path: root/third_party/libwebrtc/moz-patch-stack/0042.patch

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 415ad0640a..1e8cff5441 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 17cf859ed8..b743b550ba 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -105,7 +105,8 @@ 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;
@@ -541,7 +542,8 @@ 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)),
@@ -580,6 +582,7 @@ 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));
@@ -1122,13 +1125,15 @@ 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 a596ef5fd4..9c6ceb2403 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 3ed4e5bb18..a3cfd903f4 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),