Move BWE_TEST_LOGGING_COMPILE_TIME_ENABLE code to the worker thread.

This is part of moving calls to GetSendRates() to the worker.

Change-Id: Ifb93096a863ddf2669237e7f44af296d0e086b20
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177661
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31582}

modules/rtp_rtcp/source/rtp_rtcp_impl2.cc

@@ -289,6 +289,10 @@ void ModuleRtpRtcpImpl2::SetCsrcs(const std::vector<uint32_t>& csrcs) {
 // TODO(pbos): Handle media and RTX streams separately (separate RTCP
 // feedbacks).
 RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
+  // TODO(bugs.webrtc.org/11581): Called by potentially multiple threads.
+  // "Send*" methods and on the ProcessThread. Make sure it's only called on the
+  // construction thread.
+
   RTCPSender::FeedbackState state;
   // This is called also when receiver_only is true. Hence below
   // checks that rtp_sender_ exists.
@@ -653,6 +657,7 @@ void ModuleRtpRtcpImpl2::BitrateSent(uint32_t* total_rate,
                                      uint32_t* video_rate,
                                      uint32_t* fec_rate,
                                      uint32_t* nack_rate) const {
+  RTC_DCHECK_RUN_ON(worker_queue_);
   RtpSendRates send_rates = rtp_sender_->packet_sender.GetSendRates();
   *total_rate = send_rates.Sum().bps<uint32_t>();
   if (video_rate)
@@ -663,6 +668,7 @@ void ModuleRtpRtcpImpl2::BitrateSent(uint32_t* total_rate,
 }
 
 RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
+  RTC_DCHECK_RUN_ON(worker_queue_);
   return rtp_sender_->packet_sender.GetSendRates();
 }
 

modules/rtp_rtcp/source/rtp_sender_egress.cc

@@ -17,7 +17,6 @@
 #include "absl/strings/match.h"
 #include "api/transport/field_trial_based_config.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/task_utils/to_queued_task.h"
 
@@ -70,7 +69,9 @@ RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
       packet_history_(packet_history),
       transport_(config.outgoing_transport),
       event_log_(config.event_log),
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
       is_audio_(config.audio),
+#endif
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
       transport_feedback_observer_(config.transport_feedback_callback),
       send_side_delay_observer_(config.send_side_delay_observer),
@@ -110,30 +111,18 @@ void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
   RTC_DCHECK_RUN_ON(&pacer_checker_);
   RTC_DCHECK(packet);
 
-  const uint32_t packet_ssrc = packet->Ssrc();
   RTC_DCHECK(packet->packet_type().has_value());
   RTC_DCHECK(HasCorrectSsrc(*packet));
-  int64_t now_ms = clock_->TimeInMilliseconds();
 
-  if (is_audio_) {
+  const uint32_t packet_ssrc = packet->Ssrc();
+  const int64_t now_ms = clock_->TimeInMilliseconds();
+
 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+  worker_queue_->PostTask(
-                                    GetSendRates().Sum().kbps(), packet_ssrc);
+      ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc]() {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        BweTestLoggingPlot(now_ms, packet_ssrc);
-        1, "AudioNackBitrate_kbps", now_ms,
+      }));
-        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
-        packet_ssrc);
 #endif
-  } else {
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
-                                    GetSendRates().Sum().kbps(), packet_ssrc);
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
-        1, "VideoNackBitrate_kbps", now_ms,
-        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
-        packet_ssrc);
-#endif
-  }
 
   if (need_rtp_packet_infos_ &&
       packet->packet_type() == RtpPacketToSend::Type::kVideo) {
@@ -226,11 +215,12 @@ void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
     RtpPacketMediaType packet_type = *packet->packet_type();
     RtpPacketCounter counter(*packet);
     size_t size = packet->size();
-    worker_queue_->PostTask(ToQueuedTask(
+    worker_queue_->PostTask(
-        task_safety_, [this, now_ms, ssrc = packet->Ssrc(), packet_type,
+        ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc, packet_type,
                                     counter = std::move(counter), size]() {
           RTC_DCHECK_RUN_ON(worker_queue_);
-          UpdateRtpStats(now_ms, ssrc, packet_type, std::move(counter), size);
+          UpdateRtpStats(now_ms, packet_ssrc, packet_type, std::move(counter),
+                         size);
         }));
   }
 }
@@ -516,4 +506,25 @@ void RtpSenderEgress::PeriodicUpdate() {
       send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
 }
 
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+void RtpSenderEgress::BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+
+  const auto rates = GetSendRates();
+  if (is_audio_) {
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+                                    rates.Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "AudioNackBitrate_kbps", now_ms,
+        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
+  } else {
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
+                                    rates.Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "VideoNackBitrate_kbps", now_ms,
+        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
+  }
+}
+#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+
 }  // namespace webrtc

modules/rtp_rtcp/source/rtp_sender_egress.h

@@ -20,6 +20,7 @@
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "api/task_queue/task_queue_base.h"
 #include "api/units/data_rate.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -110,6 +111,9 @@ class RtpSenderEgress {
                       RtpPacketMediaType packet_type,
                       RtpPacketCounter counter,
                       size_t packet_size);
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  void BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc);
+#endif
 
   // Called on a timer, once a second, on the worker_queue_.
   void PeriodicUpdate();
@@ -125,7 +129,9 @@ class RtpSenderEgress {
   RtpPacketHistory* const packet_history_;
   Transport* const transport_;
   RtcEventLog* const event_log_;
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
   const bool is_audio_;
+#endif
   const bool need_rtp_packet_infos_;
 
   TransportFeedbackObserver* const transport_feedback_observer_;