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Set RtcpSender transport at construction.

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BUG=

Review URL: https://codereview.webrtc.org/1365043002

Cr-Commit-Position: refs/heads/master@{#10106}
This commit is contained in:
sprang 2015-09-29 04:45:43 -07:00 committed by Commit bot
parent 38502a7be4
commit 86fd9ed6f9
8 changed files with 33 additions and 49 deletions
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1
webrtc/modules/modules.gyp
View File

@ -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': [

9
webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
View File

@ -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 @@ class RtcpFormatRembTest : public ::testing::Test {
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 @@ void RtcpFormatRembTest::SetUp() {
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() {

27
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -135,13 +135,12 @@ RTCPSender::RTCPSender(
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 @@ RTCPSender::RTCPSender(
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(
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 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
}
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::AllVolatileFlagsConsumed() const {
}
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 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
Transport* const transport_;
bool send_failure_;
} sender(cbTransport_);
} sender(transport_);
uint8_t buffer[IP_PACKET_SIZE];
return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&

8
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -74,11 +74,10 @@ public:
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 @@ private:
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_);

10
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -230,11 +230,10 @@ class RtcpSenderTest : public ::testing::Test {
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, TestSendTimeOfXrRrtr) {
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());

6
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -72,7 +72,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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);

3
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -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 @@ class RtpRtcpAPITest : public ::testing::Test {
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 @@ class RtpRtcpAPITest : public ::testing::Test {
uint16_t test_sequence_number_;
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
};
TEST_F(RtpRtcpAPITest, Basic) {

18
webrtc/video/video_send_stream_tests.cc
View File

@ -329,8 +329,8 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
FakeReceiveStatistics lossy_receive_stats(
kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
&lossy_receive_stats, nullptr,
&transport_adapter_);
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
@ -411,8 +411,7 @@ void VideoSendStreamTest::TestNackRetransmission(
nacked_sequence_number_ = nack_sequence_number;
NullReceiveStatistics null_stats;
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &null_stats,
nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
nullptr, &transport_adapter_);
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
@ -598,8 +597,8 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
FakeReceiveStatistics lossy_receive_stats(
kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
&lossy_receive_stats, nullptr,
&transport_adapter_);
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
@ -828,8 +827,8 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
EXCLUSIVE_LOCKS_REQUIRED(crit_) {
FakeReceiveStatistics receive_stats(
kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr,
&transport_adapter_);
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
@ -888,8 +887,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
nullptr);
EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
nullptr, &transport_adapter_);
rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);