| /* |
| * Copyright (c) 2015 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 "call/rtp_video_sender.h" |
| |
| #include <atomic> |
| #include <memory> |
| #include <string> |
| |
| #include "call/rtp_transport_controller_send.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/byte_io.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/nack.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "modules/video_coding/fec_controller_default.h" |
| #include "modules/video_coding/include/video_codec_interface.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/rate_limiter.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/mock_transport.h" |
| #include "test/scenario/scenario.h" |
| #include "test/time_controller/simulated_time_controller.h" |
| #include "video/call_stats.h" |
| #include "video/send_delay_stats.h" |
| #include "video/send_statistics_proxy.h" |
| |
| using ::testing::_; |
| using ::testing::Invoke; |
| using ::testing::NiceMock; |
| using ::testing::SaveArg; |
| using ::testing::Unused; |
| |
| namespace webrtc { |
| namespace { |
| const int8_t kPayloadType = 96; |
| const uint32_t kSsrc1 = 12345; |
| const uint32_t kSsrc2 = 23456; |
| const uint32_t kRtxSsrc1 = 34567; |
| const uint32_t kRtxSsrc2 = 45678; |
| const int16_t kInitialPictureId1 = 222; |
| const int16_t kInitialPictureId2 = 44; |
| const int16_t kInitialTl0PicIdx1 = 99; |
| const int16_t kInitialTl0PicIdx2 = 199; |
| const int64_t kRetransmitWindowSizeMs = 500; |
| const int kTransportsSequenceExtensionId = 7; |
| |
| class MockRtcpIntraFrameObserver : public RtcpIntraFrameObserver { |
| public: |
| MOCK_METHOD1(OnReceivedIntraFrameRequest, void(uint32_t)); |
| }; |
| |
| RtpSenderObservers CreateObservers( |
| RtcpRttStats* rtcp_rtt_stats, |
| RtcpIntraFrameObserver* intra_frame_callback, |
| RtcpStatisticsCallback* rtcp_stats, |
| ReportBlockDataObserver* report_block_data_observer, |
| StreamDataCountersCallback* rtp_stats, |
| BitrateStatisticsObserver* bitrate_observer, |
| FrameCountObserver* frame_count_observer, |
| RtcpPacketTypeCounterObserver* rtcp_type_observer, |
| SendSideDelayObserver* send_delay_observer, |
| SendPacketObserver* send_packet_observer) { |
| RtpSenderObservers observers; |
| observers.rtcp_rtt_stats = rtcp_rtt_stats; |
| observers.intra_frame_callback = intra_frame_callback; |
| observers.rtcp_loss_notification_observer = nullptr; |
| observers.rtcp_stats = rtcp_stats; |
| observers.report_block_data_observer = report_block_data_observer; |
| observers.rtp_stats = rtp_stats; |
| observers.bitrate_observer = bitrate_observer; |
| observers.frame_count_observer = frame_count_observer; |
| observers.rtcp_type_observer = rtcp_type_observer; |
| observers.send_delay_observer = send_delay_observer; |
| observers.send_packet_observer = send_packet_observer; |
| return observers; |
| } |
| |
| BitrateConstraints GetBitrateConfig() { |
| BitrateConstraints bitrate_config; |
| bitrate_config.min_bitrate_bps = 30000; |
| bitrate_config.start_bitrate_bps = 300000; |
| bitrate_config.max_bitrate_bps = 3000000; |
| return bitrate_config; |
| } |
| |
| VideoSendStream::Config CreateVideoSendStreamConfig( |
| Transport* transport, |
| const std::vector<uint32_t>& ssrcs, |
| const std::vector<uint32_t>& rtx_ssrcs, |
| int payload_type) { |
| VideoSendStream::Config config(transport); |
| config.rtp.ssrcs = ssrcs; |
| config.rtp.rtx.ssrcs = rtx_ssrcs; |
| config.rtp.payload_type = payload_type; |
| config.rtp.rtx.payload_type = payload_type + 1; |
| config.rtp.nack.rtp_history_ms = 1000; |
| config.rtp.extensions.emplace_back(RtpExtension::kTransportSequenceNumberUri, |
| kTransportsSequenceExtensionId); |
| return config; |
| } |
| |
| class RtpVideoSenderTestFixture { |
| public: |
| RtpVideoSenderTestFixture( |
| const std::vector<uint32_t>& ssrcs, |
| const std::vector<uint32_t>& rtx_ssrcs, |
| int payload_type, |
| const std::map<uint32_t, RtpPayloadState>& suspended_payload_states, |
| FrameCountObserver* frame_count_observer) |
| : time_controller_(Timestamp::ms(1000000)), |
| config_(CreateVideoSendStreamConfig(&transport_, |
| ssrcs, |
| rtx_ssrcs, |
| payload_type)), |
| send_delay_stats_(time_controller_.GetClock()), |
| bitrate_config_(GetBitrateConfig()), |
| transport_controller_(time_controller_.GetClock(), |
| &event_log_, |
| nullptr, |
| nullptr, |
| bitrate_config_, |
| ProcessThread::Create("PacerThread"), |
| time_controller_.GetTaskQueueFactory(), |
| &field_trials_), |
| process_thread_(ProcessThread::Create("test_thread")), |
| call_stats_(time_controller_.GetClock(), process_thread_.get()), |
| stats_proxy_(time_controller_.GetClock(), |
| config_, |
| VideoEncoderConfig::ContentType::kRealtimeVideo), |
| retransmission_rate_limiter_(time_controller_.GetClock(), |
| kRetransmitWindowSizeMs) { |
| std::map<uint32_t, RtpState> suspended_ssrcs; |
| router_ = std::make_unique<RtpVideoSender>( |
| time_controller_.GetClock(), suspended_ssrcs, suspended_payload_states, |
| config_.rtp, config_.rtcp_report_interval_ms, &transport_, |
| CreateObservers(&call_stats_, &encoder_feedback_, &stats_proxy_, |
| &stats_proxy_, &stats_proxy_, &stats_proxy_, |
| frame_count_observer, &stats_proxy_, &stats_proxy_, |
| &send_delay_stats_), |
| &transport_controller_, &event_log_, &retransmission_rate_limiter_, |
| std::make_unique<FecControllerDefault>(time_controller_.GetClock()), |
| nullptr, CryptoOptions{}); |
| } |
| RtpVideoSenderTestFixture( |
| const std::vector<uint32_t>& ssrcs, |
| const std::vector<uint32_t>& rtx_ssrcs, |
| int payload_type, |
| const std::map<uint32_t, RtpPayloadState>& suspended_payload_states) |
| : RtpVideoSenderTestFixture(ssrcs, |
| rtx_ssrcs, |
| payload_type, |
| suspended_payload_states, |
| /*frame_count_observer=*/nullptr) {} |
| |
| RtpVideoSender* router() { return router_.get(); } |
| MockTransport& transport() { return transport_; } |
| void AdvanceTime(TimeDelta delta) { time_controller_.AdvanceTime(delta); } |
| |
| private: |
| NiceMock<MockTransport> transport_; |
| NiceMock<MockRtcpIntraFrameObserver> encoder_feedback_; |
| GlobalSimulatedTimeController time_controller_; |
| RtcEventLogNull event_log_; |
| VideoSendStream::Config config_; |
| SendDelayStats send_delay_stats_; |
| BitrateConstraints bitrate_config_; |
| const FieldTrialBasedConfig field_trials_; |
| RtpTransportControllerSend transport_controller_; |
| std::unique_ptr<ProcessThread> process_thread_; |
| CallStats call_stats_; |
| SendStatisticsProxy stats_proxy_; |
| RateLimiter retransmission_rate_limiter_; |
| std::unique_ptr<RtpVideoSender> router_; |
| }; |
| } // namespace |
| |
| TEST(RtpVideoSenderTest, SendOnOneModule) { |
| constexpr uint8_t kPayload = 'a'; |
| EncodedImage encoded_image; |
| encoded_image.SetTimestamp(1); |
| encoded_image.capture_time_ms_ = 2; |
| encoded_image._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1)); |
| |
| RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}); |
| EXPECT_NE( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| test.router()->SetActive(true); |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| test.router()->SetActive(false); |
| EXPECT_NE( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| test.router()->SetActive(true); |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| } |
| |
| TEST(RtpVideoSenderTest, SendSimulcastSetActive) { |
| constexpr uint8_t kPayload = 'a'; |
| EncodedImage encoded_image_1; |
| encoded_image_1.SetTimestamp(1); |
| encoded_image_1.capture_time_ms_ = 2; |
| encoded_image_1._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image_1.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1)); |
| |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, {}); |
| |
| CodecSpecificInfo codec_info; |
| codec_info.codecType = kVideoCodecVP8; |
| |
| test.router()->SetActive(true); |
| EXPECT_EQ(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_1, &codec_info, nullptr) |
| .error); |
| |
| EncodedImage encoded_image_2(encoded_image_1); |
| encoded_image_2.SetSpatialIndex(1); |
| EXPECT_EQ(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_2, &codec_info, nullptr) |
| .error); |
| |
| // Inactive. |
| test.router()->SetActive(false); |
| EXPECT_NE(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_1, &codec_info, nullptr) |
| .error); |
| EXPECT_NE(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_2, &codec_info, nullptr) |
| .error); |
| } |
| |
| // Tests how setting individual rtp modules to active affects the overall |
| // behavior of the payload router. First sets one module to active and checks |
| // that outgoing data can be sent on this module, and checks that no data can |
| // be sent if both modules are inactive. |
| TEST(RtpVideoSenderTest, SendSimulcastSetActiveModules) { |
| constexpr uint8_t kPayload = 'a'; |
| EncodedImage encoded_image_1; |
| encoded_image_1.SetTimestamp(1); |
| encoded_image_1.capture_time_ms_ = 2; |
| encoded_image_1._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image_1.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1)); |
| |
| EncodedImage encoded_image_2(encoded_image_1); |
| encoded_image_2.SetSpatialIndex(1); |
| |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, {}); |
| CodecSpecificInfo codec_info; |
| codec_info.codecType = kVideoCodecVP8; |
| |
| // Only setting one stream to active will still set the payload router to |
| // active and allow sending data on the active stream. |
| std::vector<bool> active_modules({true, false}); |
| test.router()->SetActiveModules(active_modules); |
| EXPECT_EQ(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_1, &codec_info, nullptr) |
| .error); |
| |
| // Setting both streams to inactive will turn the payload router to |
| // inactive. |
| active_modules = {false, false}; |
| test.router()->SetActiveModules(active_modules); |
| // An incoming encoded image will not ask the module to send outgoing data |
| // because the payload router is inactive. |
| EXPECT_NE(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_1, &codec_info, nullptr) |
| .error); |
| EXPECT_NE(EncodedImageCallback::Result::OK, |
| test.router() |
| ->OnEncodedImage(encoded_image_1, &codec_info, nullptr) |
| .error); |
| } |
| |
| TEST(RtpVideoSenderTest, CreateWithNoPreviousStates) { |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, {}); |
| test.router()->SetActive(true); |
| |
| std::map<uint32_t, RtpPayloadState> initial_states = |
| test.router()->GetRtpPayloadStates(); |
| EXPECT_EQ(2u, initial_states.size()); |
| EXPECT_NE(initial_states.find(kSsrc1), initial_states.end()); |
| EXPECT_NE(initial_states.find(kSsrc2), initial_states.end()); |
| } |
| |
| TEST(RtpVideoSenderTest, CreateWithPreviousStates) { |
| const int64_t kState1SharedFrameId = 123; |
| const int64_t kState2SharedFrameId = 234; |
| RtpPayloadState state1; |
| state1.picture_id = kInitialPictureId1; |
| state1.tl0_pic_idx = kInitialTl0PicIdx1; |
| state1.shared_frame_id = kState1SharedFrameId; |
| RtpPayloadState state2; |
| state2.picture_id = kInitialPictureId2; |
| state2.tl0_pic_idx = kInitialTl0PicIdx2; |
| state2.shared_frame_id = kState2SharedFrameId; |
| std::map<uint32_t, RtpPayloadState> states = {{kSsrc1, state1}, |
| {kSsrc2, state2}}; |
| |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, states); |
| test.router()->SetActive(true); |
| |
| std::map<uint32_t, RtpPayloadState> initial_states = |
| test.router()->GetRtpPayloadStates(); |
| EXPECT_EQ(2u, initial_states.size()); |
| EXPECT_EQ(kInitialPictureId1, initial_states[kSsrc1].picture_id); |
| EXPECT_EQ(kInitialTl0PicIdx1, initial_states[kSsrc1].tl0_pic_idx); |
| EXPECT_EQ(kInitialPictureId2, initial_states[kSsrc2].picture_id); |
| EXPECT_EQ(kInitialTl0PicIdx2, initial_states[kSsrc2].tl0_pic_idx); |
| EXPECT_EQ(kState2SharedFrameId, initial_states[kSsrc1].shared_frame_id); |
| EXPECT_EQ(kState2SharedFrameId, initial_states[kSsrc2].shared_frame_id); |
| } |
| |
| TEST(RtpVideoSenderTest, FrameCountCallbacks) { |
| class MockFrameCountObserver : public FrameCountObserver { |
| public: |
| MOCK_METHOD2(FrameCountUpdated, |
| void(const FrameCounts& frame_counts, uint32_t ssrc)); |
| } callback; |
| |
| RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}, |
| &callback); |
| |
| constexpr uint8_t kPayload = 'a'; |
| EncodedImage encoded_image; |
| encoded_image.SetTimestamp(1); |
| encoded_image.capture_time_ms_ = 2; |
| encoded_image._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1)); |
| |
| encoded_image._frameType = VideoFrameType::kVideoFrameKey; |
| |
| // No callbacks when not active. |
| EXPECT_CALL(callback, FrameCountUpdated).Times(0); |
| EXPECT_NE( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| ::testing::Mock::VerifyAndClearExpectations(&callback); |
| |
| test.router()->SetActive(true); |
| |
| FrameCounts frame_counts; |
| EXPECT_CALL(callback, FrameCountUpdated(_, kSsrc1)) |
| .WillOnce(SaveArg<0>(&frame_counts)); |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| EXPECT_EQ(1, frame_counts.key_frames); |
| EXPECT_EQ(0, frame_counts.delta_frames); |
| |
| ::testing::Mock::VerifyAndClearExpectations(&callback); |
| |
| encoded_image._frameType = VideoFrameType::kVideoFrameDelta; |
| EXPECT_CALL(callback, FrameCountUpdated(_, kSsrc1)) |
| .WillOnce(SaveArg<0>(&frame_counts)); |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| EXPECT_EQ(1, frame_counts.key_frames); |
| EXPECT_EQ(1, frame_counts.delta_frames); |
| } |
| |
| // Integration test verifying that ack of packet via TransportFeedback means |
| // that the packet is removed from RtpPacketHistory and won't be retransmitted |
| // again. |
| TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) { |
| const int64_t kTimeoutMs = 500; |
| |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, {}); |
| test.router()->SetActive(true); |
| |
| constexpr uint8_t kPayload = 'a'; |
| EncodedImage encoded_image; |
| encoded_image.SetTimestamp(1); |
| encoded_image.capture_time_ms_ = 2; |
| encoded_image._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1)); |
| |
| // Send two tiny images, mapping to two RTP packets. Capture sequence numbers. |
| rtc::Event event; |
| std::vector<uint16_t> rtp_sequence_numbers; |
| std::vector<uint16_t> transport_sequence_numbers; |
| EXPECT_CALL(test.transport(), SendRtp) |
| .Times(2) |
| .WillRepeatedly( |
| [&event, &rtp_sequence_numbers, &transport_sequence_numbers]( |
| const uint8_t* packet, size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber()); |
| transport_sequence_numbers.push_back(options.packet_id); |
| if (transport_sequence_numbers.size() == 2) { |
| event.Set(); |
| } |
| return true; |
| }); |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| encoded_image.SetTimestamp(2); |
| encoded_image.capture_time_ms_ = 3; |
| EXPECT_EQ( |
| EncodedImageCallback::Result::OK, |
| test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error); |
| |
| test.AdvanceTime(TimeDelta::ms(33)); |
| |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| |
| // Construct a NACK message for requesting retransmission of both packet. |
| rtcp::Nack nack; |
| nack.SetMediaSsrc(kSsrc1); |
| nack.SetPacketIds(rtp_sequence_numbers); |
| rtc::Buffer nack_buffer = nack.Build(); |
| |
| std::vector<uint16_t> retransmitted_rtp_sequence_numbers; |
| EXPECT_CALL(test.transport(), SendRtp) |
| .Times(2) |
| .WillRepeatedly([&event, &retransmitted_rtp_sequence_numbers]( |
| const uint8_t* packet, size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1); |
| // Capture the retransmitted sequence number from the RTX header. |
| rtc::ArrayView<const uint8_t> payload = rtp_packet.payload(); |
| retransmitted_rtp_sequence_numbers.push_back( |
| ByteReader<uint16_t>::ReadBigEndian(payload.data())); |
| if (retransmitted_rtp_sequence_numbers.size() == 2) { |
| event.Set(); |
| } |
| return true; |
| }); |
| test.router()->DeliverRtcp(nack_buffer.data(), nack_buffer.size()); |
| test.AdvanceTime(TimeDelta::ms(33)); |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| |
| // Verify that both packets were retransmitted. |
| EXPECT_EQ(retransmitted_rtp_sequence_numbers, rtp_sequence_numbers); |
| |
| // Simulate transport feedback indicating fist packet received, next packet |
| // lost (not other way around as that would trigger early retransmit). |
| StreamFeedbackObserver::StreamPacketInfo lost_packet_feedback; |
| lost_packet_feedback.rtp_sequence_number = rtp_sequence_numbers[0]; |
| lost_packet_feedback.ssrc = kSsrc1; |
| lost_packet_feedback.received = false; |
| |
| StreamFeedbackObserver::StreamPacketInfo received_packet_feedback; |
| received_packet_feedback.rtp_sequence_number = rtp_sequence_numbers[1]; |
| received_packet_feedback.ssrc = kSsrc1; |
| received_packet_feedback.received = true; |
| |
| test.router()->OnPacketFeedbackVector( |
| {lost_packet_feedback, received_packet_feedback}); |
| |
| // Advance time to make sure retransmission would be allowed and try again. |
| // This time the retransmission should not happen for the first packet since |
| // the history has been notified of the ack and removed the packet. The |
| // second packet, included in the feedback but not marked as received, should |
| // still be retransmitted. |
| test.AdvanceTime(TimeDelta::ms(33)); |
| EXPECT_CALL(test.transport(), SendRtp) |
| .WillOnce([&event, &lost_packet_feedback](const uint8_t* packet, |
| size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1); |
| // Capture the retransmitted sequence number from the RTX header. |
| rtc::ArrayView<const uint8_t> payload = rtp_packet.payload(); |
| EXPECT_EQ(lost_packet_feedback.rtp_sequence_number, |
| ByteReader<uint16_t>::ReadBigEndian(payload.data())); |
| event.Set(); |
| return true; |
| }); |
| test.router()->DeliverRtcp(nack_buffer.data(), nack_buffer.size()); |
| test.AdvanceTime(TimeDelta::ms(33)); |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| } |
| |
| // This tests that we utilize transport wide feedback to retransmit lost |
| // packets. This is tested by dropping all ordirary packets from a "lossy" |
| // stream send along with an secondary untouched stream. The transport wide |
| // feedback packets from the secondary stream allows the sending side to |
| // detect and retreansmit the lost packets from the lossy stream. |
| TEST(RtpVideoSenderTest, RetransmitsOnTransportWideLossInfo) { |
| int rtx_packets; |
| test::Scenario s(test_info_); |
| test::CallClientConfig call_conf; |
| // Keeping the bitrate fixed to avoid RTX due to probing. |
| call_conf.transport.rates.max_rate = DataRate::kbps(300); |
| call_conf.transport.rates.start_rate = DataRate::kbps(300); |
| test::NetworkSimulationConfig net_conf; |
| net_conf.bandwidth = DataRate::kbps(300); |
| auto send_node = s.CreateSimulationNode(net_conf); |
| auto* route = s.CreateRoutes(s.CreateClient("send", call_conf), {send_node}, |
| s.CreateClient("return", call_conf), |
| {s.CreateSimulationNode(net_conf)}); |
| |
| test::VideoStreamConfig lossy_config; |
| lossy_config.source.framerate = 5; |
| auto* lossy = s.CreateVideoStream(route->forward(), lossy_config); |
| // The secondary stream acts a driver for transport feedback messages, |
| // ensuring that lost packets on the lossy stream are retransmitted. |
| s.CreateVideoStream(route->forward(), test::VideoStreamConfig()); |
| |
| send_node->router()->SetFilter([&](const EmulatedIpPacket& packet) { |
| RtpPacket rtp; |
| if (rtp.Parse(packet.data)) { |
| // Drops all regular packets for the lossy stream and counts all RTX |
| // packets. Since no packets are let trough, NACKs can't be triggered |
| // by the receiving side. |
| if (lossy->send()->UsingSsrc(rtp.Ssrc())) { |
| return false; |
| } else if (lossy->send()->UsingRtxSsrc(rtp.Ssrc())) { |
| ++rtx_packets; |
| } |
| } |
| return true; |
| }); |
| |
| // Run for a short duration and reset counters to avoid counting RTX packets |
| // from initial probing. |
| s.RunFor(TimeDelta::seconds(1)); |
| rtx_packets = 0; |
| int decoded_baseline = lossy->receive()->GetStats().frames_decoded; |
| s.RunFor(TimeDelta::seconds(1)); |
| // We expect both that RTX packets were sent and that an appropriate number of |
| // frames were received. This is somewhat redundant but reduces the risk of |
| // false positives in future regressions (e.g. RTX is send due to probing). |
| EXPECT_GE(rtx_packets, 1); |
| int frames_decoded = |
| lossy->receive()->GetStats().frames_decoded - decoded_baseline; |
| EXPECT_EQ(frames_decoded, 5); |
| } |
| |
| // Integration test verifying that retransmissions are sent for packets which |
| // can be detected as lost early, using transport wide feedback. |
| TEST(RtpVideoSenderTest, EarlyRetransmits) { |
| const int64_t kTimeoutMs = 500; |
| |
| RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2}, |
| kPayloadType, {}); |
| test.router()->SetActive(true); |
| |
| const uint8_t kPayload[1] = {'a'}; |
| EncodedImage encoded_image; |
| encoded_image.SetTimestamp(1); |
| encoded_image.capture_time_ms_ = 2; |
| encoded_image._frameType = VideoFrameType::kVideoFrameKey; |
| encoded_image.SetEncodedData( |
| EncodedImageBuffer::Create(kPayload, sizeof(kPayload))); |
| encoded_image.SetSpatialIndex(0); |
| |
| CodecSpecificInfo codec_specific; |
| codec_specific.codecType = VideoCodecType::kVideoCodecGeneric; |
| |
| // Send two tiny images, mapping to single RTP packets. Capture sequence |
| // numbers. |
| rtc::Event event; |
| uint16_t frame1_rtp_sequence_number = 0; |
| uint16_t frame1_transport_sequence_number = 0; |
| EXPECT_CALL(test.transport(), SendRtp) |
| .WillOnce([&event, &frame1_rtp_sequence_number, |
| &frame1_transport_sequence_number]( |
| const uint8_t* packet, size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| frame1_rtp_sequence_number = rtp_packet.SequenceNumber(); |
| frame1_transport_sequence_number = options.packet_id; |
| EXPECT_EQ(rtp_packet.Ssrc(), kSsrc1); |
| event.Set(); |
| return true; |
| }); |
| EXPECT_EQ(test.router() |
| ->OnEncodedImage(encoded_image, &codec_specific, nullptr) |
| .error, |
| EncodedImageCallback::Result::OK); |
| |
| test.AdvanceTime(TimeDelta::ms(33)); |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| |
| uint16_t frame2_rtp_sequence_number = 0; |
| uint16_t frame2_transport_sequence_number = 0; |
| encoded_image.SetSpatialIndex(1); |
| EXPECT_CALL(test.transport(), SendRtp) |
| .WillOnce([&event, &frame2_rtp_sequence_number, |
| &frame2_transport_sequence_number]( |
| const uint8_t* packet, size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| frame2_rtp_sequence_number = rtp_packet.SequenceNumber(); |
| frame2_transport_sequence_number = options.packet_id; |
| EXPECT_EQ(rtp_packet.Ssrc(), kSsrc2); |
| event.Set(); |
| return true; |
| }); |
| EXPECT_EQ(test.router() |
| ->OnEncodedImage(encoded_image, &codec_specific, nullptr) |
| .error, |
| EncodedImageCallback::Result::OK); |
| test.AdvanceTime(TimeDelta::ms(33)); |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| |
| EXPECT_NE(frame1_transport_sequence_number, frame2_transport_sequence_number); |
| |
| // Inject a transport feedback where the packet for the first frame is lost, |
| // expect a retransmission for it. |
| EXPECT_CALL(test.transport(), SendRtp) |
| .WillOnce([&event, &frame1_rtp_sequence_number]( |
| const uint8_t* packet, size_t length, |
| const PacketOptions& options) { |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1); |
| |
| // Retransmitted sequence number from the RTX header should match |
| // the lost packet. |
| rtc::ArrayView<const uint8_t> payload = rtp_packet.payload(); |
| EXPECT_EQ(ByteReader<uint16_t>::ReadBigEndian(payload.data()), |
| frame1_rtp_sequence_number); |
| event.Set(); |
| return true; |
| }); |
| |
| StreamFeedbackObserver::StreamPacketInfo first_packet_feedback; |
| first_packet_feedback.rtp_sequence_number = frame1_rtp_sequence_number; |
| first_packet_feedback.ssrc = kSsrc1; |
| first_packet_feedback.received = false; |
| |
| StreamFeedbackObserver::StreamPacketInfo second_packet_feedback; |
| second_packet_feedback.rtp_sequence_number = frame2_rtp_sequence_number; |
| second_packet_feedback.ssrc = kSsrc2; |
| second_packet_feedback.received = true; |
| |
| test.router()->OnPacketFeedbackVector( |
| {first_packet_feedback, second_packet_feedback}); |
| |
| // Wait for pacer to run and send the RTX packet. |
| test.AdvanceTime(TimeDelta::ms(33)); |
| ASSERT_TRUE(event.Wait(kTimeoutMs)); |
| } |
| |
| TEST(RtpVideoSenderTest, CanSetZeroBitrateWithOverhead) { |
| test::ScopedFieldTrials trials("WebRTC-SendSideBwe-WithOverhead/Enabled/"); |
| RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}); |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::Zero(); |
| update.packet_loss_ratio = 0; |
| update.round_trip_time = TimeDelta::Zero(); |
| |
| test.router()->OnBitrateUpdated(update, /*framerate*/ 0); |
| } |
| |
| TEST(RtpVideoSenderTest, CanSetZeroBitrateWithoutOverhead) { |
| RtpVideoSenderTestFixture test({kSsrc1}, {kRtxSsrc1}, kPayloadType, {}); |
| |
| BitrateAllocationUpdate update; |
| update.target_bitrate = DataRate::Zero(); |
| update.packet_loss_ratio = 0; |
| update.round_trip_time = TimeDelta::Zero(); |
| |
| test.router()->OnBitrateUpdated(update, /*framerate*/ 0); |
| } |
| } // namespace webrtc |