From b1ba85385e424ad3cbac4dd2a1c7a1e7771b3be3 Mon Sep 17 00:00:00 2001 From: Jianhui Dai Date: Fri, 13 May 2022 10:40:25 +0800 Subject: [PATCH] Eliminate unnecessary `RTC_TRACE_EVENTS_ENABLED` MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:14073 Change-Id: I6365cc17393be52c11312dfa954783a3e135cb8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262263 Reviewed-by: Erik Språng Reviewed-by: Johannes Kron Commit-Queue: Harald Alvestrand Reviewed-by: Henrik Boström Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#36929} --- modules/pacing/BUILD.gn | 1 + modules/pacing/packet_router.cc | 4 ++-- modules/pacing/task_queue_paced_sender.cc | 6 ++---- modules/rtp_rtcp/source/rtp_sender_audio.cc | 7 +------ modules/rtp_rtcp/source/rtp_sender_video.cc | 6 +----- pc/rtc_stats_integrationtest.cc | 12 ++++++------ 6 files changed, 13 insertions(+), 23 deletions(-) diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn index 848a477a82..e7c6a43a28 100644 --- a/modules/pacing/BUILD.gn +++ b/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", diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc index fcc7ee3449..4254a636d3 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 @@ std::vector> 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; } diff --git a/modules/pacing/task_queue_paced_sender.cc b/modules/pacing/task_queue_paced_sender.cc index b95bed28bb..db8e87ae6e 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::SetPacingRates(DataRate pacing_rate, void TaskQueuePacedSender::EnqueuePackets( std::vector> 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; diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc index c0a8075306..244f644bd1 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 @@ 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 diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc index 614a3862b0..05428ff289 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/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 payload, RTPVideoHeader video_header, absl::optional 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 cdb75ada65..598395af94 100644 --- a/pc/rtc_stats_integrationtest.cc +++ b/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