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}
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 848a477..e7c6a43 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -57,6 +57,7 @@
"../../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",
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index fcc7ee3..4254a63 100644
--- a/modules/pacing/packet_router.cc
+++ b/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 @@
}
}
-#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;
}
diff --git a/modules/pacing/task_queue_paced_sender.cc b/modules/pacing/task_queue_paced_sender.cc
index b95bed2..db8e87a 100644
--- a/modules/pacing/task_queue_paced_sender.cc
+++ b/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::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 @@
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;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index c0a8075..244f644 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/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 @@
}
RTC_CHECK_NOTREACHED();
}
-#endif
constexpr char kIncludeCaptureClockOffset[] =
"WebRTC-IncludeCaptureClockOffset";
@@ -166,10 +163,8 @@
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
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 614a386..05428ff 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -97,8 +97,7 @@
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 @@
return "";
}
}
-#endif
bool IsNoopDelay(const VideoPlayoutDelay& delay) {
return delay.min_ms == -1 && delay.max_ms == -1;
@@ -477,10 +475,8 @@
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)
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index cdb75ad..598395a 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -1169,7 +1169,7 @@
#if RTC_TRACE_EVENTS_ENABLED
EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
- #endif
+#endif
}
TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) {
@@ -1180,7 +1180,7 @@
#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 @@
// 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 @@
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
