path: root/third_party/libwebrtc/moz-patch-stack/0048.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                      |  3 ++-
 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, 56 insertions(+), 10 deletions(-)

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 0bb1168384..7063f40186 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -47,6 +47,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
     }
   }
   ss << ']';
+  ss << ", rtcp_event_observer: "
+     << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
   ss << '}';
   return ss.str();
 }
@@ -81,7 +83,8 @@ 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 b95d98c20c..50bc94fe1f 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -102,7 +102,8 @@ 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)),
@@ -586,6 +588,7 @@ 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));
@@ -1119,13 +1122,15 @@ 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 b47a4b5b97..dd3ca1af83 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -186,7 +186,8 @@ 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 1228861c42..6fc93b2d9a 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 87df97cbdd..838dfcf135 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -153,6 +153,8 @@ std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const {
       ss << ", ";
   }
   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 cda8b1f6af..eeb7d14cc3 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -19,6 +19,7 @@
 #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"
@@ -234,6 +235,8 @@ 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 43bba3e57a..882f861d0b 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -211,6 +211,14 @@ class RtcpBandwidthObserver {
   virtual ~RtcpBandwidthObserver() {}
 };
 
+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 68171d1c2a..69d62ead5a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -145,6 +145,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
       rtp_rtcp_(owner),
       registered_ssrcs_(false, config),
       rtcp_bandwidth_observer_(config.bandwidth_callback),
+      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),
@@ -178,6 +179,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
       rtp_rtcp_(owner),
       registered_ssrcs_(true, config),
       rtcp_bandwidth_observer_(config.bandwidth_callback),
+      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),
@@ -848,6 +850,10 @@ void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
     return;
   }
 
+  if (rtcp_event_observer_) {
+    rtcp_event_observer_->OnRtcpBye();
+  }
+
   // Clear our lists.
   rtts_.erase(bye.sender_ssrc());
   EraseIf(received_report_blocks_, [&](const auto& elem) {
@@ -1265,12 +1271,20 @@ 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 6912912cfc..a05a69059a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -385,6 +385,7 @@ class RTCPReceiver final {
   RegisteredSsrcs registered_ssrcs_;
 
   RtcpBandwidthObserver* const rtcp_bandwidth_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 c6854937cb..b988c7805d 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -73,6 +73,9 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
     // stream.
     RtcpBandwidthObserver* bandwidth_callback = 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 eed9770d93..c7b5e7bc7c 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;
 
@@ -276,7 +278,8 @@ 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),
-- 
2.34.1