firefox-main/third_party/libwebrtc/moz-patch-stack/0040.patch

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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
---
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<voe::ChannelReceiveInterface> 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 832f86aaab..2f8ae83c03 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -133,7 +133,8 @@ class ChannelReceive : public ChannelReceiveInterface,
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
- scoped_refptr<FrameTransformerInterface> frame_transformer);
+ scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer);
~ChannelReceive() override;
void SetSink(AudioSinkInterface* sink) override;
@@ -565,7 +566,8 @@ ChannelReceive::ChannelReceive(
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
- scoped_refptr<FrameTransformerInterface> frame_transformer)
+ scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer)
: env_(env),
worker_thread_(TaskQueueBase::Current()),
rtp_receive_statistics_(ReceiveStatistics::Create(&env_.clock())),
@@ -600,6 +602,7 @@ 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));
@@ -1189,13 +1192,14 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const CryptoOptions& crypto_options,
- scoped_refptr<FrameTransformerInterface> frame_transformer) {
+ scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer) {
return std::make_unique<ChannelReceive>(
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<ChannelReceiveInterface> CreateChannelReceive(
std::optional<AudioCodecPairId> codec_pair_id,
scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- scoped_refptr<FrameTransformerInterface> frame_transformer);
+ 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 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 3bfb35297f..20c91982c1 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -169,6 +169,8 @@ 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 <vector>
#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<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 aae48bc268..c66505656d 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<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 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 9aa6b79551..5fb1f3114c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -129,7 +129,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> 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<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;
auto rtp_rtcp = std::make_unique<ModuleRtpRtcpImpl2>(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),