| /* |
| * Copyright 2018 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <memory> |
| |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/test/simulated_network.h" |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" |
| #include "api/video/video_bitrate_allocation.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" |
| #include "rtc_base/rate_limiter.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "system_wrappers/include/sleep.h" |
| #include "test/call_test.h" |
| #include "test/fake_encoder.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| #include "test/rtcp_packet_parser.h" |
| #include "test/rtp_rtcp_observer.h" |
| #include "test/video_encoder_proxy_factory.h" |
| |
| namespace webrtc { |
| namespace { |
| enum : int { // The first valid value is 1. |
| kAbsSendTimeExtensionId = 1, |
| kTransportSequenceNumberId, |
| }; |
| } // namespace |
| |
| class BandwidthEndToEndTest : public test::CallTest { |
| public: |
| BandwidthEndToEndTest() = default; |
| }; |
| |
| TEST_F(BandwidthEndToEndTest, ReceiveStreamSendsRemb) { |
| class RembObserver : public test::EndToEndTest { |
| public: |
| RembObserver() : EndToEndTest(kDefaultTimeoutMs) {} |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->rtp.extensions.clear(); |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); |
| (*receive_configs)[0].rtp.transport_cc = false; |
| } |
| |
| Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
| test::RtcpPacketParser parser; |
| EXPECT_TRUE(parser.Parse(packet, length)); |
| |
| if (parser.remb()->num_packets() > 0) { |
| EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc()); |
| EXPECT_LT(0U, parser.remb()->bitrate_bps()); |
| EXPECT_EQ(1U, parser.remb()->ssrcs().size()); |
| EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]); |
| observation_complete_.Set(); |
| } |
| |
| return SEND_PACKET; |
| } |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out while waiting for a " |
| "receiver RTCP REMB packet to be " |
| "sent."; |
| } |
| } test; |
| |
| RunBaseTest(&test); |
| } |
| |
| class BandwidthStatsTest : public test::EndToEndTest { |
| public: |
| BandwidthStatsTest(bool send_side_bwe, TaskQueueBase* task_queue) |
| : EndToEndTest(test::CallTest::kDefaultTimeoutMs), |
| sender_call_(nullptr), |
| receiver_call_(nullptr), |
| has_seen_pacer_delay_(false), |
| send_side_bwe_(send_side_bwe), |
| task_queue_(task_queue) {} |
| |
| ~BandwidthStatsTest() override { |
| // Block until all already posted tasks run to avoid races when such task |
| // accesses `this`. |
| SendTask(RTC_FROM_HERE, task_queue_, [] {}); |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->rtp.extensions.clear(); |
| if (!send_side_bwe_) { |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); |
| (*receive_configs)[0].rtp.transport_cc = false; |
| } else { |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberId)); |
| (*receive_configs)[0].rtp.transport_cc = true; |
| } |
| |
| // Force a too high encoder bitrate to make sure we get pacer delay. |
| encoder_config->number_of_streams = 1; |
| encoder_config->max_bitrate_bps = kMaxBitrateBps * 2; |
| encoder_config->simulcast_layers[0].min_bitrate_bps = kMaxBitrateBps * 2; |
| encoder_config->simulcast_layers[0].target_bitrate_bps = kMaxBitrateBps * 2; |
| encoder_config->simulcast_layers[0].max_bitrate_bps = kMaxBitrateBps * 2; |
| } |
| |
| void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) override { |
| bitrate_config->max_bitrate_bps = kMaxBitrateBps; |
| } |
| |
| // Called on the pacer thread. |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| // Stats need to be fetched on the thread where the caller objects were |
| // constructed. |
| task_queue_->PostTask(ToQueuedTask([this]() { |
| if (!sender_call_ || !receiver_call_) { |
| return; |
| } |
| |
| Call::Stats sender_stats = sender_call_->GetStats(); |
| if (!has_seen_pacer_delay_) { |
| has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0; |
| } |
| |
| if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) { |
| Call::Stats receiver_stats = receiver_call_->GetStats(); |
| if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0) { |
| observation_complete_.Set(); |
| } |
| } |
| })); |
| |
| return SEND_PACKET; |
| } |
| |
| void OnCallsCreated(Call* sender_call, Call* receiver_call) override { |
| sender_call_ = sender_call; |
| receiver_call_ = receiver_call; |
| } |
| |
| void OnStreamsStopped() override { |
| sender_call_ = nullptr; |
| receiver_call_ = nullptr; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out while waiting for " |
| "non-zero bandwidth stats."; |
| } |
| |
| private: |
| static const int kMaxBitrateBps = 3000000; |
| Call* sender_call_; |
| Call* receiver_call_; |
| bool has_seen_pacer_delay_; |
| const bool send_side_bwe_; |
| TaskQueueBase* const task_queue_; |
| }; |
| |
| TEST_F(BandwidthEndToEndTest, VerifySendSideBweStats) { |
| BandwidthStatsTest test(true, task_queue()); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(BandwidthEndToEndTest, VerifyRecvSideBweStats) { |
| BandwidthStatsTest test(false, task_queue()); |
| RunBaseTest(&test); |
| } |
| |
| // Verifies that it's possible to limit the send BWE by sending a REMB. |
| // This is verified by allowing the send BWE to ramp-up to >1000 kbps, |
| // then have the test generate a REMB of 500 kbps and verify that the send BWE |
| // is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the |
| // test verifies that the send BWE ramps back up to exactly 1000 kbps. |
| TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) { |
| class BweObserver : public test::EndToEndTest { |
| public: |
| explicit BweObserver(TaskQueueBase* task_queue) |
| : EndToEndTest(kDefaultTimeoutMs), |
| sender_call_(nullptr), |
| clock_(Clock::GetRealTimeClock()), |
| sender_ssrc_(0), |
| remb_bitrate_bps_(1000000), |
| receive_transport_(nullptr), |
| state_(kWaitForFirstRampUp), |
| retransmission_rate_limiter_(clock_, 1000), |
| task_queue_(task_queue) {} |
| |
| void OnStreamsStopped() override { rtp_rtcp_ = nullptr; } |
| |
| std::unique_ptr<test::PacketTransport> CreateReceiveTransport( |
| TaskQueueBase* task_queue) override { |
| auto receive_transport = std::make_unique<test::PacketTransport>( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig()))); |
| receive_transport_ = receive_transport.get(); |
| return receive_transport; |
| } |
| |
| void ModifySenderBitrateConfig( |
| BitrateConstraints* bitrate_config) override { |
| // Set a high start bitrate to reduce the test completion time. |
| bitrate_config->start_bitrate_bps = remb_bitrate_bps_; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| ASSERT_EQ(1u, send_config->rtp.ssrcs.size()); |
| sender_ssrc_ = send_config->rtp.ssrcs[0]; |
| |
| encoder_config->max_bitrate_bps = 2000000; |
| |
| ASSERT_EQ(1u, receive_configs->size()); |
| RtpRtcpInterface::Configuration config; |
| config.receiver_only = true; |
| config.clock = clock_; |
| config.outgoing_transport = receive_transport_; |
| config.retransmission_rate_limiter = &retransmission_rate_limiter_; |
| config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc; |
| rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config); |
| rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc); |
| rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize); |
| } |
| |
| void OnCallsCreated(Call* sender_call, Call* receiver_call) override { |
| RTC_DCHECK(sender_call); |
| sender_call_ = sender_call; |
| task_queue_->PostTask(ToQueuedTask([this]() { PollStats(); })); |
| } |
| |
| void PollStats() { |
| Call::Stats stats = sender_call_->GetStats(); |
| switch (state_) { |
| case kWaitForFirstRampUp: |
| if (stats.send_bandwidth_bps >= remb_bitrate_bps_) { |
| state_ = kWaitForRemb; |
| remb_bitrate_bps_ /= 2; |
| rtp_rtcp_->SetRemb( |
| remb_bitrate_bps_, |
| std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1)); |
| rtp_rtcp_->SendRTCP(kRtcpRr); |
| } |
| break; |
| |
| case kWaitForRemb: |
| if (stats.send_bandwidth_bps == remb_bitrate_bps_) { |
| state_ = kWaitForSecondRampUp; |
| remb_bitrate_bps_ *= 2; |
| rtp_rtcp_->SetRemb( |
| remb_bitrate_bps_, |
| std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1)); |
| rtp_rtcp_->SendRTCP(kRtcpRr); |
| } |
| break; |
| |
| case kWaitForSecondRampUp: |
| if (stats.send_bandwidth_bps == remb_bitrate_bps_) { |
| observation_complete_.Set(); |
| return; |
| } |
| break; |
| } |
| |
| task_queue_->PostDelayedTask(ToQueuedTask([this] { PollStats(); }), 1000); |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) |
| << "Timed out while waiting for bitrate to change according to REMB."; |
| } |
| |
| private: |
| enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp }; |
| |
| Call* sender_call_; |
| Clock* const clock_; |
| uint32_t sender_ssrc_; |
| int remb_bitrate_bps_; |
| std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; |
| test::PacketTransport* receive_transport_; |
| TestState state_; |
| RateLimiter retransmission_rate_limiter_; |
| TaskQueueBase* const task_queue_; |
| } test(task_queue()); |
| |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) { |
| // If these fields trial are on, we get lower bitrates than expected by this |
| // test, due to the packetization overhead and encoder pushback. |
| webrtc::test::ScopedFieldTrials field_trials( |
| std::string(field_trial::GetFieldTrialString()) + |
| "WebRTC-VideoRateControl/bitrate_adjuster:false/"); |
| class EncoderRateStatsTest : public test::EndToEndTest, |
| public test::FakeEncoder { |
| public: |
| explicit EncoderRateStatsTest(TaskQueueBase* task_queue) |
| : EndToEndTest(kDefaultTimeoutMs), |
| FakeEncoder(Clock::GetRealTimeClock()), |
| task_queue_(task_queue), |
| send_stream_(nullptr), |
| encoder_factory_(this), |
| bitrate_allocator_factory_( |
| CreateBuiltinVideoBitrateAllocatorFactory()), |
| bitrate_kbps_(0) {} |
| |
| void OnVideoStreamsCreated(VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStreamInterface*>& |
| receive_streams) override { |
| send_stream_ = send_stream; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->encoder_settings.encoder_factory = &encoder_factory_; |
| send_config->encoder_settings.bitrate_allocator_factory = |
| bitrate_allocator_factory_.get(); |
| RTC_DCHECK_EQ(1, encoder_config->number_of_streams); |
| } |
| |
| void SetRates(const RateControlParameters& parameters) override { |
| // Make sure not to trigger on any default zero bitrates. |
| if (parameters.bitrate.get_sum_bps() == 0) |
| return; |
| MutexLock lock(&mutex_); |
| bitrate_kbps_ = parameters.bitrate.get_sum_kbps(); |
| observation_complete_.Set(); |
| } |
| |
| void PerformTest() override { |
| ASSERT_TRUE(Wait()) |
| << "Timed out while waiting for encoder SetRates() call."; |
| |
| SendTask(RTC_FROM_HERE, task_queue_, [this]() { |
| WaitForEncoderTargetBitrateMatchStats(); |
| send_stream_->Stop(); |
| WaitForStatsReportZeroTargetBitrate(); |
| send_stream_->Start(); |
| WaitForEncoderTargetBitrateMatchStats(); |
| }); |
| } |
| |
| void WaitForEncoderTargetBitrateMatchStats() { |
| for (int i = 0; i < kDefaultTimeoutMs; ++i) { |
| VideoSendStream::Stats stats = send_stream_->GetStats(); |
| { |
| MutexLock lock(&mutex_); |
| if ((stats.target_media_bitrate_bps + 500) / 1000 == |
| static_cast<int>(bitrate_kbps_)) { |
| return; |
| } |
| } |
| SleepMs(1); |
| } |
| FAIL() |
| << "Timed out waiting for stats reporting the currently set bitrate."; |
| } |
| |
| void WaitForStatsReportZeroTargetBitrate() { |
| for (int i = 0; i < kDefaultTimeoutMs; ++i) { |
| if (send_stream_->GetStats().target_media_bitrate_bps == 0) { |
| return; |
| } |
| SleepMs(1); |
| } |
| FAIL() << "Timed out waiting for stats reporting zero bitrate."; |
| } |
| |
| private: |
| TaskQueueBase* const task_queue_; |
| Mutex mutex_; |
| VideoSendStream* send_stream_; |
| test::VideoEncoderProxyFactory encoder_factory_; |
| std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_; |
| uint32_t bitrate_kbps_ RTC_GUARDED_BY(mutex_); |
| } test(task_queue()); |
| |
| RunBaseTest(&test); |
| } |
| } // namespace webrtc |
