Add thread accessors to Call.

Classes associated with the Call instance, need access to these threads
and/or awareness, for checking for thread correctness.

Bug: webrtc:11993
Change-Id: I93bcee0657875f211be2ec959b96f818fa9fd8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215584
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33772}

call/call.cc

@@ -292,6 +292,9 @@ class Call final : public webrtc::Call,
 
   const WebRtcKeyValueConfig& trials() const override;
 
+  TaskQueueBase* network_thread() const override;
+  TaskQueueBase* worker_thread() const override;
+
   // Implements PacketReceiver.
   DeliveryStatus DeliverPacket(MediaType media_type,
                                rtc::CopyOnWriteBuffer packet,
@@ -1164,6 +1167,14 @@ const WebRtcKeyValueConfig& Call::trials() const {
   return *config_.trials;
 }
 
+TaskQueueBase* Call::network_thread() const {
+  return network_thread_;
+}
+
+TaskQueueBase* Call::worker_thread() const {
+  return worker_thread_;
+}
+
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   RTC_DCHECK_RUN_ON(network_thread_);
   RTC_DCHECK(media == MediaType::AUDIO || media == MediaType::VIDEO);

call/call.h

@@ -17,6 +17,7 @@
 
 #include "api/adaptation/resource.h"
 #include "api/media_types.h"
+#include "api/task_queue/task_queue_base.h"
 #include "call/audio_receive_stream.h"
 #include "call/audio_send_stream.h"
 #include "call/call_config.h"
@@ -158,6 +159,9 @@ class Call {
 
   virtual const WebRtcKeyValueConfig& trials() const = 0;
 
+  virtual TaskQueueBase* network_thread() const = 0;
+  virtual TaskQueueBase* worker_thread() const = 0;
+
   virtual ~Call() {}
 };
 

call/degraded_call.cc

@@ -270,6 +270,14 @@ const WebRtcKeyValueConfig& DegradedCall::trials() const {
   return call_->trials();
 }
 
+TaskQueueBase* DegradedCall::network_thread() const {
+  return call_->network_thread();
+}
+
+TaskQueueBase* DegradedCall::worker_thread() const {
+  return call_->worker_thread();
+}
+
 void DegradedCall::SignalChannelNetworkState(MediaType media,
                                              NetworkState state) {
   call_->SignalChannelNetworkState(media, state);

call/degraded_call.h

@@ -87,6 +87,9 @@ class DegradedCall : public Call, private PacketReceiver {
 
   const WebRtcKeyValueConfig& trials() const override;
 
+  TaskQueueBase* network_thread() const override;
+  TaskQueueBase* worker_thread() const override;
+
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
   void OnAudioTransportOverheadChanged(
       int transport_overhead_per_packet) override;

media/engine/fake_webrtc_call.cc

@@ -17,6 +17,7 @@
 #include "media/base/rtp_utils.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/gunit.h"
+#include "rtc_base/thread.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
@@ -377,7 +378,13 @@ void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {
 }
 
 FakeCall::FakeCall()
-    : audio_network_state_(webrtc::kNetworkUp),
+    : FakeCall(rtc::Thread::Current(), rtc::Thread::Current()) {}
+
+FakeCall::FakeCall(webrtc::TaskQueueBase* worker_thread,
+                   webrtc::TaskQueueBase* network_thread)
+    : network_thread_(network_thread),
+      worker_thread_(worker_thread),
+      audio_network_state_(webrtc::kNetworkUp),
       video_network_state_(webrtc::kNetworkUp),
       num_created_send_streams_(0),
       num_created_receive_streams_(0) {}
@@ -612,6 +619,14 @@ webrtc::Call::Stats FakeCall::GetStats() const {
   return stats_;
 }
 
+webrtc::TaskQueueBase* FakeCall::network_thread() const {
+  return network_thread_;
+}
+
+webrtc::TaskQueueBase* FakeCall::worker_thread() const {
+  return worker_thread_;
+}
+
 void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
                                          webrtc::NetworkState state) {
   switch (media) {

media/engine/fake_webrtc_call.h

@@ -282,6 +282,8 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
  public:
   FakeCall();
+  FakeCall(webrtc::TaskQueueBase* worker_thread,
+           webrtc::TaskQueueBase* network_thread);
   ~FakeCall() override;
 
   webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
@@ -364,12 +366,18 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
     return trials_;
   }
 
+  webrtc::TaskQueueBase* network_thread() const override;
+  webrtc::TaskQueueBase* worker_thread() const override;
+
   void SignalChannelNetworkState(webrtc::MediaType media,
                                  webrtc::NetworkState state) override;
   void OnAudioTransportOverheadChanged(
       int transport_overhead_per_packet) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
+  webrtc::TaskQueueBase* const network_thread_;
+  webrtc::TaskQueueBase* const worker_thread_;
+
   ::testing::NiceMock<webrtc::MockRtpTransportControllerSend>
       transport_controller_send_;
 

pc/channel_manager_unittest.cc

@@ -61,7 +61,7 @@ class ChannelManagerTest : public ::testing::Test {
                                             false,
                                             worker_,
                                             network_.get())),
-        fake_call_() {
+        fake_call_(worker_, network_.get()) {
     network_->SetName("Network", this);
     network_->Start();
   }

pc/rtp_sender_receiver_unittest.cc

@@ -107,7 +107,7 @@ class RtpSenderReceiverTest
         // Create fake media engine/etc. so we can create channels to use to
         // test RtpSenders/RtpReceivers.
         media_engine_(new cricket::FakeMediaEngine()),
-        fake_call_(),
+        fake_call_(worker_thread_, network_thread_),
         local_stream_(MediaStream::Create(kStreamId1)) {
     worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
       channel_manager_ = cricket::ChannelManager::Create(