From: "Byron Campen [:bwc]" Date: Fri, 19 Feb 2021 15:56:00 -0600 Subject: Bug 1654112 - Get RTCP BYE and RTP timeout handling working again (from Bug 1595479) r=mjf,dminor Differential Revision: https://phabricator.services.mozilla.com/D106145 Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0b311007c033e83824f5f6996a70ab9e870f31f --- audio/audio_receive_stream.cc | 4 +++- audio/channel_receive.cc | 12 ++++++++---- audio/channel_receive.h | 4 +++- call/audio_receive_stream.h | 3 +++ call/video_receive_stream.cc | 2 ++ call/video_receive_stream.h | 3 +++ modules/rtp_rtcp/include/rtp_rtcp_defines.h | 8 ++++++++ modules/rtp_rtcp/source/rtcp_receiver.cc | 18 ++++++++++++++++-- modules/rtp_rtcp/source/rtcp_receiver.h | 1 + modules/rtp_rtcp/source/rtp_rtcp_interface.h | 3 +++ video/rtp_video_stream_receiver2.cc | 7 +++++-- 11 files changed, 55 insertions(+), 10 deletions(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 1c4d634729..a748c4f398 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -57,6 +57,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const { << (rtcp_mode == RtcpMode::kCompound ? "compound" : (rtcp_mode == RtcpMode::kReducedSize ? "reducedSize" : "off")); + ss << ", rtcp_event_observer: " + << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr"); ss << '}'; return ss.str(); } @@ -90,7 +92,7 @@ std::unique_ptr CreateChannelReceive( config.jitter_buffer_min_delay_ms, config.enable_non_sender_rtt, config.decoder_factory, config.codec_pair_id, std::move(config.frame_decryptor), config.crypto_options, - std::move(config.frame_transformer)); + std::move(config.frame_transformer), config.rtp.rtcp_event_observer); } } // namespace diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index 724fb32ff3..ca1e41b63e 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -133,7 +133,8 @@ class ChannelReceive : public ChannelReceiveInterface, std::optional codec_pair_id, scoped_refptr frame_decryptor, const CryptoOptions& crypto_options, - scoped_refptr frame_transformer); + scoped_refptr frame_transformer, + RtcpEventObserver* rtcp_event_observer); ~ChannelReceive() override; void SetSink(AudioSinkInterface* sink) override; @@ -564,7 +565,8 @@ ChannelReceive::ChannelReceive( std::optional codec_pair_id, scoped_refptr frame_decryptor, const CryptoOptions& crypto_options, - scoped_refptr frame_transformer) + scoped_refptr frame_transformer, + RtcpEventObserver* rtcp_event_observer) : env_(env), worker_thread_(TaskQueueBase::Current()), rtp_receive_statistics_(ReceiveStatistics::Create(&env_.clock())), @@ -599,6 +601,7 @@ ChannelReceive::ChannelReceive( configuration.local_media_ssrc = local_ssrc; configuration.rtcp_packet_type_counter_observer = this; configuration.non_sender_rtt_measurement = enable_non_sender_rtt; + configuration.rtcp_event_observer = rtcp_event_observer; if (frame_transformer) InitFrameTransformerDelegate(std::move(frame_transformer)); @@ -1188,13 +1191,14 @@ std::unique_ptr CreateChannelReceive( std::optional codec_pair_id, scoped_refptr frame_decryptor, const CryptoOptions& crypto_options, - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer, + RtcpEventObserver* rtcp_event_observer) { return std::make_unique( env, neteq_factory, audio_device_module, rtcp_send_transport, local_ssrc, remote_ssrc, jitter_buffer_max_packets, jitter_buffer_fast_playout, jitter_buffer_min_delay_ms, enable_non_sender_rtt, decoder_factory, codec_pair_id, std::move(frame_decryptor), crypto_options, - std::move(frame_transformer)); + std::move(frame_transformer), rtcp_event_observer); } } // namespace voe diff --git a/audio/channel_receive.h b/audio/channel_receive.h index 6179c4f8f0..d2316adb12 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -39,6 +39,7 @@ #include "call/rtp_packet_sink_interface.h" #include "call/syncable.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { @@ -179,7 +180,8 @@ std::unique_ptr CreateChannelReceive( std::optional codec_pair_id, scoped_refptr frame_decryptor, const webrtc::CryptoOptions& crypto_options, - scoped_refptr frame_transformer); + scoped_refptr frame_transformer, + RtcpEventObserver* rtcp_event_observer); } // namespace voe } // namespace webrtc diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 4ae9ba04de..098307f135 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -22,6 +22,7 @@ #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_format.h" #include "api/call/transport.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "api/crypto/crypto_options.h" #include "api/crypto/frame_decryptor_interface.h" #include "api/frame_transformer_interface.h" @@ -130,6 +131,8 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface { // See NackConfig for description. NackConfig nack; RtcpMode rtcp_mode = RtcpMode::kCompound; + + RtcpEventObserver* rtcp_event_observer = nullptr; } rtp; // Receive-side RTT. diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc index c16cd59b4d..94a76388fb 100644 --- a/call/video_receive_stream.cc +++ b/call/video_receive_stream.cc @@ -165,6 +165,8 @@ std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const { ss << pt << ", "; } ss << "}"; + ss << ", rtcp_event_observer: " + << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr"); ss << "}"; return ss.str(); } diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h index c69ec1a674..b07f08eddf 100644 --- a/call/video_receive_stream.h +++ b/call/video_receive_stream.h @@ -22,6 +22,7 @@ #include #include "api/call/transport.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "api/crypto/crypto_options.h" #include "api/crypto/frame_decryptor_interface.h" #include "api/frame_transformer_interface.h" @@ -271,6 +272,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface { // meta data is expected to be present in generic frame descriptor // RTP header extension). std::set raw_payload_types; + + RtcpEventObserver* rtcp_event_observer = nullptr; } rtp; // Transport for outgoing packets (RTCP). diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h index a888e4f3d8..4783206052 100644 --- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h +++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h @@ -180,6 +180,14 @@ class NetworkLinkRtcpObserver { virtual void OnRttUpdate(Timestamp /* receive_time */, TimeDelta /* rtt */) {} }; +class RtcpEventObserver { + public: + virtual void OnRtcpBye() = 0; + virtual void OnRtcpTimeout() = 0; + + virtual ~RtcpEventObserver() {} +}; + // NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType! static constexpr size_t kNumMediaTypes = 5; enum class RtpPacketMediaType : size_t { diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc index c5b3606c44..9e4aadde46 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -166,6 +166,7 @@ RTCPReceiver::RTCPReceiver(const Environment& env, rtp_rtcp_(owner), registered_ssrcs_(false, config), network_link_rtcp_observer_(config.network_link_rtcp_observer), + rtcp_event_observer_(config.rtcp_event_observer), rtcp_intra_frame_observer_(config.intra_frame_callback), rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer), network_state_estimate_observer_(config.network_state_estimate_observer), @@ -196,6 +197,7 @@ RTCPReceiver::RTCPReceiver(const Environment& env, rtp_rtcp_(owner), registered_ssrcs_(true, config), network_link_rtcp_observer_(config.network_link_rtcp_observer), + rtcp_event_observer_(config.rtcp_event_observer), rtcp_intra_frame_observer_(config.intra_frame_callback), rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer), network_state_estimate_observer_(config.network_state_estimate_observer), @@ -809,6 +811,10 @@ bool RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) { return false; } + if (rtcp_event_observer_) { + rtcp_event_observer_->OnRtcpBye(); + } + // Clear our lists. rtts_.erase(bye.sender_ssrc()); EraseIf(received_report_blocks_, [&](const auto& elem) { @@ -1252,12 +1258,20 @@ std::vector RTCPReceiver::TmmbrReceived() { } bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) { - return ResetTimestampIfExpired(now, last_received_rb_, report_interval_); + bool result = ResetTimestampIfExpired(now, last_received_rb_, report_interval_); + if (result && rtcp_event_observer_) { + rtcp_event_observer_->OnRtcpTimeout(); + } + return result; } bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) { - return ResetTimestampIfExpired(now, last_increased_sequence_number_, + bool result = ResetTimestampIfExpired(now, last_increased_sequence_number_, report_interval_); + if (result && rtcp_event_observer_) { + rtcp_event_observer_->OnRtcpTimeout(); + } + return result; } } // namespace webrtc diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h index 8fc8ea4bf6..9b9ddb4987 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver.h +++ b/modules/rtp_rtcp/source/rtcp_receiver.h @@ -371,6 +371,7 @@ class RTCPReceiver final { RegisteredSsrcs registered_ssrcs_; NetworkLinkRtcpObserver* const network_link_rtcp_observer_; + RtcpEventObserver* const rtcp_event_observer_; RtcpIntraFrameObserver* const rtcp_intra_frame_observer_; RtcpLossNotificationObserver* const rtcp_loss_notification_observer_; NetworkStateEstimateObserver* const network_state_estimate_observer_; diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h index 40836198de..d2304e87db 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h +++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h @@ -71,6 +71,9 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface { // bandwidth estimation related message. NetworkLinkRtcpObserver* network_link_rtcp_observer = nullptr; + // Called when we receive a RTCP bye or timeout + RtcpEventObserver* rtcp_event_observer = nullptr; + NetworkStateEstimateObserver* network_state_estimate_observer = nullptr; // DEPRECATED, transport_feedback_callback is no longer invoked by the RTP diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc index 7a06f1a820..8de1772b98 100644 --- a/video/rtp_video_stream_receiver2.cc +++ b/video/rtp_video_stream_receiver2.cc @@ -129,7 +129,8 @@ std::unique_ptr CreateRtpRtcpModule( RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, RtcpCnameCallback* rtcp_cname_callback, bool non_sender_rtt_measurement, - uint32_t local_ssrc) { + uint32_t local_ssrc, + RtcpEventObserver* rtcp_event_observer) { RtpRtcpInterface::Configuration configuration; configuration.audio = false; configuration.receiver_only = true; @@ -140,6 +141,7 @@ std::unique_ptr CreateRtpRtcpModule( rtcp_packet_type_counter_observer; configuration.rtcp_cname_callback = rtcp_cname_callback; configuration.local_media_ssrc = local_ssrc; + configuration.rtcp_event_observer = rtcp_event_observer; configuration.non_sender_rtt_measurement = non_sender_rtt_measurement; auto rtp_rtcp = std::make_unique(env, configuration); @@ -313,7 +315,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2( rtcp_packet_type_counter_observer, rtcp_cname_callback, config_.rtp.rtcp_xr.receiver_reference_time_report, - config_.rtp.local_ssrc)), + config_.rtp.local_ssrc, + config_.rtp.rtcp_event_observer)), nack_periodic_processor_(nack_periodic_processor), complete_frame_callback_(complete_frame_callback), keyframe_request_method_(config_.rtp.keyframe_method),