diff options
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0042.patch')
| -rw-r--r-- | third_party/libwebrtc/moz-patch-stack/0042.patch | 302 |
1 files changed, 302 insertions, 0 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0042.patch b/third_party/libwebrtc/moz-patch-stack/0042.patch new file mode 100644 index 0000000000..be51a5a8b6 --- /dev/null +++ b/third_party/libwebrtc/moz-patch-stack/0042.patch @@ -0,0 +1,302 @@ +From: "Byron Campen [:bwc]" <docfaraday@gmail.com> +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 | 5 ++++- + audio/channel_receive.cc | 13 +++++++++---- + 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, 57 insertions(+), 10 deletions(-) + +diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc +index 978bbb25b2..655b2761ac 100644 +--- a/audio/audio_receive_stream.cc ++++ b/audio/audio_receive_stream.cc +@@ -39,6 +39,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const { + ss << "{remote_ssrc: " << remote_ssrc; + ss << ", local_ssrc: " << local_ssrc; + ss << ", nack: " << nack.ToString(); ++ ss << ", rtcp_event_observer: " ++ << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr"); + ss << '}'; + return ss.str(); + } +@@ -73,7 +75,8 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive( + config.jitter_buffer_fast_accelerate, 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)); ++ config.crypto_options, std::move(config.frame_transformer), ++ config.rtp.rtcp_event_observer); + } + } // namespace + +diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc +index f5d214f672..c714b1dd4d 100644 +--- a/audio/channel_receive.cc ++++ b/audio/channel_receive.cc +@@ -104,7 +104,8 @@ class ChannelReceive : public ChannelReceiveInterface, + absl::optional<AudioCodecPairId> codec_pair_id, + rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, + const webrtc::CryptoOptions& crypto_options, +- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); ++ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, ++ RtcpEventObserver* rtcp_event_observer); + ~ChannelReceive() override; + + void SetSink(AudioSinkInterface* sink) override; +@@ -538,7 +539,8 @@ ChannelReceive::ChannelReceive( + absl::optional<AudioCodecPairId> codec_pair_id, + rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, + const webrtc::CryptoOptions& crypto_options, +- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) ++ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, ++ RtcpEventObserver* rtcp_event_observer) + : worker_thread_(TaskQueueBase::Current()), + event_log_(rtc_event_log), + rtp_receive_statistics_(ReceiveStatistics::Create(clock)), +@@ -584,6 +586,7 @@ 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)); +@@ -1111,13 +1114,15 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive( + absl::optional<AudioCodecPairId> codec_pair_id, + rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, + const webrtc::CryptoOptions& crypto_options, +- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) { ++ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, ++ RtcpEventObserver* rtcp_event_observer) { + return std::make_unique<ChannelReceive>( + clock, neteq_factory, audio_device_module, rtcp_send_transport, + rtc_event_log, 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_decryptor), crypto_options, std::move(frame_transformer), ++ rtcp_event_observer); + } + + } // namespace voe +diff --git a/audio/channel_receive.h b/audio/channel_receive.h +index ab69103269..5713d97aaa 100644 +--- a/audio/channel_receive.h ++++ b/audio/channel_receive.h +@@ -28,6 +28,7 @@ + #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" + #include "modules/rtp_rtcp/source/source_tracker.h" + #include "system_wrappers/include/clock.h" + +@@ -186,7 +187,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive( + absl::optional<AudioCodecPairId> codec_pair_id, + rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, + const webrtc::CryptoOptions& crypto_options, +- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); ++ rtc::scoped_refptr<FrameTransformerInterface> 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 4879311fdb..88b74b44ac 100644 +--- a/call/audio_receive_stream.h ++++ b/call/audio_receive_stream.h +@@ -19,6 +19,7 @@ + #include "absl/types/optional.h" + #include "api/audio_codecs/audio_decoder_factory.h" + #include "api/call/transport.h" ++#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" + #include "api/crypto/crypto_options.h" + #include "api/rtp_parameters.h" + #include "call/receive_stream.h" +@@ -117,6 +118,8 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface { + + // See NackConfig for description. + NackConfig nack; ++ ++ RtcpEventObserver* rtcp_event_observer = nullptr; + } rtp; + + // Receive-side RTT. +diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc +index 8d88ce23c6..9ee9ed3e76 100644 +--- a/call/video_receive_stream.cc ++++ b/call/video_receive_stream.cc +@@ -161,6 +161,8 @@ 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 a1fc204e7c..01ac7e0ba4 100644 +--- a/call/video_receive_stream.h ++++ b/call/video_receive_stream.h +@@ -20,6 +20,7 @@ + #include <vector> + + #include "api/call/transport.h" ++#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" + #include "api/crypto/crypto_options.h" + #include "api/rtp_headers.h" + #include "api/rtp_parameters.h" +@@ -241,6 +242,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface { + // meta data is expected to be present in generic frame descriptor + // RTP header extension). + std::set<int> 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 249cf835ba..de85abd4ae 100644 +--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h ++++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h +@@ -173,6 +173,14 @@ 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 a98b200c05..e2ad674012 100644 +--- a/modules/rtp_rtcp/source/rtcp_receiver.cc ++++ b/modules/rtp_rtcp/source/rtcp_receiver.cc +@@ -144,6 +144,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config, + 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), +@@ -171,6 +172,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config, + 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), +@@ -778,6 +780,10 @@ 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) { +@@ -1199,12 +1205,20 @@ std::vector<rtcp::TmmbItem> 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 e748b257e8..36e117af55 100644 +--- a/modules/rtp_rtcp/source/rtcp_receiver.h ++++ b/modules/rtp_rtcp/source/rtcp_receiver.h +@@ -362,6 +362,7 @@ 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 0bdd389795..2c56dccd2a 100644 +--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h ++++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h +@@ -74,6 +74,9 @@ 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; + TransportFeedbackObserver* transport_feedback_callback = nullptr; + VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr; +diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc +index d12e833cab..2ea8ce8c62 100644 +--- a/video/rtp_video_stream_receiver2.cc ++++ b/video/rtp_video_stream_receiver2.cc +@@ -83,7 +83,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule( + RtcpCnameCallback* rtcp_cname_callback, + bool non_sender_rtt_measurement, + uint32_t local_ssrc, +- RtcEventLog* rtc_event_log) { ++ RtcEventLog* rtc_event_log, ++ RtcpEventObserver* rtcp_event_observer) { + RtpRtcpInterface::Configuration configuration; + configuration.clock = clock; + configuration.audio = false; +@@ -95,6 +96,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> 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; + configuration.event_log = rtc_event_log; + +@@ -275,7 +277,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2( + rtcp_cname_callback, + config_.rtp.rtcp_xr.receiver_reference_time_report, + config_.rtp.local_ssrc, +- event_log)), ++ event_log, ++ config_.rtp.rtcp_event_observer)), + nack_periodic_processor_(nack_periodic_processor), + complete_frame_callback_(complete_frame_callback), + keyframe_request_method_(config_.rtp.keyframe_method), |