commit b27578801673769f8a1393c2bd11b8c2fd7fb850
author Piotr (Peter) Slatala <psla@webrtc.org> Tue Dec 18 19:17:09 2018
committer Commit Bot <commit-bot@chromium.org> Wed Dec 19 21:19:01 2018
tree e500daf543e305eb3b17aaaaf461002ef7e9bdb1
parent 322bba7b7db2c47b509e0c4db33f6069e2eb3547

Register stat callbacks after rate observer is registered.

Currently the stats callback is registered too early.
For now we ignore media transport for these callbacks (it was ignored
already), and we will introduce changes to media transport in the
future.

Bug: webrtc:9719
Bug: chromium:906998
Bug: chromium:906533
Change-Id: I24c0265d46ec2eb35743de6cd96a11d8c41fefbe
Reviewed-on: https://webrtc-review.googlesource.com/c/114904
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26062}

call/call.cc

diff --git a/call/call.cc b/call/call.cc
index 5811f5c..0bd0a5c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -455,15 +455,6 @@
   RTC_DCHECK(config.event_log != nullptr);
   transport_send_ = std::move(transport_send);
   transport_send_ptr_ = transport_send_.get();
-
-  call_stats_->RegisterStatsObserver(&receive_side_cc_);
-  call_stats_->RegisterStatsObserver(transport_send_->GetCallStatsObserver());
-
-  module_process_thread_->RegisterModule(
-      receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
-  module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
-  module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
-  module_process_thread_->Start();
 }
 
 Call::~Call() {
@@ -475,13 +466,16 @@
   RTC_CHECK(audio_receive_streams_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
-  module_process_thread_->DeRegisterModule(
-      receive_side_cc_.GetRemoteBitrateEstimator(true));
-  module_process_thread_->DeRegisterModule(&receive_side_cc_);
-  module_process_thread_->DeRegisterModule(call_stats_.get());
-  module_process_thread_->Stop();
-  call_stats_->DeregisterStatsObserver(&receive_side_cc_);
-  call_stats_->DeregisterStatsObserver(transport_send_->GetCallStatsObserver());
+  if (!media_transport_) {
+    module_process_thread_->DeRegisterModule(
+        receive_side_cc_.GetRemoteBitrateEstimator(true));
+    module_process_thread_->DeRegisterModule(&receive_side_cc_);
+    module_process_thread_->DeRegisterModule(call_stats_.get());
+    module_process_thread_->Stop();
+    call_stats_->DeregisterStatsObserver(&receive_side_cc_);
+    call_stats_->DeregisterStatsObserver(
+        transport_send_->GetCallStatsObserver());
+  }
 
   int64_t first_sent_packet_ms = transport_send_->GetFirstPacketTimeMs();
   // Only update histograms after process threads have been shut down, so that
@@ -504,9 +498,21 @@
   is_target_rate_observer_registered_ = true;
 
   if (media_transport_) {
+    // TODO(bugs.webrtc.org/9719): We should report call_stats_ from
+    // media transport (at least Rtt). We should extend media transport
+    // interface to include "receive_side bwe" if needed.
     media_transport_->AddTargetTransferRateObserver(this);
   } else {
     transport_send_ptr_->RegisterTargetTransferRateObserver(this);
+
+    call_stats_->RegisterStatsObserver(&receive_side_cc_);
+    call_stats_->RegisterStatsObserver(transport_send_->GetCallStatsObserver());
+
+    module_process_thread_->RegisterModule(
+        receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
+    module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
+    module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
+    module_process_thread_->Start();
   }
 }
 
@@ -695,6 +701,7 @@
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+  RegisterRateObserver();
   event_log_->Log(absl::make_unique<RtcEventAudioReceiveStreamConfig>(
       CreateRtcLogStreamConfig(config)));
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
@@ -849,6 +856,8 @@
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
 
+  RegisterRateObserver();
+
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       &video_receiver_controller_, num_cpu_cores_,
       transport_send_ptr_->packet_router(), std::move(configuration),