Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 297 insertions, 0 deletions

diff --git a/third_party/libwebrtc/moz-patch-stack/0048.patch b/third_party/libwebrtc/moz-patch-stack/0048.patch
new file mode 100644
index 0000000000..641a244353
--- /dev/null
+++ b/third_party/libwebrtc/moz-patch-stack/0048.patch
@@ -0,0 +1,297 @@
+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
+