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Eliminate unnecessary RTC_TRACE_EVENTS_ENABLED

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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}
This commit is contained in:
Jianhui Dai
2022-05-13 10:40:25 +08:00
committed by WebRTC LUCI CQ
parent d68a06aedf
commit b1ba85385e
6 changed files with 13 additions and 23 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -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

6
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -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)