Eliminate unnecessary RTC_TRACE_EVENTS_ENABLED

Bug: webrtc:14073
Change-Id: I6365cc17393be52c11312dfa954783a3e135cb8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262263
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36929}

modules/pacing/BUILD.gn

@@ -57,6 +57,7 @@ rtc_library("pacing") {
     "../../rtc_base:timeutils",
     "../../rtc_base/experiments:field_trial_parser",
     "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:unused",
     "../../rtc_base/task_utils:to_queued_task",
     "../../system_wrappers",
     "../../system_wrappers:metrics",

modules/pacing/packet_router.cc

@@ -23,6 +23,7 @@
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/system/unused.h"
 #include "rtc_base/time_utils.h"
 #include "rtc_base/trace_event.h"
 
@@ -214,14 +215,13 @@ std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
     }
   }
 
-#if RTC_TRACE_EVENTS_ENABLED
   for (auto& packet : padding_packets) {
+    RTC_UNUSED(packet);
     TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
                  "PacketRouter::GeneratePadding::Loop", "sequence_number",
                  packet->SequenceNumber(), "rtp_timestamp",
                  packet->Timestamp());
   }
-#endif
 
   return padding_packets;
 }

modules/pacing/task_queue_paced_sender.cc

@@ -18,6 +18,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/experiments/field_trial_units.h"
+#include "rtc_base/system/unused.h"
 #include "rtc_base/trace_event.h"
 
 namespace webrtc {
@@ -129,16 +130,15 @@ void TaskQueuePacedSender::SetPacingRates(DataRate pacing_rate,
 
 void TaskQueuePacedSender::EnqueuePackets(
     std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
-#if RTC_TRACE_EVENTS_ENABLED
   TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
                "TaskQueuePacedSender::EnqueuePackets");
   for (auto& packet : packets) {
+    RTC_UNUSED(packet);
     TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
                  "TaskQueuePacedSender::EnqueuePackets::Loop",
                  "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
                  packet->Timestamp());
   }
-#endif
 
   task_queue_.PostTask([this, packets_ = std::move(packets)]() mutable {
     RTC_DCHECK_RUN_ON(&task_queue_);
@@ -224,10 +224,8 @@ void TaskQueuePacedSender::MaybeProcessPackets(
     Timestamp scheduled_process_time) {
   RTC_DCHECK_RUN_ON(&task_queue_);
 
-#if RTC_TRACE_EVENTS_ENABLED
   TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
                "TaskQueuePacedSender::MaybeProcessPackets");
-#endif
 
   if (is_shutdown_ || !is_started_) {
     return;

modules/rtp_rtcp/source/rtp_sender_audio.cc

@@ -35,9 +35,7 @@
 namespace webrtc {
 
 namespace {
-
-#if RTC_TRACE_EVENTS_ENABLED
-const char* FrameTypeToString(AudioFrameType frame_type) {
+[[maybe_unused]] const char* FrameTypeToString(AudioFrameType frame_type) {
   switch (frame_type) {
     case AudioFrameType::kEmptyFrame:
       return "empty";
@@ -48,7 +46,6 @@ const char* FrameTypeToString(AudioFrameType frame_type) {
   }
   RTC_CHECK_NOTREACHED();
 }
-#endif
 
 constexpr char kIncludeCaptureClockOffset[] =
     "WebRTC-IncludeCaptureClockOffset";
@@ -166,10 +163,8 @@ bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
                                const uint8_t* payload_data,
                                size_t payload_size,
                                int64_t absolute_capture_timestamp_ms) {
-#if RTC_TRACE_EVENTS_ENABLED
   TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
                           FrameTypeToString(frame_type));
-  #endif
 
   // From RFC 4733:
   // A source has wide latitude as to how often it sends event updates. A

modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -97,8 +97,7 @@ bool IsBaseLayer(const RTPVideoHeader& video_header) {
   return true;
 }
 
-#if RTC_TRACE_EVENTS_ENABLED
-const char* FrameTypeToString(VideoFrameType frame_type) {
+[[maybe_unused]] const char* FrameTypeToString(VideoFrameType frame_type) {
   switch (frame_type) {
     case VideoFrameType::kEmptyFrame:
       return "empty";
@@ -111,7 +110,6 @@ const char* FrameTypeToString(VideoFrameType frame_type) {
       return "";
   }
 }
-#endif
 
 bool IsNoopDelay(const VideoPlayoutDelay& delay) {
   return delay.min_ms == -1 && delay.max_ms == -1;
@@ -477,10 +475,8 @@ bool RTPSenderVideo::SendVideo(
     rtc::ArrayView<const uint8_t> payload,
     RTPVideoHeader video_header,
     absl::optional<int64_t> expected_retransmission_time_ms) {
-#if RTC_TRACE_EVENTS_ENABLED
   TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
                           FrameTypeToString(video_header.frame_type));
-#endif
   RTC_CHECK_RUNS_SERIALIZED(&send_checker_);
 
   if (video_header.frame_type == VideoFrameType::kEmptyFrame)

pc/rtc_stats_integrationtest.cc

@@ -1169,7 +1169,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
 
 #if RTC_TRACE_EVENTS_ENABLED
   EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
-  #endif
+#endif
 }
 
 TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) {
@@ -1180,7 +1180,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) {
 
 #if RTC_TRACE_EVENTS_ENABLED
   EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
-  #endif
+#endif
 }
 
 // These tests exercise the integration of the stats selection algorithm inside
@@ -1260,10 +1260,10 @@ TEST_F(RTCStatsIntegrationTest,
   // Any pending stats requests should have completed in the act of destroying
   // the peer connection.
   ASSERT_TRUE(stats_obtainer->report());
-  #if RTC_TRACE_EVENTS_ENABLED
+#if RTC_TRACE_EVENTS_ENABLED
   EXPECT_EQ(stats_obtainer->report()->ToJson(),
             RTCStatsReportTraceListener::last_trace());
-  #endif
+#endif
 }
 
 TEST_F(RTCStatsIntegrationTest, GetsStatsWhileClosingPeerConnection) {
@@ -1275,10 +1275,10 @@ TEST_F(RTCStatsIntegrationTest, GetsStatsWhileClosingPeerConnection) {
   caller_->pc()->Close();
 
   ASSERT_TRUE(stats_obtainer->report());
-  #if RTC_TRACE_EVENTS_ENABLED
+#if RTC_TRACE_EVENTS_ENABLED
   EXPECT_EQ(stats_obtainer->report()->ToJson(),
             RTCStatsReportTraceListener::last_trace());
-  #endif
+#endif
 }
 
 // GetStatsReferencedIds() is optimized to recognize what is or isn't a