diff --git a/audio/channel_send.cc b/audio/channel_send.cc index d729b9f0ec..876ee69095 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -701,7 +701,7 @@ ChannelSend::ChannelSend(Clock* clock, configuration.extmap_allow_mixed = extmap_allow_mixed; configuration.rtcp_report_interval_ms = rtcp_report_interval_ms; - configuration.media_send_ssrc = ssrc; + configuration.local_media_ssrc = ssrc; _rtpRtcpModule = RtpRtcp::Create(configuration); _rtpRtcpModule->SetSendingMediaStatus(false); diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc index f466cadccc..9ffa515ba3 100644 --- a/call/flexfec_receive_stream_impl.cc +++ b/call/flexfec_receive_stream_impl.cc @@ -131,7 +131,7 @@ std::unique_ptr CreateRtpRtcpModule( configuration.receive_statistics = receive_statistics; configuration.outgoing_transport = config.rtcp_send_transport; configuration.rtt_stats = rtt_stats; - configuration.media_send_ssrc = config.local_ssrc; + configuration.local_media_ssrc = config.local_ssrc; return RtpRtcp::Create(configuration); } diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc index 77f2ba9f0e..7e4a2ad2ec 100644 --- a/call/rtp_video_sender.cc +++ b/call/rtp_video_sender.cc @@ -113,11 +113,11 @@ std::vector CreateRtpStreamSenders( RTC_DCHECK(rtp_config.rtx.ssrcs.empty() || rtp_config.rtx.ssrcs.size() == rtp_config.rtx.ssrcs.size()); for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) { - configuration.media_send_ssrc = rtp_config.ssrcs[i]; + configuration.local_media_ssrc = rtp_config.ssrcs[i]; bool enable_flexfec = flexfec_sender != nullptr && std::find(flexfec_protected_ssrcs.begin(), flexfec_protected_ssrcs.end(), - *configuration.media_send_ssrc) != + *configuration.local_media_ssrc) != flexfec_protected_ssrcs.end(); configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr; auto playout_delay_oracle = absl::make_unique(); diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h index 6a2d91b3d6..5ace64b717 100644 --- a/modules/rtp_rtcp/include/rtp_rtcp.h +++ b/modules/rtp_rtcp/include/rtp_rtcp.h @@ -122,11 +122,18 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { // defaults to webrtc::FieldTrialBasedConfig. const WebRtcKeyValueConfig* field_trials = nullptr; - // SSRCs for sending media and retransmission, respectively. + // SSRCs for media and retransmission, respectively. // FlexFec SSRC is fetched from |flexfec_sender|. + // |media_send_ssrc| has been deprecated, use local_media_ssrc instead. absl::optional media_send_ssrc; + absl::optional local_media_ssrc; absl::optional rtx_send_ssrc; + // TODO(bugs.webrtc.org/10774): Remove this fallback. + absl::optional get_local_media_ssrc() const { + return local_media_ssrc ? local_media_ssrc : media_send_ssrc; + } + private: RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); }; diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 1f5d64ac30..363fa7960e 100644 --- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -135,7 +135,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test { configuration.receive_statistics = receive_statistics_.get(); configuration.outgoing_transport = &transport_; configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; - configuration.media_send_ssrc = kTestSsrc; + configuration.local_media_ssrc = kTestSsrc; rtp_rtcp_module_ = RtpRtcp::Create(configuration); rtp_sender_video_ = absl::make_unique( &fake_clock, rtp_rtcp_module_->RtpSender(), nullptr, diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index 69cb44ff50..20cfb8f24e 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -137,7 +137,8 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config, ? config.rtcp_report_interval_ms : (config.audio ? kDefaultAudioReportInterval : kDefaultVideoReportInterval)), - main_ssrc_(config.media_send_ssrc.value_or(0)), + // TODO(bugs.webrtc.org/10774): Remove fallback. + main_ssrc_(config.get_local_media_ssrc().value_or(0)), remote_ssrc_(0), remote_sender_rtp_time_(0), xr_rrtr_status_(false), @@ -152,8 +153,8 @@ RTCPReceiver::RTCPReceiver(const RtpRtcp::Configuration& config, num_skipped_packets_(0), last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) { RTC_DCHECK(owner); - if (config.media_send_ssrc) { - registered_ssrcs_.insert(*config.media_send_ssrc); + if (config.get_local_media_ssrc()) { + registered_ssrcs_.insert(*config.get_local_media_ssrc()); } if (config.rtx_send_ssrc) { registered_ssrcs_.insert(*config.rtx_send_ssrc); diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index e9c6e2cfec..3eff3e483a 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -152,7 +152,7 @@ class RtcpReceiverTest : public ::testing::Test { config.bitrate_allocation_observer = &bitrate_allocation_observer_; config.rtcp_report_interval_ms = kRtcpIntervalMs; - config.media_send_ssrc = kReceiverMainSsrc; + config.local_media_ssrc = kReceiverMainSsrc; config.rtx_send_ssrc = kReceiverExtraSsrc; return config; }(), diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc index aedca53518..a54b451ba0 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/modules/rtp_rtcp/source/rtcp_sender.cc @@ -130,7 +130,7 @@ RTCPSender::RTCPSender(const RtpRtcp::Configuration& config) timestamp_offset_(0), last_rtp_timestamp_(0), last_frame_capture_time_ms_(-1), - ssrc_(config.media_send_ssrc.value_or(0)), + ssrc_(config.get_local_media_ssrc().value_or(0)), remote_ssrc_(0), receive_statistics_(config.receive_statistics), diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index 09cdff17a2..a077836925 100644 --- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc @@ -92,7 +92,7 @@ class RtcpSenderTest : public ::testing::Test { configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; configuration.rtcp_report_interval_ms = 1000; configuration.receive_statistics = receive_statistics_.get(); - configuration.media_send_ssrc = kSenderSsrc; + configuration.local_media_ssrc = kSenderSsrc; return configuration; } @@ -195,7 +195,7 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) { config.receive_statistics = receive_statistics_.get(); config.outgoing_transport = &test_transport_; config.rtcp_report_interval_ms = 1000; - config.media_send_ssrc = kSenderSsrc; + config.local_media_ssrc = kSenderSsrc; rtcp_sender_.reset(new RTCPSender(config)); rtcp_sender_->SetRemoteSSRC(kRemoteSsrc); rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize); @@ -217,7 +217,7 @@ TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) { config.receive_statistics = receive_statistics_.get(); config.outgoing_transport = &test_transport_; config.rtcp_report_interval_ms = 1000; - config.media_send_ssrc = kSenderSsrc; + config.local_media_ssrc = kSenderSsrc; rtcp_sender_.reset(new RTCPSender(config)); rtcp_sender_->SetRemoteSSRC(kRemoteSsrc); rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound); @@ -695,7 +695,7 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) { config.receive_statistics = receive_statistics_.get(); config.outgoing_transport = &mock_transport; config.rtcp_report_interval_ms = 1000; - config.media_send_ssrc = kSenderSsrc; + config.local_media_ssrc = kSenderSsrc; rtcp_sender_.reset(new RTCPSender(config)); rtcp_sender_->SetRemoteSSRC(kRemoteSsrc); @@ -827,7 +827,7 @@ TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportWhenSsrcSetOnConstruction) { TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportOnFirstSetSsrc) { // Set up without first SSRC not set at construction. RtpRtcp::Configuration configuration = GetDefaultConfig(); - configuration.media_send_ssrc = absl::nullopt; + configuration.local_media_ssrc = absl::nullopt; rtcp_sender_.reset(new RTCPSender(configuration)); rtcp_sender_->SetRemoteSSRC(kRemoteSsrc); diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index f4553e111f..e6f8db130a 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -162,7 +162,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver { config.rtcp_packet_type_counter_observer = this; config.rtt_stats = &rtt_stats_; config.rtcp_report_interval_ms = rtcp_report_interval_ms_; - config.media_send_ssrc = kSenderSsrc; + config.local_media_ssrc = kSenderSsrc; impl_.reset(new ModuleRtpRtcpImpl(config)); impl_->SetRTCPStatus(RtcpMode::kCompound); diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index a29cb2455a..f7ee2634e5 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/modules/rtp_rtcp/source/rtp_sender.cc @@ -175,7 +175,7 @@ RTPSender::RTPSender(const RtpRtcp::Configuration& config) bitrate_callback_(config.send_bitrate_observer), // RTP variables sequence_number_forced_(false), - ssrc_(config.media_send_ssrc), + ssrc_(config.get_local_media_ssrc()), ssrc_has_acked_(false), rtx_ssrc_has_acked_(false), last_rtp_timestamp_(0), diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index dea2a38742..1dad5b71ed 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -70,7 +70,7 @@ class RtpSenderAudioTest : public ::testing::Test { config.audio = true; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; return config; }()), rtp_sender_audio_(&fake_clock_, &rtp_sender_) { diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc index dd36dc29dd..d50528093f 100644 --- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -245,7 +245,7 @@ class RtpSenderTest : public ::testing::TestWithParam { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.rtx_send_ssrc = kRtxSsrc; config.flexfec_sender = &flexfec_sender_; config.transport_sequence_number_allocator = &seq_num_allocator_; @@ -426,7 +426,7 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) { config.clock = &fake_clock_; config.outgoing_transport = &transport; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.event_log = &mock_rtc_event_log_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; rtp_sender_ = absl::make_unique(config); @@ -476,7 +476,7 @@ TEST_P(RtpSenderTestWithoutPacer, RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.transport_sequence_number_allocator = &seq_num_allocator_; config.transport_feedback_callback = &feedback_observer_; config.event_log = &mock_rtc_event_log_; @@ -515,7 +515,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.transport_sequence_number_allocator = &seq_num_allocator_; config.transport_feedback_callback = &feedback_observer_; config.event_log = &mock_rtc_event_log_; @@ -557,7 +557,7 @@ TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.transport_sequence_number_allocator = &seq_num_allocator_; config.transport_feedback_callback = &feedback_observer_; config.event_log = &mock_rtc_event_log_; @@ -617,7 +617,7 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.send_side_delay_observer = &send_side_delay_observer_; config.event_log = &mock_rtc_event_log_; rtp_sender_ = absl::make_unique(config); @@ -707,7 +707,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) { config.clock = &fake_clock_; config.outgoing_transport = &transport_; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.transport_sequence_number_allocator = &seq_num_allocator_; config.transport_feedback_callback = &feedback_observer_; config.event_log = &mock_rtc_event_log_; @@ -1244,7 +1244,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) { config.clock = &fake_clock_; config.outgoing_transport = &transport_; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.send_packet_observer = &send_packet_observer_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; rtp_sender_ = absl::make_unique(config); @@ -1280,7 +1280,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) { config.clock = &fake_clock_; config.outgoing_transport = &transport; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.rtx_send_ssrc = kRtxSsrc; config.event_log = &mock_rtc_event_log_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; @@ -1448,7 +1448,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) { config.clock = &fake_clock_; config.outgoing_transport = &transport_; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.flexfec_sender = &flexfec_sender_; config.transport_sequence_number_allocator = &seq_num_allocator_; config.event_log = &mock_rtc_event_log_; @@ -1562,7 +1562,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) { config.event_log = &mock_rtc_event_log_; config.send_packet_observer = &send_packet_observer_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; rtp_sender_ = absl::make_unique(config); rtp_sender_->SetSequenceNumber(kSeqNum); rtp_sender_->SetStorePacketsStatus(true, 10); @@ -1723,7 +1723,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.flexfec_sender = &flexfec_sender; config.transport_sequence_number_allocator = &seq_num_allocator_; config.event_log = &mock_rtc_event_log_; @@ -1992,7 +1992,7 @@ TEST_P(RtpSenderTest, FecOverheadRate) { config.clock = &fake_clock_; config.outgoing_transport = &transport_; config.paced_sender = &mock_paced_sender_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.flexfec_sender = &flexfec_sender; config.transport_sequence_number_allocator = &seq_num_allocator_; config.event_log = &mock_rtc_event_log_; @@ -2075,7 +2075,7 @@ TEST_P(RtpSenderTest, BitrateCallbacks) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.send_bitrate_observer = &callback; config.retransmission_rate_limiter = &retransmission_rate_limiter_; rtp_sender_ = absl::make_unique(config); @@ -2314,7 +2314,7 @@ TEST_P(RtpSenderTest, OnOverheadChanged) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.retransmission_rate_limiter = &retransmission_rate_limiter_; config.overhead_observer = &mock_overhead_observer; rtp_sender_ = absl::make_unique(config); @@ -2337,7 +2337,7 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.retransmission_rate_limiter = &retransmission_rate_limiter_; config.overhead_observer = &mock_overhead_observer; rtp_sender_ = absl::make_unique(config); @@ -2560,7 +2560,7 @@ TEST_P(RtpSenderTest, TrySendPacketUpdatesStats) { RtpRtcp::Configuration config; config.clock = &fake_clock_; config.outgoing_transport = &transport_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; config.rtx_send_ssrc = kRtxSsrc; config.flexfec_sender = &flexfec_sender_; config.send_side_delay_observer = &send_side_delay_observer; diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index f19c110b68..54210c73e7 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -147,7 +147,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam { config.outgoing_transport = &transport_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; config.field_trials = &field_trials_; - config.media_send_ssrc = kSsrc; + config.local_media_ssrc = kSsrc; return config; }()), rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr, field_trials_) { diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc index e9b4131d9b..64c50d0d1f 100644 --- a/video/end_to_end_tests/bandwidth_tests.cc +++ b/video/end_to_end_tests/bandwidth_tests.cc @@ -201,7 +201,7 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) { config.clock = clock_; config.outgoing_transport = receive_transport_; config.retransmission_rate_limiter = &retransmission_rate_limiter_; - config.media_send_ssrc = (*receive_configs)[0].rtp.local_ssrc; + config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc; rtp_rtcp_ = RtpRtcp::Create(config); rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc); rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize); diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc index 696aa2c7a2..6f478f8c22 100644 --- a/video/rtp_video_stream_receiver.cc +++ b/video/rtp_video_stream_receiver.cc @@ -67,7 +67,7 @@ std::unique_ptr CreateRtpRtcpModule( configuration.rtt_stats = rtt_stats; configuration.rtcp_packet_type_counter_observer = rtcp_packet_type_counter_observer; - configuration.media_send_ssrc = local_ssrc; + configuration.local_media_ssrc = local_ssrc; std::unique_ptr rtp_rtcp = RtpRtcp::Create(configuration); rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);