diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h index fcad876a4a..25be175cd7 100644 --- a/modules/rtp_rtcp/include/rtp_rtcp.h +++ b/modules/rtp_rtcp/include/rtp_rtcp.h @@ -28,7 +28,6 @@ #include "modules/rtp_rtcp/include/rtp_packet_pacer.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" -#include "modules/rtp_rtcp/source/rtp_sender.h" #include "rtc_base/constructor_magic.h" #include "rtc_base/deprecation.h" @@ -41,6 +40,7 @@ class RateLimiter; class ReceiveStatisticsProvider; class RemoteBitrateEstimator; class RtcEventLog; +class RTPSender; class Transport; class VideoBitrateAllocationObserver; @@ -52,6 +52,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { public: struct Configuration { Configuration(); + Configuration(Configuration&& rhs); // True for a audio version of the RTP/RTCP module object false will create // a video version. @@ -120,6 +121,11 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { // defaults to webrtc::FieldTrialBasedConfig. const WebRtcKeyValueConfig* field_trials = nullptr; + // SSRCs for sending media and retransmission, respectively. + // FlexFec SSRC is fetched from |flexfec_sender|. + absl::optional media_send_ssrc; + absl::optional rtx_send_ssrc; + private: RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); }; @@ -193,6 +199,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { uint32_t SSRC() const override = 0; // Sets SSRC, default is a random number. + // TODO(bugs.webrtc.org/10774): Remove. virtual void SetSSRC(uint32_t ssrc) = 0; // Sets the value for sending in the RID (and Repaired) RTP header extension. @@ -220,6 +227,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, // only the SSRC is set. + // TODO(bugs.webrtc.org/10774): Remove. virtual void SetRtxSsrc(uint32_t ssrc) = 0; // Sets the payload type to use when sending RTX packets. Note that this diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index f8a0d709ae..21b85a19d5 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -40,6 +40,7 @@ constexpr int32_t kDefaultAudioReportInterval = 5000; } // namespace RtpRtcp::Configuration::Configuration() = default; +RtpRtcp::Configuration::Configuration(Configuration&& rhs) = default; std::unique_ptr RtpRtcp::Create(const Configuration& configuration) { RTC_DCHECK(configuration.clock); @@ -94,27 +95,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) ack_observer_(configuration.ack_observer), rtt_stats_(configuration.rtt_stats), rtt_ms_(0) { - FieldTrialBasedConfig default_trials; if (!configuration.receiver_only) { - rtp_sender_.reset(new RTPSender( - configuration.audio, configuration.clock, - configuration.outgoing_transport, configuration.paced_sender, - configuration.flexfec_sender - ? absl::make_optional(configuration.flexfec_sender->ssrc()) - : absl::nullopt, - configuration.transport_sequence_number_allocator, - configuration.transport_feedback_callback, - configuration.send_bitrate_observer, - configuration.send_side_delay_observer, configuration.event_log, - configuration.send_packet_observer, - configuration.retransmission_rate_limiter, - configuration.overhead_observer, - configuration.populate_network2_timestamp, - configuration.frame_encryptor, configuration.require_frame_encryption, - configuration.extmap_allow_mixed, - configuration.field_trials ? *configuration.field_trials - : default_trials)); - + rtp_sender_.reset(new RTPSender(configuration)); // Make sure rtcp sender use same timestamp offset as rtp sender. rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset()); } diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index 75b8a3dccf..a932fab24f 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/modules/rtp_rtcp/source/rtp_sender.cc @@ -18,6 +18,7 @@ #include "absl/memory/memory.h" #include "absl/strings/match.h" #include "api/array_view.h" +#include "api/transport/field_trial_based_config.h" #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" #include "logging/rtc_event_log/rtc_event_log.h" #include "modules/rtp_rtcp/include/rtp_cvo.h" @@ -124,8 +125,96 @@ RtpPacketSender::Priority PacketTypeToPriority(RtpPacketToSend::Type type) { return RtpPacketSender::Priority::kLowPriority; } +bool IsEnabled(absl::string_view name, + const WebRtcKeyValueConfig* field_trials) { + FieldTrialBasedConfig default_trials; + auto& trials = field_trials ? *field_trials : default_trials; + return trials.Lookup(name).find("Enabled") == 0; +} + +bool IsDisabled(absl::string_view name, + const WebRtcKeyValueConfig* field_trials) { + FieldTrialBasedConfig default_trials; + auto& trials = field_trials ? *field_trials : default_trials; + return trials.Lookup(name).find("Disabled") == 0; +} + } // namespace +RTPSender::RTPSender(const RtpRtcp::Configuration& config) + : clock_(config.clock), + random_(clock_->TimeInMicroseconds()), + audio_configured_(config.audio), + flexfec_ssrc_(config.flexfec_sender + ? absl::make_optional(config.flexfec_sender->ssrc()) + : absl::nullopt), + paced_sender_(config.paced_sender), + transport_sequence_number_allocator_( + config.transport_sequence_number_allocator), + transport_feedback_observer_(config.transport_feedback_callback), + transport_(config.outgoing_transport), + sending_media_(true), // Default to sending media. + force_part_of_allocation_(false), + max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP. + last_payload_type_(-1), + rtp_header_extension_map_(config.extmap_allow_mixed), + packet_history_(clock_), + flexfec_packet_history_(clock_), + // Statistics + send_delays_(), + max_delay_it_(send_delays_.end()), + sum_delays_ms_(0), + total_packet_send_delay_ms_(0), + rtp_stats_callback_(nullptr), + total_bitrate_sent_(kBitrateStatisticsWindowMs, + RateStatistics::kBpsScale), + nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale), + send_side_delay_observer_(config.send_side_delay_observer), + event_log_(config.event_log), + send_packet_observer_(config.send_packet_observer), + bitrate_callback_(config.send_bitrate_observer), + // RTP variables + sequence_number_forced_(false), + ssrc_(config.media_send_ssrc), + last_rtp_timestamp_(0), + capture_time_ms_(0), + last_timestamp_time_ms_(0), + media_has_been_sent_(false), + last_packet_marker_bit_(false), + csrcs_(), + rtx_(kRtxOff), + ssrc_rtx_(config.rtx_send_ssrc), + rtp_overhead_bytes_per_packet_(0), + retransmission_rate_limiter_(config.retransmission_rate_limiter), + overhead_observer_(config.overhead_observer), + populate_network2_timestamp_(config.populate_network2_timestamp), + send_side_bwe_with_overhead_( + IsEnabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)), + legacy_packet_history_storage_mode_( + IsEnabled("WebRTC-UseRtpPacketHistoryLegacyStorageMode", + config.field_trials)), + payload_padding_prefer_useful_packets_( + !IsDisabled("WebRTC-PayloadPadding-UseMostUsefulPacket", + config.field_trials)) { + // This random initialization is not intended to be cryptographic strong. + timestamp_offset_ = random_.Rand(); + // Random start, 16 bits. Can't be 0. + sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber); + sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber); + + // Store FlexFEC packets in the packet history data structure, so they can + // be found when paced. + if (flexfec_ssrc_) { + RtpPacketHistory::StorageMode storage_mode = + legacy_packet_history_storage_mode_ + ? RtpPacketHistory::StorageMode::kStore + : RtpPacketHistory::StorageMode::kStoreAndCull; + + flexfec_packet_history_.SetStorePacketsStatus( + storage_mode, kMinFlexfecPacketsToStoreForPacing); + } +} + RTPSender::RTPSender( bool audio, Clock* clock, diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h index c931a15c9a..c191694de5 100644 --- a/modules/rtp_rtcp/source/rtp_sender.h +++ b/modules/rtp_rtcp/source/rtp_sender.h @@ -25,6 +25,7 @@ #include "modules/rtp_rtcp/include/flexfec_sender.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/include/rtp_packet_pacer.h" +#include "modules/rtp_rtcp/include/rtp_rtcp.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_history.h" #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" @@ -45,6 +46,9 @@ class RtpPacketToSend; class RTPSender { public: + explicit RTPSender(const RtpRtcp::Configuration& config); + + // TODO(bugs.webrtc.org/10774): Remove once downstream projects are fixed. RTPSender(bool audio, Clock* clock, Transport* transport, @@ -83,6 +87,7 @@ class RTPSender { uint32_t TimestampOffset() const; void SetTimestampOffset(uint32_t timestamp); + // TODO(bugs.webrtc.org/10774): Remove. void SetSSRC(uint32_t ssrc); void SetRid(const std::string& rid); @@ -129,8 +134,9 @@ class RTPSender { // RTX. void SetRtxStatus(int mode); int RtxStatus() const; - uint32_t RtxSsrc() const; + + // TODO(bugs.webrtc.org/10774): Remove. void SetRtxSsrc(uint32_t ssrc); void SetRtxPayloadType(int payload_type, int associated_payload_type); diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index d698bd7c09..22289946bf 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -64,26 +64,15 @@ class RtpSenderAudioTest : public ::testing::Test { public: RtpSenderAudioTest() : fake_clock_(kStartTime), - rtp_sender_(true, - &fake_clock_, - &transport_, - nullptr, - absl::nullopt, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - false, - nullptr, - false, - false, - FieldTrialBasedConfig()), + rtp_sender_([&] { + RtpRtcp::Configuration config; + config.audio = true; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + return config; + }()), rtp_sender_audio_(&fake_clock_, &rtp_sender_) { - rtp_sender_.SetSSRC(kSsrc); rtp_sender_.SetSequenceNumber(kSeqNum); } diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc index fd60b73aa3..dad0d74102 100644 --- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -195,6 +195,14 @@ class RtpSenderTest : public ::testing::TestWithParam { mock_rtc_event_log_(), mock_paced_sender_(), retransmission_rate_limiter_(&fake_clock_, 1000), + flexfec_sender_(0, + kFlexFecSsrc, + kSsrc, + "", + std::vector(), + std::vector(), + nullptr, + &fake_clock_), rtp_sender_(), transport_(), kMarkerBit(true), @@ -204,16 +212,21 @@ class RtpSenderTest : public ::testing::TestWithParam { void SetUp() override { SetUpRtpSender(true, false); } void SetUpRtpSender(bool pacer, bool populate_network2) { - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr, - kFlexFecSsrc, &seq_num_allocator_, nullptr, nullptr, nullptr, - &mock_rtc_event_log_, &send_packet_observer_, - &retransmission_rate_limiter_, nullptr, populate_network2, nullptr, - false, false, FieldTrialBasedConfig())); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.rtx_send_ssrc = kRtxSsrc; + config.flexfec_sender = &flexfec_sender_; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + config.paced_sender = pacer ? &mock_paced_sender_ : nullptr; + config.populate_network2_timestamp = populate_network2; + rtp_sender_.reset(new RTPSender(config)); rtp_sender_->SetSequenceNumber(kSeqNum); rtp_sender_->SetTimestampOffset(0); - rtp_sender_->SetSSRC(kSsrc); - rtp_sender_->SetRtxSsrc(kRtxSsrc); } SimulatedClock fake_clock_; @@ -223,6 +236,7 @@ class RtpSenderTest : public ::testing::TestWithParam { StrictMock send_packet_observer_; StrictMock feedback_observer_; RateLimiter retransmission_rate_limiter_; + FlexfecSender flexfec_sender_; std::unique_ptr rtp_sender_; LoopbackTransportTest transport_; const bool kMarkerBit; @@ -345,14 +359,17 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) { TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) { MockTransport transport; - const bool kEnableAudio = true; - rtp_sender_.reset(new RTPSender( - kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, - absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, - nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false, - false, FieldTrialBasedConfig())); + RtpRtcp::Configuration config; + config.audio = true; + config.clock = &fake_clock_; + config.outgoing_transport = &transport; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.event_log = &mock_rtc_event_log_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetTimestampOffset(0); - rtp_sender_->SetSSRC(kSsrc); std::unique_ptr audio_packet = rtp_sender_->AllocatePacket(); // Padding on audio stream allowed regardless of marker in the last packet. @@ -393,13 +410,18 @@ TEST_P(RtpSenderTestWithoutPacer, TransportFeedbackObserverGetsCorrectByteCount) { constexpr int kRtpOverheadBytesPerPacket = 12 + 8; NiceMock mock_overhead_observer; - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, - &mock_rtc_event_log_, nullptr, - &retransmission_rate_limiter_, &mock_overhead_observer, - false, nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.transport_feedback_callback = &feedback_observer_; + config.event_log = &mock_rtc_event_log_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + config.overhead_observer = &mock_overhead_observer; + rtp_sender_ = absl::make_unique(config); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId)); @@ -427,13 +449,17 @@ TEST_P(RtpSenderTestWithoutPacer, } TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) { - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, - &mock_rtc_event_log_, &send_packet_observer_, - &retransmission_rate_limiter_, nullptr, false, nullptr, - false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.transport_feedback_callback = &feedback_observer_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId)); @@ -465,13 +491,16 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) { } TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) { - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, - &mock_rtc_event_log_, &send_packet_observer_, - &retransmission_rate_limiter_, nullptr, false, nullptr, - false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.transport_feedback_callback = &feedback_observer_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); SendGenericPacket(); @@ -521,12 +550,15 @@ TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) { TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) { StrictMock send_side_delay_observer_; - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - nullptr, nullptr, nullptr, &send_side_delay_observer_, - &mock_rtc_event_log_, nullptr, nullptr, nullptr, false, - nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.send_side_delay_observer = &send_side_delay_observer_; + config.event_log = &mock_rtc_event_log_; + rtp_sender_ = absl::make_unique(config); + PlayoutDelayOracle playout_delay_oracle; RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr, &playout_delay_oracle, nullptr, false, false, @@ -608,14 +640,19 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendPacketUpdated) { } TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) { - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt, - &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, - &mock_rtc_event_log_, &send_packet_observer_, - &retransmission_rate_limiter_, nullptr, false, nullptr, false, false, - FieldTrialBasedConfig())); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.transport_feedback_callback = &feedback_observer_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); - rtp_sender_->SetSSRC(kSsrc); rtp_sender_->SetStorePacketsStatus(true, 10); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, @@ -1004,13 +1041,16 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) { } TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) { - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt, - nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr, - nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr, - false, nullptr, false, false, FieldTrialBasedConfig())); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); - rtp_sender_->SetSSRC(kSsrc); EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId)); @@ -1035,13 +1075,17 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) { test::ScopedFieldTrials field_trials( "WebRTC-PayloadPadding-UseMostUsefulPacket/Disabled/"); MockTransport transport; - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt, - nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr, - &retransmission_rate_limiter_, nullptr, false, nullptr, false, false, - FieldTrialBasedConfig())); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.rtx_send_ssrc = kRtxSsrc; + config.event_log = &mock_rtc_event_log_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); - rtp_sender_->SetSSRC(kSsrc); rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload); uint16_t seq_num = kSeqNum; @@ -1054,7 +1098,6 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) { rtp_header_len += 4; // 4 extra bytes common to all extension headers. rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); - rtp_sender_->SetRtxSsrc(1234); const size_t kNumPayloadSizes = 10; const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, @@ -1113,13 +1156,17 @@ TEST_P(RtpSenderTest, SendRedundantPayloadsUsefulPadding) { test::ScopedFieldTrials field_trials( "WebRTC-PayloadPadding-UseMostUsefulPacket/Enabled/"); MockTransport transport; - rtp_sender_ = absl::make_unique( - false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt, - nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr, - &retransmission_rate_limiter_, nullptr, false, nullptr, false, false, - FieldTrialBasedConfig()); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.rtx_send_ssrc = kRtxSsrc; + config.event_log = &mock_rtc_event_log_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); - rtp_sender_->SetSSRC(kSsrc); rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload); uint16_t seq_num = kSeqNum; @@ -1132,7 +1179,6 @@ TEST_P(RtpSenderTest, SendRedundantPayloadsUsefulPadding) { rtp_header_len += 4; // 4 extra bytes common to all extension headers. rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); - rtp_sender_->SetRtxSsrc(1234); const size_t kNumPayloadSizes = 10; const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700, @@ -1259,21 +1305,25 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) { constexpr uint32_t kTimestamp = 1234; constexpr int kMediaPayloadType = 127; constexpr int kFlexfecPayloadType = 118; - constexpr uint32_t kMediaSsrc = 1234; - constexpr uint32_t kFlexfecSsrc = 5678; const std::vector kNoRtpExtensions; const std::vector kNoRtpExtensionSizes; - FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, - kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes, + FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexFecSsrc, kSsrc, kNoMid, + kNoRtpExtensions, kNoRtpExtensionSizes, nullptr /* rtp_state */, &fake_clock_); // Reset |rtp_sender_| to use FlexFEC. - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc, - &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_, - &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false, - nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kMediaSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.flexfec_sender = &flexfec_sender_; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); rtp_sender_->SetStorePacketsStatus(true, 10); @@ -1291,12 +1341,11 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) { params.fec_mask_type = kFecMaskRandom; rtp_sender_video.SetFecParameters(params, params); - EXPECT_CALL(mock_paced_sender_, - InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc, kSeqNum, - _, _, false)); + EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority, + kSsrc, kSeqNum, _, _, false)); uint16_t flexfec_seq_num; EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority, - kFlexfecSsrc, _, _, _, false)) + kFlexFecSsrc, _, _, _, false)) .WillOnce(SaveArg<2>(&flexfec_seq_num)); RTPVideoHeader video_header; @@ -1309,43 +1358,47 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) { LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing))) .Times(2); EXPECT_EQ(RtpPacketSendResult::kSuccess, - rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum, + rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, fake_clock_.TimeInMilliseconds(), false, PacedPacketInfo())); EXPECT_EQ(RtpPacketSendResult::kSuccess, - rtp_sender_->TimeToSendPacket(kFlexfecSsrc, flexfec_seq_num, + rtp_sender_->TimeToSendPacket(kFlexFecSsrc, flexfec_seq_num, fake_clock_.TimeInMilliseconds(), false, PacedPacketInfo())); ASSERT_EQ(2, transport_.packets_sent()); const RtpPacketReceived& media_packet = transport_.sent_packets_[0]; EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType()); EXPECT_EQ(kSeqNum, media_packet.SequenceNumber()); - EXPECT_EQ(kMediaSsrc, media_packet.Ssrc()); + EXPECT_EQ(kSsrc, media_packet.Ssrc()); const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1]; EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType()); EXPECT_EQ(flexfec_seq_num, flexfec_packet.SequenceNumber()); - EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc()); + EXPECT_EQ(kFlexFecSsrc, flexfec_packet.Ssrc()); } TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) { constexpr uint32_t kTimestamp = 1234; constexpr int kMediaPayloadType = 127; constexpr int kFlexfecPayloadType = 118; - constexpr uint32_t kMediaSsrc = 1234; constexpr uint32_t kFlexfecSsrc = 5678; const std::vector kNoRtpExtensions; const std::vector kNoRtpExtensionSizes; - FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, - kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes, + FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kSsrc, kNoMid, + kNoRtpExtensions, kNoRtpExtensionSizes, nullptr /* rtp_state */, &fake_clock_); // Reset |rtp_sender_| to use FlexFEC. - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(), - &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_, - &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false, - nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kMediaSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.flexfec_sender = &flexfec_sender; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); PlayoutDelayOracle playout_delay_oracle; @@ -1374,7 +1427,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) { ASSERT_EQ(2, transport_.packets_sent()); const RtpPacketReceived& media_packet = transport_.sent_packets_[0]; EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType()); - EXPECT_EQ(kMediaSsrc, media_packet.Ssrc()); + EXPECT_EQ(kSsrc, media_packet.Ssrc()); const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1]; EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType()); EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc()); @@ -1434,7 +1487,6 @@ TEST_P(RtpSenderTestWithoutPacer, RidIncludedOnRtxSentPackets) { rtp_sender_->SetSendingMediaStatus(true); rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); - rtp_sender_->SetRtxSsrc(kRtxSsrc); rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload); rtp_sender_->SetStorePacketsStatus(true, 10); @@ -1461,22 +1513,25 @@ TEST_P(RtpSenderTest, FecOverheadRate) { constexpr uint32_t kTimestamp = 1234; constexpr int kMediaPayloadType = 127; constexpr int kFlexfecPayloadType = 118; - constexpr uint32_t kMediaSsrc = 1234; - constexpr uint32_t kFlexfecSsrc = 5678; const std::vector kNoRtpExtensions; const std::vector kNoRtpExtensionSizes; - FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, - kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes, + FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexFecSsrc, kSsrc, kNoMid, + kNoRtpExtensions, kNoRtpExtensionSizes, nullptr /* rtp_state */, &fake_clock_); // Reset |rtp_sender_| to use FlexFEC. - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, &mock_paced_sender_, - flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr, - &mock_rtc_event_log_, &send_packet_observer_, - &retransmission_rate_limiter_, nullptr, false, nullptr, false, false, - FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kMediaSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.paced_sender = &mock_paced_sender_; + config.media_send_ssrc = kSsrc; + config.flexfec_sender = &flexfec_sender; + config.transport_sequence_number_allocator = &seq_num_allocator_; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); + rtp_sender_->SetSequenceNumber(kSeqNum); PlayoutDelayOracle playout_delay_oracle; @@ -1543,12 +1598,14 @@ TEST_P(RtpSenderTest, BitrateCallbacks) { uint32_t total_bitrate_; uint32_t retransmit_bitrate_; } callback; - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - nullptr, nullptr, &callback, nullptr, nullptr, nullptr, - &retransmission_rate_limiter_, nullptr, false, nullptr, - false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.send_bitrate_observer = &callback; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + rtp_sender_ = absl::make_unique(config); PlayoutDelayOracle playout_delay_oracle; RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr, @@ -1712,8 +1769,6 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) { TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) { // XXX const char* kPayloadName = "GENERIC"; const uint8_t kPayloadType = 127; - rtp_sender_->SetSSRC(1234); - rtp_sender_->SetRtxSsrc(kRtxSsrc); rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType); rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); @@ -1783,12 +1838,13 @@ TEST_P(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) { TEST_P(RtpSenderTest, OnOverheadChanged) { MockOverheadObserver mock_overhead_observer; - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, - &retransmission_rate_limiter_, &mock_overhead_observer, - false, nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + config.overhead_observer = &mock_overhead_observer; + rtp_sender_ = absl::make_unique(config); // RTP overhead is 12B. EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1); @@ -1805,12 +1861,13 @@ TEST_P(RtpSenderTest, OnOverheadChanged) { TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) { MockOverheadObserver mock_overhead_observer; - rtp_sender_.reset( - new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt, - nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, - &retransmission_rate_limiter_, &mock_overhead_observer, - false, nullptr, false, false, FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + config.overhead_observer = &mock_overhead_observer; + rtp_sender_ = absl::make_unique(config); EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); SendGenericPacket(); @@ -2026,13 +2083,17 @@ TEST_P(RtpSenderTest, TrySendPacketUpdatesStats) { const size_t kPayloadSize = 1000; StrictMock send_side_delay_observer; - rtp_sender_.reset(new RTPSender( - false, &fake_clock_, &transport_, nullptr, kFlexFecSsrc, nullptr, nullptr, - nullptr, &send_side_delay_observer, &mock_rtc_event_log_, - &send_packet_observer_, nullptr, nullptr, false, nullptr, false, false, - FieldTrialBasedConfig())); - rtp_sender_->SetSSRC(kSsrc); - rtp_sender_->SetRtxSsrc(kRtxSsrc); + + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.media_send_ssrc = kSsrc; + config.rtx_send_ssrc = kRtxSsrc; + config.flexfec_sender = &flexfec_sender_; + config.send_side_delay_observer = &send_side_delay_observer; + config.event_log = &mock_rtc_event_log_; + config.send_packet_observer = &send_packet_observer_; + rtp_sender_ = absl::make_unique(config); ASSERT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( kRtpExtensionTransportSequenceNumber, kTransportSequenceNumberExtensionId)); diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index e6468aaeb1..2589ad2715 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -140,29 +140,18 @@ class RtpSenderVideoTest : public ::testing::TestWithParam { : field_trials_(GetParam()), fake_clock_(kStartTime), retransmission_rate_limiter_(&fake_clock_, 1000), - // TODO(pbos): Set up to use pacer. - rtp_sender_(false, - &fake_clock_, - &transport_, - nullptr, - absl::nullopt, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - nullptr, - &retransmission_rate_limiter_, - nullptr, - false, - nullptr, - false, - false, - field_trials_), + rtp_sender_([&] { + RtpRtcp::Configuration config; + config.clock = &fake_clock_; + config.outgoing_transport = &transport_; + config.retransmission_rate_limiter = &retransmission_rate_limiter_; + config.field_trials = &field_trials_; + config.media_send_ssrc = kSsrc; + return config; + }()), rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr, field_trials_) { rtp_sender_.SetSequenceNumber(kSeqNum); rtp_sender_.SetTimestampOffset(0); - rtp_sender_.SetSSRC(kSsrc); rtp_sender_video_.RegisterPayloadType(kPayload, "generic", /*raw_payload=*/false);