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
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
index 20be2d5..0d6c174 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
+++ b/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) {
- return new FecReceiverImpl(id, callback);
+FecReceiver* FecReceiver::Create(RtpData* callback) {
+ return new FecReceiverImpl(callback);
}
-FecReceiverImpl::FecReceiverImpl(const int32_t id, RtpData* callback)
- : id_(id),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+FecReceiverImpl::FecReceiverImpl(RtpData* callback)
+ : 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 @@
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_,
- "Corrupt payload found in %s", __FUNCTION__);
+ LOG(LS_WARNING) << "Corrupt payload found.";
delete received_packet;
return -1;
}
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 0342123..b876bed 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -25,7 +25,7 @@
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 @@
virtual int32_t ProcessReceivedFec() OVERRIDE;
private:
- int id_;
scoped_ptr<CriticalSectionWrapper> crit_sect_;
RtpData* recovered_packet_callback_;
ForwardErrorCorrection* fec_;
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index 2e8846c..0b12449 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -39,8 +39,8 @@
class ReceiverFecTest : public ::testing::Test {
protected:
virtual void SetUp() {
- fec_ = new ForwardErrorCorrection(0);
- receiver_fec_ = FecReceiver::Create(0, &rtp_data_callback_);
+ fec_ = new ForwardErrorCorrection();
+ receiver_fec_ = FecReceiver::Create(&rtp_data_callback_);
generator_ = new FrameGenerator();
}
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index af2cb9e..31303c8 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/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::RecoveredPacket::RecoveredPacket() {}
ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() {}
-ForwardErrorCorrection::ForwardErrorCorrection(int32_t id)
- : id_(id),
- generated_fec_packets_(kMaxMediaPackets),
+ForwardErrorCorrection::ForwardErrorCorrection()
+ : generated_fec_packets_(kMaxMediaPackets),
fec_packet_received_(false) {}
ForwardErrorCorrection::~ForwardErrorCorrection() {}
@@ -112,43 +111,23 @@
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 @@
assert(media_packet);
if (media_packet->length < kRtpHeaderSize) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s media packet (%d bytes) is smaller than RTP header",
- __FUNCTION__, media_packet->length);
+ LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
+ << "is smaller than RTP header.";
return -1;
}
// Ensure our FEC packets will fit in a typical MTU.
if (media_packet->length + PacketOverhead() + kTransportOverhead >
IP_PACKET_SIZE) {
- WEBRTC_TRACE(
- kTraceError, kTraceRtpRtcp, id_,
- "%s media packet (%d bytes) with overhead is larger than MTU(%d)",
- __FUNCTION__, media_packet->length, IP_PACKET_SIZE);
- return -1;
+ LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
+ << "with overhead is larger than " << IP_PACKET_SIZE;
}
media_list_it++;
}
@@ -582,9 +557,7 @@
}
if (fec_packet->protected_pkt_list.empty()) {
// All-zero packet mask; we can discard this FEC packet.
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
- "FEC packet %u has an all-zero packet mask.",
- fec_packet->seq_num, __FUNCTION__);
+ LOG(LS_WARNING) << "FEC packet has an all-zero packet mask.";
delete fec_packet;
} else {
AssignRecoveredPackets(fec_packet, recovered_packet_list);
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
index 8910fe4..bb790f3 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -117,8 +117,7 @@
typedef std::list<ReceivedPacket*> ReceivedPacketList;
typedef std::list<RecoveredPacket*> RecoveredPacketList;
- // \param[in] id Module ID
- ForwardErrorCorrection(int32_t id);
+ ForwardErrorCorrection();
virtual ~ForwardErrorCorrection();
@@ -304,7 +303,6 @@
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_;
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 8c6cc54..34e479a 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -164,7 +164,7 @@
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_(),
diff --git a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
index ada7d70..baa3827 100644
--- a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
@@ -39,7 +39,7 @@
class ProducerFecTest : public ::testing::Test {
protected:
virtual void SetUp() {
- fec_ = new ForwardErrorCorrection(0);
+ fec_ = new ForwardErrorCorrection();
producer_ = new ProducerFec(fec_);
generator_ = new FrameGenerator;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
index 0c3197f..a127bc1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/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 @@
uint16_t* len,
uint16_t max_len) const {
if (*len + Length() > max_len) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max packet size reached, skipped SR.");
+ LOG(LS_WARNING) << "Max packet size reached.";
return;
}
CreateSenderReport(sr_, packet, len);
@@ -244,8 +243,7 @@
void SenderReport::WithReportBlock(ReportBlock* block) {
assert(block);
if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max report block size reached.");
+ LOG(LS_WARNING) << "Max report blocks reached.";
return;
}
report_blocks_.push_back(block);
@@ -256,8 +254,7 @@
uint16_t* len,
uint16_t max_len) const {
if (*len + Length() > max_len) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max packet size reached, skipped RR.");
+ LOG(LS_WARNING) << "Max packet size reached.";
return;
}
CreateReceiverReport(rr_, packet, len);
@@ -267,8 +264,7 @@
void ReceiverReport::WithReportBlock(ReportBlock* block) {
assert(block);
if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max report block size reached.");
+ LOG(LS_WARNING) << "Max report blocks reached.";
return;
}
report_blocks_.push_back(block);
@@ -277,8 +273,7 @@
void Bye::Create(uint8_t* packet, uint16_t* len, uint16_t max_len) const {
if (*len + Length() > max_len) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max packet size reached, skipped BYE.");
+ LOG(LS_WARNING) << "Max packet size reached.";
return;
}
CreateBye(bye_, csrcs_, packet, len);
@@ -286,8 +281,7 @@
void Bye::WithCsrc(uint32_t csrc) {
if (csrcs_.size() >= kMaxNumberOfCsrcs) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max CSRC size reached.");
+ LOG(LS_WARNING) << "Max CSRC size reached.";
return;
}
csrcs_.push_back(csrc);
@@ -295,8 +289,7 @@
void Fir::Create(uint8_t* packet, uint16_t* len, uint16_t max_len) const {
if (*len + Length() > max_len) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Max packet size reached, skipped FIR.");
+ LOG(LS_WARNING) << "Max packet size reached.";
return;
}
CreateFirRequest(fir_, fir_item_, packet, len);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 4012a81..21ccf67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/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 @@
_lastIncreasedSequenceNumberMs(0),
stats_callback_(NULL) {
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
RTCPReceiver::~RTCPReceiver() {
@@ -82,8 +81,6 @@
delete first->second;
_receivedCnameMap.erase(first);
}
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id,
- "%s deleted", __FUNCTION__);
}
void
@@ -178,8 +175,7 @@
RTCPReportBlockInformation* reportBlock =
GetReportBlockInformation(remoteSSRC);
if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tfailed to GetReportBlockInformation(%u)", remoteSSRC);
+ LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC;
return -1;
}
reportBlock->RTT = 0;
@@ -282,22 +278,14 @@
return true;
}
-int32_t
-RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const
-{
- if(senderInfo == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
- 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;
+int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const {
+ assert(senderInfo);
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ if (_lastReceivedSRNTPsecs == 0) {
+ return -1;
+ }
+ memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
+ return 0;
}
// statistics
@@ -518,8 +506,8 @@
RTCPReportBlockInformation* reportBlock =
CreateReportBlockInformation(remoteSSRC);
if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC);
+ LOG(LS_WARNING) << "Failed to CreateReportBlockInformation("
+ << remoteSSRC << ")";
return;
}
@@ -779,9 +767,6 @@
}
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 @@
TMMBRSet* boundingSet = NULL;
numBoundingSet = FindTMMBRBoundingSet(boundingSet);
if (numBoundingSet == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Failed to find TMMBR bounding set.");
+ LOG(LS_WARNING) << "Failed to find TMMBR bounding set.";
return -1;
}
// Set bounding set
@@ -1369,8 +1353,6 @@
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 @@
// 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 @@
}
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming NACK length:%d",
- rtcpPacketInformation.nackSequenceNumbers.size());
+ LOG(LS_INFO) << "Incoming NACK length: "
+ << rtcpPacketInformation.nackSequenceNumbers.size();
_rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers);
}
}
@@ -1429,13 +1407,11 @@
if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) {
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming PLI from SSRC:0x%x",
- rtcpPacketInformation.remoteSSRC);
+ LOG(LS_INFO) << "Incoming PLI from SSRC "
+ << rtcpPacketInformation.remoteSSRC;
} else {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming FIR from SSRC:0x%x",
- rtcpPacketInformation.remoteSSRC);
+ LOG(LS_INFO) << "Incoming FIR from SSRC "
+ << rtcpPacketInformation.remoteSSRC;
}
_cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc);
}
@@ -1450,9 +1426,8 @@
}
if (_cbRtcpBandwidthObserver) {
if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming REMB:%d",
- rtcpPacketInformation.receiverEstimatedMaxBitrate);
+ LOG(LS_INFO) << "Incoming REMB: "
+ << rtcpPacketInformation.receiverEstimatedMaxBitrate;
_cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(
rtcpPacketInformation.receiverEstimatedMaxBitrate);
}
@@ -1548,9 +1523,6 @@
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();
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 6e6edf8..d73de9c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/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 @@
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 @@
}
delete _criticalSectionTransport;
delete _criticalSectionRTCPSender;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
}
int32_t
@@ -427,7 +423,8 @@
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 @@
uint32_t SSRC,
std::map<uint32_t, RTCPReportBlock*>* report_blocks,
const RTCPReportBlock* reportBlock) {
- if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
+ assert(reportBlock);
if (report_blocks->size() >= RTCP_MAX_REPORT_BLOCKS) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
+ LOG(LS_WARNING) << "Too many report blocks.";
return -1;
}
std::map<uint32_t, RTCPReportBlock*>::iterator it =
@@ -677,7 +669,7 @@
// 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 @@
// sanity
if(pos + 12 + lengthCname >= IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
+ LOG(LS_WARNING) << "Failed to build SDEC.";
return -2;
}
// SDEC Source Description
@@ -913,7 +904,9 @@
{
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 @@
// 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 @@
// 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 @@
// 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 @@
numOfNackFields++;
}
if (i != nackSize) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Nack list to large for one packet.");
+ LOG(LS_WARNING) << "Nack list to large for one packet.";
}
rtcpbuffer[nackSizePos] = static_cast<uint8_t>(2 + numOfNackFields);
*nackString = stringBuilder.GetResult();
@@ -1715,8 +1707,7 @@
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_method == kRtcpOff)
{
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "%s invalid state", __FUNCTION__);
+ LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
return -1;
}
}
@@ -2128,13 +2119,7 @@
RTCPSender::SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize],
const uint8_t arrLength)
{
- if(arrLength > kRtpCsrcSize)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- assert(false);
- return -1;
- }
-
+ assert(arrLength <= kRtpCsrcSize);
CriticalSectionScoped lock(_criticalSectionRTCPSender);
for(int i = 0; i < arrLength;i++)
@@ -2153,7 +2138,7 @@
{
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 @@
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,
- "%s invalid argument", __FUNCTION__);
+ LOG(LS_WARNING) << "Can't fit all report blocks.";
return -1;
}
pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, internal_report_blocks_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 8474390..dfb655c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -278,7 +278,7 @@
: 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(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
index 904156e..fa84762 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -41,7 +41,7 @@
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 @@
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;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index d048725..bb24d4d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/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 @@
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;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index 73b2326..fb43563 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/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() {
- 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::Allocate(uint16_t number_to_store) {
assert(number_to_store > 0);
- CriticalSectionScoped cs(critsect_);
- if (store_) {
- if (number_to_store != stored_packets_.size()) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "SetStorePacketsStatus already set, number: %d",
- number_to_store);
- }
- return;
- }
-
+ assert(!store_);
store_ = true;
stored_packets_.resize(number_to_store);
stored_seq_nums_.resize(number_to_store);
@@ -68,7 +67,6 @@
}
void RTPPacketHistory::Free() {
- CriticalSectionScoped cs(critsect_);
if (!store_) {
return;
}
@@ -133,8 +131,8 @@
VerifyAndAllocatePacketLength(max_packet_length);
if (packet_length > max_packet_length_) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, -1,
- "Failed to store RTP packet, length: %d", packet_length);
+ LOG(LS_WARNING) << "Failed to store RTP packet with length: "
+ << packet_length;
return -1;
}
@@ -169,25 +167,20 @@
assert(packet);
assert(rtp_header_length > 3);
-
- if (rtp_header_length > max_packet_length_) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "Failed to replace RTP packet, length: %d", rtp_header_length);
- return -1;
- }
+ assert(rtp_header_length <= max_packet_length_);
int32_t index = 0;
bool found = FindSeqNum(sequence_number, &index);
if (!found) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "No match for getting seqNum %u", sequence_number);
+ LOG(LS_WARNING)
+ << "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,
- "No match for getting seqNum %u, len %d", sequence_number, length);
+ LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number
+ << ", len " << length;
return -1;
}
assert(stored_seq_nums_[index] == sequence_number);
@@ -225,6 +218,7 @@
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 @@
int32_t index = 0;
bool found = FindSeqNum(sequence_number, &index);
if (!found) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "No match for getting seqNum %u", sequence_number);
+ LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number;
return false;
}
uint16_t length = stored_lengths_.at(index);
- if (length == 0 || length > max_packet_length_) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "No match for getting seqNum %u, len %d", sequence_number, length);
- return false;
- }
-
- if (length > *packet_length) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Input buffer too short for packet %u", sequence_number);
+ assert(length <= max_packet_length_);
+ if (length == 0) {
+ LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number
+ << ", len " << length;
return false;
}
@@ -255,8 +243,6 @@
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;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 785e499..a657d41 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/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 @@
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 Free();
+ void Allocate(uint16_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+ 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;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 1682b7c..1072518 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -103,19 +103,6 @@
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 @@
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 @@
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 @@
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,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 1c3b990..61190f1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/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 @@
case 77: // 205 Transport layer FB message.
case 78: // 206 Payload-specific FB message.
case 79: // 207 Extended report.
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s invalid payloadtype:%d",
- __FUNCTION__, payload_type);
+ LOG(LS_ERROR) << "Can't register invalid receiver payload type: "
+ << payload_type;
return -1;
default:
break;
@@ -94,9 +91,7 @@
return 0;
}
}
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s invalid argument payload_type:%d already registered",
- __FUNCTION__, payload_type);
+ LOG(LS_ERROR) << "Payload type already registered: " << payload_type;
return -1;
}
@@ -138,14 +133,8 @@
const int8_t payload_type) {
CriticalSectionScoped cs(crit_sect_.get());
ModuleRTPUtility::PayloadTypeMap::iterator it =
- payload_type_map_.find(payload_type);
-
- if (it == payload_type_map_.end()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s failed to find payload_type:%d",
- __FUNCTION__, payload_type);
- return -1;
- }
+ payload_type_map_.find(payload_type);
+ assert(it != payload_type_map_.end());
delete it->second;
payload_type_map_.erase(it);
return 0;
@@ -194,11 +183,7 @@
const uint8_t channels,
const uint32_t rate,
int8_t* payload_type) const {
- if (payload_type == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
+ assert(payload_type);
size_t payload_name_length = strlen(payload_name);
CriticalSectionScoped cs(crit_sect_.get());
@@ -296,8 +281,7 @@
(*restored_packet)[1] |= kRtpMarkerBitMask; // Marker bit is set.
}
} else {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
- "Incorrect RTX configuration, dropping packet.");
+ LOG(LS_WARNING) << "Incorrect RTX configuration, dropping packet.";
return false;
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 96fa80a..c03ffcd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -32,8 +32,7 @@
void SetUp() {
// Note: the payload registry takes ownership of the strategy.
mock_payload_strategy_ = new testing::NiceMock<MockRTPPayloadStrategy>();
- rtp_payload_registry_.reset(
- new RTPPayloadRegistry(123, mock_payload_strategy_));
+ rtp_payload_registry_.reset(new RTPPayloadRegistry(mock_payload_strategy_));
}
protected:
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 1345485..c8104cc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/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 @@
specific_payload.Audio.frequency,
specific_payload.Audio.channels,
specific_payload.Audio.rate)) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- id,
- "Failed to create video decoder for payload type:%d",
- payload_type);
+ LOG(LS_ERROR) << "Failed to create decoder for payload type: "
+ << payload_name << "/" << payload_type;
return -1;
}
return 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 9a27681..d92618f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/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 @@
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 @@
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 @@
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 @@
if (created_new_payload) {
if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_name, payload_type,
frequency) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s failed to register payload",
- __FUNCTION__);
+ LOG(LS_ERROR) << "Failed to register payload: " << payload_name << "/"
+ << payload_type;
return -1;
}
}
@@ -182,19 +178,12 @@
PayloadUnion payload_specific,
bool in_order) {
// Sanity check.
- if (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];
- }
+ assert(payload_length >= 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 @@
&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_,
- "%s received invalid payloadtype",
- __FUNCTION__);
+ LOG(LS_WARNING) << "Receiving invalid payload type.";
return false;
}
@@ -347,9 +331,8 @@
id_, rtp_header.payloadType, payload_name,
rtp_header.payload_type_frequency, channels, rate)) {
// New stream, same codec.
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "Failed to create decoder for payload type:%d",
- rtp_header.payloadType);
+ LOG(LS_ERROR) << "Failed to create decoder for payload type: "
+ << rtp_header.payloadType;
}
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index d8a2257..09c9b6f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -26,8 +26,7 @@
// This class is not thread-safe and must be protected by its caller.
class RTPReceiverStrategy {
public:
- static RTPReceiverStrategy* CreateVideoStrategy(int32_t id,
- RtpData* data_callback);
+ static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
static RTPReceiverStrategy* CreateAudioStrategy(
int32_t id, RtpData* data_callback,
RtpAudioFeedback* incoming_messages_callback);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index b733cdb..5bb519f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/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) {
- return new RTPReceiverVideo(id, data_callback);
+ RtpData* data_callback) {
+ return new RTPReceiverVideo(data_callback);
}
-RTPReceiverVideo::RTPReceiverVideo(int32_t id, RtpData* data_callback)
- : RTPReceiverStrategy(data_callback),
- id_(id) {}
+RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
+ : RTPReceiverStrategy(data_callback) {}
RTPReceiverVideo::~RTPReceiverVideo() {
}
@@ -93,11 +92,8 @@
// For video we just go with default values.
if (-1 == callback->OnInitializeDecoder(
id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- id,
- "Failed to create video decoder for payload type:%d",
- payload_type);
+ LOG(LS_ERROR) << "Failed to created decoder for payload type: "
+ << payload_type;
return -1;
}
return 0;
@@ -111,13 +107,6 @@
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 @@
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;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index ab69b40..4d81cb3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -22,7 +22,7 @@
class RTPReceiverVideo : public RTPReceiverStrategy {
public:
- RTPReceiverVideo(const int32_t id, RtpData* data_callback);
+ RTPReceiverVideo(RtpData* data_callback);
virtual ~RTPReceiverVideo();
@@ -80,8 +80,6 @@
RtpVideoCodecTypes video_type,
int64_t now_ms,
bool is_first_packet);
-
- int32_t id_;
};
} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 575da60..dc65336 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -110,13 +110,9 @@
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 @@
}
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::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 @@
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_,
- "IncomingRtcpPacket invalid RTCP packet");
+ LOG(LS_WARNING) << "Incoming invalid RTCP packet";
return -1;
}
RTCPHelp::RTCPPacketInformation rtcp_packet_information;
@@ -318,14 +286,6 @@
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(
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::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 @@
}
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::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 @@
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 @@
}
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_,
- "Failed to send RTCP BYE");
+ LOG(LS_WARNING) << "Failed to send RTCP BYE";
}
collision_detected_ = false;
@@ -525,25 +430,15 @@
}
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 @@
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 @@
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 @@
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 @@
}
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 @@
}
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 @@
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::SetMaxTransferUnit(const uint16_t mtu) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetMaxTransferUnit(%u)",
- mtu);
if (mtu > IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
- "Invalid in argument to SetMaxTransferUnit(%u)", mtu);
+ LOG(LS_ERROR) << "Invalid mtu: " << mtu;
return -1;
}
return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
@@ -813,7 +670,6 @@
}
RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "RTCP()");
if (rtcp_sender_.Status() != kRtcpOff) {
return rtcp_receiver_.Status();
}
@@ -822,8 +678,6 @@
// 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 @@
}
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 @@
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 @@
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 @@
// 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 @@
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 @@
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::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 @@
// (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 @@
// (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 @@
// 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::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 @@
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 @@
// 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 @@
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 @@
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::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 @@
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 @@
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.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 29e4616..1e5fb3c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/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 @@
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 @@
}
delete audio_;
delete video_;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
}
void RTPSender::SetTargetSendBitrate(const uint32_t bits) {
@@ -290,16 +287,12 @@
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",
- __FUNCTION__);
+ LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
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 @@
CriticalSectionScoped cs(send_critsect_);
if (payload_type < 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)",
- payload_type);
+ LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
return -1;
}
if (audio_configured_) {
@@ -373,8 +365,7 @@
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_,
- "\tpayloadType:%d not registered", payload_type);
+ LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
return -1;
}
payload_type_ = payload_type;
@@ -403,9 +394,7 @@
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
- "%s invalid argument failed to find payload_type:%d",
- __FUNCTION__, payload_type);
+ LOG(LS_ERROR) << "Don't send data with unknown payload type.";
return -1;
}
@@ -616,8 +605,6 @@
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 @@
}
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_,
- "Transport failed to send packet");
+ LOG(LS_WARNING) << "Transport failed to send packet";
return false;
}
return true;
@@ -676,11 +662,8 @@
// Enough bandwidth to send NACK?
if (!ProcessNACKBitRate(now)) {
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- id_,
- "NACK bitrate reached. Skip sending NACK response. Target %d",
- target_bitrate_kbps);
+ LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
+ << target_bitrate_kbps;
return;
}
@@ -695,9 +678,8 @@
continue;
} else if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
- "Failed resending RTP packet %d, Discard rest of packets",
- *it);
+ LOG(LS_WARNING) << "Failed resending RTP packet " << *it
+ << ", Discard rest of packets";
break;
}
// Delay bandwidth estimate (RTT * BW).
@@ -1258,39 +1240,36 @@
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionTransmissionTimeOffset);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, not registered.");
+ LOG(LS_WARNING)
+ << "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_,
- "Failed to update transmission time offset, invalid length.");
+ LOG(LS_WARNING)
+ << "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(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update transmission time offset, hdr extension not found.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update transmission time offset, no id.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update transmission time offset.");
+ LOG(LS_WARNING) << "Failed to update transmission time offset.";
return false;
}
// Update transmission offset field (converting to a 90 kHz timestamp).
@@ -1311,37 +1290,31 @@
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAudioLevel);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, not registered.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update audio level, invalid length.");
+ LOG(LS_WARNING) << "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(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update audio level, hdr extension not found.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update audio level, no id.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update audio level.");
+ LOG(LS_WARNING) << "Failed to update audio level.";
return false;
}
rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
@@ -1358,38 +1331,33 @@
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAbsoluteSendTime);
if (extension_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, not registered.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update absolute send time, invalid length.");
+ LOG(LS_WARNING) << "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(
- kTraceStream, kTraceRtpRtcp, id_,
- "Failed to update absolute send time, hdr extension not found.");
+ LOG(LS_WARNING)
+ << "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_,
- "Failed to update absolute send time, no id.");
+ LOG(LS_WARNING) << "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_,
- "Failed to update absolute send time.");
+ LOG(LS_WARNING) << "Failed to update absolute send time.";
return false;
}
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 10bc252..5d8ae16 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/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 @@
ForwardErrorCorrection::Packet* pkt;
};
-RTPSenderVideo::RTPSenderVideo(const int32_t id,
- Clock* clock,
+RTPSenderVideo::RTPSenderVideo(Clock* clock,
RTPSenderInterface* rtpSender)
- : _id(id),
- _rtpSender(*rtpSender),
+ : _rtpSender(*rtpSender),
_sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_videoType(kRtpVideoGeneric),
_videoCodecInformation(NULL),
@@ -43,7 +41,7 @@
_retransmissionSettings(kRetransmitBaseLayer),
// Generic FEC
- _fec(id),
+ _fec(),
_fecEnabled(false),
_payloadTypeRED(-1),
_payloadTypeFEC(-1),
@@ -329,8 +327,6 @@
{
return retVal;
}
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "%s(timestamp:%u)",
- __FUNCTION__, captureTimeStamp);
return 0;
}
@@ -476,9 +472,9 @@
rtpHeaderLength, captureTimeStamp,
capture_time_ms, storage, protect))
{
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "RTPSenderVideo::SendVP8 failed to send packet number"
- " %d", _rtpSender.SequenceNumber());
+ LOG(LS_WARNING)
+ << "RTPSenderVideo::SendVP8 failed to send packet number "
+ << _rtpSender.SequenceNumber();
}
}
TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 4c406d7..daa730e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -31,7 +31,7 @@
class RTPSenderVideo
{
public:
- RTPSenderVideo(const int32_t id, Clock* clock,
+ RTPSenderVideo(Clock* clock,
RTPSenderInterface* rtpSender);
virtual ~RTPSenderVideo();
@@ -112,7 +112,6 @@
const RTPVideoTypeHeader* rtpTypeHdr);
private:
- int32_t _id;
RTPSenderInterface& _rtpSender;
CriticalSectionWrapper* _sendVideoCritsect;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index 2aa218b..3f99845 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/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 @@
ptr++;
if (id == 15) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Ext id: 15 encountered, parsing terminated.");
+ LOG(LS_WARNING)
+ << "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,
- "Failed to find extension id: %d", id);
+ LOG(LS_WARNING) << "Failed to find extension id: " << id;
} else {
switch (type) {
case kRtpExtensionTransmissionTimeOffset: {
if (len != 2) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Incorrect transmission time offset len: %d", len);
+ LOG(LS_WARNING) << "Incorrect transmission time offset len: "
+ << len;
return;
}
// 0 1 2 3
@@ -502,8 +501,7 @@
}
case kRtpExtensionAudioLevel: {
if (len != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Incorrect audio level len: %d", len);
+ LOG(LS_WARNING) << "Incorrect audio level len: " << len;
return;
}
// 0 1 2 3
@@ -525,8 +523,7 @@
}
case kRtpExtensionAbsoluteSendTime: {
if (len != 2) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Incorrect absolute send time len: %d", len);
+ LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
return;
}
// 0 1 2 3
@@ -543,8 +540,7 @@
break;
}
default: {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "Extension type not implemented.");
+ LOG(LS_WARNING) << "Extension type not implemented: " << type;
return;
}
}
@@ -570,17 +566,12 @@
return num_zero_bytes;
}
-// RTP payload parser
RTPPayloadParser::RTPPayloadParser(const RtpVideoCodecTypes videoType,
const uint8_t* payloadData,
- uint16_t payloadDataLength,
- int32_t id)
- :
- _id(id),
- _dataPtr(payloadData),
- _dataLength(payloadDataLength),
- _videoType(videoType) {
-}
+ uint16_t payloadDataLength)
+ : _dataPtr(payloadData),
+ _dataLength(payloadDataLength),
+ _videoType(videoType) {}
RTPPayloadParser::~RTPPayloadParser() {
}
@@ -655,8 +646,7 @@
}
if (dataLength <= 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Error parsing VP8 payload descriptor; payload too short");
+ LOG(LS_ERROR) << "Error parsing VP8 payload descriptor!";
return false;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index 8002273..732301f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -166,8 +166,8 @@
public:
RTPPayloadParser(const RtpVideoCodecTypes payloadType,
const uint8_t* payloadData,
- const uint16_t payloadDataLength, // Length w/o padding.
- const int32_t id);
+ // Length w/o padding.
+ const uint16_t payloadDataLength);
~RTPPayloadParser();
@@ -202,7 +202,6 @@
int dataLength) const;
private:
- int32_t _id;
const uint8_t* _dataPtr;
const uint16_t _dataLength;
const RtpVideoCodecTypes _videoType;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
index 02a89fc..d33eaf4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility_unittest.cc
@@ -76,7 +76,7 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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));
diff --git a/webrtc/modules/rtp_rtcp/source/ssrc_database.cc b/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
index 1e57970..df09b01 100644
--- a/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
+++ b/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 @@
_ssrcVector = new uint32_t[10];
#endif
_critSect = CriticalSectionWrapper::CreateCriticalSection();
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s created", __FUNCTION__);
}
SSRCDatabase::~SSRCDatabase()
@@ -197,8 +194,6 @@
_ssrcMap.clear();
#endif
delete _critSect;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s deleted", __FUNCTION__);
}
uint32_t SSRCDatabase::GenerateRandom()