Thread annotate RTCPSender.
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 88463e4..0514277 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -97,7 +97,6 @@
rtcp_receiver_ = new RTCPReceiver(0, system_clock_, dummy_rtp_rtcp_impl_);
test_transport_ = new TestTransport(rtcp_receiver_);
- EXPECT_EQ(0, rtcp_sender_->Init());
EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index fa129ab..2cf7e1c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -188,61 +188,6 @@
}
int32_t
-RTCPSender::Init()
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- _method = kRtcpOff;
- _cbTransport = NULL;
- _usingNack = false;
- _sending = false;
- _sendTMMBN = false;
- _TMMBR = false;
- _IJ = false;
- _REMB = false;
- _sendREMB = false;
- last_rtp_timestamp_ = 0;
- last_frame_capture_time_ms_ = -1;
- start_timestamp_ = -1;
- _SSRC = 0;
- _remoteSSRC = 0;
- _cameraDelayMS = 0;
- _sequenceNumberFIR = 0;
- _tmmbr_Send = 0;
- _packetOH_Send = 0;
- _nextTimeToSendRTCP = 0;
- _CSRCs = 0;
- _appSend = false;
- _appSubType = 0;
-
- if(_appData)
- {
- delete [] _appData;
- _appData = NULL;
- }
- _appLength = 0;
-
- xrSendReceiverReferenceTimeEnabled_ = false;
-
- _xrSendVoIPMetric = false;
-
- memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric));
- memset(_CNAME, 0, sizeof(_CNAME));
- memset(_lastSendReport, 0, sizeof(_lastSendReport));
- memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
- last_xr_rr_.clear();
-
- memset(&packet_type_counter_, 0, sizeof(packet_type_counter_));
- return 0;
-}
-
-void
-RTCPSender::ChangeUniqueId(const int32_t id)
-{
- _id = id;
-}
-
-int32_t
RTCPSender::RegisterSendTransport(Transport* outgoingTransport)
{
CriticalSectionScoped lock(_criticalSectionTransport);
@@ -381,6 +326,7 @@
}
void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
start_timestamp_ = start_timestamp;
}
@@ -686,13 +632,9 @@
// the frame being captured at this moment. We are calculating that
// timestamp as the last frame's timestamp + the time since the last frame
// was captured.
- {
- // Needs protection since this method is called on the process thread.
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- RTPtime = start_timestamp_ + last_rtp_timestamp_ + (
- _clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
- (feedback_state.frequency_hz / 1000);
- }
+ RTPtime = start_timestamp_ + last_rtp_timestamp_ +
+ (_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
+ (feedback_state.frequency_hz / 1000);
// Add sender data
// Save for our length field
@@ -2102,6 +2044,7 @@
int32_t
RTCPSender::SetCSRCStatus(const bool include)
{
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
_includeCSRCs = include;
return 0;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 71d92c8..fad3b5e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -23,6 +23,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -74,10 +75,6 @@
ReceiveStatistics* receive_statistics);
virtual ~RTCPSender();
- void ChangeUniqueId(const int32_t id);
-
- int32_t Init();
-
int32_t RegisterSendTransport(Transport* outgoingTransport);
RTCPMethod Status() const;
@@ -184,13 +181,12 @@
private:
int32_t SendToNetwork(const uint8_t* dataBuffer, const uint16_t length);
- void UpdatePacketRate();
-
int32_t WriteAllReportBlocksToBuffer(uint8_t* rtcpbuffer,
int pos,
uint8_t& numberOfReportBlocks,
const uint32_t NTPsec,
- const uint32_t NTPfrac);
+ const uint32_t NTPfrac)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t WriteReportBlocksToBuffer(
uint8_t* rtcpbuffer,
@@ -211,12 +207,14 @@
uint8_t* rtcpbuffer,
int& pos,
uint32_t NTPsec,
- uint32_t NTPfrac);
+ uint32_t NTPfrac)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t BuildRR(uint8_t* rtcpbuffer,
int& pos,
const uint32_t NTPsec,
- const uint32_t NTPfrac);
+ const uint32_t NTPfrac)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int PrepareRTCP(
const FeedbackState& feedback_state,
@@ -233,117 +231,136 @@
int32_t BuildExtendedJitterReport(
uint8_t* rtcpbuffer,
int& pos,
- const uint32_t jitterTransmissionTimeOffset);
+ const uint32_t jitterTransmissionTimeOffset)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
- int32_t BuildSDEC(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildPLI(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildREMB(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildTMMBR(ModuleRtpRtcpImpl* module,
- uint8_t* rtcpbuffer,
- int& pos);
- int32_t BuildTMMBN(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildAPP(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildBYE(uint8_t* rtcpbuffer, int& pos);
- int32_t BuildFIR(uint8_t* rtcpbuffer, int& pos, bool repeat);
- int32_t BuildSLI(uint8_t* rtcpbuffer,
- int& pos,
- const uint8_t pictureID);
+ int32_t BuildSDEC(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildPLI(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildREMB(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildTMMBR(ModuleRtpRtcpImpl* module, uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildAPP(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildBYE(uint8_t* rtcpbuffer, int& pos)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildFIR(uint8_t* rtcpbuffer, int& pos, bool repeat)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
+ int32_t BuildSLI(uint8_t* rtcpbuffer, int& pos, const uint8_t pictureID)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t BuildRPSI(uint8_t* rtcpbuffer,
int& pos,
const uint64_t pictureID,
- const uint8_t payloadType);
+ const uint8_t payloadType)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t BuildNACK(uint8_t* rtcpbuffer,
int& pos,
const int32_t nackSize,
const uint16_t* nackList,
- std::string* nackString);
-
+ std::string* nackString)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t BuildReceiverReferenceTime(uint8_t* buffer,
int& pos,
uint32_t ntp_sec,
- uint32_t ntp_frac);
+ uint32_t ntp_frac)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
int32_t BuildDlrr(uint8_t* buffer,
int& pos,
- const RtcpReceiveTimeInfo& info);
+ const RtcpReceiveTimeInfo& info)
+ EXCLUSIVE_LOCKS_REQUIRED(_criticalSectionRTCPSender);
private:
- int32_t _id;
+ const int32_t _id;
const bool _audio;
- Clock* _clock;
- RTCPMethod _method;
+ Clock* const _clock;
+ RTCPMethod _method GUARDED_BY(_criticalSectionRTCPSender);
CriticalSectionWrapper* _criticalSectionTransport;
- Transport* _cbTransport;
+ Transport* _cbTransport GUARDED_BY(_criticalSectionTransport);
CriticalSectionWrapper* _criticalSectionRTCPSender;
- bool _usingNack;
- bool _sending;
- bool _sendTMMBN;
- bool _REMB;
- bool _sendREMB;
- bool _TMMBR;
- bool _IJ;
+ bool _usingNack GUARDED_BY(_criticalSectionRTCPSender);
+ bool _sending GUARDED_BY(_criticalSectionRTCPSender);
+ bool _sendTMMBN GUARDED_BY(_criticalSectionRTCPSender);
+ bool _REMB GUARDED_BY(_criticalSectionRTCPSender);
+ bool _sendREMB GUARDED_BY(_criticalSectionRTCPSender);
+ bool _TMMBR GUARDED_BY(_criticalSectionRTCPSender);
+ bool _IJ GUARDED_BY(_criticalSectionRTCPSender);
- int64_t _nextTimeToSendRTCP;
+ int64_t _nextTimeToSendRTCP GUARDED_BY(_criticalSectionRTCPSender);
- uint32_t start_timestamp_;
- uint32_t last_rtp_timestamp_;
- int64_t last_frame_capture_time_ms_;
- uint32_t _SSRC;
- uint32_t _remoteSSRC; // SSRC that we receive on our RTP channel
- char _CNAME[RTCP_CNAME_SIZE];
+ uint32_t start_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t last_rtp_timestamp_ GUARDED_BY(_criticalSectionRTCPSender);
+ int64_t last_frame_capture_time_ms_ GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _SSRC GUARDED_BY(_criticalSectionRTCPSender);
+ // SSRC that we receive on our RTP channel
+ uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPSender);
+ char _CNAME[RTCP_CNAME_SIZE] GUARDED_BY(_criticalSectionRTCPSender);
+ ReceiveStatistics* receive_statistics_
+ GUARDED_BY(_criticalSectionRTCPSender);
+ std::map<uint32_t, RTCPReportBlock*> internal_report_blocks_
+ GUARDED_BY(_criticalSectionRTCPSender);
+ std::map<uint32_t, RTCPReportBlock*> external_report_blocks_
+ GUARDED_BY(_criticalSectionRTCPSender);
+ std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs
+ GUARDED_BY(_criticalSectionRTCPSender);
- ReceiveStatistics* receive_statistics_;
- std::map<uint32_t, RTCPReportBlock*> internal_report_blocks_;
- std::map<uint32_t, RTCPReportBlock*> external_report_blocks_;
- std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
-
- int32_t _cameraDelayMS;
+ int32_t _cameraDelayMS GUARDED_BY(_criticalSectionRTCPSender);
// Sent
- uint32_t _lastSendReport[RTCP_NUMBER_OF_SR]; // allow packet loss and RTT above 1 sec
- uint32_t _lastRTCPTime[RTCP_NUMBER_OF_SR];
+ uint32_t _lastSendReport[RTCP_NUMBER_OF_SR] GUARDED_BY(
+ _criticalSectionRTCPSender); // allow packet loss and RTT above 1 sec
+ uint32_t _lastRTCPTime[RTCP_NUMBER_OF_SR] GUARDED_BY(
+ _criticalSectionRTCPSender);
// Sent XR receiver reference time report.
// <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
- std::map<uint32_t, int64_t> last_xr_rr_;
+ std::map<uint32_t, int64_t> last_xr_rr_
+ GUARDED_BY(_criticalSectionRTCPSender);
// send CSRCs
- uint8_t _CSRCs;
- uint32_t _CSRC[kRtpCsrcSize];
- bool _includeCSRCs;
+ uint8_t _CSRCs GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _CSRC[kRtpCsrcSize] GUARDED_BY(_criticalSectionRTCPSender);
+ bool _includeCSRCs GUARDED_BY(_criticalSectionRTCPSender);
// Full intra request
- uint8_t _sequenceNumberFIR;
+ uint8_t _sequenceNumberFIR GUARDED_BY(_criticalSectionRTCPSender);
// REMB
- uint8_t _lengthRembSSRC;
- uint8_t _sizeRembSSRC;
- uint32_t* _rembSSRC;
- uint32_t _rembBitrate;
+ uint8_t _lengthRembSSRC GUARDED_BY(_criticalSectionRTCPSender);
+ uint8_t _sizeRembSSRC GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t* _rembSSRC GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _rembBitrate GUARDED_BY(_criticalSectionRTCPSender);
- TMMBRHelp _tmmbrHelp;
- uint32_t _tmmbr_Send;
- uint32_t _packetOH_Send;
+ TMMBRHelp _tmmbrHelp GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _tmmbr_Send GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _packetOH_Send GUARDED_BY(_criticalSectionRTCPSender);
// APP
- bool _appSend;
- uint8_t _appSubType;
- uint32_t _appName;
- uint8_t* _appData;
- uint16_t _appLength;
+ bool _appSend GUARDED_BY(_criticalSectionRTCPSender);
+ uint8_t _appSubType GUARDED_BY(_criticalSectionRTCPSender);
+ uint32_t _appName GUARDED_BY(_criticalSectionRTCPSender);
+ uint8_t* _appData GUARDED_BY(_criticalSectionRTCPSender);
+ uint16_t _appLength GUARDED_BY(_criticalSectionRTCPSender);
// True if sending of XR Receiver reference time report is enabled.
- bool xrSendReceiverReferenceTimeEnabled_;
+ bool xrSendReceiverReferenceTimeEnabled_
+ GUARDED_BY(_criticalSectionRTCPSender);
// XR VoIP metric
- bool _xrSendVoIPMetric;
- RTCPVoIPMetric _xrVoIPMetric;
+ bool _xrSendVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
+ RTCPVoIPMetric _xrVoIPMetric GUARDED_BY(_criticalSectionRTCPSender);
- RtcpPacketTypeCounter packet_type_counter_;
+ RtcpPacketTypeCounter packet_type_counter_
+ GUARDED_BY(_criticalSectionRTCPSender);
};
} // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index dfb655c..cba1c34 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -304,7 +304,6 @@
rtcp_receiver_ = new RTCPReceiver(0, &clock_, rtp_rtcp_impl_);
test_transport_->SetRTCPReceiver(rtcp_receiver_);
// Initialize
- EXPECT_EQ(0, rtcp_sender_->Init());
EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
}
~RtcpSenderTest() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 0c771c5..34a08a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -47,13 +47,12 @@
if (configuration.clock) {
return new ModuleRtpRtcpImpl(configuration);
} else {
+ // No clock implementation provided, use default clock.
RtpRtcp::Configuration configuration_copy;
memcpy(&configuration_copy, &configuration,
sizeof(RtpRtcp::Configuration));
configuration_copy.clock = Clock::GetRealTimeClock();
- ModuleRtpRtcpImpl* rtp_rtcp_instance =
- new ModuleRtpRtcpImpl(configuration_copy);
- return rtp_rtcp_instance;
+ return new ModuleRtpRtcpImpl(configuration_copy);
}
}
