ModuleRtcRtcpImpl2: remove Module inheritance.upstream-master

This change achieves an Idle Wakeup savings of 200 Hz.

ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.

Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}

1 files changed, 4 insertions, 19 deletions

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 47afc7982b..52dd528504 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -69,7 +69,6 @@ class ChannelSend : public ChannelSendInterface,
 
   ChannelSend(Clock* clock,
               TaskQueueFactory* task_queue_factory,
-              ProcessThread* module_process_thread,
               Transport* rtp_transport,
               RtcpRttStats* rtcp_rtt_stats,
               RtcEventLog* rtc_event_log,
@@ -180,7 +179,6 @@ class ChannelSend : public ChannelSendInterface,
   // voe::Channel into parts with single-threaded semantics, and thereby reduce
   // the need for locks.
   SequenceChecker worker_thread_checker_;
-  SequenceChecker module_process_thread_checker_;
   // Methods accessed from audio and video threads are checked for sequential-
   // only access. We don't necessarily own and control these threads, so thread
   // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
@@ -200,7 +198,6 @@ class ChannelSend : public ChannelSendInterface,
   uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
 
   // uses
-  ProcessThread* const _moduleProcessThreadPtr;
   RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
   bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_);
   bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
@@ -445,7 +442,6 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
 ChannelSend::ChannelSend(
     Clock* clock,
     TaskQueueFactory* task_queue_factory,
-    ProcessThread* module_process_thread,
     Transport* rtp_transport,
     RtcpRttStats* rtcp_rtt_stats,
     RtcEventLog* rtc_event_log,
@@ -459,7 +455,6 @@ ChannelSend::ChannelSend(
     : event_log_(rtc_event_log),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
-      _moduleProcessThreadPtr(module_process_thread),
       input_mute_(false),
       previous_frame_muted_(false),
       _includeAudioLevelIndication(false),
@@ -475,9 +470,6 @@ ChannelSend::ChannelSend(
           TaskQueueFactory::Priority::NORMAL)),
       fixing_timestamp_stall_(
           !field_trial::IsDisabled("WebRTC-Audio-FixTimestampStall")) {
-  RTC_DCHECK(module_process_thread);
-  module_process_thread_checker_.Detach();
-
   audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
 
   RtpRtcpInterface::Configuration configuration;
@@ -504,8 +496,6 @@ ChannelSend::ChannelSend(
   rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock,
                                                        rtp_rtcp_->RtpSender());
 
-  _moduleProcessThreadPtr->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
-
   // Ensure that RTCP is enabled by default for the created channel.
   rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
 
@@ -525,9 +515,6 @@ ChannelSend::~ChannelSend() {
   StopSend();
   int error = audio_coding_->RegisterTransportCallback(NULL);
   RTC_DCHECK_EQ(0, error);
-
-  if (_moduleProcessThreadPtr)
-    _moduleProcessThreadPtr->DeRegisterModule(rtp_rtcp_.get());
 }
 
 void ChannelSend::StartSend() {
@@ -858,7 +845,6 @@ ANAStats ChannelSend::GetANAStatistics() const {
 }
 
 RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
-  RTC_DCHECK(module_process_thread_checker_.IsCurrent());
   return rtp_rtcp_.get();
 }
 
@@ -930,7 +916,6 @@ void ChannelSend::InitFrameTransformerDelegate(
 std::unique_ptr<ChannelSendInterface> CreateChannelSend(
     Clock* clock,
     TaskQueueFactory* task_queue_factory,
-    ProcessThread* module_process_thread,
     Transport* rtp_transport,
     RtcpRttStats* rtcp_rtt_stats,
     RtcEventLog* rtc_event_log,
@@ -942,10 +927,10 @@ std::unique_ptr<ChannelSendInterface> CreateChannelSend(
     rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
     TransportFeedbackObserver* feedback_observer) {
   return std::make_unique<ChannelSend>(
-      clock, task_queue_factory, module_process_thread, rtp_transport,
-      rtcp_rtt_stats, rtc_event_log, frame_encryptor, crypto_options,
-      extmap_allow_mixed, rtcp_report_interval_ms, ssrc,
-      std::move(frame_transformer), feedback_observer);
+      clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
+      frame_encryptor, crypto_options, extmap_allow_mixed,
+      rtcp_report_interval_ms, ssrc, std::move(frame_transformer),
+      feedback_observer);
 }
 
 }  // namespace voe