| /* |
| * 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/test/simulated_network.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "modules/include/module_common_types_public.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "modules/video_coding/codecs/vp8/include/vp8.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "test/call_test.h" |
| #include "test/gtest.h" |
| #include "test/rtcp_packet_parser.h" |
| |
| namespace webrtc { |
| namespace { |
| enum : int { // The first valid value is 1. |
| kTransportSequenceNumberExtensionId = 1, |
| }; |
| } // namespace |
| |
| class RtpRtcpEndToEndTest : public test::CallTest { |
| protected: |
| void RespectsRtcpMode(RtcpMode rtcp_mode); |
| void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp); |
| }; |
| |
| void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) { |
| static const int kNumCompoundRtcpPacketsToObserve = 10; |
| class RtcpModeObserver : public test::EndToEndTest { |
| public: |
| explicit RtcpModeObserver(RtcpMode rtcp_mode) |
| : EndToEndTest(kDefaultTimeoutMs), |
| rtcp_mode_(rtcp_mode), |
| sent_rtp_(0), |
| sent_rtcp_(0) {} |
| |
| private: |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| MutexLock lock(&mutex_); |
| if (++sent_rtp_ % 3 == 0) |
| return DROP_PACKET; |
| |
| return SEND_PACKET; |
| } |
| |
| Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
| MutexLock lock(&mutex_); |
| ++sent_rtcp_; |
| test::RtcpPacketParser parser; |
| EXPECT_TRUE(parser.Parse(packet, length)); |
| |
| EXPECT_EQ(0, parser.sender_report()->num_packets()); |
| |
| switch (rtcp_mode_) { |
| case RtcpMode::kCompound: |
| // TODO(holmer): We shouldn't send transport feedback alone if |
| // compound RTCP is negotiated. |
| if (parser.receiver_report()->num_packets() == 0 && |
| parser.transport_feedback()->num_packets() == 0) { |
| ADD_FAILURE() << "Received RTCP packet without receiver report for " |
| "RtcpMode::kCompound."; |
| observation_complete_.Set(); |
| } |
| |
| if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve) |
| observation_complete_.Set(); |
| |
| break; |
| case RtcpMode::kReducedSize: |
| if (parser.receiver_report()->num_packets() == 0) |
| observation_complete_.Set(); |
| break; |
| case RtcpMode::kOff: |
| RTC_NOTREACHED(); |
| break; |
| } |
| |
| return SEND_PACKET; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) |
| << (rtcp_mode_ == RtcpMode::kCompound |
| ? "Timed out before observing enough compound packets." |
| : "Timed out before receiving a non-compound RTCP packet."); |
| } |
| |
| RtcpMode rtcp_mode_; |
| Mutex mutex_; |
| // Must be protected since RTCP can be sent by both the process thread |
| // and the pacer thread. |
| int sent_rtp_ RTC_GUARDED_BY(&mutex_); |
| int sent_rtcp_ RTC_GUARDED_BY(&mutex_); |
| } test(rtcp_mode); |
| |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RtpRtcpEndToEndTest, UsesRtcpCompoundMode) { |
| RespectsRtcpMode(RtcpMode::kCompound); |
| } |
| |
| TEST_F(RtpRtcpEndToEndTest, UsesRtcpReducedSizeMode) { |
| RespectsRtcpMode(RtcpMode::kReducedSize); |
| } |
| |
| void RtpRtcpEndToEndTest::TestRtpStatePreservation( |
| bool use_rtx, |
| bool provoke_rtcpsr_before_rtp) { |
| // This test uses other VideoStream settings than the the default settings |
| // implemented in DefaultVideoStreamFactory. Therefore this test implements |
| // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created |
| // in ModifyVideoConfigs. |
| class VideoStreamFactory |
| : public VideoEncoderConfig::VideoStreamFactoryInterface { |
| public: |
| VideoStreamFactory() {} |
| |
| private: |
| std::vector<VideoStream> CreateEncoderStreams( |
| int width, |
| int height, |
| const VideoEncoderConfig& encoder_config) override { |
| std::vector<VideoStream> streams = |
| test::CreateVideoStreams(width, height, encoder_config); |
| |
| if (encoder_config.number_of_streams > 1) { |
| // Lower bitrates so that all streams send initially. |
| RTC_DCHECK_EQ(3, encoder_config.number_of_streams); |
| for (size_t i = 0; i < encoder_config.number_of_streams; ++i) { |
| streams[i].min_bitrate_bps = 10000; |
| streams[i].target_bitrate_bps = 15000; |
| streams[i].max_bitrate_bps = 20000; |
| } |
| } else { |
| // Use the same total bitrates when sending a single stream to avoid |
| // lowering |
| // the bitrate estimate and requiring a subsequent rampup. |
| streams[0].min_bitrate_bps = 3 * 10000; |
| streams[0].target_bitrate_bps = 3 * 15000; |
| streams[0].max_bitrate_bps = 3 * 20000; |
| } |
| return streams; |
| } |
| }; |
| |
| class RtpSequenceObserver : public test::RtpRtcpObserver { |
| public: |
| explicit RtpSequenceObserver(bool use_rtx) |
| : test::RtpRtcpObserver(kDefaultTimeoutMs), |
| ssrcs_to_observe_(kNumSimulcastStreams) { |
| for (size_t i = 0; i < kNumSimulcastStreams; ++i) { |
| ssrc_is_rtx_[kVideoSendSsrcs[i]] = false; |
| if (use_rtx) |
| ssrc_is_rtx_[kSendRtxSsrcs[i]] = true; |
| } |
| } |
| |
| void ResetExpectedSsrcs(size_t num_expected_ssrcs) { |
| MutexLock lock(&mutex_); |
| ssrc_observed_.clear(); |
| ssrcs_to_observe_ = num_expected_ssrcs; |
| } |
| |
| private: |
| void ValidateTimestampGap(uint32_t ssrc, |
| uint32_t timestamp, |
| bool only_padding) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) { |
| static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90; |
| auto timestamp_it = last_observed_timestamp_.find(ssrc); |
| if (timestamp_it == last_observed_timestamp_.end()) { |
| EXPECT_FALSE(only_padding); |
| last_observed_timestamp_[ssrc] = timestamp; |
| } else { |
| // Verify timestamps are reasonably close. |
| uint32_t latest_observed = timestamp_it->second; |
| // Wraparound handling is unnecessary here as long as an int variable |
| // is used to store the result. |
| int32_t timestamp_gap = timestamp - latest_observed; |
| EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap) |
| << "Gap in timestamps (" << latest_observed << " -> " << timestamp |
| << ") too large for SSRC: " << ssrc << "."; |
| timestamp_it->second = timestamp; |
| } |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| const uint32_t ssrc = rtp_packet.Ssrc(); |
| const int64_t sequence_number = |
| seq_numbers_unwrapper_.Unwrap(rtp_packet.SequenceNumber()); |
| const uint32_t timestamp = rtp_packet.Timestamp(); |
| const bool only_padding = rtp_packet.payload_size() == 0; |
| |
| EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end()) |
| << "Received SSRC that wasn't configured: " << ssrc; |
| |
| static const int64_t kMaxSequenceNumberGap = 100; |
| std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc]; |
| if (seq_numbers->empty()) { |
| seq_numbers->push_back(sequence_number); |
| } else { |
| // We shouldn't get replays of previous sequence numbers. |
| for (int64_t observed : *seq_numbers) { |
| EXPECT_NE(observed, sequence_number) |
| << "Received sequence number " << sequence_number << " for SSRC " |
| << ssrc << " 2nd time."; |
| } |
| // Verify sequence numbers are reasonably close. |
| int64_t latest_observed = seq_numbers->back(); |
| int64_t sequence_number_gap = sequence_number - latest_observed; |
| EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap) |
| << "Gap in sequence numbers (" << latest_observed << " -> " |
| << sequence_number << ") too large for SSRC: " << ssrc << "."; |
| seq_numbers->push_back(sequence_number); |
| if (seq_numbers->size() >= kMaxSequenceNumberGap) { |
| seq_numbers->pop_front(); |
| } |
| } |
| |
| if (!ssrc_is_rtx_[ssrc]) { |
| MutexLock lock(&mutex_); |
| ValidateTimestampGap(ssrc, timestamp, only_padding); |
| |
| // Wait for media packets on all ssrcs. |
| if (!ssrc_observed_[ssrc] && !only_padding) { |
| ssrc_observed_[ssrc] = true; |
| if (--ssrcs_to_observe_ == 0) |
| observation_complete_.Set(); |
| } |
| } |
| |
| return SEND_PACKET; |
| } |
| |
| Action OnSendRtcp(const uint8_t* packet, size_t length) override { |
| test::RtcpPacketParser rtcp_parser; |
| rtcp_parser.Parse(packet, length); |
| if (rtcp_parser.sender_report()->num_packets() > 0) { |
| uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc(); |
| uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp(); |
| |
| MutexLock lock(&mutex_); |
| ValidateTimestampGap(ssrc, rtcp_timestamp, false); |
| } |
| return SEND_PACKET; |
| } |
| |
| SequenceNumberUnwrapper seq_numbers_unwrapper_; |
| std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_; |
| std::map<uint32_t, uint32_t> last_observed_timestamp_; |
| std::map<uint32_t, bool> ssrc_is_rtx_; |
| |
| Mutex mutex_; |
| size_t ssrcs_to_observe_ RTC_GUARDED_BY(mutex_); |
| std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(mutex_); |
| } observer(use_rtx); |
| |
| std::unique_ptr<test::PacketTransport> send_transport; |
| std::unique_ptr<test::PacketTransport> receive_transport; |
| |
| VideoEncoderConfig one_stream; |
| |
| SendTask( |
| RTC_FROM_HERE, task_queue(), |
| [this, &observer, &send_transport, &receive_transport, &one_stream, |
| use_rtx]() { |
| CreateCalls(); |
| |
| send_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), sender_call_.get(), &observer, |
| test::PacketTransport::kSender, payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig()))); |
| receive_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), nullptr, &observer, test::PacketTransport::kReceiver, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig()))); |
| send_transport->SetReceiver(receiver_call_->Receiver()); |
| receive_transport->SetReceiver(sender_call_->Receiver()); |
| |
| CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get()); |
| |
| if (use_rtx) { |
| for (size_t i = 0; i < kNumSimulcastStreams; ++i) { |
| GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
| } |
| GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType; |
| } |
| |
| GetVideoEncoderConfig()->video_stream_factory = |
| new rtc::RefCountedObject<VideoStreamFactory>(); |
| // Use the same total bitrates when sending a single stream to avoid |
| // lowering the bitrate estimate and requiring a subsequent rampup. |
| one_stream = GetVideoEncoderConfig()->Copy(); |
| // one_stream.streams.resize(1); |
| one_stream.number_of_streams = 1; |
| CreateMatchingReceiveConfigs(receive_transport.get()); |
| |
| CreateVideoStreams(); |
| CreateFrameGeneratorCapturer(30, 1280, 720); |
| |
| Start(); |
| }); |
| |
| EXPECT_TRUE(observer.Wait()) |
| << "Timed out waiting for all SSRCs to send packets."; |
| |
| // Test stream resetting more than once to make sure that the state doesn't |
| // get set once (this could be due to using std::map::insert for instance). |
| for (size_t i = 0; i < 3; ++i) { |
| SendTask(RTC_FROM_HERE, task_queue(), [&]() { |
| DestroyVideoSendStreams(); |
| |
| // Re-create VideoSendStream with only one stream. |
| CreateVideoSendStream(one_stream); |
| GetVideoSendStream()->Start(); |
| if (provoke_rtcpsr_before_rtp) { |
| // Rapid Resync Request forces sending RTCP Sender Report back. |
| // Using this request speeds up this test because then there is no need |
| // to wait for a second for periodic Sender Report. |
| rtcp::RapidResyncRequest force_send_sr_back_request; |
| rtc::Buffer packet = force_send_sr_back_request.Build(); |
| static_cast<webrtc::test::DirectTransport*>(receive_transport.get()) |
| ->SendRtcp(packet.data(), packet.size()); |
| } |
| CreateFrameGeneratorCapturer(30, 1280, 720); |
| }); |
| |
| observer.ResetExpectedSsrcs(1); |
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet."; |
| |
| // Reconfigure back to use all streams. |
| SendTask(RTC_FROM_HERE, task_queue(), [this]() { |
| GetVideoSendStream()->ReconfigureVideoEncoder( |
| GetVideoEncoderConfig()->Copy()); |
| }); |
| observer.ResetExpectedSsrcs(kNumSimulcastStreams); |
| EXPECT_TRUE(observer.Wait()) |
| << "Timed out waiting for all SSRCs to send packets."; |
| |
| // Reconfigure down to one stream. |
| SendTask(RTC_FROM_HERE, task_queue(), [this, &one_stream]() { |
| GetVideoSendStream()->ReconfigureVideoEncoder(one_stream.Copy()); |
| }); |
| observer.ResetExpectedSsrcs(1); |
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet."; |
| |
| // Reconfigure back to use all streams. |
| SendTask(RTC_FROM_HERE, task_queue(), [this]() { |
| GetVideoSendStream()->ReconfigureVideoEncoder( |
| GetVideoEncoderConfig()->Copy()); |
| }); |
| observer.ResetExpectedSsrcs(kNumSimulcastStreams); |
| EXPECT_TRUE(observer.Wait()) |
| << "Timed out waiting for all SSRCs to send packets."; |
| } |
| |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &send_transport, &receive_transport]() { |
| Stop(); |
| DestroyStreams(); |
| send_transport.reset(); |
| receive_transport.reset(); |
| DestroyCalls(); |
| }); |
| } |
| |
| TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpState) { |
| TestRtpStatePreservation(false, false); |
| } |
| |
| TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) { |
| TestRtpStatePreservation(true, false); |
| } |
| |
| TEST_F(RtpRtcpEndToEndTest, |
| RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) { |
| TestRtpStatePreservation(true, true); |
| } |
| |
| // See https://bugs.chromium.org/p/webrtc/issues/detail?id=9648. |
| TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) { |
| class RtpSequenceObserver : public test::RtpRtcpObserver { |
| public: |
| RtpSequenceObserver() |
| : test::RtpRtcpObserver(kDefaultTimeoutMs), |
| num_flexfec_packets_sent_(0) {} |
| |
| void ResetPacketCount() { |
| MutexLock lock(&mutex_); |
| num_flexfec_packets_sent_ = 0; |
| } |
| |
| private: |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| MutexLock lock(&mutex_); |
| |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| const uint16_t sequence_number = rtp_packet.SequenceNumber(); |
| const uint32_t timestamp = rtp_packet.Timestamp(); |
| const uint32_t ssrc = rtp_packet.Ssrc(); |
| |
| if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) { |
| return SEND_PACKET; |
| } |
| EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent."; |
| |
| ++num_flexfec_packets_sent_; |
| |
| // If this is the first packet, we have nothing to compare to. |
| if (!last_observed_sequence_number_) { |
| last_observed_sequence_number_.emplace(sequence_number); |
| last_observed_timestamp_.emplace(timestamp); |
| |
| return SEND_PACKET; |
| } |
| |
| // Verify continuity and monotonicity of RTP sequence numbers. |
| EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1), |
| sequence_number); |
| last_observed_sequence_number_.emplace(sequence_number); |
| |
| // Timestamps should be non-decreasing... |
| const bool timestamp_is_same_or_newer = |
| timestamp == *last_observed_timestamp_ || |
| IsNewerTimestamp(timestamp, *last_observed_timestamp_); |
| EXPECT_TRUE(timestamp_is_same_or_newer); |
| // ...but reasonably close in time. |
| const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency; |
| EXPECT_TRUE(IsNewerTimestamp( |
| *last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp)); |
| last_observed_timestamp_.emplace(timestamp); |
| |
| // Pass test when enough packets have been let through. |
| if (num_flexfec_packets_sent_ >= 10) { |
| observation_complete_.Set(); |
| } |
| |
| return SEND_PACKET; |
| } |
| |
| absl::optional<uint16_t> last_observed_sequence_number_ |
| RTC_GUARDED_BY(mutex_); |
| absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_); |
| size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(mutex_); |
| Mutex mutex_; |
| } observer; |
| |
| static constexpr int kFrameMaxWidth = 320; |
| static constexpr int kFrameMaxHeight = 180; |
| static constexpr int kFrameRate = 15; |
| |
| std::unique_ptr<test::PacketTransport> send_transport; |
| std::unique_ptr<test::PacketTransport> receive_transport; |
| test::FunctionVideoEncoderFactory encoder_factory( |
| []() { return VP8Encoder::Create(); }); |
| |
| SendTask(RTC_FROM_HERE, task_queue(), [&]() { |
| CreateCalls(); |
| |
| BuiltInNetworkBehaviorConfig lossy_delayed_link; |
| lossy_delayed_link.loss_percent = 2; |
| lossy_delayed_link.queue_delay_ms = 50; |
| |
| send_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), sender_call_.get(), &observer, |
| test::PacketTransport::kSender, payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(lossy_delayed_link))); |
| send_transport->SetReceiver(receiver_call_->Receiver()); |
| |
| BuiltInNetworkBehaviorConfig flawless_link; |
| receive_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), nullptr, &observer, test::PacketTransport::kReceiver, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(flawless_link))); |
| receive_transport->SetReceiver(sender_call_->Receiver()); |
| |
| // For reduced flakyness, we use a real VP8 encoder together with NACK |
| // and RTX. |
| const int kNumVideoStreams = 1; |
| const int kNumFlexfecStreams = 1; |
| CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams, |
| send_transport.get()); |
| |
| GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory; |
| GetVideoSendConfig()->rtp.payload_name = "VP8"; |
| GetVideoSendConfig()->rtp.payload_type = kVideoSendPayloadType; |
| GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); |
| GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType; |
| GetVideoEncoderConfig()->codec_type = kVideoCodecVP8; |
| |
| CreateMatchingReceiveConfigs(receive_transport.get()); |
| video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0]; |
| video_receive_configs_[0] |
| .rtp.rtx_associated_payload_types[kSendRtxPayloadType] = |
| kVideoSendPayloadType; |
| |
| // The matching FlexFEC receive config is not created by |
| // CreateMatchingReceiveConfigs since this is not a test::BaseTest. |
| // Set up the receive config manually instead. |
| FlexfecReceiveStream::Config flexfec_receive_config( |
| receive_transport.get()); |
| flexfec_receive_config.payload_type = |
| GetVideoSendConfig()->rtp.flexfec.payload_type; |
| flexfec_receive_config.remote_ssrc = GetVideoSendConfig()->rtp.flexfec.ssrc; |
| flexfec_receive_config.protected_media_ssrcs = |
| GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs; |
| flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc; |
| flexfec_receive_config.transport_cc = true; |
| flexfec_receive_config.rtp_header_extensions.emplace_back( |
| RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId); |
| flexfec_receive_configs_.push_back(flexfec_receive_config); |
| |
| CreateFlexfecStreams(); |
| CreateVideoStreams(); |
| |
| // RTCP might be disabled if the network is "down". |
| sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp); |
| receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp); |
| |
| CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight); |
| |
| Start(); |
| }); |
| |
| // Initial test. |
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets."; |
| |
| SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() { |
| // Ensure monotonicity when the VideoSendStream is restarted. |
| Stop(); |
| observer.ResetPacketCount(); |
| Start(); |
| }); |
| |
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets."; |
| |
| SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() { |
| // Ensure monotonicity when the VideoSendStream is recreated. |
| DestroyVideoSendStreams(); |
| observer.ResetPacketCount(); |
| CreateVideoSendStreams(); |
| GetVideoSendStream()->Start(); |
| CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight); |
| }); |
| |
| EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets."; |
| |
| // Cleanup. |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &send_transport, &receive_transport]() { |
| Stop(); |
| DestroyStreams(); |
| send_transport.reset(); |
| receive_transport.reset(); |
| DestroyCalls(); |
| }); |
| } |
| } // namespace webrtc |
