commit 90738ddb4eb05fae8ef036d96dc75b6412b63924
author Tommi <tommi@webrtc.org> Mon May 31 15:36:47 2021
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Mon May 31 17:10:23 2021
tree 431bfc88e03fa5223ac396dc3357c9fc9989c9ea
parent 27d2be3583fefa258903f04084c86a192f4ced12

Split VideoReceiveStream2 init into worker / network steps.

This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.

Inject Call* instead of worker_thread(), which will simplify upcoming
work that needs to access the network_thread() as well.

This is related to:
https://webrtc-review.googlesource.com/c/src/+/220608
https://webrtc-review.googlesource.com/c/src/+/220609

Bug: webrtc:11993
Change-Id: I72769fd61de84967d9a645750c40d01660a2716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220764
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34172}

audio/audio_receive_stream.cc

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index ef69e73..aecc246 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -126,7 +126,7 @@
   RTC_DCHECK(audio_state_);
   RTC_DCHECK(channel_receive_);
 
-  network_thread_checker_.Detach();
+  packet_sequence_checker_.Detach();
 
   RTC_DCHECK(packet_router);
   // Configure bandwidth estimation.
@@ -157,14 +157,14 @@
 
 void AudioReceiveStream::RegisterWithTransport(
     RtpStreamReceiverControllerInterface* receiver_controller) {
-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   RTC_DCHECK(!rtp_stream_receiver_);
   rtp_stream_receiver_ = receiver_controller->CreateReceiver(
       config_.rtp.remote_ssrc, channel_receive_.get());
 }
 
 void AudioReceiveStream::UnregisterFromTransport() {
-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   rtp_stream_receiver_.reset();
 }
 
@@ -395,7 +395,7 @@
 }
 
 void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   channel_receive_->SetAssociatedSendChannel(
       send_stream ? send_stream->GetChannel() : nullptr);
   associated_send_stream_ = send_stream;
@@ -416,7 +416,7 @@
 
 const AudioSendStream* AudioReceiveStream::GetAssociatedSendStreamForTesting()
     const {
-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
   return associated_send_stream_;
 }
 

audio/audio_receive_stream.h

diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 40749cc..87c82cc 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -125,18 +125,18 @@
   // thread, but still serves as a mechanism of grouping together concepts
   // that belong to the network thread. Once the packets are fully delivered
   // on the network thread, this comment will be deleted.
-  RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
   webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   SourceTracker source_tracker_;
   const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
   AudioSendStream* associated_send_stream_
-      RTC_GUARDED_BY(network_thread_checker_) = nullptr;
+      RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
 
   bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
 
   std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_
-      RTC_GUARDED_BY(network_thread_checker_);
+      RTC_GUARDED_BY(packet_sequence_checker_);
 };
 }  // namespace internal
 }  // namespace webrtc