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Add NACK rate throttling for audio channels.

Browse Source 
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}
This commit is contained in:
Erik Språng
2016-07-29 12:59:36 +02:00
parent f3882571b0
commit 737336d37a
18 changed files with 218 additions and 73 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/call/rtc_event_log_unittest.cc
View File

@ -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"
@ -111,6 +112,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
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
@ -123,7 +125,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
nullptr, // SendSideDelayObserver*
nullptr, // RtcEventLog*
nullptr, // SendPacketObserver*
nullptr); // NackRateLimiter*
&retranmission_rate_limiter);
std::vector<uint32_t> csrcs;
for (unsigned i = 0; i < csrcs_count; i++) {

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

@ -175,7 +175,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
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 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
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 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
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) {

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

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

32
webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
View File

@ -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"
@ -80,25 +81,25 @@ class RtcpReceiverTest : public ::testing::Test {
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_);
}
~RtcpReceiverTest() {
delete rtcp_receiver_;
delete rtp_rtcp_impl_;
delete test_transport_;
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() {}
// Injects an RTCP packet into the receiver.
// Returns 0 for OK, non-0 for failure.
int InjectRtcpPacket(const uint8_t* packet,
@ -142,12 +143,13 @@ class RtcpReceiverTest : public ::testing::Test {
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_;
};

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

@ -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 @@ class RtcpSenderTest : public ::testing::Test {
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 @@ class RtcpSenderTest : public ::testing::Test {
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) {

2
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -127,6 +127,8 @@ RTPSender::RTPSender(
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.

24
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -1013,7 +1013,7 @@ TEST_F(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
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 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
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 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
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 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
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 @@ class RtpSenderAudioTest : public RtpSenderTest {
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);
}
};

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -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 @@ int32_t TestRtpReceiver::OnReceivedPayloadData(
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 @@ class RtpRtcpAPITest : public ::testing::Test {
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 @@ class RtpRtcpAPITest : public ::testing::Test {
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAPITest, Basic) {

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -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 RTPCallback : public NullRtpFeedback {
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 @@ class RtpRtcpAudioTest : public ::testing::Test {
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 @@ class RtpRtcpAudioTest : public ::testing::Test {
uint16_t test_sequence_number;
uint32_t test_CSRC[webrtc::kRtpCsrcSize];
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAudioTest, Basic) {

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
View File

@ -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 TestRtpFeedback : public NullRtpFeedback {
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 @@ class RtpRtcpRtcpTest : public ::testing::Test {
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 @@ class RtpRtcpRtcpTest : public ::testing::Test {
uint16_t test_sequence_number;
std::vector<uint32_t> test_csrcs;
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
View File

@ -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 @@ class RtpRtcpVideoTest : public ::testing::Test {
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 @@ class RtpRtcpVideoTest : public ::testing::Test {
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 @@ class RtpRtcpVideoTest : public ::testing::Test {
uint8_t video_frame_[65000];
size_t payload_data_length_;
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpVideoTest, BasicVideo) {

79
webrtc/video/end_to_end_tests.cc
View File

@ -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 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
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 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) {
poller_thread_(&BitrateStatsPollingThread,
this,
"BitrateStatsPollingThread"),
state_(kWaitForFirstRampUp) {}
state_(kWaitForFirstRampUp),
retransmission_rate_limiter_(clock_, 1000) {}
~BweObserver() {}
@ -1847,6 +1922,7 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) {
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 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) {
rtc::Event event_;
rtc::PlatformThread poller_thread_;
TestState state_;
RateLimiter retransmission_rate_limiter_;
} test;
RunBaseTest(&test);

11
webrtc/video/rtp_stream_receiver.cc
View File

@ -39,7 +39,8 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
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 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
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 @@ RtpStreamReceiver::RtpStreamReceiver(
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 @@ RtpStreamReceiver::RtpStreamReceiver(
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);

3
webrtc/video/rtp_stream_receiver.h
View File

@ -62,7 +62,8 @@ class RtpStreamReceiver : public RtpData, public RtpFeedback,
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);

7
webrtc/video/video_receive_stream.cc
View File

@ -14,6 +14,7 @@
#include <set>
#include <string>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
@ -162,7 +163,8 @@ VideoReceiveStream::VideoReceiveStream(
call_stats_(call_stats),
video_receiver_(clock_, nullptr, this, this, this),
stats_proxy_(&config_, clock_),
rtp_stream_receiver_(&video_receiver_,
rtp_stream_receiver_(
&video_receiver_,
congestion_controller_->GetRemoteBitrateEstimator(
UseSendSideBwe(config_)),
&transport_adapter_,
@ -172,7 +174,8 @@ VideoReceiveStream::VideoReceiveStream(
remb,
&config_,
&stats_proxy_,
process_thread_),
process_thread_,
congestion_controller_->GetRetransmissionRateLimiter()),
vie_sync_(&video_receiver_) {
LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();

26
webrtc/video/video_send_stream_tests.cc
View File

@ -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 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
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 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
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 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
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 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
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 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
public:
EncoderStateObserver()
: SendTest(kDefaultTimeoutMs),
stream_(nullptr),
initialized_(false),
callback_registered_(false),
num_releases_(0),
@ -1624,7 +1629,8 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
: 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 @@ class VideoCodecConfigObserver : public test::SendTest,
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 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
: 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 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
public:
ScreencastTargetBitrateTest()
: SendTest(kDefaultTimeoutMs),
test::FakeEncoder(Clock::GetRealTimeClock()) {}
test::FakeEncoder(Clock::GetRealTimeClock()),
send_stream_(nullptr) {}
private:
int32_t Encode(const VideoFrame& input_image,

18
webrtc/voice_engine/channel.cc
View File

@ -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 voe {
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 @@ Channel::Channel(int32_t channelId,
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 @@ Channel::Channel(int32_t channelId,
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 @@ void Channel::SetBitRate(int bitrate_bps) {
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 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
// 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;

3
webrtc/voice_engine/channel.h
View File

@ -36,7 +36,6 @@
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace rtc {
class TimestampWrapAroundHandler;
}
@ -47,6 +46,7 @@ class Config;
class FileWrapper;
class PacketRouter;
class ProcessThread;
class RateLimiter;
class ReceiveStatistics;
class RemoteNtpTimeEstimator;
class RtcEventLog;
@ -589,6 +589,7 @@ class Channel
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_;