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Thread annotate RTCPSender.

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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
This commit is contained in:
pbos@webrtc.org 2014-07-11 15:36:26 +00:00
parent 336e8e8f50
commit 180e516bef
5 changed files with 100 additions and 143 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
View File

@ -97,7 +97,6 @@ void RtcpFormatRembTest::SetUp() {
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_));
}

67
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -187,61 +187,6 @@ RTCPSender::~RTCPSender() {
delete _criticalSectionRTCPSender;
}
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)
{
@ -381,6 +326,7 @@ RTCPSender::SetIJStatus(const bool enable)
}
void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
start_timestamp_ = start_timestamp;
}
@ -686,13 +632,9 @@ int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state,
// 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 @@ RTCPSender::SendToNetwork(const uint8_t* dataBuffer,
int32_t
RTCPSender::SetCSRCStatus(const bool include)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_includeCSRCs = include;
return 0;
}

169
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -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 @@ public:
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 @@ public:
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 @@ private:
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 @@ private:
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

1
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -304,7 +304,6 @@ class RtcpSenderTest : public ::testing::Test {
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() {

5
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -47,13 +47,12 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
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);
}
}