Add NACK rate throttling for audio channels.
Not really used for audio today (already in place for video), but should
still function anyway.
BUG=
R=henrika@webrtc.org, minyue@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2181383002 .
Cr-Commit-Position: refs/heads/master@{#13571}
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index f8b555b..57d18b7 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -20,6 +20,7 @@
#include "webrtc/base/buffer.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/random.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/call/rtc_event_log_parser.h"
@@ -111,19 +112,20 @@
Random* prng) {
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
Clock* clock = Clock::GetRealTimeClock();
+ RateLimiter retranmission_rate_limiter(clock, 1000);
- RTPSender rtp_sender(false, // bool audio
- clock, // Clock* clock
- nullptr, // Transport*
- nullptr, // PacedSender*
- nullptr, // PacketRouter*
- nullptr, // SendTimeObserver*
- nullptr, // BitrateStatisticsObserver*
- nullptr, // FrameCountObserver*
- nullptr, // SendSideDelayObserver*
- nullptr, // RtcEventLog*
- nullptr, // SendPacketObserver*
- nullptr); // NackRateLimiter*
+ RTPSender rtp_sender(false, // bool audio
+ clock, // Clock* clock
+ nullptr, // Transport*
+ nullptr, // PacedSender*
+ nullptr, // PacketRouter*
+ nullptr, // SendTimeObserver*
+ nullptr, // BitrateStatisticsObserver*
+ nullptr, // FrameCountObserver*
+ nullptr, // SendSideDelayObserver*
+ nullptr, // RtcEventLog*
+ nullptr, // SendPacketObserver*
+ &retranmission_rate_limiter);
std::vector<uint32_t> csrcs;
for (unsigned i = 0; i < csrcs_count; i++) {
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index b12c08e..a0e0cec 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -175,7 +175,7 @@
receiver_(),
payload_data_length(sizeof(payload_data)),
fake_clock(123456),
- retranmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
+ retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
~RtpRtcpRtxNackTest() {}
void SetUp() override {
@@ -185,7 +185,7 @@
receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
configuration.receive_statistics = receive_statistics_.get();
configuration.outgoing_transport = &transport_;
- configuration.retransmission_rate_limiter = &retranmission_rate_limiter_;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
@@ -292,7 +292,7 @@
uint8_t payload_data[65000];
size_t payload_data_length;
SimulatedClock fake_clock;
- RateLimiter retranmission_rate_limiter_;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpRtxNackTest, LongNackList) {
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 bbfb52c..0e0d7f5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -12,6 +12,7 @@
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
@@ -69,7 +70,8 @@
remote_bitrate_observer_(),
remote_bitrate_estimator_(
new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
- system_clock_)) {}
+ system_clock_)),
+ retransmission_rate_limiter_(Clock::GetRealTimeClock(), 1000) {}
void SetUp() override;
void TearDown() override;
@@ -83,6 +85,7 @@
test::NullTransport null_transport_;
MockRemoteBitrateObserver remote_bitrate_observer_;
std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+ RateLimiter retransmission_rate_limiter_;
};
void RtcpFormatRembTest::SetUp() {
@@ -91,6 +94,7 @@
configuration.clock = system_clock_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
configuration.outgoing_transport = &null_transport_;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
rtcp_receiver_ = new RTCPReceiver(system_clock_, false, nullptr, nullptr,
nullptr, nullptr, dummy_rtp_rtcp_impl_);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 924d009..cdb71be 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -13,6 +13,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
@@ -80,24 +81,24 @@
remote_bitrate_observer_(),
remote_bitrate_estimator_(
new RemoteBitrateEstimatorSingleStream(&remote_bitrate_observer_,
- &system_clock_)) {
- test_transport_ = new TestTransport();
+ &system_clock_)),
+ retransmission_rate_limiter_(&system_clock_, 1000) {
+ test_transport_.reset(new TestTransport());
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &system_clock_;
- configuration.outgoing_transport = test_transport_;
+ configuration.outgoing_transport = test_transport_.get();
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
- rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
- rtcp_receiver_ = new RTCPReceiver(&system_clock_, false, nullptr, nullptr,
- nullptr, nullptr, rtp_rtcp_impl_);
- test_transport_->SetRTCPReceiver(rtcp_receiver_);
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
+ rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
+ rtcp_receiver_.reset(new RTCPReceiver(&system_clock_, false, nullptr,
+ nullptr, nullptr, nullptr,
+ rtp_rtcp_impl_.get()));
+ test_transport_->SetRTCPReceiver(rtcp_receiver_.get());
}
- ~RtcpReceiverTest() {
- delete rtcp_receiver_;
- delete rtp_rtcp_impl_;
- delete test_transport_;
- }
+
+ ~RtcpReceiverTest() {}
// Injects an RTCP packet into the receiver.
// Returns 0 for OK, non-0 for failure.
@@ -142,12 +143,13 @@
OverUseDetectorOptions over_use_detector_options_;
SimulatedClock system_clock_;
- ModuleRtpRtcpImpl* rtp_rtcp_impl_;
- RTCPReceiver* rtcp_receiver_;
- TestTransport* test_transport_;
+ std::unique_ptr<TestTransport> test_transport_;
+ std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
+ std::unique_ptr<RTCPReceiver> rtcp_receiver_;
RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
MockRemoteBitrateObserver remote_bitrate_observer_;
std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+ RateLimiter retransmission_rate_limiter_;
};
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index ec83308..270088e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -13,6 +13,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -229,11 +230,13 @@
protected:
RtcpSenderTest()
: clock_(1335900000),
- receive_statistics_(ReceiveStatistics::Create(&clock_)) {
+ receive_statistics_(ReceiveStatistics::Create(&clock_)),
+ retransmission_rate_limiter_(&clock_, 1000) {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &clock_;
configuration.outgoing_transport = &test_transport_;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
@@ -265,6 +268,7 @@
std::unique_ptr<ReceiveStatistics> receive_statistics_;
std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
std::unique_ptr<RTCPSender> rtcp_sender_;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtcpSenderTest, SetRtcpStatus) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 0afc8d2..3f2371e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -127,6 +127,8 @@
csrcs_(),
rtx_(kRtxOff),
retransmission_rate_limiter_(retransmission_rate_limiter) {
+ RTC_DCHECK(retransmission_rate_limiter_ != nullptr);
+
// We need to seed the random generator for BuildPaddingPacket() below.
// TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac
// early on in the process.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 302f6dd..fed767b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1013,7 +1013,7 @@
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_,
nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
- nullptr, nullptr, &send_packet_observer_, nullptr));
+ nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
@@ -1034,7 +1034,8 @@
MockTransport transport;
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
- nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr, nullptr));
+ nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
+ &retransmission_rate_limiter_));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@@ -1178,9 +1179,10 @@
FrameCounts frame_counts_;
} callback;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
- nullptr, &callback, nullptr, nullptr, nullptr, nullptr));
+ rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
+ &mock_paced_sender_, nullptr, nullptr,
+ nullptr, &callback, nullptr, nullptr, nullptr,
+ &retransmission_rate_limiter_));
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
@@ -1239,9 +1241,9 @@
uint32_t total_bitrate_;
uint32_t retransmit_bitrate_;
} callback;
- rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
- nullptr, nullptr, &callback, nullptr, nullptr,
- nullptr, nullptr, nullptr));
+ rtp_sender_.reset(new RTPSender(
+ false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, &callback,
+ nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_));
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
// number of packets selected so that we fill (but don't overflow) the one
@@ -1296,9 +1298,9 @@
void SetUp() override {
payload_ = kAudioPayload;
- rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
- nullptr, nullptr, nullptr, nullptr, nullptr,
- nullptr, nullptr, nullptr));
+ rtp_sender_.reset(new RTPSender(
+ true, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
+ nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index d4c1111..ab666ce 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -14,6 +14,7 @@
#include <memory>
#include <vector>
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/test/null_transport.h"
namespace webrtc {
@@ -80,7 +81,8 @@
class RtpRtcpAPITest : public ::testing::Test {
protected:
- RtpRtcpAPITest() : fake_clock_(123456) {
+ RtpRtcpAPITest()
+ : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {
test_csrcs_.push_back(1234);
test_csrcs_.push_back(2345);
test_ssrc_ = 3456;
@@ -94,6 +96,7 @@
configuration.audio = true;
configuration.clock = &fake_clock_;
configuration.outgoing_transport = &null_transport_;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
@@ -110,6 +113,7 @@
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAPITest, Basic) {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 0fb9845..f9e5001 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -15,6 +15,7 @@
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -77,7 +78,8 @@
class RtpRtcpAudioTest : public ::testing::Test {
protected:
- RtpRtcpAudioTest() : fake_clock(123456) {
+ RtpRtcpAudioTest()
+ : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
test_CSRC[0] = 1234;
test_CSRC[2] = 2345;
test_ssrc = 3456;
@@ -106,6 +108,7 @@
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
@@ -152,6 +155,7 @@
uint16_t test_sequence_number;
uint32_t test_CSRC[webrtc::kRtpCsrcSize];
SimulatedClock fake_clock;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAudioTest, Basic) {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index c1359df..f33507e 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -14,6 +14,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -66,7 +67,8 @@
class RtpRtcpRtcpTest : public ::testing::Test {
protected:
- RtpRtcpRtcpTest() : fake_clock(123456) {
+ RtpRtcpRtcpTest()
+ : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
test_csrcs.push_back(1234);
test_csrcs.push_back(2345);
test_ssrc = 3456;
@@ -91,6 +93,7 @@
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
configuration.intra_frame_callback = myRTCPFeedback1;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_payload_registry1_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
@@ -197,6 +200,7 @@
uint16_t test_sequence_number;
std::vector<uint32_t> test_csrcs;
SimulatedClock fake_clock;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index d84ff37..e784386 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -15,6 +15,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ -38,7 +39,8 @@
test_ssrc_(3456),
test_timestamp_(4567),
test_sequence_number_(2345),
- fake_clock(123456) {}
+ fake_clock(123456),
+ retransmission_rate_limiter_(&fake_clock, 1000) {}
~RtpRtcpVideoTest() {}
virtual void SetUp() {
@@ -49,6 +51,7 @@
configuration.audio = false;
configuration.clock = &fake_clock;
configuration.outgoing_transport = transport_;
+ configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
@@ -139,6 +142,7 @@
uint8_t video_frame_[65000];
size_t payload_data_length_;
SimulatedClock fake_clock;
+ RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpVideoTest, BasicVideo) {
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 034d063..b2a29d2 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -19,12 +19,15 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
@@ -487,6 +490,77 @@
RunBaseTest(&test);
}
+TEST_F(EndToEndTest, ReceivesNackAndRetransmitsAudio) {
+ class NackObserver : public test::EndToEndTest {
+ public:
+ NackObserver()
+ : EndToEndTest(kLongTimeoutMs),
+ local_ssrc_(0),
+ remote_ssrc_(0),
+ receive_transport_(nullptr) {}
+
+ private:
+ size_t GetNumVideoStreams() const override { return 0; }
+ size_t GetNumAudioStreams() const override { return 1; }
+
+ test::PacketTransport* CreateReceiveTransport() override {
+ test::PacketTransport* receive_transport = new test::PacketTransport(
+ nullptr, this, test::PacketTransport::kReceiver,
+ FakeNetworkPipe::Config());
+ receive_transport_ = receive_transport;
+ return receive_transport;
+ }
+
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+
+ if (!sequence_number_to_retransmit_) {
+ sequence_number_to_retransmit_ =
+ rtc::Optional<uint16_t>(header.sequenceNumber);
+
+ // Don't ask for retransmission straight away, may be deduped in pacer.
+ } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
+ observation_complete_.Set();
+ } else {
+ // Send a NACK as often as necessary until retransmission is received.
+ rtcp::Nack nack;
+ nack.From(local_ssrc_);
+ nack.To(remote_ssrc_);
+ uint16_t nack_list[] = {*sequence_number_to_retransmit_};
+ nack.WithList(nack_list, 1);
+ rtc::Buffer buffer = nack.Build();
+
+ EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
+ }
+
+ return SEND_PACKET;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+ (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+ local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
+ remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
+ }
+
+ void PerformTest() override {
+ EXPECT_TRUE(Wait())
+ << "Timed out waiting for packets to be NACKed, retransmitted and "
+ "rendered.";
+ }
+
+ uint32_t local_ssrc_;
+ uint32_t remote_ssrc_;
+ Transport* receive_transport_;
+ rtc::Optional<uint16_t> sequence_number_to_retransmit_;
+ } test;
+
+ RunBaseTest(&test);
+}
+
TEST_F(EndToEndTest, CanReceiveFec) {
class FecRenderObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
@@ -1807,7 +1881,8 @@
poller_thread_(&BitrateStatsPollingThread,
this,
"BitrateStatsPollingThread"),
- state_(kWaitForFirstRampUp) {}
+ state_(kWaitForFirstRampUp),
+ retransmission_rate_limiter_(clock_, 1000) {}
~BweObserver() {}
@@ -1847,6 +1922,7 @@
config.receiver_only = true;
config.clock = clock_;
config.outgoing_transport = receive_transport_;
+ config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
@@ -1919,6 +1995,7 @@
rtc::Event event_;
rtc::PlatformThread poller_thread_;
TestState state_;
+ RateLimiter retransmission_rate_limiter_;
} test;
RunBaseTest(&test);
diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc
index 4caf55a..e7367d8 100644
--- a/webrtc/video/rtp_stream_receiver.cc
+++ b/webrtc/video/rtp_stream_receiver.cc
@@ -39,7 +39,8 @@
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpPacketSender* paced_sender,
- TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
+ TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+ RateLimiter* retransmission_rate_limiter) {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = true;
@@ -58,7 +59,7 @@
configuration.send_packet_observer = nullptr;
configuration.bandwidth_callback = nullptr;
configuration.transport_feedback_callback = nullptr;
- configuration.retransmission_rate_limiter = nullptr;
+ configuration.retransmission_rate_limiter = retransmission_rate_limiter;
std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
rtp_rtcp->SetSendingStatus(false);
@@ -80,7 +81,8 @@
VieRemb* remb,
const VideoReceiveStream::Config* config,
ReceiveStatisticsProxy* receive_stats_proxy,
- ProcessThread* process_thread)
+ ProcessThread* process_thread,
+ RateLimiter* retransmission_rate_limiter)
: clock_(Clock::GetRealTimeClock()),
config_(*config),
video_receiver_(video_receiver),
@@ -106,7 +108,8 @@
receive_stats_proxy,
remote_bitrate_estimator_,
paced_sender,
- packet_router)) {
+ packet_router,
+ retransmission_rate_limiter)) {
packet_router_->AddRtpModule(rtp_rtcp_.get());
rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
diff --git a/webrtc/video/rtp_stream_receiver.h b/webrtc/video/rtp_stream_receiver.h
index efe7181..2af15b0 100644
--- a/webrtc/video/rtp_stream_receiver.h
+++ b/webrtc/video/rtp_stream_receiver.h
@@ -62,7 +62,8 @@
VieRemb* remb,
const VideoReceiveStream::Config* config,
ReceiveStatisticsProxy* receive_stats_proxy,
- ProcessThread* process_thread);
+ ProcessThread* process_thread,
+ RateLimiter* retransmission_rate_limiter);
~RtpStreamReceiver();
bool SetReceiveCodec(const VideoCodec& video_codec);
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 609a789..bc51269 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -14,6 +14,7 @@
#include <set>
#include <string>
+#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
@@ -162,17 +163,19 @@
call_stats_(call_stats),
video_receiver_(clock_, nullptr, this, this, this),
stats_proxy_(&config_, clock_),
- rtp_stream_receiver_(&video_receiver_,
- congestion_controller_->GetRemoteBitrateEstimator(
- UseSendSideBwe(config_)),
- &transport_adapter_,
- call_stats_->rtcp_rtt_stats(),
- congestion_controller_->pacer(),
- congestion_controller_->packet_router(),
- remb,
- &config_,
- &stats_proxy_,
- process_thread_),
+ rtp_stream_receiver_(
+ &video_receiver_,
+ congestion_controller_->GetRemoteBitrateEstimator(
+ UseSendSideBwe(config_)),
+ &transport_adapter_,
+ call_stats_->rtcp_rtt_stats(),
+ congestion_controller_->pacer(),
+ congestion_controller_->packet_router(),
+ remb,
+ &config_,
+ &stats_proxy_,
+ process_thread_,
+ congestion_controller_->GetRetransmissionRateLimiter()),
vie_sync_(&video_receiver_) {
LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 3a8acbc..623cd34 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/common_video/include/frame_callback.h"
@@ -819,13 +820,13 @@
RembObserver()
: SendTest(kDefaultTimeoutMs),
clock_(Clock::GetRealTimeClock()),
+ stream_(nullptr),
test_state_(kBeforeSuspend),
rtp_count_(0),
last_sequence_number_(0),
suspended_frame_count_(0),
low_remb_bps_(0),
- high_remb_bps_(0) {
- }
+ high_remb_bps_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -1049,8 +1050,9 @@
public:
BitrateObserver()
: SendTest(kDefaultTimeoutMs),
- bitrate_capped_(false) {
- }
+ retranmission_rate_limiter_(Clock::GetRealTimeClock(), 1000),
+ stream_(nullptr),
+ bitrate_capped_(false) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -1092,6 +1094,7 @@
stream_ = send_stream;
RtpRtcp::Configuration config;
config.outgoing_transport = feedback_transport_.get();
+ config.retransmission_rate_limiter = &retranmission_rate_limiter_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
@@ -1114,6 +1117,7 @@
std::unique_ptr<RtpRtcp> rtp_rtcp_;
std::unique_ptr<internal::TransportAdapter> feedback_transport_;
+ RateLimiter retranmission_rate_limiter_;
VideoSendStream* stream_;
bool bitrate_capped_;
} test;
@@ -1504,6 +1508,7 @@
public:
EncoderStateObserver()
: SendTest(kDefaultTimeoutMs),
+ stream_(nullptr),
initialized_(false),
callback_registered_(false),
num_releases_(0),
@@ -1624,7 +1629,8 @@
: SendTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
init_encode_event_(false, false),
- num_initializations_(0) {}
+ num_initializations_(0),
+ stream_(nullptr) {}
private:
void ModifyVideoConfigs(
@@ -1689,7 +1695,8 @@
video_codec_type_(video_codec_type),
codec_name_(codec_name),
init_encode_event_(false, false),
- num_initializations_(0) {
+ num_initializations_(0),
+ stream_(nullptr) {
memset(&encoder_settings_, 0, sizeof(encoder_settings_));
}
@@ -1931,7 +1938,9 @@
: SendTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
init_encode_event_(false, false),
- num_initializations_(0) {}
+ num_initializations_(0),
+ call_(nullptr),
+ send_stream_(nullptr) {}
private:
int32_t InitEncode(const VideoCodec* codecSettings,
@@ -2045,7 +2054,8 @@
public:
ScreencastTargetBitrateTest()
: SendTest(kDefaultTimeoutMs),
- test::FakeEncoder(Clock::GetRealTimeClock()) {}
+ test::FakeEncoder(Clock::GetRealTimeClock()),
+ send_stream_(nullptr) {}
private:
int32_t Encode(const VideoFrame& input_image,
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index dcedd75..f5be7ef 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -17,6 +17,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/rate_limiter.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call/rtc_event_log.h"
@@ -47,6 +48,9 @@
namespace {
+constexpr int64_t kMaxRetransmissionWindowMs = 1000;
+constexpr int64_t kMinRetransmissionWindowMs = 30;
+
bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
acm2::RentACodec* rac,
const CodecInst& ci) {
@@ -902,6 +906,8 @@
feedback_observer_proxy_(new TransportFeedbackProxy()),
seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
+ retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
+ kMaxRetransmissionWindowMs)),
decoder_factory_(decoder_factory) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::Channel() - ctor");
@@ -933,6 +939,8 @@
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
configuration.event_log = &(*event_log_proxy_);
+ configuration.retransmission_rate_limiter =
+ retransmission_rate_limiter_.get();
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -1352,6 +1360,7 @@
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
"Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
audio_coding_->SetBitRate(bitrate_bps);
+ retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
}
void Channel::OnIncomingFractionLoss(int fraction_lost) {
@@ -1710,6 +1719,15 @@
// Waiting for valid RTT.
return 0;
}
+
+ int64_t nack_window_ms = rtt;
+ if (nack_window_ms < kMinRetransmissionWindowMs) {
+ nack_window_ms = kMinRetransmissionWindowMs;
+ } else if (nack_window_ms > kMaxRetransmissionWindowMs) {
+ nack_window_ms = kMaxRetransmissionWindowMs;
+ }
+ retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
+
uint32_t ntp_secs = 0;
uint32_t ntp_frac = 0;
uint32_t rtp_timestamp = 0;
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index b51b7ec..34e5c5a 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -36,7 +36,6 @@
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace rtc {
-
class TimestampWrapAroundHandler;
}
@@ -47,6 +46,7 @@
class FileWrapper;
class PacketRouter;
class ProcessThread;
+class RateLimiter;
class ReceiveStatistics;
class RemoteNtpTimeEstimator;
class RtcEventLog;
@@ -589,6 +589,7 @@
std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
+ std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
// TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
