Set RtcpSender transport at construction.
BUG=
Review URL: https://codereview.webrtc.org/1365043002
Cr-Commit-Position: refs/heads/master@{#10106}
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 38c1c04..ee5fe13 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -93,6 +93,7 @@
'<(webrtc_root)/test/test.gyp:frame_generator',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
'<(webrtc_root)/tools/tools.gyp:agc_test_utils',
],
'sources': [
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 3f0b047..9652a53 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -18,6 +18,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/test/null_transport.h"
#include "webrtc/typedefs.h"
namespace {
@@ -79,6 +80,7 @@
RTCPSender* rtcp_sender_;
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
+ test::NullTransport null_transport_;
MockRemoteBitrateObserver remote_bitrate_observer_;
rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
};
@@ -88,14 +90,13 @@
configuration.audio = false;
configuration.clock = system_clock_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
+ configuration.outgoing_transport = &null_transport_;
dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
- rtcp_sender_ =
- new RTCPSender(false, system_clock_, receive_statistics_.get(), nullptr);
rtcp_receiver_ = new RTCPReceiver(system_clock_, false, nullptr, nullptr,
nullptr, nullptr, dummy_rtp_rtcp_impl_);
test_transport_ = new TestTransport(rtcp_receiver_);
-
- EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
+ rtcp_sender_ = new RTCPSender(false, system_clock_, receive_statistics_.get(),
+ nullptr, test_transport_);
}
void RtcpFormatRembTest::TearDown() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 111757b..f250e29 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -135,13 +135,12 @@
bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics,
- RtcpPacketTypeCounterObserver* packet_type_counter_observer)
+ RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+ Transport* outgoing_transport)
: audio_(audio),
clock_(clock),
method_(kRtcpOff),
- critical_section_transport_(
- CriticalSectionWrapper::CreateCriticalSection()),
- cbTransport_(nullptr),
+ transport_(outgoing_transport),
critical_section_rtcp_sender_(
CriticalSectionWrapper::CreateCriticalSection()),
@@ -173,6 +172,7 @@
packet_type_counter_observer_(packet_type_counter_observer) {
memset(last_send_report_, 0, sizeof(last_send_report_));
memset(last_rtcp_time_, 0, sizeof(last_rtcp_time_));
+ RTC_DCHECK(transport_ != nullptr);
builders_[kRtcpSr] = &RTCPSender::BuildSR;
builders_[kRtcpRr] = &RTCPSender::BuildRR;
@@ -196,12 +196,6 @@
RTCPSender::~RTCPSender() {
}
-int32_t RTCPSender::RegisterSendTransport(Transport* outgoingTransport) {
- CriticalSectionScoped lock(critical_section_transport_.get());
- cbTransport_ = outgoingTransport;
- return 0;
-}
-
RTCPMethod RTCPSender::Status() const {
CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
return method_;
@@ -1115,11 +1109,8 @@
}
int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
- CriticalSectionScoped lock(critical_section_transport_.get());
- if (cbTransport_) {
- if (cbTransport_->SendRtcp(dataBuffer, length))
- return 0;
- }
+ if (transport_->SendRtcp(dataBuffer, length))
+ return 0;
return -1;
}
@@ -1210,10 +1201,6 @@
}
bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
- CriticalSectionScoped lock(critical_section_transport_.get());
- if (!cbTransport_)
- return false;
-
class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
public:
Sender(Transport* transport)
@@ -1226,7 +1213,7 @@
Transport* const transport_;
bool send_failure_;
- } sender(cbTransport_);
+ } sender(transport_);
uint8_t buffer[IP_PACKET_SIZE];
return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 2051102..2f27989 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -74,11 +74,10 @@
RTCPSender(bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics,
- RtcpPacketTypeCounterObserver* packet_type_counter_observer);
+ RtcpPacketTypeCounterObserver* packet_type_counter_observer,
+ Transport* outgoing_transport);
virtual ~RTCPSender();
- int32_t RegisterSendTransport(Transport* outgoingTransport);
-
RTCPMethod Status() const;
void SetRTCPStatus(RTCPMethod method);
@@ -228,8 +227,7 @@
Clock* const clock_;
RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_);
- rtc::scoped_ptr<CriticalSectionWrapper> critical_section_transport_;
- Transport* cbTransport_ GUARDED_BY(critical_section_transport_);
+ Transport* const transport_;
rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 33c3f84..19dbc4d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -230,11 +230,10 @@
configuration.outgoing_transport = &test_transport_;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
- rtcp_sender_.reset(
- new RTCPSender(false, &clock_, receive_statistics_.get(), nullptr));
+ rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+ nullptr, &test_transport_));
rtcp_sender_->SetSSRC(kSenderSsrc);
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
- EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
}
void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
@@ -667,10 +666,9 @@
TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
RtcpPacketTypeCounterObserverImpl observer;
- rtcp_sender_.reset(
- new RTCPSender(false, &clock_, receive_statistics_.get(), &observer));
+ rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
+ &observer, &test_transport_));
rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
- EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(&test_transport_));
rtcp_sender_->SetRTCPStatus(kRtcpNonCompound);
EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
EXPECT_EQ(1, parser()->pli()->num_packets());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index ae4caf7..2e65452 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -72,7 +72,8 @@
rtcp_sender_(configuration.audio,
configuration.clock,
configuration.receive_statistics,
- configuration.rtcp_packet_type_counter_observer),
+ configuration.rtcp_packet_type_counter_observer,
+ configuration.outgoing_transport),
rtcp_receiver_(configuration.clock,
configuration.receiver_only,
configuration.rtcp_packet_type_counter_observer,
@@ -99,9 +100,6 @@
rtt_ms_(0) {
send_video_codec_.codecType = kVideoCodecUnknown;
- // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
- rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
-
// Make sure that RTCP objects are aware of our SSRC.
uint32_t SSRC = rtp_sender_.SSRC();
rtcp_sender_.SetSSRC(SSRC);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 1540e55..f75113a 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -9,6 +9,7 @@
*/
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
+#include "webrtc/test/null_transport.h"
#include <algorithm>
#include <vector>
@@ -90,6 +91,7 @@
RtpRtcp::Configuration configuration;
configuration.audio = true;
configuration.clock = &fake_clock_;
+ configuration.outgoing_transport = &null_transport_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
@@ -105,6 +107,7 @@
uint16_t test_sequence_number_;
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
+ test::NullTransport null_transport_;
};
TEST_F(RtpRtcpAPITest, Basic) {
