Adds ability to delay pacer start until media is added.

This avoids the pacer thread waking up at 5ms interval if a
PeerConnection is created without actually using media.

The TaskQueuePacedSender solves the problem too, this CL is mostly a
safeguard in case we still find issues when turning it on...

Can be turned off by setting field trial "WebRTC-LazyPacerStart" to
"Disabled".

Bug: webrtc:10809
Change-Id: I8501106e608eccb14487576f24bdceaf3f324d80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183982
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32101}

call/call.cc

@@ -306,7 +306,9 @@ class Call final : public webrtc::Call,
   void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
-  void RegisterRateObserver();
+  // Ensure that necessary process threads are started, and any required
+  // callbacks have been registered.
+  void EnsureStarted() RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
 
   rtc::TaskQueue* send_transport_queue() const {
     return transport_send_ptr_->GetWorkerQueue();
@@ -433,8 +435,7 @@ class Call final : public webrtc::Call,
   // last ensures that it is destroyed first and any running tasks are finished.
   std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
 
-  bool is_target_rate_observer_registered_ RTC_GUARDED_BY(worker_thread_) =
+  bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;
-      false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
@@ -655,19 +656,18 @@ Call::~Call() {
   UpdateHistograms();
 }
 
-void Call::RegisterRateObserver() {
+void Call::EnsureStarted() {
-  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (is_started_) {
-
-  if (is_target_rate_observer_registered_)
     return;
-
+  }
-  is_target_rate_observer_registered_ = true;
+  is_started_ = true;
 
   // This call seems to kick off a number of things, so probably better left
   // off being kicked off on request rather than in the ctor.
   transport_send_ptr_->RegisterTargetTransferRateObserver(this);
 
   module_process_thread_->EnsureStarted();
+  transport_send_ptr_->EnsureStarted();
 }
 
 void Call::SetClientBitratePreferences(const BitrateSettings& preferences) {
@@ -762,7 +762,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK_RUN_ON(worker_thread_);
 
-  RegisterRateObserver();
+  EnsureStarted();
 
   // Stream config is logged in AudioSendStream::ConfigureStream, as it may
   // change during the stream's lifetime.
@@ -822,7 +822,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_RUN_ON(worker_thread_);
-  RegisterRateObserver();
+  EnsureStarted();
   event_log_->Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
       CreateRtcLogStreamConfig(config)));
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
@@ -877,7 +877,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK_RUN_ON(worker_thread_);
 
-  RegisterRateObserver();
+  EnsureStarted();
 
   video_send_delay_stats_->AddSsrcs(config);
   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
@@ -976,7 +976,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
   receive_side_cc_.SetSendPeriodicFeedback(
       SendPeriodicFeedback(configuration.rtp.extensions));
 
-  RegisterRateObserver();
+  EnsureStarted();
 
   TaskQueueBase* current = GetCurrentTaskQueueOrThread();
   RTC_CHECK(current);

call/rtp_transport_controller_send.cc

@@ -82,6 +82,7 @@ RtpTransportControllerSend::RtpTransportControllerSend(
     : clock_(clock),
       event_log_(event_log),
       bitrate_configurator_(bitrate_config),
+      process_thread_started_(false),
       process_thread_(std::move(process_thread)),
       use_task_queue_pacer_(IsEnabled(trials, "WebRTC-TaskQueuePacer")),
       process_thread_pacer_(use_task_queue_pacer_
@@ -130,15 +131,13 @@ RtpTransportControllerSend::RtpTransportControllerSend(
   pacer()->SetPacingRates(
       DataRate::BitsPerSec(bitrate_config.start_bitrate_bps), DataRate::Zero());
 
-  if (!use_task_queue_pacer_) {
+  if (!absl::StartsWith(trials->Lookup("WebRTC-LazyPacerStart"), "Disabled")) {
-    process_thread_->Start();
+    EnsureStarted();
   }
 }
 
 RtpTransportControllerSend::~RtpTransportControllerSend() {
-  if (!use_task_queue_pacer_) {
+  process_thread_->Stop();
-    process_thread_->Stop();
-  }
 }
 
 RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
@@ -491,6 +490,13 @@ void RtpTransportControllerSend::IncludeOverheadInPacedSender() {
   pacer()->SetIncludeOverhead();
 }
 
+void RtpTransportControllerSend::EnsureStarted() {
+  if (!use_task_queue_pacer_ && !process_thread_started_) {
+    process_thread_started_ = true;
+    process_thread_->Start();
+  }
+}
+
 void RtpTransportControllerSend::OnReceivedEstimatedBitrate(uint32_t bitrate) {
   RemoteBitrateReport msg;
   msg.receive_time = Timestamp::Millis(clock_->TimeInMilliseconds());

call/rtp_transport_controller_send.h

@@ -110,6 +110,7 @@ class RtpTransportControllerSend final
 
   void AccountForAudioPacketsInPacedSender(bool account_for_audio) override;
   void IncludeOverheadInPacedSender() override;
+  void EnsureStarted() override;
 
   // Implements RtcpBandwidthObserver interface
   void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
@@ -151,6 +152,7 @@ class RtpTransportControllerSend final
   std::vector<std::unique_ptr<RtpVideoSenderInterface>> video_rtp_senders_;
   RtpBitrateConfigurator bitrate_configurator_;
   std::map<std::string, rtc::NetworkRoute> network_routes_;
+  bool process_thread_started_;
   const std::unique_ptr<ProcessThread> process_thread_;
   const bool use_task_queue_pacer_;
   std::unique_ptr<PacedSender> process_thread_pacer_;

call/rtp_transport_controller_send_interface.h

@@ -154,6 +154,8 @@ class RtpTransportControllerSendInterface {
 
   virtual void AccountForAudioPacketsInPacedSender(bool account_for_audio) = 0;
   virtual void IncludeOverheadInPacedSender() = 0;
+
+  virtual void EnsureStarted() = 0;
 };
 
 }  // namespace webrtc

call/rtp_video_sender_unittest.cc

@@ -143,6 +143,7 @@ class RtpVideoSenderTestFixture {
                      VideoEncoderConfig::ContentType::kRealtimeVideo),
         retransmission_rate_limiter_(time_controller_.GetClock(),
                                      kRetransmitWindowSizeMs) {
+    transport_controller_.EnsureStarted();
     std::map<uint32_t, RtpState> suspended_ssrcs;
     router_ = std::make_unique<RtpVideoSender>(
         time_controller_.GetClock(), suspended_ssrcs, suspended_payload_states,

call/test/mock_rtp_transport_controller_send.h

@@ -99,6 +99,7 @@ class MockRtpTransportControllerSend
   MOCK_METHOD(void, AccountForAudioPacketsInPacedSender, (bool), (override));
   MOCK_METHOD(void, IncludeOverheadInPacedSender, (), (override));
   MOCK_METHOD(void, OnReceivedPacket, (const ReceivedPacket&), (override));
+  MOCK_METHOD(void, EnsureStarted, (), (override));
 };
 }  // namespace webrtc
 #endif  // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_