Add support for transport wide sequence numbers

webrtc/video_engine/vie_channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 68 matching lines @@
 };
 
 ViEChannel::ViEChannel(int32_t channel_id,
                        int32_t engine_id,
                        uint32_t number_of_cores,
                        const Config& config,
                        Transport* transport,
                        ProcessThread* module_process_thread,
                        RtcpIntraFrameObserver* intra_frame_observer,
                        RtcpBandwidthObserver* bandwidth_observer,
+                       SendTimeObserver* send_time_observer,
                        RemoteBitrateEstimator* remote_bitrate_estimator,
                        RtcpRttStats* rtt_stats,
                        PacedSender* paced_sender,
                        PacketRouter* packet_router,
                        size_t max_rtp_streams,
                        bool sender,
                        bool disable_default_encoder)
     : channel_id_(channel_id),
       engine_id_(engine_id),
       number_of_cores_(number_of_cores),
       sender_(sender),
       module_process_thread_(module_process_thread),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
       send_payload_router_(new PayloadRouter()),
       vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
       vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
                                      nullptr,
                                      nullptr)),
       vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
       vie_sync_(vcm_),
       stats_observer_(new ChannelStatsObserver(this)),
       vcm_receive_stats_callback_(NULL),
       incoming_video_stream_(nullptr),
       codec_observer_(NULL),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
+      send_time_observer_(send_time_observer),
       decoder_reset_(true),
       disable_default_encoder_(disable_default_encoder),
       nack_history_size_sender_(kSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       report_block_stats_sender_(new ReportBlockStats()),
       rtp_rtcp_modules_(
           CreateRtpRtcpModules(ViEModuleId(engine_id_, channel_id_),
                                !sender,
                                vie_receiver_.GetReceiveStatistics(),
                                transport,
                                sender ? intra_frame_observer_ : nullptr,
                                sender ? bandwidth_observer_.get() : nullptr,
+                               sender ? send_time_observer_ : nullptr,
                                rtt_stats_,
                                &rtcp_packet_type_counter_observer_,
                                remote_bitrate_estimator,
                                paced_sender_,
+                               sender_ ? packet_router_ : nullptr,
                                &send_bitrate_observer_,
                                &send_frame_count_observer_,
                                &send_side_delay_observer_,
                                max_rtp_streams)),
       num_active_rtp_rtcp_modules_(1) {
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]);
   vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
 }
 
 int32_t ViEChannel::Init() {
@@ skipping 49 matching lines @@
 }
 
 ViEChannel::~ViEChannel() {
   UpdateHistograms();
   // Make sure we don't get more callbacks from the RTP module.
   module_process_thread_->DeRegisterModule(
       vie_receiver_.GetReceiveStatistics());
   module_process_thread_->DeRegisterModule(vcm_);
   module_process_thread_->DeRegisterModule(&vie_sync_);
   send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
+  if (sender_ && packet_router_) {
+    for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
+      packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+  }
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    packet_router_->RemoveRtpModule(rtp_rtcp);
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
   if (decode_thread_) {
     StopDecodeThread();
   }
   // Release modules.
   VideoCodingModule::Destroy(vcm_);
 }
 
@@ skipping 136 matching lines @@
   }
   // Update the RTP module with the settings.
   // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
   // set explicitly.
   // The first layer is always active, so the first module can be checked for
   // sending status.
   bool is_sending = rtp_rtcp_modules_[0]->Sending();
   bool router_was_active = send_payload_router_->active();
   send_payload_router_->set_active(false);
   send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>());
-  for (RtpRtcp* module : rtp_rtcp_modules_)
-    packet_router_->RemoveRtpModule(module);
 
   std::vector<RtpRtcp*> registered_modules;
   std::vector<RtpRtcp*> deregistered_modules;
   size_t num_active_modules = video_codec.numberOfSimulcastStreams > 0
                                    ? video_codec.numberOfSimulcastStreams
                                    : 1;
   size_t num_prev_active_modules;
   {
     // Cache which modules are active so StartSend can know which ones to start.
     CriticalSectionScoped cs(crit_.get());
@@ skipping 20 matching lines @@
     }
     rtp_rtcp->SetSendingStatus(is_sending);
     rtp_rtcp->SetSendingMediaStatus(is_sending);
   }
 
   // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
   // modules can be deleted after this step.
   vie_receiver_.RegisterRtpRtcpModules(registered_modules);
 
   // Update the packet and payload routers with the sending RtpRtcp modules.
-  std::list<RtpRtcp*> active_send_modules;
-  for (RtpRtcp* rtp_rtcp : registered_modules) {
-    packet_router_->AddRtpModule(rtp_rtcp);
-    active_send_modules.push_back(rtp_rtcp);
+  if (sender_) {
+    std::list<RtpRtcp*> active_send_modules;
+    for (RtpRtcp* rtp_rtcp : registered_modules)
+      active_send_modules.push_back(rtp_rtcp);
+    send_payload_router_->SetSendingRtpModules(active_send_modules);
   }
-  send_payload_router_->SetSendingRtpModules(active_send_modules);
 
   if (router_was_active)
     send_payload_router_->set_active(true);
 
   // Deregister previously registered modules.
-  for (size_t i = num_active_modules; i < num_prev_active_modules; ++i)
+  for (size_t i = num_active_modules; i < num_prev_active_modules; ++i) {
     module_process_thread_->DeRegisterModule(rtp_rtcp_modules_[i]);
+    if (sender_ && packet_router_)
+      packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]);
+  }
   // Register new active modules.
-  for (size_t i = num_prev_active_modules; i < num_active_modules; ++i)
+  for (size_t i = num_prev_active_modules; i < num_active_modules; ++i) {
     module_process_thread_->RegisterModule(rtp_rtcp_modules_[i]);
+    if (sender_ && packet_router_)
+      packet_router_->AddRtpModule(rtp_rtcp_modules_[i]);
+  }
   return 0;
 }
 
 int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
   DCHECK(!sender_);
   if (!vie_receiver_.SetReceiveCodec(video_codec)) {
     return -1;
   }
 
   if (video_codec.codecType != kVideoCodecRED &&
@@ skipping 263 matching lines @@
     error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
         kRtpExtensionVideoRotation, id);
   }
   return error;
 }
 
 int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) {
   return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1;
 }
 
+int ViEChannel::SetSendTransportSequenceNumber(bool enable, int id) {
+  // Disable any previous registrations of this extension to avoid errors.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->DeregisterSendRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber);
+  }
+  if (!enable)
+    return 0;
+  // Enable the extension.
+  int error = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    error |= rtp_rtcp->RegisterSendRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber, id);
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) {
+  return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1;
+}
+
 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
   rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable);
 }
 
 void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
   DCHECK(paced_sender_ && "No paced sender registered.");
   paced_sender_->SetStatus(enable);
 }
 
 void ViEChannel::EnableTMMBR(bool enable) {
@@ skipping 426 matching lines @@
   return 0;
 }
 
 std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
     int32_t id,
     bool receiver_only,
     ReceiveStatistics* receive_statistics,
     Transport* outgoing_transport,
     RtcpIntraFrameObserver* intra_frame_callback,
     RtcpBandwidthObserver* bandwidth_callback,
+    SendTimeObserver* send_time_callback,
     RtcpRttStats* rtt_stats,
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
     RemoteBitrateEstimator* remote_bitrate_estimator,
     PacedSender* paced_sender,
+    PacketRouter* packet_router,
     BitrateStatisticsObserver* send_bitrate_observer,
     FrameCountObserver* send_frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
     size_t num_modules) {
   DCHECK_GT(num_modules, 0u);
   RtpRtcp::Configuration configuration;
   ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
   configuration.id = id;
   configuration.audio = false;
   configuration.receiver_only = receiver_only;
   configuration.receive_statistics = receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = intra_frame_callback;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
   configuration.paced_sender = paced_sender;
+  configuration.packet_router = packet_router;
   configuration.send_bitrate_observer = send_bitrate_observer;
   configuration.send_frame_count_observer = send_frame_count_observer;
   configuration.send_side_delay_observer = send_side_delay_observer;
   configuration.bandwidth_callback = bandwidth_callback;
+  configuration.send_time_callback = send_time_callback;
 
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(kRtcpCompound);
     modules.push_back(rtp_rtcp);
     // Receive statistics and remote bitrate estimator should only be set for
     // the primary (first) module.
@@ skipping 84 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   vcm_receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc