Use separate rtp module lists for send and receive in PacketRouter.
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.
Also moves sending transport feedback to the pacer thread.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1628683002
Cr-Commit-Position: refs/heads/master@{#11443}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index ebe2713..b731126 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -117,8 +117,8 @@
}
}
// Configure bandwidth estimation.
- channel_proxy_->SetCongestionControlObjects(
- nullptr, nullptr, congestion_controller->packet_router());
+ channel_proxy_->RegisterReceiverCongestionControlObjects(
+ congestion_controller->packet_router());
if (config.combined_audio_video_bwe) {
if (UseSendSideBwe(config)) {
remote_bitrate_estimator_ =
@@ -134,7 +134,7 @@
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
- channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+ channel_proxy_->ResetCongestionControlObjects();
if (remote_bitrate_estimator_) {
remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index b241bed..fd88fc8 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -94,13 +94,12 @@
EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
kTransportSequenceNumberId))
.Times(1);
- EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
- nullptr, nullptr, &packet_router_))
+ EXPECT_CALL(*channel_proxy_,
+ RegisterReceiverCongestionControlObjects(&packet_router_))
.Times(1);
EXPECT_CALL(congestion_controller_, packet_router())
.WillOnce(Return(&packet_router_));
- EXPECT_CALL(*channel_proxy_,
- SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
.Times(1);
return channel_proxy_;
}));
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 35a6552..62121c6 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -68,7 +68,7 @@
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
- channel_proxy_->SetCongestionControlObjects(
+ channel_proxy_->RegisterSenderCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(),
congestion_controller->packet_router());
@@ -92,7 +92,7 @@
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
- channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+ channel_proxy_->ResetCongestionControlObjects();
}
void AudioSendStream::Start() {
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 466c157..a01ef02 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -83,13 +83,12 @@
kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
- SetCongestionControlObjects(
+ RegisterSenderCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
congestion_controller_.packet_router()))
.Times(1);
- EXPECT_CALL(*channel_proxy_,
- SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
.Times(1);
return channel_proxy_;
}));
diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc
index 0267d79..e548a96 100644
--- a/webrtc/call/congestion_controller.cc
+++ b/webrtc/call/congestion_controller.cc
@@ -171,9 +171,9 @@
call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
pacer_thread_->RegisterModule(pacer_.get());
+ pacer_thread_->RegisterModule(remote_estimator_proxy_.get());
pacer_thread_->Start();
- process_thread->RegisterModule(remote_estimator_proxy_.get());
process_thread->RegisterModule(remote_bitrate_estimator_.get());
process_thread->RegisterModule(bitrate_controller_.get());
}
@@ -181,9 +181,9 @@
CongestionController::~CongestionController() {
pacer_thread_->Stop();
pacer_thread_->DeRegisterModule(pacer_.get());
+ pacer_thread_->DeRegisterModule(remote_estimator_proxy_.get());
process_thread_->DeRegisterModule(bitrate_controller_.get());
process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
- process_thread_->DeRegisterModule(remote_estimator_proxy_.get());
call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
if (transport_feedback_adapter_.get())
call_stats_->DeregisterStatsObserver(transport_feedback_adapter_.get());
diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index 5fd3508..e98ab0c 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -18,33 +18,64 @@
namespace webrtc {
+namespace {
+void AddModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
+ RTC_DCHECK(std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module) ==
+ rtp_modules->end());
+ rtp_modules->push_back(rtp_module);
+}
+
+void RemoveModule(RtpRtcp* rtp_module, std::list<RtpRtcp*>* rtp_modules) {
+ RTC_DCHECK(std::find(rtp_modules->begin(), rtp_modules->end(), rtp_module) !=
+ rtp_modules->end());
+ rtp_modules->remove(rtp_module);
+}
+
+bool SendFeedback(rtcp::TransportFeedback* packet,
+ std::list<RtpRtcp*>* rtp_modules) {
+ for (auto* rtp_module : *rtp_modules) {
+ packet->WithPacketSenderSsrc(rtp_module->SSRC());
+ if (rtp_module->SendFeedbackPacket(*packet))
+ return true;
+ }
+ return false;
+}
+}
+
PacketRouter::PacketRouter() : transport_seq_(0) {
+ pacer_thread_checker_.DetachFromThread();
}
PacketRouter::~PacketRouter() {
- RTC_DCHECK(rtp_modules_.empty());
+ RTC_DCHECK(send_rtp_modules_.empty());
+ RTC_DCHECK(recv_rtp_modules_.empty());
}
-void PacketRouter::AddRtpModule(RtpRtcp* rtp_module) {
- rtc::CritScope cs(&modules_lock_);
- RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) ==
- rtp_modules_.end());
- rtp_modules_.push_back(rtp_module);
+void PacketRouter::AddRtpModule(RtpRtcp* rtp_module, bool sender) {
+ rtc::CritScope cs(&modules_crit_);
+ if (sender) {
+ AddModule(rtp_module, &send_rtp_modules_);
+ } else {
+ AddModule(rtp_module, &recv_rtp_modules_);
+ }
}
-void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
- rtc::CritScope cs(&modules_lock_);
- auto it = std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module);
- RTC_DCHECK(it != rtp_modules_.end());
- rtp_modules_.erase(it);
+void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module, bool sender) {
+ rtc::CritScope cs(&modules_crit_);
+ if (sender) {
+ RemoveModule(rtp_module, &send_rtp_modules_);
+ } else {
+ RemoveModule(rtp_module, &recv_rtp_modules_);
+ }
}
bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission) {
- rtc::CritScope cs(&modules_lock_);
- for (auto* rtp_module : rtp_modules_) {
+ RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
+ rtc::CritScope cs(&modules_crit_);
+ for (auto* rtp_module : send_rtp_modules_) {
if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
return rtp_module->TimeToSendPacket(ssrc, sequence_number,
capture_timestamp, retransmission);
@@ -54,9 +85,10 @@
}
size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send) {
+ RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
size_t total_bytes_sent = 0;
- rtc::CritScope cs(&modules_lock_);
- for (RtpRtcp* module : rtp_modules_) {
+ rtc::CritScope cs(&modules_crit_);
+ for (RtpRtcp* module : send_rtp_modules_) {
if (module->SendingMedia()) {
size_t bytes_sent =
module->TimeToSendPadding(bytes_to_send - total_bytes_sent);
@@ -91,12 +123,12 @@
}
bool PacketRouter::SendFeedback(rtcp::TransportFeedback* packet) {
- rtc::CritScope cs(&modules_lock_);
- for (auto* rtp_module : rtp_modules_) {
- packet->WithPacketSenderSsrc(rtp_module->SSRC());
- if (rtp_module->SendFeedbackPacket(*packet))
- return true;
- }
+ RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
+ rtc::CritScope cs(&modules_crit_);
+ if (::webrtc::SendFeedback(packet, &recv_rtp_modules_))
+ return true;
+ if (::webrtc::SendFeedback(packet, &send_rtp_modules_))
+ return true;
return false;
}
diff --git a/webrtc/modules/pacing/packet_router.h b/webrtc/modules/pacing/packet_router.h
index edef1aa..dd58faf 100644
--- a/webrtc/modules/pacing/packet_router.h
+++ b/webrtc/modules/pacing/packet_router.h
@@ -17,6 +17,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -36,8 +37,8 @@
PacketRouter();
virtual ~PacketRouter();
- void AddRtpModule(RtpRtcp* rtp_module);
- void RemoveRtpModule(RtpRtcp* rtp_module);
+ void AddRtpModule(RtpRtcp* rtp_module, bool sender);
+ void RemoveRtpModule(RtpRtcp* rtp_module, bool sender);
// Implements PacedSender::Callback.
bool TimeToSendPacket(uint32_t ssrc,
@@ -54,9 +55,10 @@
virtual bool SendFeedback(rtcp::TransportFeedback* packet);
private:
- rtc::CriticalSection modules_lock_;
- // Map from ssrc to sending rtp module.
- std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
+ rtc::ThreadChecker pacer_thread_checker_;
+ rtc::CriticalSection modules_crit_;
+ std::list<RtpRtcp*> send_rtp_modules_ GUARDED_BY(modules_crit_);
+ std::list<RtpRtcp*> recv_rtp_modules_ GUARDED_BY(modules_crit_);
volatile int transport_seq_;
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index 31acf44..e5e0589 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -16,6 +16,7 @@
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/base/scoped_ptr.h"
using ::testing::_;
@@ -35,8 +36,8 @@
TEST_F(PacketRouterTest, TimeToSendPacket) {
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
- packet_router_->AddRtpModule(&rtp_1);
- packet_router_->AddRtpModule(&rtp_2);
+ packet_router_->AddRtpModule(&rtp_1, true);
+ packet_router_->AddRtpModule(&rtp_2, true);
const uint16_t kSsrc1 = 1234;
uint16_t sequence_number = 17;
@@ -88,7 +89,7 @@
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
timestamp, retransmission));
- packet_router_->RemoveRtpModule(&rtp_1);
+ packet_router_->RemoveRtpModule(&rtp_1, true);
// rtp_1 has been removed, try sending a packet on that ssrc and make sure
// it is dropped as expected by not expecting any calls to rtp_1.
@@ -98,7 +99,7 @@
EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
timestamp, retransmission));
- packet_router_->RemoveRtpModule(&rtp_2);
+ packet_router_->RemoveRtpModule(&rtp_2, true);
}
TEST_F(PacketRouterTest, TimeToSendPadding) {
@@ -109,8 +110,8 @@
EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
MockRtpRtcp rtp_2;
EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
- packet_router_->AddRtpModule(&rtp_1);
- packet_router_->AddRtpModule(&rtp_2);
+ packet_router_->AddRtpModule(&rtp_1, true);
+ packet_router_->AddRtpModule(&rtp_2, true);
// Default configuration, sending padding on all modules sending media,
// ordered by SSRC.
@@ -146,7 +147,7 @@
EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
- packet_router_->RemoveRtpModule(&rtp_1);
+ packet_router_->RemoveRtpModule(&rtp_1, true);
// rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked
// to send by not expecting any calls. Instead verify rtp_2 is called.
@@ -154,7 +155,7 @@
EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
- packet_router_->RemoveRtpModule(&rtp_2);
+ packet_router_->RemoveRtpModule(&rtp_2, true);
}
TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
@@ -169,4 +170,19 @@
EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
}
}
+
+TEST_F(PacketRouterTest, SendFeedback) {
+ MockRtpRtcp rtp_1;
+ MockRtpRtcp rtp_2;
+ packet_router_->AddRtpModule(&rtp_1, false);
+ packet_router_->AddRtpModule(&rtp_2, true);
+
+ rtcp::TransportFeedback feedback;
+ EXPECT_CALL(rtp_1, SendFeedbackPacket(_)).Times(1);
+ packet_router_->SendFeedback(&feedback);
+ packet_router_->RemoveRtpModule(&rtp_1, false);
+ EXPECT_CALL(rtp_2, SendFeedbackPacket(_)).Times(1);
+ packet_router_->SendFeedback(&feedback);
+ packet_router_->RemoveRtpModule(&rtp_2, true);
+}
} // namespace webrtc
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index a0ec4b2..f5c8733 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -29,10 +29,13 @@
MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));
- MOCK_METHOD3(SetCongestionControlObjects,
+ MOCK_METHOD3(RegisterSenderCongestionControlObjects,
void(RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
- PacketRouter* seq_num_allocator));
+ PacketRouter* packet_router));
+ MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
+ void(PacketRouter* packet_router));
+ MOCK_METHOD0(ResetCongestionControlObjects, void());
MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
diff --git a/webrtc/video/vie_channel.cc b/webrtc/video/vie_channel.cc
index b1b0f0c..7c13a7c 100644
--- a/webrtc/video/vie_channel.cc
+++ b/webrtc/video/vie_channel.cc
@@ -150,7 +150,7 @@
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
}
- packet_router_->AddRtpModule(rtp_rtcp_modules_[0]);
+ packet_router_->AddRtpModule(rtp_rtcp_modules_[0], sender_);
if (sender_) {
std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]);
send_payload_router_->SetSendingRtpModules(send_rtp_modules);
@@ -179,7 +179,7 @@
module_process_thread_->DeRegisterModule(&vie_sync_);
send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
- packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+ packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i], sender_);
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
module_process_thread_->DeRegisterModule(rtp_rtcp);
delete rtp_rtcp;
@@ -400,12 +400,12 @@
// Deregister previously registered modules.
for (size_t i = num_active_modules; i < num_prev_active_modules; ++i) {
module_process_thread_->DeRegisterModule(rtp_rtcp_modules_[i]);
- packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+ packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i], sender_);
}
// Register new active modules.
for (size_t i = num_prev_active_modules; i < num_active_modules; ++i) {
module_process_thread_->RegisterModule(rtp_rtcp_modules_[i]);
- packet_router_->AddRtpModule(rtp_rtcp_modules_[i]);
+ packet_router_->AddRtpModule(rtp_rtcp_modules_[i], sender_);
}
return 0;
}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d5f10bf..148ef81 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -124,6 +124,12 @@
rtp_packet_sender_ = rtp_packet_sender;
}
+ bool HasPacketSender() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_);
+ return rtp_packet_sender_ != nullptr;
+ }
+
// Implements RtpPacketSender.
void InsertPacket(Priority priority,
uint32_t ssrc,
@@ -813,12 +819,9 @@
network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
associate_send_channel_(ChannelOwner(nullptr)),
pacing_enabled_(config.Get<VoicePacing>().enabled),
- feedback_observer_proxy_(pacing_enabled_ ? new TransportFeedbackProxy()
- : nullptr),
- seq_num_allocator_proxy_(
- pacing_enabled_ ? new TransportSequenceNumberProxy() : nullptr),
- rtp_packet_sender_proxy_(pacing_enabled_ ? new RtpPacketSenderProxy()
- : nullptr) {
+ feedback_observer_proxy_(new TransportFeedbackProxy()),
+ seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
+ rtp_packet_sender_proxy_(new RtpPacketSenderProxy()) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Channel() - ctor");
AudioCodingModule::Config acm_config;
@@ -843,10 +846,12 @@
configuration.audio_messages = this;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.bandwidth_callback = rtcp_observer_.get();
- configuration.paced_sender = rtp_packet_sender_proxy_.get();
- configuration.transport_sequence_number_allocator =
- seq_num_allocator_proxy_.get();
- configuration.transport_feedback_callback = feedback_observer_proxy_.get();
+ if (pacing_enabled_) {
+ configuration.paced_sender = rtp_packet_sender_proxy_.get();
+ configuration.transport_sequence_number_allocator =
+ seq_num_allocator_proxy_.get();
+ configuration.transport_feedback_callback = feedback_observer_proxy_.get();
+ }
configuration.event_log = event_log;
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
@@ -2582,32 +2587,40 @@
RTC_DCHECK(ret);
}
-void Channel::SetCongestionControlObjects(
+void Channel::RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
- RTC_DCHECK(packet_router != nullptr || packet_router_ != nullptr);
- if (transport_feedback_observer) {
- RTC_DCHECK(feedback_observer_proxy_.get());
- feedback_observer_proxy_->SetTransportFeedbackObserver(
- transport_feedback_observer);
- }
- if (rtp_packet_sender) {
- RTC_DCHECK(rtp_packet_sender_proxy_.get());
- rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
- }
- if (seq_num_allocator_proxy_.get()) {
- seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
- }
- _rtpRtcpModule->SetStorePacketsStatus(rtp_packet_sender != nullptr, 600);
- if (packet_router != nullptr) {
- packet_router->AddRtpModule(_rtpRtcpModule.get());
- } else {
- packet_router_->RemoveRtpModule(_rtpRtcpModule.get());
- }
+ RTC_DCHECK(rtp_packet_sender);
+ RTC_DCHECK(transport_feedback_observer);
+ RTC_DCHECK(packet_router && !packet_router_);
+ feedback_observer_proxy_->SetTransportFeedbackObserver(
+ transport_feedback_observer);
+ seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
+ rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
+ _rtpRtcpModule->SetStorePacketsStatus(true, 600);
+ packet_router->AddRtpModule(_rtpRtcpModule.get(), true);
packet_router_ = packet_router;
}
+void Channel::RegisterReceiverCongestionControlObjects(
+ PacketRouter* packet_router) {
+ RTC_DCHECK(packet_router && !packet_router_);
+ packet_router->AddRtpModule(_rtpRtcpModule.get(), false);
+ packet_router_ = packet_router;
+}
+
+void Channel::ResetCongestionControlObjects() {
+ RTC_DCHECK(packet_router_);
+ _rtpRtcpModule->SetStorePacketsStatus(false, 600);
+ feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr);
+ seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr);
+ const bool sender = rtp_packet_sender_proxy_->HasPacketSender();
+ packet_router_->RemoveRtpModule(_rtpRtcpModule.get(), sender);
+ packet_router_ = nullptr;
+ rtp_packet_sender_proxy_->SetPacketSender(nullptr);
+}
+
void Channel::SetRTCPStatus(bool enable) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetRTCPStatus()");
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 14bfc2b..889214b 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -339,10 +339,12 @@
void EnableSendTransportSequenceNumber(int id);
void EnableReceiveTransportSequenceNumber(int id);
- void SetCongestionControlObjects(
- RtpPacketSender* rtp_packet_sender,
- TransportFeedbackObserver* transport_feedback_observer,
- PacketRouter* packet_router);
+ void RegisterSenderCongestionControlObjects(
+ RtpPacketSender* rtp_packet_sender,
+ TransportFeedbackObserver* transport_feedback_observer,
+ PacketRouter* packet_router);
+ void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
+ void ResetCongestionControlObjects();
void SetRTCPStatus(bool enable);
int GetRTCPStatus(bool& enabled);
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 15a2151..1e2281a 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -79,15 +79,26 @@
channel()->EnableReceiveTransportSequenceNumber(id);
}
-void ChannelProxy::SetCongestionControlObjects(
+void ChannelProxy::RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- channel()->SetCongestionControlObjects(
+ channel()->RegisterSenderCongestionControlObjects(
rtp_packet_sender, transport_feedback_observer, packet_router);
}
+void ChannelProxy::RegisterReceiverCongestionControlObjects(
+ PacketRouter* packet_router) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel()->RegisterReceiverCongestionControlObjects(packet_router);
+}
+
+void ChannelProxy::ResetCongestionControlObjects() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel()->ResetCongestionControlObjects();
+}
+
CallStatistics ChannelProxy::GetRTCPStatistics() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
CallStatistics stats = {0};
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 9a93160..9d6839c 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -51,10 +51,13 @@
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
virtual void EnableSendTransportSequenceNumber(int id);
virtual void EnableReceiveTransportSequenceNumber(int id);
- virtual void SetCongestionControlObjects(
+ virtual void RegisterSenderCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router);
+ virtual void RegisterReceiverCongestionControlObjects(
+ PacketRouter* packet_router);
+ virtual void ResetCongestionControlObjects();
virtual CallStatistics GetRTCPStatistics() const;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
