   FecPacketList fec_packet_list_;
   bool fec_packet_received_;

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

@@ -164,7 +164,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
 class RtpRtcpRtxNackTest : public ::testing::Test {
  protected:
   RtpRtcpRtxNackTest()
-      : rtp_payload_registry_(0, RTPPayloadStrategy::CreateStrategy(false)),
+      : rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         rtp_rtcp_module_(NULL),
         transport_(kTestSsrc + 1),
         receiver_(),

webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc

@@ -39,7 +39,7 @@ void VerifyHeader(uint16_t seq_num,
 class ProducerFecTest : public ::testing::Test {
  protected:
   virtual void SetUp() {
-    fec_ = new ForwardErrorCorrection(0);
+    fec_ = new ForwardErrorCorrection();
     producer_ = new ProducerFec(fec_);
     generator_ = new FrameGenerator;
   }

webrtc/modules/rtp_rtcp/source/rtcp_packet.cc

@@ -11,7 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 namespace rtcp {
@@ -233,8 +233,7 @@ void SenderReport::Create(uint8_t* packet,
                           uint16_t* len,
                           uint16_t max_len) const {
   if (*len + Length() > max_len) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max packet size reached.";
-                 "Max packet size reached, skipped SR.");
     return;
   }
   CreateSenderReport(sr_, packet, len);
@@ -244,8 +243,7 @@ void SenderReport::Create(uint8_t* packet,
 void SenderReport::WithReportBlock(ReportBlock* block) {
   assert(block);
   if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max report blocks reached.";
-                 "Max report block size reached.");
     return;
   }
   report_blocks_.push_back(block);
@@ -256,8 +254,7 @@ void ReceiverReport::Create(uint8_t* packet,
                             uint16_t* len,
                             uint16_t max_len) const {
   if (*len + Length() > max_len) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max packet size reached.";
-                 "Max packet size reached, skipped RR.");
     return;
   }
   CreateReceiverReport(rr_, packet, len);
@@ -267,8 +264,7 @@ void ReceiverReport::Create(uint8_t* packet,
 void ReceiverReport::WithReportBlock(ReportBlock* block) {
   assert(block);
   if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max report blocks reached.";
-                 "Max report block size reached.");
     return;
   }
   report_blocks_.push_back(block);
@@ -277,8 +273,7 @@ void ReceiverReport::WithReportBlock(ReportBlock* block) {
 
 void Bye::Create(uint8_t* packet, uint16_t* len, uint16_t max_len) const {
   if (*len + Length() > max_len) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max packet size reached.";
-                 "Max packet size reached, skipped BYE.");
     return;
   }
   CreateBye(bye_, csrcs_, packet, len);
@@ -286,8 +281,7 @@ void Bye::Create(uint8_t* packet, uint16_t* len, uint16_t max_len) const {
 
 void Bye::WithCsrc(uint32_t csrc) {
   if (csrcs_.size() >= kMaxNumberOfCsrcs) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max CSRC size reached.";
-                 "Max CSRC size reached.");
     return;
   }
   csrcs_.push_back(csrc);
@@ -295,8 +289,7 @@ void Bye::WithCsrc(uint32_t csrc) {
 
 void Fir::Create(uint8_t* packet, uint16_t* len, uint16_t max_len) const {
   if (*len + Length() > max_len) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Max packet size reached.";
-                 "Max packet size reached, skipped FIR.");
     return;
   }
   CreateFirRequest(fir_, fir_item_, packet, len);

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

@@ -18,7 +18,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -57,7 +57,6 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock,
     _lastIncreasedSequenceNumberMs(0),
     stats_callback_(NULL) {
     memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
-    WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
 }
 
 RTCPReceiver::~RTCPReceiver() {
@@ -82,8 +81,6 @@ RTCPReceiver::~RTCPReceiver() {
     delete first->second;
     _receivedCnameMap.erase(first);
   }
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id,
-               "%s deleted", __FUNCTION__);
 }
 
 void
@@ -178,8 +175,7 @@ int32_t RTCPReceiver::ResetRTT(const uint32_t remoteSSRC) {
   RTCPReportBlockInformation* reportBlock =
       GetReportBlockInformation(remoteSSRC);
   if (reportBlock == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC;
-                 "\tfailed to GetReportBlockInformation(%u)", remoteSSRC);
     return -1;
   }
   reportBlock->RTT = 0;
@@ -282,22 +278,14 @@ bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
   return true;
 }
 
-int32_t
+int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const {
-RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const
+  assert(senderInfo);
-{
+  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-    if(senderInfo == NULL)
+  if (_lastReceivedSRNTPsecs == 0) {
-    {
+    return -1;
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+  }
-        return -1;
+  memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
-    }
+  return 0;
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-    if(_lastReceivedSRNTPsecs == 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__);
-        return -1;
-    }
-    memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
-    return 0;
 }
 
 // statistics
@@ -518,8 +506,8 @@ void RTCPReceiver::HandleReportBlock(
   RTCPReportBlockInformation* reportBlock =
       CreateReportBlockInformation(remoteSSRC);
   if (reportBlock == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Failed to CreateReportBlockInformation("
-                 "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC);
+                    << remoteSSRC << ")";
     return;
   }
 
@@ -779,9 +767,6 @@ int32_t RTCPReceiver::BoundingSet(bool &tmmbrOwner, TMMBRSet* boundingSetRec) {
   }
   RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
   if (receiveInfo == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
-                 "%s failed to get RTCPReceiveInformation",
-                 __FUNCTION__);
     return -1;
   }
   if (receiveInfo->TmmbnBoundingSet.lengthOfSet() > 0) {
@@ -1348,8 +1333,7 @@ int32_t RTCPReceiver::UpdateTMMBR() {
   TMMBRSet* boundingSet = NULL;
   numBoundingSet = FindTMMBRBoundingSet(boundingSet);
   if (numBoundingSet == -1) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Failed to find TMMBR bounding set.";
-                 "Failed to find TMMBR bounding set.");
     return -1;
   }
   // Set bounding set
@@ -1369,8 +1353,6 @@ int32_t RTCPReceiver::UpdateTMMBR() {
     CriticalSectionScoped lock(_criticalSectionFeedbacks);
     if (_cbRtcpBandwidthObserver) {
         _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000);
-      WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
-                   "Set TMMBR request:%d kbps", bitrate);
     }
   }
   return 0;
@@ -1395,9 +1377,6 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
   // Process TMMBR and REMB first to avoid multiple callbacks
   // to OnNetworkChanged.
   if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr) {
-    WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
-                 "SIG [RTCP] Incoming TMMBR to id:%d", _id);
-
     // Might trigger a OnReceivedBandwidthEstimateUpdate.
     UpdateTMMBR();
   }
@@ -1412,9 +1391,8 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
   }
   if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
     if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
-      WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+      LOG(LS_INFO) << "Incoming NACK length: "
-                   "SIG [RTCP] Incoming NACK length:%d",
+                   << rtcpPacketInformation.nackSequenceNumbers.size();
-                   rtcpPacketInformation.nackSequenceNumbers.size());
       _rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers);
     }
   }
@@ -1429,13 +1407,11 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
       if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
           (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) {
         if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) {
-          WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+          LOG(LS_INFO) << "Incoming PLI from SSRC "
-                       "SIG [RTCP] Incoming PLI from SSRC:0x%x",
+                       << rtcpPacketInformation.remoteSSRC;
-                       rtcpPacketInformation.remoteSSRC);
         } else {
-          WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+          LOG(LS_INFO) << "Incoming FIR from SSRC "
-                       "SIG [RTCP] Incoming FIR from SSRC:0x%x",
+                       << rtcpPacketInformation.remoteSSRC;
-                       rtcpPacketInformation.remoteSSRC);
         }
         _cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc);
       }
@@ -1450,9 +1426,8 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
     }
     if (_cbRtcpBandwidthObserver) {
       if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
+        LOG(LS_INFO) << "Incoming REMB: "
-                     "SIG [RTCP] Incoming REMB:%d",
+                     << rtcpPacketInformation.receiverEstimatedMaxBitrate;
-                     rtcpPacketInformation.receiverEstimatedMaxBitrate);
         _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(
             rtcpPacketInformation.receiverEstimatedMaxBitrate);
       }
@@ -1548,9 +1523,6 @@ int32_t RTCPReceiver::TMMBRReceived(const uint32_t size,
     while (receiveInfoIt != _receivedInfoMap.end()) {
       RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
       if(receiveInfo == NULL) {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
-                     "%s failed to get RTCPReceiveInformation",
-                     __FUNCTION__);
         return -1;
       }
       num += receiveInfo->TmmbrSet.lengthOfSet();

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -19,7 +19,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -161,8 +161,6 @@ RTCPSender::RTCPSender(const int32_t id,
     memset(_CNAME, 0, sizeof(_CNAME));
     memset(_lastSendReport, 0, sizeof(_lastSendReport));
     memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
-
-    WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
 }
 
 RTCPSender::~RTCPSender() {
@@ -187,8 +185,6 @@ RTCPSender::~RTCPSender() {
   }
   delete _criticalSectionTransport;
   delete _criticalSectionRTCPSender;
-
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
 }
 
 int32_t
@@ -427,7 +423,8 @@ RTCPSender::SetCameraDelay(const int32_t delayMS)
     CriticalSectionScoped lock(_criticalSectionRTCPSender);
     if(delayMS > 1000 || delayMS < -1000)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__);
+        LOG(LS_WARNING) << "Delay can't be larger than 1 second: "
+                        << delayMS << " ms";
         return -1;
     }
     _cameraDelayMS = delayMS;
@@ -631,15 +628,10 @@ int32_t RTCPSender::AddReportBlock(
     uint32_t SSRC,
     std::map<uint32_t, RTCPReportBlock*>* report_blocks,
     const RTCPReportBlock* reportBlock) {
-  if (reportBlock == NULL) {
+  assert(reportBlock);
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
-                 "%s invalid argument", __FUNCTION__);
-    return -1;
-  }
 
   if (report_blocks->size() >= RTCP_MAX_REPORT_BLOCKS) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Too many report blocks.";
-                 "%s invalid argument", __FUNCTION__);
     return -1;
   }
   std::map<uint32_t, RTCPReportBlock*>::iterator it =
@@ -677,7 +669,7 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state,
     // sanity
     if(pos + 52 >= IP_PACKET_SIZE)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+        LOG(LS_WARNING) << "Failed to build Sender Report.";
         return -2;
     }
     uint32_t RTPtime;
@@ -760,8 +752,7 @@ int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) {
 
   // sanity
   if(pos + 12 + lengthCname  >= IP_PACKET_SIZE) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Failed to build SDEC.";
-                 "%s invalid argument", __FUNCTION__);
     return -2;
   }
   // SDEC Source Description
@@ -913,7 +904,9 @@ RTCPSender::BuildExtendedJitterReport(
 {
     if (external_report_blocks_.size() > 0)
     {
-        WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented.");
+        // TODO(andresp): Remove external report blocks since they are not
+        // supported.
+        LOG(LS_ERROR) << "Handling of external report blocks not implemented.";
         return 0;
     }
 
@@ -1317,7 +1310,7 @@ RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
     // sanity
     if(pos + 12 + boundingSet->lengthOfSet()*8 >= IP_PACKET_SIZE)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+        LOG(LS_WARNING) << "Failed to build TMMBN.";
         return -2;
     }
     uint8_t FMT = 4;
@@ -1384,12 +1377,12 @@ RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos)
     // sanity
     if(_appData == NULL)
     {
-        WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
+        LOG(LS_WARNING) << "Failed to build app specific.";
         return -1;
     }
     if(pos + 12 + _appLength >= IP_PACKET_SIZE)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+        LOG(LS_WARNING) << "Failed to build app specific.";
         return -2;
     }
     rtcpbuffer[pos++]=(uint8_t)0x80 + _appSubType;
@@ -1425,7 +1418,7 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer,
     // sanity
     if(pos + 16 >= IP_PACKET_SIZE)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+        LOG(LS_WARNING) << "Failed to build NACK.";
         return -2;
     }
 
@@ -1478,8 +1471,7 @@ RTCPSender::BuildNACK(uint8_t* rtcpbuffer,
       numOfNackFields++;
     }
     if (i != nackSize) {
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+      LOG(LS_WARNING) << "Nack list to large for one packet.";
-                   "Nack list to large for one packet.");
     }
     rtcpbuffer[nackSizePos] = static_cast<uint8_t>(2 + numOfNackFields);
     *nackString = stringBuilder.GetResult();
@@ -1715,8 +1707,7 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
     CriticalSectionScoped lock(_criticalSectionRTCPSender);
     if(_method == kRtcpOff)
     {
-        WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+        LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
-                     "%s invalid state", __FUNCTION__);
         return -1;
     }
   }
@@ -2128,13 +2119,7 @@ int32_t
 RTCPSender::SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize],
                     const uint8_t arrLength)
 {
-    if(arrLength > kRtpCsrcSize)
+    assert(arrLength <= kRtpCsrcSize);
-    {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
-        assert(false);
-        return -1;
-    }
-
     CriticalSectionScoped lock(_criticalSectionRTCPSender);
 
     for(int i = 0; i < arrLength;i++)
@@ -2153,7 +2138,7 @@ RTCPSender::SetApplicationSpecificData(const uint8_t subType,
 {
     if(length %4 != 0)
     {
-        WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
+        LOG(LS_ERROR) << "Failed to SetApplicationSpecificData.";
         return -1;
     }
     CriticalSectionScoped lock(_criticalSectionRTCPSender);
@@ -2199,17 +2184,10 @@ int32_t RTCPSender::WriteAllReportBlocksToBuffer(
     uint8_t& numberOfReportBlocks,
     const uint32_t NTPsec,
     const uint32_t NTPfrac) {
-  // sanity one block
-  if(pos + 24 >= IP_PACKET_SIZE) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
-                 "%s invalid argument", __FUNCTION__);
-    return -1;
-  }
   numberOfReportBlocks = external_report_blocks_.size();
   numberOfReportBlocks += internal_report_blocks_.size();
   if ((pos + numberOfReportBlocks * 24) >= IP_PACKET_SIZE) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_WARNING) << "Can't fit all report blocks.";
-                 "%s invalid argument", __FUNCTION__);
     return -1;
   }
   pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, internal_report_blocks_);

webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc

@@ -278,7 +278,7 @@ class RtcpSenderTest : public ::testing::Test {
       : over_use_detector_options_(),
         clock_(1335900000),
         rtp_payload_registry_(new RTPPayloadRegistry(
-            0, RTPPayloadStrategy::CreateStrategy(false))),
+            RTPPayloadStrategy::CreateStrategy(false))),
         remote_bitrate_observer_(),
         remote_bitrate_estimator_(
             RemoteBitrateEstimatorFactory().Create(

webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc

@@ -41,7 +41,7 @@ template <typename T> void ClearList(std::list<T*>* my_list) {
 class RtpFecTest : public ::testing::Test {
  protected:
   RtpFecTest()
-      : fec_(new ForwardErrorCorrection(0)), ssrc_(rand()), fec_seq_num_(0) {}
+      : fec_(new ForwardErrorCorrection()), ssrc_(rand()), fec_seq_num_(0) {}
 
   ForwardErrorCorrection* fec_;
   int ssrc_;
@@ -86,43 +86,6 @@ class RtpFecTest : public ::testing::Test {
   void TearDown();
 };
 
-// TODO(marpan): Consider adding table for input/output to simplify tests.
-
-TEST_F(RtpFecTest, HandleIncorrectInputs) {
-  int kNumImportantPackets = 0;
-  bool kUseUnequalProtection = false;
-  uint8_t kProtectionFactor = 60;
-
-  // Media packet list is empty.
-  EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor,
-                                  kNumImportantPackets, kUseUnequalProtection,
-                                  webrtc::kFecMaskBursty, &fec_packet_list_));
-
-  int num_media_packets = 10;
-  ConstructMediaPackets(num_media_packets);
-
-  kNumImportantPackets = -1;
-  // Number of important packets below 0.
-  EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor,
-                                  kNumImportantPackets, kUseUnequalProtection,
-                                  webrtc::kFecMaskBursty, &fec_packet_list_));
-
-  kNumImportantPackets = 12;
-  // Number of important packets greater than number of media packets.
-  EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor,
-                                  kNumImportantPackets, kUseUnequalProtection,
-                                  webrtc::kFecMaskBursty, &fec_packet_list_));
-
-  num_media_packets = kMaxNumberMediaPackets + 1;
-  ConstructMediaPackets(num_media_packets);
-
-  kNumImportantPackets = 0;
-  // Number of media packet is above maximum allowed (kMaxNumberMediaPackets).
-  EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_, kProtectionFactor,
-                                  kNumImportantPackets, kUseUnequalProtection,
-                                  webrtc::kFecMaskBursty, &fec_packet_list_));
-}
-
 TEST_F(RtpFecTest, FecRecoveryNoLoss) {
   const int kNumImportantPackets = 0;
   const bool kUseUnequalProtection = false;

webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc

@@ -13,7 +13,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
 
@@ -60,8 +59,6 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet, int length,
 
   const bool valid_rtpheader = rtp_parser.Parse(*header, &map);
   if (!valid_rtpheader) {
-    WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1,
-                 "IncomingPacket invalid RTP header");
     return false;
   }
   return true;

webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc

@@ -18,7 +18,7 @@
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
@@ -33,13 +33,21 @@ RTPPacketHistory::RTPPacketHistory(Clock* clock)
 }
 
 RTPPacketHistory::~RTPPacketHistory() {
-  Free();
+  {
+    CriticalSectionScoped cs(critsect_);
+    Free();
+  }
   delete critsect_;
 }
 
 void RTPPacketHistory::SetStorePacketsStatus(bool enable,
                                              uint16_t number_to_store) {
+  CriticalSectionScoped cs(critsect_);
   if (enable) {
+    if (store_) {
+      LOG(LS_WARNING) << "Purging packet history in order to re-set status.";
+      Free();
+    }
     Allocate(number_to_store);
   } else {
     Free();
@@ -48,16 +56,7 @@ void RTPPacketHistory::SetStorePacketsStatus(bool enable,
 
 void RTPPacketHistory::Allocate(uint16_t number_to_store) {
   assert(number_to_store > 0);
-  CriticalSectionScoped cs(critsect_);
+  assert(!store_);
-  if (store_) {
-    if (number_to_store != stored_packets_.size()) {
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
-                   "SetStorePacketsStatus already set, number: %d",
-                   number_to_store);
-    }
-    return;
-  }
-
   store_ = true;
   stored_packets_.resize(number_to_store);
   stored_seq_nums_.resize(number_to_store);
@@ -68,7 +67,6 @@ void RTPPacketHistory::Allocate(uint16_t number_to_store) {
 }
 
 void RTPPacketHistory::Free() {
-  CriticalSectionScoped cs(critsect_);
   if (!store_) {
     return;
   }
@@ -133,8 +131,8 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
   VerifyAndAllocatePacketLength(max_packet_length);
 
   if (packet_length > max_packet_length_) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "Failed to store RTP packet with length: "
-        "Failed to store RTP packet, length: %d", packet_length);
+                    << packet_length;
     return -1;
   }
 
@@ -169,25 +167,20 @@ int32_t RTPPacketHistory::ReplaceRTPHeader(const uint8_t* packet,
 
   assert(packet);
   assert(rtp_header_length > 3);
-
+  assert(rtp_header_length <= max_packet_length_);
-  if (rtp_header_length > max_packet_length_) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
-        "Failed to replace RTP packet, length: %d", rtp_header_length);
-    return -1;
-  }
 
   int32_t index = 0;
   bool found = FindSeqNum(sequence_number, &index);
   if (!found) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING)
-        "No match for getting seqNum %u", sequence_number);
+        << "Failed to replace RTP packet due to missing sequence number.";
     return -1;
   }
 
   uint16_t length = stored_lengths_.at(index);
   if (length == 0 || length > max_packet_length_) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number
-        "No match for getting seqNum %u, len %d", sequence_number, length);
+                    << ", len " << length;
     return -1;
   }
   assert(stored_seq_nums_[index] == sequence_number);
@@ -225,6 +218,7 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
                                                uint8_t* packet,
                                                uint16_t* packet_length,
                                                int64_t* stored_time_ms) {
+  assert(*packet_length >= max_packet_length_);
   CriticalSectionScoped cs(critsect_);
   if (!store_) {
     return false;
@@ -233,21 +227,15 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
   int32_t index = 0;
   bool found = FindSeqNum(sequence_number, &index);
   if (!found) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+    LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number;
-        "No match for getting seqNum %u", sequence_number);
     return false;
   }
 
   uint16_t length = stored_lengths_.at(index);
-  if (length == 0 || length > max_packet_length_) {
+  assert(length <= max_packet_length_);
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+  if (length == 0) {
-        "No match for getting seqNum %u, len %d", sequence_number, length);
+    LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number
-    return false;
+                    << ", len " << length;
-  }
-
-  if (length > *packet_length) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
-                 "Input buffer too short for packet %u", sequence_number);
     return false;
   }
 
@@ -255,8 +243,6 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
   int64_t now = clock_->TimeInMilliseconds();
   if (min_elapsed_time_ms > 0 &&
       ((now - stored_send_times_.at(index)) < min_elapsed_time_ms)) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
-        "Skip getting packet %u, packet recently resent.", sequence_number);
     return false;
   }
 

webrtc/modules/rtp_rtcp/source/rtp_packet_history.h

@@ -18,6 +18,7 @@
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
 
 namespace webrtc {
 
@@ -74,8 +75,8 @@ class RTPPacketHistory {
  private:
   void GetPacket(int index, uint8_t* packet, uint16_t* packet_length,
                  int64_t* stored_time_ms) const;
-  void Allocate(uint16_t number_to_store);
+  void Allocate(uint16_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
-  void Free();
+  void Free() EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
   void VerifyAndAllocatePacketLength(uint16_t packet_length);
   bool FindSeqNum(uint16_t sequence_number, int32_t* index) const;
   int FindBestFittingPacket(uint16_t size) const;

webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc

@@ -103,19 +103,6 @@ TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) {
                                     kAllowRetransmission));
 }
 
-TEST_F(RtpPacketHistoryTest, GetRtpPacket_TooSmallBuffer) {
-  hist_->SetStorePacketsStatus(true, 10);
-  uint16_t len = 0;
-  int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-  CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-  EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
-                                   capture_time_ms, kAllowRetransmission));
-  uint16_t len_out = len - 1;
-  int64_t time;
-  EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_,
-                                              &len_out, &time));
-}
-
 TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
   hist_->SetStorePacketsStatus(true, 10);
   uint16_t len = kMaxPacketLength;
@@ -161,8 +148,8 @@ TEST_F(RtpPacketHistoryTest, ReplaceRtpHeader) {
   uint16_t len = 0;
   int64_t capture_time_ms = 1;
   CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
+
   // Replace should fail, packet is not stored.
-  EXPECT_EQ(-1, hist_->ReplaceRTPHeader(packet_, kSeqNum, len));
   EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
                                    capture_time_ms, kAllowRetransmission));
 
@@ -181,10 +168,6 @@ TEST_F(RtpPacketHistoryTest, ReplaceRtpHeader) {
     EXPECT_EQ(packet_[i], packet_out_[i]);
   }
 
-  // Replace should fail, too large length.
-  EXPECT_EQ(-1, hist_->ReplaceRTPHeader(packet_, kSeqNum,
-      kMaxPacketLength + 1));
-
   // Replace should fail, packet is not stored.
   len = 0;
   CreateRtpPacket(kSeqNum + 1, kSsrc, kPayload, kTimestamp, packet_, &len);
@@ -236,10 +219,10 @@ TEST_F(RtpPacketHistoryTest, MinResendTime) {
                                    capture_time_ms, kAllowRetransmission));
 
   int64_t time;
+  len = kMaxPacketLength;
   EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len,
                                              &time));
   fake_clock_.AdvanceTimeMilliseconds(100);
-
   // Time has elapsed.
   len = kMaxPacketLength;
   EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum, 100, false, packet_, &len,

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

@@ -10,15 +10,13 @@
 
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
 RTPPayloadRegistry::RTPPayloadRegistry(
-    const int32_t id,
     RTPPayloadStrategy* rtp_payload_strategy)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      id_(id),
       rtp_payload_strategy_(rtp_payload_strategy),
       red_payload_type_(-1),
       ulpfec_payload_type_(-1),
@@ -60,9 +58,8 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
     case 77:        //  205 Transport layer FB message.
     case 78:        //  206 Payload-specific FB message.
     case 79:        //  207 Extended report.
-      WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+      LOG(LS_ERROR) << "Can't register invalid receiver payload type: "
-                   "%s invalid payloadtype:%d",
+                    << payload_type;
-                   __FUNCTION__, payload_type);
       return -1;
     default:
       break;
@@ -94,9 +91,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
         return 0;
       }
     }
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+    LOG(LS_ERROR) << "Payload type already registered: " << payload_type;
-                 "%s invalid argument payload_type:%d already registered",
-                 __FUNCTION__, payload_type);
     return -1;
   }
 
@@ -138,14 +133,8 @@ int32_t RTPPayloadRegistry::DeRegisterReceivePayload(
     const int8_t payload_type) {
   CriticalSectionScoped cs(crit_sect_.get());
   ModuleRTPUtility::PayloadTypeMap::iterator it =
-    payload_type_map_.find(payload_type);
+      payload_type_map_.find(payload_type);
-
+  assert(it != payload_type_map_.end());
-  if (it == payload_type_map_.end()) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "%s failed to find payload_type:%d",
-                 __FUNCTION__, payload_type);
-    return -1;
-  }
   delete it->second;
   payload_type_map_.erase(it);
   return 0;
@@ -194,11 +183,7 @@ int32_t RTPPayloadRegistry::ReceivePayloadType(
     const uint8_t channels,
     const uint32_t rate,
     int8_t* payload_type) const {
-  if (payload_type == NULL) {
+  assert(payload_type);
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                 "%s invalid argument", __FUNCTION__);
-    return -1;
-  }
   size_t payload_name_length = strlen(payload_name);
 
   CriticalSectionScoped cs(crit_sect_.get());
@@ -296,8 +281,7 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet,
         (*restored_packet)[1] |= kRtpMarkerBitMask;  // Marker bit is set.
       }
     } else {
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+      LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
-                   "Incorrect RTX configuration, dropping packet.");
       return false;
     }
   }

webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc

@@ -32,8 +32,7 @@ class RtpPayloadRegistryTest : public ::testing::Test {
   void SetUp() {
     // Note: the payload registry takes ownership of the strategy.
     mock_payload_strategy_ = new testing::NiceMock<MockRTPPayloadStrategy>();
-    rtp_payload_registry_.reset(
+    rtp_payload_registry_.reset(new RTPPayloadRegistry(mock_payload_strategy_));
-        new RTPPayloadRegistry(123, mock_payload_strategy_));
   }
 
  protected:

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

@@ -15,7 +15,7 @@
 #include <string.h>  // memcpy()
 
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -277,11 +277,8 @@ int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
                                           specific_payload.Audio.frequency,
                                           specific_payload.Audio.channels,
                                           specific_payload.Audio.rate)) {
-    WEBRTC_TRACE(kTraceError,
+    LOG(LS_ERROR) << "Failed to create decoder for payload type: "
-                 kTraceRtpRtcp,
+                  << payload_name << "/" << payload_type;
-                 id,
-                 "Failed to create video decoder for payload type:%d",
-                 payload_type);
     return -1;
   }
   return 0;

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

@@ -18,7 +18,7 @@
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
@@ -39,7 +39,7 @@ RtpReceiver* RtpReceiver::CreateVideoReceiver(
   return new RtpReceiverImpl(
       id, clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
       rtp_payload_registry,
-      RTPReceiverStrategy::CreateVideoStrategy(id, incoming_payload_callback));
+      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
@@ -87,8 +87,6 @@ RtpReceiverImpl::RtpReceiverImpl(int32_t id,
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
-
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
 }
 
 RtpReceiverImpl::~RtpReceiverImpl() {
@@ -96,7 +94,6 @@ RtpReceiverImpl::~RtpReceiverImpl() {
     cb_rtp_feedback_->OnIncomingCSRCChanged(id_, current_remote_csrc_[i],
                                             false);
   }
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
 }
 
 RTPReceiverStrategy* RtpReceiverImpl::GetMediaReceiver() const {
@@ -127,9 +124,8 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
   if (created_new_payload) {
     if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
                                                      frequency) != 0) {
-      WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+      LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
-                   "%s failed to register payload",
+                 << payload_type;
-                   __FUNCTION__);
       return -1;
     }
   }
@@ -182,19 +178,12 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   PayloadUnion payload_specific,
   bool in_order) {
   // Sanity check.
-  if (payload_length  < 0) {
+  assert(payload_length >= 0);
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+
-                 "%s invalid argument",
-                 __FUNCTION__);
-    return false;
-  }
-  int8_t first_payload_byte = 0;
-  if (payload_length > 0) {
-    first_payload_byte = payload[0];
-  }
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);
 
+  int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
   bool should_reset_statistics = false;
 
@@ -205,14 +194,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
                           &should_reset_statistics) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
-      WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
-                   "%s received keepalive",
-                   __FUNCTION__);
       return true;
     }
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Receiving invalid payload type.";
-                 "%s received invalid payloadtype",
-                 __FUNCTION__);
     return false;
   }
 
@@ -347,9 +331,8 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
         id_, rtp_header.payloadType, payload_name,
         rtp_header.payload_type_frequency, channels, rate)) {
       // New stream, same codec.
-      WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+      LOG(LS_ERROR) << "Failed to create decoder for payload type: "
-                   "Failed to create decoder for payload type:%d",
+                    << rtp_header.payloadType;
-                   rtp_header.payloadType);
     }
   }
 }

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h

@@ -26,8 +26,7 @@ class TelephoneEventHandler;
 // This class is not thread-safe and must be protected by its caller.
 class RTPReceiverStrategy {
  public:
-  static RTPReceiverStrategy* CreateVideoStrategy(int32_t id,
+  static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
-                                                  RtpData* data_callback);
   static RTPReceiverStrategy* CreateAudioStrategy(
       int32_t id, RtpData* data_callback,
       RtpAudioFeedback* incoming_messages_callback);

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc

@@ -17,19 +17,18 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 
 RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
-    int32_t id, RtpData* data_callback) {
+    RtpData* data_callback) {
-  return new RTPReceiverVideo(id, data_callback);
+  return new RTPReceiverVideo(data_callback);
 }
 
-RTPReceiverVideo::RTPReceiverVideo(int32_t id, RtpData* data_callback)
+RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
-    : RTPReceiverStrategy(data_callback),
+    : RTPReceiverStrategy(data_callback) {}
-      id_(id) {}
 
 RTPReceiverVideo::~RTPReceiverVideo() {
 }
@@ -93,11 +92,8 @@ int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
   // For video we just go with default values.
   if (-1 == callback->OnInitializeDecoder(
       id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
-    WEBRTC_TRACE(kTraceError,
+    LOG(LS_ERROR) << "Failed to created decoder for payload type: "
-                 kTraceRtpRtcp,
+                  << payload_type;
-                 id,
-                 "Failed to create video decoder for payload type:%d",
-                 payload_type);
     return -1;
   }
   return 0;
@@ -111,13 +107,6 @@ int32_t RTPReceiverVideo::ParseVideoCodecSpecific(
     RtpVideoCodecTypes video_type,
     int64_t now_ms,
     bool is_first_packet) {
-  WEBRTC_TRACE(kTraceStream,
-               kTraceRtpRtcp,
-               id_,
-               "%s(timestamp:%u)",
-               __FUNCTION__,
-               rtp_header->header.timestamp);
-
   switch (rtp_header->type.Video.codec) {
     case kRtpVideoGeneric:
       rtp_header->type.Video.isFirstPacket = is_first_packet;
@@ -170,13 +159,8 @@ int32_t RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
                                           const uint8_t* payload_data,
                                           uint16_t payload_data_length) {
   ModuleRTPUtility::RTPPayload parsed_packet;
-  uint32_t id;
-  {
-    CriticalSectionScoped cs(crit_sect_.get());
-    id = id_;
-  }
   ModuleRTPUtility::RTPPayloadParser rtp_payload_parser(
-      kRtpVideoVp8, payload_data, payload_data_length, id);
+      kRtpVideoVp8, payload_data, payload_data_length);
 
   if (!rtp_payload_parser.Parse(parsed_packet))
     return -1;

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h

@@ -22,7 +22,7 @@ namespace webrtc {
 
 class RTPReceiverVideo : public RTPReceiverStrategy {
  public:
-  RTPReceiverVideo(const int32_t id, RtpData* data_callback);
+  RTPReceiverVideo(RtpData* data_callback);
 
   virtual ~RTPReceiverVideo();
 
@@ -80,8 +80,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
       RtpVideoCodecTypes video_type,
       int64_t now_ms,
       bool is_first_packet);
-
-  int32_t id_;
 };
 }  // namespace webrtc
 

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -110,13 +110,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
   SetRtcpReceiverSsrcs(SSRC);
-
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s created", __FUNCTION__);
 }
 
 ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
-
   // All child modules MUST be deleted before deleting the default.
   assert(child_modules_.empty());
 
@@ -134,12 +130,6 @@ ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
 }
 
 void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "RegisterChildModule(module:0x%x)",
-               module);
-
   CriticalSectionScoped lock(
       critical_section_module_ptrs_.get());
   CriticalSectionScoped double_lock(
@@ -153,11 +143,6 @@ void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
 }
 
 void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* remove_module) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "DeRegisterChildModule(module:0x%x)", remove_module);
-
   CriticalSectionScoped lock(
       critical_section_module_ptrs_.get());
   CriticalSectionScoped double_lock(
@@ -282,29 +267,12 @@ void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) {
 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
     const uint8_t* rtcp_packet,
     const uint16_t length) {
-  WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
-               "IncomingRtcpPacket(packet_length:%u)", length);
-  // Minimum RTP is 12 bytes.
-  // Minimum RTCP is 8 bytes (RTCP BYE).
-  if (length == 8) {
-    WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1,
-                 "IncomingRtcpPacket invalid length");
-    return false;
-  }
-  // Check RTP version.
-  const uint8_t version = rtcp_packet[0] >> 6;
-  if (version != 2) {
-    WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, -1,
-                 "IncomingRtcpPacket invalid RTP version");
-    return false;
-  }
   // Allow receive of non-compound RTCP packets.
   RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true);
 
   const bool valid_rtcpheader = rtcp_parser.IsValid();
   if (!valid_rtcpheader) {
-    WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Incoming invalid RTCP packet";
-                 "IncomingRtcpPacket invalid RTCP packet");
     return -1;
   }
   RTCPHelp::RTCPPacketInformation rtcp_packet_information;
@@ -318,14 +286,6 @@ int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const CodecInst& voice_codec) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "RegisterSendPayload(pl_name:%s pl_type:%d frequency:%u)",
-               voice_codec.plname,
-               voice_codec.pltype,
-               voice_codec.plfreq);
-
   return rtp_sender_.RegisterPayload(
            voice_codec.plname,
            voice_codec.pltype,
@@ -336,13 +296,6 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const VideoCodec& video_codec) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "RegisterSendPayload(pl_name:%s pl_type:%d)",
-               video_codec.plName,
-               video_codec.plType);
-
   send_video_codec_ = video_codec;
   {
     // simulcast_ is accessed when accessing child_modules_, so this write needs
@@ -359,11 +312,6 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
 
 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(
     const int8_t payload_type) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "DeRegisterSendPayload(%d)", payload_type);
-
   return rtp_sender_.DeRegisterSendPayload(payload_type);
 }
 
@@ -372,53 +320,34 @@ int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
 }
 
 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "StartTimestamp()");
-
   return rtp_sender_.StartTimestamp();
 }
 
 // Configure start timestamp, default is a random number.
 int32_t ModuleRtpRtcpImpl::SetStartTimestamp(
     const uint32_t timestamp) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetStartTimestamp(%d)",
-               timestamp);
   rtcp_sender_.SetStartTimestamp(timestamp);
   rtp_sender_.SetStartTimestamp(timestamp, true);
   return 0;  // TODO(pwestin): change to void.
 }
 
 uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SequenceNumber()");
-
   return rtp_sender_.SequenceNumber();
 }
 
 // Set SequenceNumber, default is a random number.
 int32_t ModuleRtpRtcpImpl::SetSequenceNumber(
     const uint16_t seq_num) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetSequenceNumber(%d)",
-               seq_num);
-
   rtp_sender_.SetSequenceNumber(seq_num);
   return 0;  // TODO(pwestin): change to void.
 }
 
 uint32_t ModuleRtpRtcpImpl::SSRC() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SSRC()");
-
   return rtp_sender_.SSRC();
 }
 
 // Configure SSRC, default is a random number.
 int32_t ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetSSRC(%d)", ssrc);
-
   rtp_sender_.SetSSRC(ssrc);
   rtcp_sender_.SetSSRC(ssrc);
   SetRtcpReceiverSsrcs(ssrc);
@@ -434,20 +363,12 @@ int32_t ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
 
 int32_t ModuleRtpRtcpImpl::CSRCs(
   uint32_t arr_of_csrc[kRtpCsrcSize]) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CSRCs()");
-
   return rtp_sender_.CSRCs(arr_of_csrc);
 }
 
 int32_t ModuleRtpRtcpImpl::SetCSRCs(
     const uint32_t arr_of_csrc[kRtpCsrcSize],
     const uint8_t arr_length) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetCSRCs(arr_length:%d)",
-               arr_length);
-
   if (IsDefaultModule()) {
     // For default we need to update all child modules too.
     CriticalSectionScoped lock(critical_section_module_ptrs_.get());
@@ -461,10 +382,6 @@ int32_t ModuleRtpRtcpImpl::SetCSRCs(
       it++;
     }
   } else {
-    for (int i = 0; i < arr_length; ++i) {
-      WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "\tidx:%d CSRC:%u", i,
-                   arr_of_csrc[i]);
-    }
     rtcp_sender_.SetCSRCs(arr_of_csrc, arr_length);
     rtp_sender_.SetCSRCs(arr_of_csrc, arr_length);
   }
@@ -472,35 +389,23 @@ int32_t ModuleRtpRtcpImpl::SetCSRCs(
 }
 
 uint32_t ModuleRtpRtcpImpl::PacketCountSent() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "PacketCountSent()");
   return rtp_sender_.Packets();
 }
 
 uint32_t ModuleRtpRtcpImpl::ByteCountSent() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ByteCountSent()");
   return rtp_sender_.Bytes();
 }
 
 int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "CurrentSendFrequencyHz()");
   return rtp_sender_.SendPayloadFrequency();
 }
 
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
-  if (sending) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetSendingStatus(sending)");
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetSendingStatus(stopped)");
-  }
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false
     RTCPSender::FeedbackState feedback_state(this);
     if (rtcp_sender_.SetSendingStatus(feedback_state, sending) != 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+      LOG(LS_WARNING) << "Failed to send RTCP BYE";
-                   "Failed to send RTCP BYE");
     }
 
     collision_detected_ = false;
@@ -525,25 +430,15 @@ int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
 }
 
 bool ModuleRtpRtcpImpl::Sending() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "Sending()");
   return rtcp_sender_.Sending();
 }
 
 int32_t ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
-  if (sending) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetSendingMediaStatus(sending)");
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetSendingMediaStatus(stopped)");
-  }
   rtp_sender_.SetSendingMediaStatus(sending);
   return 0;
 }
 
 bool ModuleRtpRtcpImpl::SendingMedia() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "Sending()");
-
   if (!IsDefaultModule()) {
     return rtp_sender_.SendingMedia();
   }
@@ -569,13 +464,6 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     uint32_t payload_size,
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_hdr) {
-  WEBRTC_TRACE(
-    kTraceStream,
-    kTraceRtpRtcp,
-    id_,
-    "SendOutgoingData(frame_type:%d payload_type:%d time_stamp:%u size:%u)",
-    frame_type, payload_type, time_stamp, payload_size);
-
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
 
   if (!IsDefaultModule()) {
@@ -619,11 +507,6 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     if (it == child_modules_.end()) {
       return -1;
     }
-    WEBRTC_TRACE(kTraceModuleCall,
-                 kTraceRtpRtcp,
-                 id_,
-                 "SendOutgoingData(SimulcastIdx:%u size:%u, ssrc:0x%x)",
-                 idx, payload_size, (*it)->rtp_sender_.SSRC());
     return (*it)->SendOutgoingData(frame_type,
                                    payload_type,
                                    time_stamp,
@@ -656,13 +539,6 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission) {
-  WEBRTC_TRACE(
-    kTraceStream,
-    kTraceRtpRtcp,
-    id_,
-    "TimeToSendPacket(ssrc:0x%x sequence_number:%u capture_time_ms:%ll)",
-    ssrc, sequence_number, capture_time_ms);
-
   if (!IsDefaultModule()) {
     // Don't send from default module.
     if (SendingMedia() && ssrc == rtp_sender_.SSRC()) {
@@ -686,9 +562,6 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
 }
 
 int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) {
-  WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "TimeToSendPadding(bytes: %d)",
-               bytes);
-
   if (!IsDefaultModule()) {
     // Don't send from default module.
     if (SendingMedia()) {
@@ -721,16 +594,10 @@ bool ModuleRtpRtcpImpl::GetSendSideDelay(int* avg_send_delay_ms,
 }
 
 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "MaxPayloadLength()");
   return rtp_sender_.MaxPayloadLength();
 }
 
 uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "MaxDataPayloadLength()");
-
   // Assuming IP/UDP.
   uint16_t min_data_payload_length = IP_PACKET_SIZE - 28;
 
@@ -763,13 +630,6 @@ int32_t ModuleRtpRtcpImpl::SetTransportOverhead(
     const bool tcp,
     const bool ipv6,
     const uint8_t authentication_overhead) {
-  WEBRTC_TRACE(
-    kTraceModuleCall,
-    kTraceRtpRtcp,
-    id_,
-    "SetTransportOverhead(TCP:%d, IPV6:%d authentication_overhead:%u)",
-    tcp, ipv6, authentication_overhead);
-
   uint16_t packet_overhead = 0;
   if (ipv6) {
     packet_overhead = 40;
@@ -801,11 +661,8 @@ int32_t ModuleRtpRtcpImpl::SetTransportOverhead(
 }
 
 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetMaxTransferUnit(%u)",
-               mtu);
   if (mtu > IP_PACKET_SIZE) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+    LOG(LS_ERROR) << "Invalid mtu: " << mtu;
-                 "Invalid in argument to SetMaxTransferUnit(%u)", mtu);
     return -1;
   }
   return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
@@ -813,7 +670,6 @@ int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
 }
 
 RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTCP()");
   if (rtcp_sender_.Status() != kRtcpOff) {
     return rtcp_receiver_.Status();
   }
@@ -822,8 +678,6 @@ RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
 
 // Configure RTCP status i.e on/off.
 int32_t ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPStatus(%d)",
-               method);
   if (rtcp_sender_.SetRTCPStatus(method) == 0) {
     return rtcp_receiver_.SetRTCPStatus(method);
   }
@@ -837,34 +691,26 @@ uint32_t ModuleRtpRtcpImpl::LastSendReport(
 }
 
 int32_t ModuleRtpRtcpImpl::SetCNAME(const char c_name[RTCP_CNAME_SIZE]) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetCNAME(%s)", c_name);
   return rtcp_sender_.SetCNAME(c_name);
 }
 
 int32_t ModuleRtpRtcpImpl::CNAME(char c_name[RTCP_CNAME_SIZE]) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "CNAME()");
   return rtcp_sender_.CNAME(c_name);
 }
 
 int32_t ModuleRtpRtcpImpl::AddMixedCNAME(
   const uint32_t ssrc,
   const char c_name[RTCP_CNAME_SIZE]) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "AddMixedCNAME(SSRC:%u)", ssrc);
   return rtcp_sender_.AddMixedCNAME(ssrc, c_name);
 }
 
 int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "RemoveMixedCNAME(SSRC:%u)", ssrc);
   return rtcp_sender_.RemoveMixedCNAME(ssrc);
 }
 
 int32_t ModuleRtpRtcpImpl::RemoteCNAME(
     const uint32_t remote_ssrc,
     char c_name[RTCP_CNAME_SIZE]) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "RemoteCNAME(SSRC:%u)", remote_ssrc);
   return rtcp_receiver_.CNAME(remote_ssrc, c_name);
 }
 
@@ -874,7 +720,6 @@ int32_t ModuleRtpRtcpImpl::RemoteNTP(
     uint32_t* rtcp_arrival_time_secs,
     uint32_t* rtcp_arrival_time_frac,
     uint32_t* rtcp_timestamp) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteNTP()");
   return rtcp_receiver_.NTP(received_ntpsecs,
                             received_ntpfrac,
                             rtcp_arrival_time_secs,
@@ -888,21 +733,16 @@ int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
                                uint16_t* avg_rtt,
                                uint16_t* min_rtt,
                                uint16_t* max_rtt) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTT()");
   return rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
 }
 
 // Reset RoundTripTime statistics.
 int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "ResetRTT(SSRC:%u)",
-               remote_ssrc);
   return rtcp_receiver_.ResetRTT(remote_ssrc);
 }
 
 // Reset RTP data counters for the sending side.
 int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "ResetSendDataCountersRTP()");
   rtp_sender_.ResetDataCounters();
   return 0;  // TODO(pwestin): change to void.
 }
@@ -910,8 +750,6 @@ int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
 // Force a send of an RTCP packet.
 // Normal SR and RR are triggered via the process function.
 int32_t ModuleRtpRtcpImpl::SendRTCP(uint32_t rtcp_packet_type) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendRTCP(0x%x)",
-               rtcp_packet_type);
   RTCPSender::FeedbackState feedback_state(this);
   return rtcp_sender_.SendRTCP(feedback_state, rtcp_packet_type);
 }
@@ -921,23 +759,16 @@ int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
     const uint32_t name,
     const uint8_t* data,
     const uint16_t length) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SetRTCPApplicationSpecificData(sub_type:%d name:0x%x)",
-               sub_type, name);
   return  rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length);
 }
 
 // (XR) VOIP metric.
 int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
   const RTCPVoIPMetric* voip_metric) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRTCPVoIPMetrics()");
-
   return  rtcp_sender_.SetRTCPVoIPMetrics(voip_metric);
 }
 
 void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SetRtcpXrRrtrStatus(%s)", enable ? "true" : "false");
   return rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
 }
 
@@ -948,7 +779,6 @@ bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
 int32_t ModuleRtpRtcpImpl::DataCountersRTP(
     uint32_t* bytes_sent,
     uint32_t* packets_sent) const {
-  WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "DataCountersRTP()");
   if (bytes_sent) {
     *bytes_sent = rtp_sender_.Bytes();
   }
@@ -959,27 +789,23 @@ int32_t ModuleRtpRtcpImpl::DataCountersRTP(
 }
 
 int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()");
   return rtcp_receiver_.SenderInfoReceived(sender_info);
 }
 
 // Received RTCP report.
 int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
     std::vector<RTCPReportBlock>* receive_blocks) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoteRTCPStat()");
   return rtcp_receiver_.StatisticsReceived(receive_blocks);
 }
 
 int32_t ModuleRtpRtcpImpl::AddRTCPReportBlock(
     const uint32_t ssrc,
     const RTCPReportBlock* report_block) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "AddRTCPReportBlock()");
   return rtcp_sender_.AddExternalReportBlock(ssrc, report_block);
 }
 
 int32_t ModuleRtpRtcpImpl::RemoveRTCPReportBlock(
   const uint32_t ssrc) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RemoveRTCPReportBlock()");
   return rtcp_sender_.RemoveExternalReportBlock(ssrc);
 }
 
@@ -992,44 +818,25 @@ void ModuleRtpRtcpImpl::GetRtcpPacketTypeCounters(
 
 // (REMB) Receiver Estimated Max Bitrate.
 bool ModuleRtpRtcpImpl::REMB() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "REMB()");
   return rtcp_sender_.REMB();
 }
 
 int32_t ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) {
-  if (enable) {
-    WEBRTC_TRACE(kTraceModuleCall,
-                 kTraceRtpRtcp,
-                 id_,
-                 "SetREMBStatus(enable)");
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall,
-                 kTraceRtpRtcp,
-                 id_,
-                 "SetREMBStatus(disable)");
-  }
   return rtcp_sender_.SetREMBStatus(enable);
 }
 
 int32_t ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate,
                                        const uint8_t number_of_ssrc,
                                        const uint32_t* ssrc) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SetREMBData(bitrate:%d,?,?)", bitrate);
   return rtcp_sender_.SetREMBData(bitrate, number_of_ssrc, ssrc);
 }
 
 // (IJ) Extended jitter report.
 bool ModuleRtpRtcpImpl::IJ() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "IJ()");
   return rtcp_sender_.IJ();
 }
 
 int32_t ModuleRtpRtcpImpl::SetIJStatus(const bool enable) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetIJStatus(%s)", enable ? "true" : "false");
   return rtcp_sender_.SetIJStatus(enable);
 }
 
@@ -1046,23 +853,14 @@ int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
 
 // (TMMBR) Temporary Max Media Bit Rate.
 bool ModuleRtpRtcpImpl::TMMBR() const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "TMMBR()");
   return rtcp_sender_.TMMBR();
 }
 
 int32_t ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
-  if (enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetTMMBRStatus(enable)");
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetTMMBRStatus(disable)");
-  }
   return rtcp_sender_.SetTMMBRStatus(enable);
 }
 
 int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetTMMBN()");
   uint32_t max_bitrate_kbit =
       rtp_sender_.MaxConfiguredBitrateVideo() / 1000;
   return rtcp_sender_.SetTMMBN(bounding_set, max_bitrate_kbit);
@@ -1070,32 +868,18 @@ int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) {
 
 // Returns the currently configured retransmission mode.
 int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SelectiveRetransmissions()");
   return rtp_sender_.SelectiveRetransmissions();
 }
 
 // Enable or disable a retransmission mode, which decides which packets will
 // be retransmitted if NACKed.
 int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetSelectiveRetransmissions(%u)",
-               settings);
   return rtp_sender_.SetSelectiveRetransmissions(settings);
 }
 
 // Send a Negative acknowledgment packet.
 int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
                                     const uint16_t size) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SendNACK(size:%u)", size);
-
   // Use RTT from RtcpRttStats class if provided.
   uint16_t rtt = rtt_ms();
   if (rtt == 0) {
@@ -1149,14 +933,6 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
 int32_t ModuleRtpRtcpImpl::SetStorePacketsStatus(
     const bool enable,
     const uint16_t number_to_store) {
-  if (enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetStorePacketsStatus(enable, number_to_store:%d)",
-                 number_to_store);
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-                 "SetStorePacketsStatus(disable)");
-  }
   rtp_sender_.SetStorePacketsStatus(enable, number_to_store);
   return 0;  // TODO(pwestin): change to void.
 }
@@ -1180,19 +956,11 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
     const uint8_t key,
     const uint16_t time_ms,
     const uint8_t level) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SendTelephoneEventOutband(key:%u, time_ms:%u, level:%u)", key,
-               time_ms, level);
   return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
 }
 
 bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
     int8_t& telephone_event) const {
-
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SendTelephoneEventActive()");
   return rtp_sender_.SendTelephoneEventActive(&telephone_event);
 }
 
@@ -1200,40 +968,23 @@ bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
 // packet in silence (CNG).
 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
     const uint16_t packet_size_samples) {
-
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetAudioPacketSize(%u)",
-               packet_size_samples);
   return rtp_sender_.SetAudioPacketSize(packet_size_samples);
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
     const uint8_t level_d_bov) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetAudioLevel(level_d_bov:%u)",
-               level_d_bov);
   return rtp_sender_.SetAudioLevel(level_d_bov);
 }
 
 // Set payload type for Redundant Audio Data RFC 2198.
 int32_t ModuleRtpRtcpImpl::SetSendREDPayloadType(
     const int8_t payload_type) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetSendREDPayloadType(%d)",
-               payload_type);
   return rtp_sender_.SetRED(payload_type);
 }
 
 // Get payload type for Redundant Audio Data RFC 2198.
 int32_t ModuleRtpRtcpImpl::SendREDPayloadType(
     int8_t& payload_type) const {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()");
   return rtp_sender_.RED(&payload_type);
 }
 
@@ -1243,8 +994,6 @@ RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const {
 
 void ModuleRtpRtcpImpl::SetTargetSendBitrate(
     const std::vector<uint32_t>& stream_bitrates) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,
-               "SetTargetSendBitrate: %ld streams", stream_bitrates.size());
   if (IsDefaultModule()) {
     CriticalSectionScoped lock(critical_section_module_ptrs_.get());
     if (simulcast_) {
@@ -1275,20 +1024,11 @@ void ModuleRtpRtcpImpl::SetTargetSendBitrate(
 
 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
     const KeyFrameRequestMethod method) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetKeyFrameRequestMethod(method:%u)",
-               method);
   key_frame_req_method_ = method;
   return 0;
 }
 
 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "RequestKeyFrame");
   switch (key_frame_req_method_) {
     case kKeyFrameReqFirRtp:
       return rtp_sender_.SendRTPIntraRequest();
@@ -1302,22 +1042,12 @@ int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
 
 int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
     const uint8_t picture_id) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SendRTCPSliceLossIndication (picture_id:%d)",
-               picture_id);
   RTCPSender::FeedbackState feedback_state(this);
   return rtcp_sender_.SendRTCP(
       feedback_state, kRtcpSli, 0, 0, false, picture_id);
 }
 
 int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) {
-  WEBRTC_TRACE(kTraceModuleCall,
-               kTraceRtpRtcp,
-               id_,
-               "SetCameraDelay(%d)",
-               delay_ms);
   if (IsDefaultModule()) {
     CriticalSectionScoped lock(critical_section_module_ptrs_.get());
     std::list<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
@@ -1337,18 +1067,6 @@ int32_t ModuleRtpRtcpImpl::SetGenericFECStatus(
     const bool enable,
     const uint8_t payload_type_red,
     const uint8_t payload_type_fec) {
-  if (enable) {
-    WEBRTC_TRACE(kTraceModuleCall,
-                 kTraceRtpRtcp,
-                 id_,
-                 "SetGenericFECStatus(enable, %u)",
-                 payload_type_red);
-  } else {
-    WEBRTC_TRACE(kTraceModuleCall,
-                 kTraceRtpRtcp,
-                 id_,
-                 "SetGenericFECStatus(disable)");
-  }
   return rtp_sender_.SetGenericFECStatus(enable,
                                          payload_type_red,
                                          payload_type_fec);
@@ -1358,8 +1076,6 @@ int32_t ModuleRtpRtcpImpl::GenericFECStatus(
     bool& enable,
     uint8_t& payload_type_red,
     uint8_t& payload_type_fec) {
-  WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "GenericFECStatus()");
-
   bool child_enabled = false;
   if (IsDefaultModule()) {
     // For default we need to check all child modules too.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -15,7 +15,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -105,9 +105,8 @@ RTPSender::RTPSender(const int32_t id,
     audio_ = new RTPSenderAudio(id, clock_, this);
     audio_->RegisterAudioCallback(audio_feedback);
   } else {
-    video_ = new RTPSenderVideo(id, clock_, this);
+    video_ = new RTPSenderVideo(clock_, this);
   }
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
 }
 
 RTPSender::~RTPSender() {
@@ -126,8 +125,6 @@ RTPSender::~RTPSender() {
   }
   delete audio_;
   delete video_;
-
-  WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
 }
 
 void RTPSender::SetTargetSendBitrate(const uint32_t bits) {
@@ -290,16 +287,12 @@ int32_t RTPSender::SetMaxPayloadLength(
     const uint16_t packet_over_head) {
   // Sanity check.
   if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "%s invalid argument",
+    LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
-                 __FUNCTION__);
     return -1;
   }
   CriticalSectionScoped cs(send_critsect_);
   max_payload_length_ = max_payload_length;
   packet_over_head_ = packet_over_head;
-
-  WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, id_, "SetMaxPayloadLength to %d.",
-               max_payload_length);
   return 0;
 }
 
@@ -350,8 +343,7 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type,
   CriticalSectionScoped cs(send_critsect_);
 
   if (payload_type < 0) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)",
+    LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
-                 payload_type);
     return -1;
   }
   if (audio_configured_) {
@@ -373,8 +365,7 @@ int32_t RTPSender::CheckPayloadType(const int8_t payload_type,
   std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
-                 "\tpayloadType:%d not registered", payload_type);
     return -1;
   }
   payload_type_ = payload_type;
@@ -403,9 +394,7 @@ int32_t RTPSender::SendOutgoingData(
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
+    LOG(LS_ERROR) << "Don't send data with unknown payload type.";
-                 "%s invalid argument failed to find payload_type:%d",
-                 __FUNCTION__, payload_type);
     return -1;
   }
 
@@ -616,8 +605,6 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
     RTPHeader header;
     if (!rtp_parser.Parse(header)) {
       assert(false);
-      WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
-                   "Failed to parse RTP header of packet to be retransmitted.");
       return -1;
     }
     if (!paced_sender_->SendPacket(PacedSender::kHighPriority,
@@ -644,10 +631,9 @@ bool RTPSender::SendPacketToNetwork(const uint8_t *packet, uint32_t size) {
   }
   TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::SendPacketToNetwork",
                        "size", size, "sent", bytes_sent);
-  // TODO(pwesin): Add a separate bitrate for sent bitrate after pacer.
+  // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
   if (bytes_sent <= 0) {
-    WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Transport failed to send packet";
-                 "Transport failed to send packet");
     return false;
   }
   return true;
@@ -676,11 +662,8 @@ void RTPSender::OnReceivedNACK(
 
   // Enough bandwidth to send NACK?
   if (!ProcessNACKBitRate(now)) {
-    WEBRTC_TRACE(kTraceStream,
+    LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
-                 kTraceRtpRtcp,
+                 << target_bitrate_kbps;
-                 id_,
-                 "NACK bitrate reached. Skip sending NACK response. Target %d",
-                 target_bitrate_kbps);
     return;
   }
 
@@ -695,9 +678,8 @@ void RTPSender::OnReceivedNACK(
       continue;
     } else if (bytes_sent < 0) {
       // Failed to send one Sequence number. Give up the rest in this nack.
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
+      LOG(LS_WARNING) << "Failed resending RTP packet " << *it
-                   "Failed resending RTP packet %d, Discard rest of packets",
+                      << ", Discard rest of packets";
-                   *it);
       break;
     }
     // Delay bandwidth estimate (RTT * BW).
@@ -1258,39 +1240,36 @@ bool RTPSender::UpdateTransmissionTimeOffset(
       rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
           kRtpExtensionTransmissionTimeOffset);
   if (extension_block_pos < 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING)
-                 "Failed to update transmission time offset, not registered.");
+        << "Failed to update transmission time offset, not registered.";
     return false;
   }
   int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
   if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength ||
       rtp_header.headerLength <
           block_pos + kTransmissionTimeOffsetLength) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING)
-                 "Failed to update transmission time offset, invalid length.");
+        << "Failed to update transmission time offset, invalid length.";
     return false;
   }
   // Verify that header contains extension.
   if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
         (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
-    WEBRTC_TRACE(
+    LOG(LS_WARNING) << "Failed to update transmission time offset, hdr "
-        kTraceStream, kTraceRtpRtcp, id_,
+                       "extension not found.";
-        "Failed to update transmission time offset, hdr extension not found.");
     return false;
   }
   // Get id.
   uint8_t id = 0;
   if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
                                       &id) != 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update transmission time offset, no id.";
-                 "Failed to update transmission time offset, no id.");
     return false;
   }
   // Verify first byte in block.
   const uint8_t first_block_byte = (id << 4) + 2;
   if (rtp_packet[block_pos] != first_block_byte) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update transmission time offset.";
-                 "Failed to update transmission time offset.");
     return false;
   }
   // Update transmission offset field (converting to a 90 kHz timestamp).
@@ -1311,37 +1290,31 @@ bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet,
       rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
           kRtpExtensionAudioLevel);
   if (extension_block_pos < 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update audio level, not registered.";
-                 "Failed to update audio level, not registered.");
     return false;
   }
   int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
   if (rtp_packet_length < block_pos + kAudioLevelLength ||
       rtp_header.headerLength < block_pos + kAudioLevelLength) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update audio level, invalid length.";
-                 "Failed to update audio level, invalid length.");
     return false;
   }
   // Verify that header contains extension.
   if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
         (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
-    WEBRTC_TRACE(
+    LOG(LS_WARNING) << "Failed to update audio level, hdr extension not found.";
-        kTraceStream, kTraceRtpRtcp, id_,
-        "Failed to update audio level, hdr extension not found.");
     return false;
   }
   // Get id.
   uint8_t id = 0;
   if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update audio level, no id.";
-                 "Failed to update audio level, no id.");
     return false;
   }
   // Verify first byte in block.
   const uint8_t first_block_byte = (id << 4) + 0;
   if (rtp_packet[block_pos] != first_block_byte) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update audio level.";
-                 "Failed to update audio level.");
     return false;
   }
   rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
@@ -1358,38 +1331,33 @@ bool RTPSender::UpdateAbsoluteSendTime(
       rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
           kRtpExtensionAbsoluteSendTime);
   if (extension_block_pos < 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update absolute send time, not registered.";
-                 "Failed to update absolute send time, not registered.");
     return false;
   }
   int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
   if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
       rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update absolute send time, invalid length.";
-                 "Failed to update absolute send time, invalid length.");
     return false;
   }
   // Verify that header contains extension.
   if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
         (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
-    WEBRTC_TRACE(
+    LOG(LS_WARNING)
-        kTraceStream, kTraceRtpRtcp, id_,
+        << "Failed to update absolute send time, hdr extension not found.";
-        "Failed to update absolute send time, hdr extension not found.");
     return false;
   }
   // Get id.
   uint8_t id = 0;
   if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
                                       &id) != 0) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update absolute send time, no id.";
-                 "Failed to update absolute send time, no id.");
     return false;
   }
   // Verify first byte in block.
   const uint8_t first_block_byte = (id << 4) + 2;
   if (rtp_packet[block_pos] != first_block_byte) {
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
+    LOG(LS_WARNING) << "Failed to update absolute send time.";
-                 "Failed to update absolute send time.");
     return false;
   }
   // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -20,7 +20,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
@@ -31,11 +31,9 @@ struct RtpPacket {
   ForwardErrorCorrection::Packet* pkt;
 };
 
-RTPSenderVideo::RTPSenderVideo(const int32_t id,
+RTPSenderVideo::RTPSenderVideo(Clock* clock,
-                               Clock* clock,
                                RTPSenderInterface* rtpSender)
-    : _id(id),
+    : _rtpSender(*rtpSender),
-      _rtpSender(*rtpSender),
       _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _videoType(kRtpVideoGeneric),
       _videoCodecInformation(NULL),
@@ -43,7 +41,7 @@ RTPSenderVideo::RTPSenderVideo(const int32_t id,
       _retransmissionSettings(kRetransmitBaseLayer),
 
       // Generic FEC
-      _fec(id),
+      _fec(),
       _fecEnabled(false),
       _payloadTypeRED(-1),
       _payloadTypeFEC(-1),
@@ -329,8 +327,6 @@ RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
     {
         return retVal;
     }
-    WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
-                 __FUNCTION__, captureTimeStamp);
     return 0;
 }
 
@@ -476,9 +472,9 @@ RTPSenderVideo::SendVP8(const FrameType frameType,
                                   rtpHeaderLength, captureTimeStamp,
                                   capture_time_ms, storage, protect))
         {
-          WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+          LOG(LS_WARNING)
-                       "RTPSenderVideo::SendVP8 failed to send packet number"
+              << "RTPSenderVideo::SendVP8 failed to send packet number "
-                       " %d", _rtpSender.SequenceNumber());
+              << _rtpSender.SequenceNumber();
         }
     }
     TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms,

webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -31,7 +31,7 @@ struct RtpPacket;
 class RTPSenderVideo
 {
 public:
-    RTPSenderVideo(const int32_t id, Clock* clock,
+    RTPSenderVideo(Clock* clock,
                    RTPSenderInterface* rtpSender);
     virtual ~RTPSenderVideo();
 
@@ -112,7 +112,6 @@ private:
                     const RTPVideoTypeHeader* rtpTypeHdr);
 
 private:
-    int32_t             _id;
     RTPSenderInterface&        _rtpSender;
 
     CriticalSectionWrapper*   _sendVideoCritsect;

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -30,7 +30,7 @@
 #endif
 
 #include "webrtc/system_wrappers/interface/tick_util.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 #if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
 #define DEBUG_PRINT(...)           \
@@ -464,22 +464,21 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
     ptr++;
 
     if (id == 15) {
-      WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+      LOG(LS_WARNING)
-                   "Ext id: 15 encountered, parsing terminated.");
+          << "RTP extension header 15 encountered. Terminate parsing.";
       return;
     }
 
     RTPExtensionType type;
     if (ptrExtensionMap->GetType(id, &type) != 0) {
       // If we encounter an unknown extension, just skip over it.
-      WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+      LOG(LS_WARNING) << "Failed to find extension id: " << id;
-                   "Failed to find extension id: %d", id);
     } else {
       switch (type) {
         case kRtpExtensionTransmissionTimeOffset: {
           if (len != 2) {
-            WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+            LOG(LS_WARNING) << "Incorrect transmission time offset len: "
-                         "Incorrect transmission time offset len: %d", len);
+                            << len;
             return;
           }
           //  0                   1                   2                   3
@@ -502,8 +501,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
         }
         case kRtpExtensionAudioLevel: {
           if (len != 0) {
-            WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+            LOG(LS_WARNING) << "Incorrect audio level len: " << len;
-                         "Incorrect audio level len: %d", len);
             return;
           }
           //  0                   1                   2                   3
@@ -525,8 +523,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
         }
         case kRtpExtensionAbsoluteSendTime: {
           if (len != 2) {
-            WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
+            LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
-                         "Incorrect absolute send time len: %d", len);
             return;
           }
           //  0                   1                   2                   3
@@ -543,8 +540,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
           break;
         }
         default: {
-          WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
+          LOG(LS_WARNING) << "Extension type not implemented: " << type;
-                       "Extension type not implemented.");
           return;
         }
       }
@@ -570,17 +566,12 @@ uint8_t RTPHeaderParser::ParsePaddingBytes(
   return num_zero_bytes;
 }
 
-// RTP payload parser
 RTPPayloadParser::RTPPayloadParser(const RtpVideoCodecTypes videoType,
                                    const uint8_t* payloadData,
-                                   uint16_t payloadDataLength,
+                                   uint16_t payloadDataLength)
-                                   int32_t id)
+    : _dataPtr(payloadData),
-  :
+      _dataLength(payloadDataLength),
-  _id(id),
+      _videoType(videoType) {}
-  _dataPtr(payloadData),
-  _dataLength(payloadDataLength),
-  _videoType(videoType) {
-}
 
 RTPPayloadParser::~RTPPayloadParser() {
 }
@@ -655,8 +646,7 @@ bool RTPPayloadParser::ParseVP8(RTPPayload& parsedPacket) const {
   }
 
   if (dataLength <= 0) {
-    WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+    LOG(LS_ERROR) << "Error parsing VP8 payload descriptor!";
-                 "Error parsing VP8 payload descriptor; payload too short");
     return false;
   }
 

webrtc/modules/rtp_rtcp/source/rtp_utility.h

@@ -166,8 +166,8 @@ namespace ModuleRTPUtility
     public:
         RTPPayloadParser(const RtpVideoCodecTypes payloadType,
                          const uint8_t* payloadData,
-                         const uint16_t payloadDataLength, // Length w/o padding.
+                         // Length w/o padding.
-                         const int32_t id);
+                         const uint16_t payloadDataLength);
 
         ~RTPPayloadParser();
 
@@ -202,7 +202,6 @@ namespace ModuleRTPUtility
                               int dataLength) const;
 
     private:
-        int32_t               _id;
         const uint8_t*        _dataPtr;
         const uint16_t        _dataLength;
         const RtpVideoCodecTypes    _videoType;

webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc

@@ -76,7 +76,7 @@ TEST(ParseVP8Test, BasicHeader) {
   payload[0] = 0x14;  // Binary 0001 0100; S = 1, PartID = 4.
   payload[1] = 0x01;  // P frame.
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -97,7 +97,7 @@ TEST(ParseVP8Test, PictureID) {
   payload[1] = 0x80;
   payload[2] = 17;
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -117,7 +117,7 @@ TEST(ParseVP8Test, PictureID) {
   // Re-use payload, but change to long PictureID.
   payload[2] = 0x80 | 17;
   payload[3] = 17;
-  RTPPayloadParser rtpPayloadParser2(kRtpVideoVp8, payload, 10, 0);
+  RTPPayloadParser rtpPayloadParser2(kRtpVideoVp8, payload, 10);
 
   ASSERT_TRUE(rtpPayloadParser2.Parse(parsedPacket));
 
@@ -136,7 +136,7 @@ TEST(ParseVP8Test, Tl0PicIdx) {
   payload[1] = 0x40;
   payload[2] = 17;
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 13, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 13);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -159,7 +159,7 @@ TEST(ParseVP8Test, TIDAndLayerSync) {
   payload[1] = 0x20;
   payload[2] = 0x80;  // TID(2) + LayerSync(false)
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -183,7 +183,7 @@ TEST(ParseVP8Test, KeyIdx) {
   payload[1] = 0x10;  // K = 1.
   payload[2] = 0x11;  // KEYIDX = 17 decimal.
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -209,7 +209,7 @@ TEST(ParseVP8Test, MultipleExtensions) {
   payload[4] = 42;           // Tl0PicIdx.
   payload[5] = 0x40 | 0x20 | 0x11;  // TID(1) + LayerSync(true) + KEYIDX(17).
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 10);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
@@ -236,7 +236,7 @@ TEST(ParseVP8Test, TooShortHeader) {
   payload[2] = 0x80 | 17;  // ... but only 2 bytes PictureID is provided.
   payload[3] = 17;  // PictureID, low 8 bits.
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, payload, 4);
 
   RTPPayload parsedPacket;
   EXPECT_FALSE(rtpPayloadParser.Parse(parsedPacket));
@@ -258,7 +258,7 @@ TEST(ParseVP8Test, TestWithPacketizer) {
   ASSERT_EQ(0, packetizer.NextPacket(packet, &send_bytes, &last));
   ASSERT_TRUE(last);
 
-  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, packet, send_bytes, 0);
+  RTPPayloadParser rtpPayloadParser(kRtpVideoVp8, packet, send_bytes);
 
   RTPPayload parsedPacket;
   ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));

webrtc/modules/rtp_rtcp/source/ssrc_database.cc

@@ -14,7 +14,6 @@
 #include <stdlib.h>
 
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/trace.h"
 
 #ifdef _WIN32
     #include <windows.h>
@@ -185,8 +184,6 @@ SSRCDatabase::SSRCDatabase()
     _ssrcVector = new uint32_t[10];
 #endif
     _critSect = CriticalSectionWrapper::CreateCriticalSection();
-
-    WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s created", __FUNCTION__);
 }
 
 SSRCDatabase::~SSRCDatabase()
@@ -197,8 +194,6 @@ SSRCDatabase::~SSRCDatabase()
     _ssrcMap.clear();
 #endif
     delete _critSect;
-
-    WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s deleted", __FUNCTION__);
 }
 
 uint32_t SSRCDatabase::GenerateRandom()

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

@@ -34,7 +34,7 @@ class RtpRtcpAPITest : public ::testing::Test {
     configuration.clock = &fake_clock;
     module = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_payload_registry_.reset(new RTPPayloadRegistry(
-            test_id, RTPPayloadStrategy::CreateStrategy(true)));
+            RTPPayloadStrategy::CreateStrategy(true)));
     rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
         test_id, &fake_clock, NULL, NULL, NULL, rtp_payload_registry_.get()));
   }

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -122,9 +122,9 @@ class RtpRtcpAudioTest : public ::testing::Test {
     receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
-        test_id, RTPPayloadStrategy::CreateStrategy(true)));
+        RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
-        test_id, RTPPayloadStrategy::CreateStrategy(true)));
+        RTPPayloadStrategy::CreateStrategy(true)));
 
     RtpRtcp::Configuration configuration;
     configuration.id = test_id;

webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

@@ -116,9 +116,9 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     configuration.intra_frame_callback = myRTCPFeedback1;
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
-            test_id, RTPPayloadStrategy::CreateStrategy(true)));
+            RTPPayloadStrategy::CreateStrategy(true)));
     rtp_payload_registry2_.reset(new RTPPayloadRegistry(
-            test_id, RTPPayloadStrategy::CreateStrategy(true)));
+            RTPPayloadStrategy::CreateStrategy(true)));
 
     module1 = RtpRtcp::CreateRtpRtcp(configuration);
 
@@ -280,7 +280,6 @@ TEST_F(RtpRtcpRtcpTest, RTCP) {
   reportBlock.lastSR = 6;
 
   // Set report blocks.
-  EXPECT_EQ(-1, module1->AddRTCPReportBlock(test_CSRC[0], NULL));
   EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[0], &reportBlock));
 
   reportBlock.lastSR= 7;
@@ -319,7 +318,6 @@ TEST_F(RtpRtcpRtcpTest, RTCP) {
 
   // get all report blocks
   std::vector<RTCPReportBlock> report_blocks;
-  EXPECT_EQ(-1, module1->RemoteRTCPStat(NULL));
   EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
   ASSERT_EQ(1u, report_blocks.size());
   const RTCPReportBlock& reportBlockReceived = report_blocks[0];

webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc

@@ -28,7 +28,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
  protected:
   RtpRtcpVideoTest()
       : test_id_(123),
-        rtp_payload_registry_(0, RTPPayloadStrategy::CreateStrategy(false)),
+        rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
         test_ssrc_(3456),
         test_timestamp_(4567),
         test_sequence_number_(2345),

webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc

@@ -105,9 +105,7 @@ TEST(FecTest, FecTest) {
   ASSERT_EQ(12, kMaxMediaPackets[1]) << "Max media packets for bursty mode not "
                                      << "equal to 12.";
 
-  uint32_t id = 0;
+  ForwardErrorCorrection fec;
-  ForwardErrorCorrection fec(id);
-
   ForwardErrorCorrection::PacketList mediaPacketList;
   ForwardErrorCorrection::PacketList fecPacketList;
   ForwardErrorCorrection::ReceivedPacketList toDecodeList;

webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc

@@ -157,7 +157,7 @@ int MTRxTxTest(CmdArgs& args)
     configuration.outgoing_transport = outgoingTransport;
     RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration);
     scoped_ptr<RTPPayloadRegistry> registry(new RTPPayloadRegistry(
-        -1, RTPPayloadStrategy::CreateStrategy(false)));
+        RTPPayloadStrategy::CreateStrategy(false)));
     scoped_ptr<RtpReceiver> rtp_receiver(
         RtpReceiver::CreateVideoReceiver(-1, Clock::GetRealTimeClock(),
                                          &dataCallback, NULL, registry.get()));

webrtc/modules/video_coding/main/test/rtp_player.cc

@@ -273,7 +273,7 @@ class SsrcHandlers {
             LostPackets* lost_packets)
         : rtp_header_parser_(RtpHeaderParser::Create()),
           rtp_payload_registry_(new RTPPayloadRegistry(
-              0, RTPPayloadStrategy::CreateStrategy(false))),
+              RTPPayloadStrategy::CreateStrategy(false))),
           rtp_module_(),
           payload_sink_(),
           ssrc_(ssrc),

webrtc/video/rampup_tests.cc

@@ -54,8 +54,7 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
         feedback_transport_(feedback_transport),
         receive_stats_(ReceiveStatistics::Create(clock)),
         payload_registry_(
-            new RTPPayloadRegistry(-1,
+            new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
-                                   RTPPayloadStrategy::CreateStrategy(false))),
         clock_(clock),
         expected_bitrate_bps_(0),
         rtx_media_ssrcs_(rtx_media_ssrcs),

webrtc/video_engine/vie_channel.cc

@@ -643,11 +643,6 @@ int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
       nack_history_size_sender_ = kSendSidePacketHistorySize;
     }
   }
-  // Setting nack_history_size_.
-  // First disabling (forcing free) and then resetting to desired value.
-  if (rtp_rtcp_->SetStorePacketsStatus(false, 0) != 0) {
-    return -1;
-  }
   if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) != 0) {
     return -1;
   }

webrtc/video_engine/vie_receiver.cc

@@ -35,13 +35,13 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
       channel_id_(channel_id),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(new RTPPayloadRegistry(
-          channel_id, RTPPayloadStrategy::CreateStrategy(false))),
+          RTPPayloadStrategy::CreateStrategy(false))),
       rtp_receiver_(RtpReceiver::CreateVideoReceiver(
           channel_id, Clock::GetRealTimeClock(), this, rtp_feedback,
           rtp_payload_registry_.get())),
       rtp_receive_statistics_(ReceiveStatistics::Create(
           Clock::GetRealTimeClock())),
-      fec_receiver_(FecReceiver::Create(channel_id, this)),
+      fec_receiver_(FecReceiver::Create(this)),
       rtp_rtcp_(NULL),
       vcm_(module_vcm),
       remote_bitrate_estimator_(remote_bitrate_estimator),

webrtc/voice_engine/channel.cc

@@ -837,8 +837,7 @@ Channel::Channel(int32_t channelId,
     _channelId(channelId),
     rtp_header_parser_(RtpHeaderParser::Create()),
     rtp_payload_registry_(
-        new RTPPayloadRegistry(channelId,
+        new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
-                               RTPPayloadStrategy::CreateStrategy(true))),
     rtp_receive_statistics_(ReceiveStatistics::Create(
         Clock::GetRealTimeClock())),
     rtp_receiver_(RtpReceiver::CreateAudioReceiver(